Formatting some files with LOG macros usage.
In order to create a clean CL to switch to RTC_ prefixed LOG macros
this CL runs `git cl format --full` on the files with LOG macros in
the following directories:
- modules/audio_device
- modules/media_file
- modules/video_capture
This CL has been automatically generated with:
for m in PLOG \
LOG_TAG \
LOG_GLEM \
LOG_GLE_EX \
LOG_GLE \
LAST_SYSTEM_ERROR \
LOG_ERRNO_EX \
LOG_ERRNO \
LOG_ERR_EX \
LOG_ERR \
LOG_V \
LOG_F \
LOG_T_F \
LOG_E \
LOG_T \
LOG_CHECK_LEVEL_V \
LOG_CHECK_LEVEL \
LOG
do
for d in media_file video_capture audio_device; do
cd modules/$d
git grep -l $m | grep -E "\.(cc|h|m|mm)$" | xargs sed -i "1 s/$/ /"
cd ../..
done
done
git cl format --full
Bug: webrtc:8452
Change-Id: I2858b6928e6bd79957f2e5e0b07028eb68a304b2
Reviewed-on: https://webrtc-review.googlesource.com/21322
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20613}diff --git a/modules/audio_device/android/audio_device_template.h b/modules/audio_device/android/audio_device_template.h
index 5be3a9e..04ff1bc 100644
--- a/modules/audio_device/android/audio_device_template.h
+++ b/modules/audio_device/android/audio_device_template.h
@@ -100,18 +100,16 @@
return 1;
}
- int32_t PlayoutDeviceName(
- uint16_t index,
- char name[kAdmMaxDeviceNameSize],
- char guid[kAdmMaxGuidSize]) override {
+ int32_t PlayoutDeviceName(uint16_t index,
+ char name[kAdmMaxDeviceNameSize],
+ char guid[kAdmMaxGuidSize]) override {
FATAL() << "Should never be called";
return -1;
}
- int32_t RecordingDeviceName(
- uint16_t index,
- char name[kAdmMaxDeviceNameSize],
- char guid[kAdmMaxGuidSize]) override {
+ int32_t RecordingDeviceName(uint16_t index,
+ char name[kAdmMaxDeviceNameSize],
+ char guid[kAdmMaxGuidSize]) override {
FATAL() << "Should never be called";
return -1;
}
@@ -215,9 +213,7 @@
return err;
}
- bool Recording() const override {
- return input_.Recording() ;
- }
+ bool Recording() const override { return input_.Recording(); }
int32_t SetAGC(bool enable) override {
if (enable) {
@@ -276,7 +272,7 @@
return output_.MinSpeakerVolume(minVolume);
}
- int32_t MicrophoneVolumeIsAvailable(bool& available) override{
+ int32_t MicrophoneVolumeIsAvailable(bool& available) override {
available = false;
return -1;
}
diff --git a/modules/audio_device/android/audio_manager.cc b/modules/audio_device/android/audio_manager.cc
index 9e38a85..6f385a3 100644
--- a/modules/audio_device/android/audio_manager.cc
+++ b/modules/audio_device/android/audio_manager.cc
@@ -107,9 +107,9 @@
// that the user explicitly selects the high-latency audio path, hence we use
// the selected |audio_layer| here to set the delay estimate.
delay_estimate_in_milliseconds_ =
- (audio_layer == AudioDeviceModule::kAndroidJavaAudio) ?
- kHighLatencyModeDelayEstimateInMilliseconds :
- kLowLatencyModeDelayEstimateInMilliseconds;
+ (audio_layer == AudioDeviceModule::kAndroidJavaAudio)
+ ? kHighLatencyModeDelayEstimateInMilliseconds
+ : kLowLatencyModeDelayEstimateInMilliseconds;
ALOGD("delay_estimate_in_milliseconds: %d", delay_estimate_in_milliseconds_);
}
@@ -201,8 +201,9 @@
ALOGD("IsLowLatencyPlayoutSupported()");
// Some devices are blacklisted for usage of OpenSL ES even if they report
// that low-latency playout is supported. See b/21485703 for details.
- return j_audio_manager_->IsDeviceBlacklistedForOpenSLESUsage() ?
- false : low_latency_playout_;
+ return j_audio_manager_->IsDeviceBlacklistedForOpenSLESUsage()
+ ? false
+ : low_latency_playout_;
}
bool AudioManager::IsLowLatencyRecordSupported() const {
diff --git a/modules/audio_device/android/audio_record_jni.cc b/modules/audio_device/android/audio_record_jni.cc
index b437644..79f8c8b 100644
--- a/modules/audio_device/android/audio_record_jni.cc
+++ b/modules/audio_device/android/audio_record_jni.cc
@@ -41,8 +41,8 @@
AudioRecordJni::JavaAudioRecord::~JavaAudioRecord() {}
-int AudioRecordJni::JavaAudioRecord::InitRecording(
- int sample_rate, size_t channels) {
+int AudioRecordJni::JavaAudioRecord::InitRecording(int sample_rate,
+ size_t channels) {
return audio_record_->CallIntMethod(init_recording_,
static_cast<jint>(sample_rate),
static_cast<jint>(channels));
@@ -83,10 +83,10 @@
RTC_CHECK(j_environment_);
JNINativeMethod native_methods[] = {
{"nativeCacheDirectBufferAddress", "(Ljava/nio/ByteBuffer;J)V",
- reinterpret_cast<void*>(
- &webrtc::AudioRecordJni::CacheDirectBufferAddress)},
+ reinterpret_cast<void*>(
+ &webrtc::AudioRecordJni::CacheDirectBufferAddress)},
{"nativeDataIsRecorded", "(IJ)V",
- reinterpret_cast<void*>(&webrtc::AudioRecordJni::DataIsRecorded)}};
+ reinterpret_cast<void*>(&webrtc::AudioRecordJni::DataIsRecorded)}};
j_native_registration_ = j_environment_->RegisterNatives(
"org/webrtc/voiceengine/WebRtcAudioRecord", native_methods,
arraysize(native_methods));
@@ -168,7 +168,7 @@
thread_checker_java_.DetachFromThread();
initialized_ = false;
recording_ = false;
- direct_buffer_address_= nullptr;
+ direct_buffer_address_ = nullptr;
return 0;
}
@@ -206,29 +206,32 @@
return j_audio_record_->EnableBuiltInNS(enable) ? 0 : -1;
}
-void JNICALL AudioRecordJni::CacheDirectBufferAddress(
- JNIEnv* env, jobject obj, jobject byte_buffer, jlong nativeAudioRecord) {
+void JNICALL AudioRecordJni::CacheDirectBufferAddress(JNIEnv* env,
+ jobject obj,
+ jobject byte_buffer,
+ jlong nativeAudioRecord) {
webrtc::AudioRecordJni* this_object =
- reinterpret_cast<webrtc::AudioRecordJni*> (nativeAudioRecord);
+ reinterpret_cast<webrtc::AudioRecordJni*>(nativeAudioRecord);
this_object->OnCacheDirectBufferAddress(env, byte_buffer);
}
-void AudioRecordJni::OnCacheDirectBufferAddress(
- JNIEnv* env, jobject byte_buffer) {
+void AudioRecordJni::OnCacheDirectBufferAddress(JNIEnv* env,
+ jobject byte_buffer) {
ALOGD("OnCacheDirectBufferAddress");
RTC_DCHECK(thread_checker_.CalledOnValidThread());
RTC_DCHECK(!direct_buffer_address_);
- direct_buffer_address_ =
- env->GetDirectBufferAddress(byte_buffer);
+ direct_buffer_address_ = env->GetDirectBufferAddress(byte_buffer);
jlong capacity = env->GetDirectBufferCapacity(byte_buffer);
ALOGD("direct buffer capacity: %lld", capacity);
direct_buffer_capacity_in_bytes_ = static_cast<size_t>(capacity);
}
-void JNICALL AudioRecordJni::DataIsRecorded(
- JNIEnv* env, jobject obj, jint length, jlong nativeAudioRecord) {
+void JNICALL AudioRecordJni::DataIsRecorded(JNIEnv* env,
+ jobject obj,
+ jint length,
+ jlong nativeAudioRecord) {
webrtc::AudioRecordJni* this_object =
- reinterpret_cast<webrtc::AudioRecordJni*> (nativeAudioRecord);
+ reinterpret_cast<webrtc::AudioRecordJni*>(nativeAudioRecord);
this_object->OnDataIsRecorded(length);
}
diff --git a/modules/audio_device/android/audio_track_jni.cc b/modules/audio_device/android/audio_track_jni.cc
index 45e59c4..89d4af0 100644
--- a/modules/audio_device/android/audio_track_jni.cc
+++ b/modules/audio_device/android/audio_track_jni.cc
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "modules/audio_device/android/audio_manager.h"
#include "modules/audio_device/android/audio_track_jni.h"
+#include "modules/audio_device/android/audio_manager.h"
#include <utility>
@@ -82,10 +82,10 @@
RTC_CHECK(j_environment_);
JNINativeMethod native_methods[] = {
{"nativeCacheDirectBufferAddress", "(Ljava/nio/ByteBuffer;J)V",
- reinterpret_cast<void*>(
- &webrtc::AudioTrackJni::CacheDirectBufferAddress)},
+ reinterpret_cast<void*>(
+ &webrtc::AudioTrackJni::CacheDirectBufferAddress)},
{"nativeGetPlayoutData", "(IJ)V",
- reinterpret_cast<void*>(&webrtc::AudioTrackJni::GetPlayoutData)}};
+ reinterpret_cast<void*>(&webrtc::AudioTrackJni::GetPlayoutData)}};
j_native_registration_ = j_environment_->RegisterNatives(
"org/webrtc/voiceengine/WebRtcAudioTrack", native_methods,
arraysize(native_methods));
@@ -122,8 +122,8 @@
RTC_DCHECK(thread_checker_.CalledOnValidThread());
RTC_DCHECK(!initialized_);
RTC_DCHECK(!playing_);
- if (!j_audio_track_->InitPlayout(
- audio_parameters_.sample_rate(), audio_parameters_.channels())) {
+ if (!j_audio_track_->InitPlayout(audio_parameters_.sample_rate(),
+ audio_parameters_.channels())) {
ALOGE("InitPlayout failed!");
return -1;
}
@@ -209,20 +209,21 @@
audio_device_buffer_->SetPlayoutChannels(channels);
}
-void JNICALL AudioTrackJni::CacheDirectBufferAddress(
- JNIEnv* env, jobject obj, jobject byte_buffer, jlong nativeAudioTrack) {
+void JNICALL AudioTrackJni::CacheDirectBufferAddress(JNIEnv* env,
+ jobject obj,
+ jobject byte_buffer,
+ jlong nativeAudioTrack) {
webrtc::AudioTrackJni* this_object =
- reinterpret_cast<webrtc::AudioTrackJni*> (nativeAudioTrack);
+ reinterpret_cast<webrtc::AudioTrackJni*>(nativeAudioTrack);
this_object->OnCacheDirectBufferAddress(env, byte_buffer);
}
-void AudioTrackJni::OnCacheDirectBufferAddress(
- JNIEnv* env, jobject byte_buffer) {
+void AudioTrackJni::OnCacheDirectBufferAddress(JNIEnv* env,
+ jobject byte_buffer) {
ALOGD("OnCacheDirectBufferAddress");
RTC_DCHECK(thread_checker_.CalledOnValidThread());
RTC_DCHECK(!direct_buffer_address_);
- direct_buffer_address_ =
- env->GetDirectBufferAddress(byte_buffer);
+ direct_buffer_address_ = env->GetDirectBufferAddress(byte_buffer);
jlong capacity = env->GetDirectBufferCapacity(byte_buffer);
ALOGD("direct buffer capacity: %lld", capacity);
direct_buffer_capacity_in_bytes_ = static_cast<size_t>(capacity);
@@ -231,10 +232,12 @@
ALOGD("frames_per_buffer: %" PRIuS, frames_per_buffer_);
}
-void JNICALL AudioTrackJni::GetPlayoutData(
- JNIEnv* env, jobject obj, jint length, jlong nativeAudioTrack) {
+void JNICALL AudioTrackJni::GetPlayoutData(JNIEnv* env,
+ jobject obj,
+ jint length,
+ jlong nativeAudioTrack) {
webrtc::AudioTrackJni* this_object =
- reinterpret_cast<webrtc::AudioTrackJni*> (nativeAudioTrack);
+ reinterpret_cast<webrtc::AudioTrackJni*>(nativeAudioTrack);
this_object->OnGetPlayoutData(static_cast<size_t>(length));
}
diff --git a/modules/audio_device/android/opensles_player.cc b/modules/audio_device/android/opensles_player.cc
index 1530741..7ac6912 100644
--- a/modules/audio_device/android/opensles_player.cc
+++ b/modules/audio_device/android/opensles_player.cc
@@ -289,10 +289,10 @@
SLDataSink audio_sink = {&locator_output_mix, nullptr};
// Define interfaces that we indend to use and realize.
- const SLInterfaceID interface_ids[] = {
- SL_IID_ANDROIDCONFIGURATION, SL_IID_BUFFERQUEUE, SL_IID_VOLUME};
- const SLboolean interface_required[] = {
- SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE};
+ const SLInterfaceID interface_ids[] = {SL_IID_ANDROIDCONFIGURATION,
+ SL_IID_BUFFERQUEUE, SL_IID_VOLUME};
+ const SLboolean interface_required[] = {SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE,
+ SL_BOOLEAN_TRUE};
// Create the audio player on the engine interface.
RETURN_ON_ERROR(
diff --git a/modules/audio_device/audio_device_buffer.h b/modules/audio_device/audio_device_buffer.h
index a68bbdf..8b8e907 100644
--- a/modules/audio_device/audio_device_buffer.h
+++ b/modules/audio_device/audio_device_buffer.h
@@ -197,7 +197,7 @@
// dynamically.
rtc::BufferT<int16_t> rec_buffer_ RTC_ACCESS_ON(recording_thread_checker_);
- // AGC parameters.
+// AGC parameters.
#if !defined(WEBRTC_WIN)
uint32_t current_mic_level_ RTC_ACCESS_ON(recording_thread_checker_);
#else
diff --git a/modules/audio_device/audio_device_impl.cc b/modules/audio_device/audio_device_impl.cc
index 12caa2e..0a669ca 100644
--- a/modules/audio_device/audio_device_impl.cc
+++ b/modules/audio_device/audio_device_impl.cc
@@ -386,7 +386,7 @@
int32_t AudioDeviceModuleImpl::SpeakerMuteIsAvailable(bool* available) {
LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
- bool isAvailable = false;
+ bool isAvailable = false;
if (audio_device_->SpeakerMuteIsAvailable(isAvailable) == -1) {
return -1;
}
diff --git a/modules/audio_device/dummy/file_audio_device.cc b/modules/audio_device/dummy/file_audio_device.cc
index 6b0ee04..6954762 100644
--- a/modules/audio_device/dummy/file_audio_device.cc
+++ b/modules/audio_device/dummy/file_audio_device.cc
@@ -26,24 +26,23 @@
kRecordingFixedSampleRate / 100 * kRecordingNumChannels * 2;
FileAudioDevice::FileAudioDevice(const char* inputFilename,
- const char* outputFilename):
- _ptrAudioBuffer(NULL),
- _recordingBuffer(NULL),
- _playoutBuffer(NULL),
- _recordingFramesLeft(0),
- _playoutFramesLeft(0),
- _recordingBufferSizeIn10MS(0),
- _recordingFramesIn10MS(0),
- _playoutFramesIn10MS(0),
- _playing(false),
- _recording(false),
- _lastCallPlayoutMillis(0),
- _lastCallRecordMillis(0),
- _outputFile(*FileWrapper::Create()),
- _inputFile(*FileWrapper::Create()),
- _outputFilename(outputFilename),
- _inputFilename(inputFilename) {
-}
+ const char* outputFilename)
+ : _ptrAudioBuffer(NULL),
+ _recordingBuffer(NULL),
+ _playoutBuffer(NULL),
+ _recordingFramesLeft(0),
+ _playoutFramesLeft(0),
+ _recordingBufferSizeIn10MS(0),
+ _recordingFramesIn10MS(0),
+ _playoutFramesIn10MS(0),
+ _playing(false),
+ _recording(false),
+ _lastCallPlayoutMillis(0),
+ _lastCallRecordMillis(0),
+ _outputFile(*FileWrapper::Create()),
+ _inputFile(*FileWrapper::Create()),
+ _outputFilename(outputFilename),
+ _inputFilename(inputFilename) {}
FileAudioDevice::~FileAudioDevice() {
delete &_outputFile;
@@ -59,9 +58,13 @@
return InitStatus::OK;
}
-int32_t FileAudioDevice::Terminate() { return 0; }
+int32_t FileAudioDevice::Terminate() {
+ return 0;
+}
-bool FileAudioDevice::Initialized() const { return true; }
+bool FileAudioDevice::Initialized() const {
+ return true;
+}
int16_t FileAudioDevice::PlayoutDevices() {
return 1;
@@ -72,8 +75,8 @@
}
int32_t FileAudioDevice::PlayoutDeviceName(uint16_t index,
- char name[kAdmMaxDeviceNameSize],
- char guid[kAdmMaxGuidSize]) {
+ char name[kAdmMaxDeviceNameSize],
+ char guid[kAdmMaxGuidSize]) {
const char* kName = "dummy_device";
const char* kGuid = "dummy_device_unique_id";
if (index < 1) {
@@ -87,8 +90,8 @@
}
int32_t FileAudioDevice::RecordingDeviceName(uint16_t index,
- char name[kAdmMaxDeviceNameSize],
- char guid[kAdmMaxGuidSize]) {
+ char name[kAdmMaxDeviceNameSize],
+ char guid[kAdmMaxGuidSize]) {
const char* kName = "dummy_device";
const char* kGuid = "dummy_device_unique_id";
if (index < 1) {
@@ -138,9 +141,9 @@
int32_t FileAudioDevice::InitPlayout() {
if (_ptrAudioBuffer) {
- // Update webrtc audio buffer with the selected parameters
- _ptrAudioBuffer->SetPlayoutSampleRate(kPlayoutFixedSampleRate);
- _ptrAudioBuffer->SetPlayoutChannels(kPlayoutNumChannels);
+ // Update webrtc audio buffer with the selected parameters
+ _ptrAudioBuffer->SetPlayoutSampleRate(kPlayoutFixedSampleRate);
+ _ptrAudioBuffer->SetPlayoutChannels(kPlayoutNumChannels);
}
return 0;
}
@@ -180,7 +183,7 @@
int32_t FileAudioDevice::StartPlayout() {
if (_playing) {
- return 0;
+ return 0;
}
_playoutFramesIn10MS = static_cast<size_t>(kPlayoutFixedSampleRate / 100);
@@ -188,7 +191,7 @@
_playoutFramesLeft = 0;
if (!_playoutBuffer) {
- _playoutBuffer = new int8_t[kPlayoutBufferSize];
+ _playoutBuffer = new int8_t[kPlayoutBufferSize];
}
if (!_playoutBuffer) {
_playing = false;
@@ -200,7 +203,7 @@
!_outputFile.OpenFile(_outputFilename.c_str(), false)) {
LOG(LS_ERROR) << "Failed to open playout file: " << _outputFilename;
_playing = false;
- delete [] _playoutBuffer;
+ delete[] _playoutBuffer;
_playoutBuffer = NULL;
return -1;
}
@@ -210,32 +213,30 @@
_ptrThreadPlay->Start();
_ptrThreadPlay->SetPriority(rtc::kRealtimePriority);
- LOG(LS_INFO) << "Started playout capture to output file: "
- << _outputFilename;
+ LOG(LS_INFO) << "Started playout capture to output file: " << _outputFilename;
return 0;
}
int32_t FileAudioDevice::StopPlayout() {
{
- rtc::CritScope lock(&_critSect);
- _playing = false;
+ rtc::CritScope lock(&_critSect);
+ _playing = false;
}
// stop playout thread first
if (_ptrThreadPlay) {
- _ptrThreadPlay->Stop();
- _ptrThreadPlay.reset();
+ _ptrThreadPlay->Stop();
+ _ptrThreadPlay.reset();
}
rtc::CritScope lock(&_critSect);
_playoutFramesLeft = 0;
- delete [] _playoutBuffer;
+ delete[] _playoutBuffer;
_playoutBuffer = NULL;
_outputFile.CloseFile();
- LOG(LS_INFO) << "Stopped playout capture to output file: "
- << _outputFilename;
+ LOG(LS_INFO) << "Stopped playout capture to output file: " << _outputFilename;
return 0;
}
@@ -247,11 +248,10 @@
_recording = true;
// Make sure we only create the buffer once.
- _recordingBufferSizeIn10MS = _recordingFramesIn10MS *
- kRecordingNumChannels *
- 2;
+ _recordingBufferSizeIn10MS =
+ _recordingFramesIn10MS * kRecordingNumChannels * 2;
if (!_recordingBuffer) {
- _recordingBuffer = new int8_t[_recordingBufferSizeIn10MS];
+ _recordingBuffer = new int8_t[_recordingBufferSizeIn10MS];
}
if (!_inputFilename.empty() &&
@@ -269,13 +269,11 @@
_ptrThreadRec->Start();
_ptrThreadRec->SetPriority(rtc::kRealtimePriority);
- LOG(LS_INFO) << "Started recording from input file: "
- << _inputFilename;
+ LOG(LS_INFO) << "Started recording from input file: " << _inputFilename;
return 0;
}
-
int32_t FileAudioDevice::StopRecording() {
{
rtc::CritScope lock(&_critSect);
@@ -283,20 +281,19 @@
}
if (_ptrThreadRec) {
- _ptrThreadRec->Stop();
- _ptrThreadRec.reset();
+ _ptrThreadRec->Stop();
+ _ptrThreadRec.reset();
}
rtc::CritScope lock(&_critSect);
_recordingFramesLeft = 0;
if (_recordingBuffer) {
- delete [] _recordingBuffer;
- _recordingBuffer = NULL;
+ delete[] _recordingBuffer;
+ _recordingBuffer = NULL;
}
_inputFile.CloseFile();
- LOG(LS_INFO) << "Stopped recording from input file: "
- << _inputFilename;
+ LOG(LS_INFO) << "Stopped recording from input file: " << _inputFilename;
return 0;
}
@@ -304,25 +301,41 @@
return _recording;
}
-int32_t FileAudioDevice::SetAGC(bool enable) { return -1; }
+int32_t FileAudioDevice::SetAGC(bool enable) {
+ return -1;
+}
-bool FileAudioDevice::AGC() const { return false; }
+bool FileAudioDevice::AGC() const {
+ return false;
+}
-int32_t FileAudioDevice::InitSpeaker() { return -1; }
+int32_t FileAudioDevice::InitSpeaker() {
+ return -1;
+}
-bool FileAudioDevice::SpeakerIsInitialized() const { return false; }
+bool FileAudioDevice::SpeakerIsInitialized() const {
+ return false;
+}
-int32_t FileAudioDevice::InitMicrophone() { return 0; }
+int32_t FileAudioDevice::InitMicrophone() {
+ return 0;
+}
-bool FileAudioDevice::MicrophoneIsInitialized() const { return true; }
+bool FileAudioDevice::MicrophoneIsInitialized() const {
+ return true;
+}
int32_t FileAudioDevice::SpeakerVolumeIsAvailable(bool& available) {
return -1;
}
-int32_t FileAudioDevice::SetSpeakerVolume(uint32_t volume) { return -1; }
+int32_t FileAudioDevice::SetSpeakerVolume(uint32_t volume) {
+ return -1;
+}
-int32_t FileAudioDevice::SpeakerVolume(uint32_t& volume) const { return -1; }
+int32_t FileAudioDevice::SpeakerVolume(uint32_t& volume) const {
+ return -1;
+}
int32_t FileAudioDevice::MaxSpeakerVolume(uint32_t& maxVolume) const {
return -1;
@@ -336,7 +349,9 @@
return -1;
}
-int32_t FileAudioDevice::SetMicrophoneVolume(uint32_t volume) { return -1; }
+int32_t FileAudioDevice::SetMicrophoneVolume(uint32_t volume) {
+ return -1;
+}
int32_t FileAudioDevice::MicrophoneVolume(uint32_t& volume) const {
return -1;
@@ -350,19 +365,29 @@
return -1;
}
-int32_t FileAudioDevice::SpeakerMuteIsAvailable(bool& available) { return -1; }
+int32_t FileAudioDevice::SpeakerMuteIsAvailable(bool& available) {
+ return -1;
+}
-int32_t FileAudioDevice::SetSpeakerMute(bool enable) { return -1; }
+int32_t FileAudioDevice::SetSpeakerMute(bool enable) {
+ return -1;
+}
-int32_t FileAudioDevice::SpeakerMute(bool& enabled) const { return -1; }
+int32_t FileAudioDevice::SpeakerMute(bool& enabled) const {
+ return -1;
+}
int32_t FileAudioDevice::MicrophoneMuteIsAvailable(bool& available) {
return -1;
}
-int32_t FileAudioDevice::SetMicrophoneMute(bool enable) { return -1; }
+int32_t FileAudioDevice::SetMicrophoneMute(bool enable) {
+ return -1;
+}
-int32_t FileAudioDevice::MicrophoneMute(bool& enabled) const { return -1; }
+int32_t FileAudioDevice::MicrophoneMute(bool& enabled) const {
+ return -1;
+}
int32_t FileAudioDevice::StereoPlayoutIsAvailable(bool& available) {
available = true;
@@ -409,81 +434,76 @@
_ptrAudioBuffer->SetPlayoutChannels(0);
}
-bool FileAudioDevice::PlayThreadFunc(void* pThis)
-{
- return (static_cast<FileAudioDevice*>(pThis)->PlayThreadProcess());
+bool FileAudioDevice::PlayThreadFunc(void* pThis) {
+ return (static_cast<FileAudioDevice*>(pThis)->PlayThreadProcess());
}
-bool FileAudioDevice::RecThreadFunc(void* pThis)
-{
- return (static_cast<FileAudioDevice*>(pThis)->RecThreadProcess());
+bool FileAudioDevice::RecThreadFunc(void* pThis) {
+ return (static_cast<FileAudioDevice*>(pThis)->RecThreadProcess());
}
-bool FileAudioDevice::PlayThreadProcess()
-{
- if (!_playing) {
- return false;
- }
- int64_t currentTime = rtc::TimeMillis();
- _critSect.Enter();
+bool FileAudioDevice::PlayThreadProcess() {
+ if (!_playing) {
+ return false;
+ }
+ int64_t currentTime = rtc::TimeMillis();
+ _critSect.Enter();
- if (_lastCallPlayoutMillis == 0 ||
- currentTime - _lastCallPlayoutMillis >= 10) {
- _critSect.Leave();
- _ptrAudioBuffer->RequestPlayoutData(_playoutFramesIn10MS);
- _critSect.Enter();
-
- _playoutFramesLeft = _ptrAudioBuffer->GetPlayoutData(_playoutBuffer);
- RTC_DCHECK_EQ(_playoutFramesIn10MS, _playoutFramesLeft);
- if (_outputFile.is_open()) {
- _outputFile.Write(_playoutBuffer, kPlayoutBufferSize);
- }
- _lastCallPlayoutMillis = currentTime;
- }
- _playoutFramesLeft = 0;
+ if (_lastCallPlayoutMillis == 0 ||
+ currentTime - _lastCallPlayoutMillis >= 10) {
_critSect.Leave();
-
- int64_t deltaTimeMillis = rtc::TimeMillis() - currentTime;
- if (deltaTimeMillis < 10) {
- SleepMs(10 - deltaTimeMillis);
- }
-
- return true;
-}
-
-bool FileAudioDevice::RecThreadProcess()
-{
- if (!_recording) {
- return false;
- }
-
- int64_t currentTime = rtc::TimeMillis();
+ _ptrAudioBuffer->RequestPlayoutData(_playoutFramesIn10MS);
_critSect.Enter();
- if (_lastCallRecordMillis == 0 ||
- currentTime - _lastCallRecordMillis >= 10) {
- if (_inputFile.is_open()) {
- if (_inputFile.Read(_recordingBuffer, kRecordingBufferSize) > 0) {
- _ptrAudioBuffer->SetRecordedBuffer(_recordingBuffer,
- _recordingFramesIn10MS);
- } else {
- _inputFile.Rewind();
- }
- _lastCallRecordMillis = currentTime;
- _critSect.Leave();
- _ptrAudioBuffer->DeliverRecordedData();
- _critSect.Enter();
+ _playoutFramesLeft = _ptrAudioBuffer->GetPlayoutData(_playoutBuffer);
+ RTC_DCHECK_EQ(_playoutFramesIn10MS, _playoutFramesLeft);
+ if (_outputFile.is_open()) {
+ _outputFile.Write(_playoutBuffer, kPlayoutBufferSize);
+ }
+ _lastCallPlayoutMillis = currentTime;
+ }
+ _playoutFramesLeft = 0;
+ _critSect.Leave();
+
+ int64_t deltaTimeMillis = rtc::TimeMillis() - currentTime;
+ if (deltaTimeMillis < 10) {
+ SleepMs(10 - deltaTimeMillis);
+ }
+
+ return true;
+}
+
+bool FileAudioDevice::RecThreadProcess() {
+ if (!_recording) {
+ return false;
+ }
+
+ int64_t currentTime = rtc::TimeMillis();
+ _critSect.Enter();
+
+ if (_lastCallRecordMillis == 0 || currentTime - _lastCallRecordMillis >= 10) {
+ if (_inputFile.is_open()) {
+ if (_inputFile.Read(_recordingBuffer, kRecordingBufferSize) > 0) {
+ _ptrAudioBuffer->SetRecordedBuffer(_recordingBuffer,
+ _recordingFramesIn10MS);
+ } else {
+ _inputFile.Rewind();
}
+ _lastCallRecordMillis = currentTime;
+ _critSect.Leave();
+ _ptrAudioBuffer->DeliverRecordedData();
+ _critSect.Enter();
}
+ }
- _critSect.Leave();
+ _critSect.Leave();
- int64_t deltaTimeMillis = rtc::TimeMillis() - currentTime;
- if (deltaTimeMillis < 10) {
- SleepMs(10 - deltaTimeMillis);
- }
+ int64_t deltaTimeMillis = rtc::TimeMillis() - currentTime;
+ if (deltaTimeMillis < 10) {
+ SleepMs(10 - deltaTimeMillis);
+ }
- return true;
+ return true;
}
} // namespace webrtc
diff --git a/modules/audio_device/dummy/file_audio_device_factory.cc b/modules/audio_device/dummy/file_audio_device_factory.cc
index 96e3eaf..1739953 100644
--- a/modules/audio_device/dummy/file_audio_device_factory.cc
+++ b/modules/audio_device/dummy/file_audio_device_factory.cc
@@ -36,7 +36,8 @@
}
void FileAudioDeviceFactory::SetFilenamesToUse(
- const char* inputAudioFilename, const char* outputAudioFilename) {
+ const char* inputAudioFilename,
+ const char* outputAudioFilename) {
#ifdef WEBRTC_DUMMY_FILE_DEVICES
RTC_DCHECK_LT(strlen(inputAudioFilename), MAX_FILENAME_LEN);
RTC_DCHECK_LT(strlen(outputAudioFilename), MAX_FILENAME_LEN);
@@ -47,8 +48,9 @@
_isConfigured = true;
#else
// Sanity: must be compiled with the right define to run this.
- printf("Trying to use dummy file devices, but is not compiled "
- "with WEBRTC_DUMMY_FILE_DEVICES. Bailing out.\n");
+ printf(
+ "Trying to use dummy file devices, but is not compiled "
+ "with WEBRTC_DUMMY_FILE_DEVICES. Bailing out.\n");
std::exit(1);
#endif
}
diff --git a/modules/audio_device/ios/audio_device_ios.mm b/modules/audio_device/ios/audio_device_ios.mm
index 07d4660..d0a9be5 100644
--- a/modules/audio_device/ios/audio_device_ios.mm
+++ b/modules/audio_device/ios/audio_device_ios.mm
@@ -34,7 +34,6 @@
#import "sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h"
#import "sdk/objc/Framework/Headers/WebRTC/RTCAudioSessionConfiguration.h"
-
namespace webrtc {
#define LOGI() LOG(LS_INFO) << "AudioDeviceIOS::"
@@ -56,7 +55,6 @@
} \
} while (0)
-
// Hardcoded delay estimates based on real measurements.
// TODO(henrika): these value is not used in combination with built-in AEC.
// Can most likely be removed.
@@ -93,8 +91,8 @@
LOG(LS_INFO) << " process ID: " << ios::GetProcessID();
LOG(LS_INFO) << " OS version: " << ios::GetOSVersionString();
LOG(LS_INFO) << " processing cores: " << ios::GetProcessorCount();
-#if defined(__IPHONE_9_0) && defined(__IPHONE_OS_VERSION_MAX_ALLOWED) \
- && __IPHONE_OS_VERSION_MAX_ALLOWED >= __IPHONE_9_0
+#if defined(__IPHONE_9_0) && defined(__IPHONE_OS_VERSION_MAX_ALLOWED) && \
+ __IPHONE_OS_VERSION_MAX_ALLOWED >= __IPHONE_9_0
LOG(LS_INFO) << " low power mode: " << ios::GetLowPowerModeEnabled();
#endif
#if TARGET_IPHONE_SIMULATOR
@@ -121,8 +119,7 @@
LOGI() << "ctor" << ios::GetCurrentThreadDescription();
io_thread_checker_.DetachFromThread();
thread_ = rtc::Thread::Current();
- audio_session_observer_ =
- [[RTCAudioSessionDelegateAdapter alloc] initWithObserver:this];
+ audio_session_observer_ = [[RTCAudioSessionDelegateAdapter alloc] initWithObserver:this];
}
AudioDeviceIOS::~AudioDeviceIOS() {
@@ -152,12 +149,9 @@
// here. They have not been set and confirmed yet since configureForWebRTC
// is not called until audio is about to start. However, it makes sense to
// store the parameters now and then verify at a later stage.
- RTCAudioSessionConfiguration* config =
- [RTCAudioSessionConfiguration webRTCConfiguration];
- playout_parameters_.reset(config.sampleRate,
- config.outputNumberOfChannels);
- record_parameters_.reset(config.sampleRate,
- config.inputNumberOfChannels);
+ RTCAudioSessionConfiguration* config = [RTCAudioSessionConfiguration webRTCConfiguration];
+ playout_parameters_.reset(config.sampleRate, config.outputNumberOfChannels);
+ record_parameters_.reset(config.sampleRate, config.inputNumberOfChannels);
// Ensure that the audio device buffer (ADB) knows about the internal audio
// parameters. Note that, even if we are unable to get a mono audio session,
// we will always tell the I/O audio unit to do a channel format conversion
@@ -235,8 +229,7 @@
if (fine_audio_buffer_) {
fine_audio_buffer_->ResetPlayout();
}
- if (!recording_ &&
- audio_unit_->GetState() == VoiceProcessingAudioUnit::kInitialized) {
+ if (!recording_ && audio_unit_->GetState() == VoiceProcessingAudioUnit::kInitialized) {
if (!audio_unit_->Start()) {
RTCLogError(@"StartPlayout failed to start audio unit.");
return -1;
@@ -269,9 +262,8 @@
average_number_of_playout_callbacks_between_glitches =
num_playout_callbacks_ / num_detected_playout_glitches_;
}
- RTC_HISTOGRAM_COUNTS_100000(
- "WebRTC.Audio.AveragePlayoutCallbacksBetweenGlitches",
- average_number_of_playout_callbacks_between_glitches);
+ RTC_HISTOGRAM_COUNTS_100000("WebRTC.Audio.AveragePlayoutCallbacksBetweenGlitches",
+ average_number_of_playout_callbacks_between_glitches);
RTCLog(@"Average number of playout callbacks between glitches: %d",
average_number_of_playout_callbacks_between_glitches);
return 0;
@@ -286,8 +278,7 @@
if (fine_audio_buffer_) {
fine_audio_buffer_->ResetRecord();
}
- if (!playing_ &&
- audio_unit_->GetState() == VoiceProcessingAudioUnit::kInitialized) {
+ if (!playing_ && audio_unit_->GetState() == VoiceProcessingAudioUnit::kInitialized) {
if (!audio_unit_->Start()) {
RTCLogError(@"StartRecording failed to start audio unit.");
return -1;
@@ -333,9 +324,8 @@
options = AVAudioSessionCategoryOptionDefaultToSpeaker;
}
NSError* error = nil;
- BOOL success = [session setCategory:AVAudioSessionCategoryPlayAndRecord
- withOptions:options
- error:&error];
+ BOOL success =
+ [session setCategory:AVAudioSessionCategoryPlayAndRecord withOptions:options error:&error];
ios::CheckAndLogError(success, error);
[session unlockForConfiguration];
return (error == nil) ? 0 : -1;
@@ -389,7 +379,9 @@
void AudioDeviceIOS::OnCanPlayOrRecordChange(bool can_play_or_record) {
RTC_DCHECK(thread_);
- thread_->Post(RTC_FROM_HERE, this, kMessageTypeCanPlayOrRecordChange,
+ thread_->Post(RTC_FROM_HERE,
+ this,
+ kMessageTypeCanPlayOrRecordChange,
new rtc::TypedMessageData<bool>(can_play_or_record));
}
@@ -406,11 +398,9 @@
RTC_DCHECK_RUN_ON(&io_thread_checker_);
OSStatus result = noErr;
// Simply return if recording is not enabled.
- if (!rtc::AtomicOps::AcquireLoad(&recording_))
- return result;
+ if (!rtc::AtomicOps::AcquireLoad(&recording_)) return result;
- const size_t num_bytes =
- num_frames * VoiceProcessingAudioUnit::kBytesPerSample;
+ const size_t num_bytes = num_frames * VoiceProcessingAudioUnit::kBytesPerSample;
// Set the size of our own audio buffer and clear it first to avoid copying
// in combination with potential reallocations.
// On real iOS devices, the size will only be set once (at first callback).
@@ -435,8 +425,7 @@
// We can make the audio unit provide a buffer instead in io_data, but we
// currently just use our own.
// TODO(henrika): should error handling be improved?
- result = audio_unit_->Render(
- flags, time_stamp, bus_number, num_frames, &audio_buffer_list);
+ result = audio_unit_->Render(flags, time_stamp, bus_number, num_frames, &audio_buffer_list);
if (result != noErr) {
RTCLogError(@"Failed to render audio.");
return result;
@@ -445,9 +434,8 @@
// Get a pointer to the recorded audio and send it to the WebRTC ADB.
// Use the FineAudioBuffer instance to convert between native buffer size
// and the 10ms buffer size used by WebRTC.
- fine_audio_buffer_->DeliverRecordedData(record_audio_buffer_,
- kFixedPlayoutDelayEstimate,
- kFixedRecordDelayEstimate);
+ fine_audio_buffer_->DeliverRecordedData(
+ record_audio_buffer_, kFixedPlayoutDelayEstimate, kFixedRecordDelayEstimate);
return noErr;
}
@@ -465,8 +453,7 @@
// Get pointer to internal audio buffer to which new audio data shall be
// written.
const size_t size_in_bytes = audio_buffer->mDataByteSize;
- RTC_CHECK_EQ(size_in_bytes / VoiceProcessingAudioUnit::kBytesPerSample,
- num_frames);
+ RTC_CHECK_EQ(size_in_bytes / VoiceProcessingAudioUnit::kBytesPerSample, num_frames);
int8_t* destination = reinterpret_cast<int8_t*>(audio_buffer->mData);
// Produce silence and give audio unit a hint about it if playout is not
// activated.
@@ -508,12 +495,11 @@
// Read decoded 16-bit PCM samples from WebRTC (using a size that matches
// the native I/O audio unit) and copy the result to the audio buffer in the
// |io_data| destination.
- fine_audio_buffer_->GetPlayoutData(
- rtc::ArrayView<int8_t>(destination, size_in_bytes));
+ fine_audio_buffer_->GetPlayoutData(rtc::ArrayView<int8_t>(destination, size_in_bytes));
return noErr;
}
-void AudioDeviceIOS::OnMessage(rtc::Message *msg) {
+void AudioDeviceIOS::OnMessage(rtc::Message* msg) {
switch (msg->message_id) {
case kMessageTypeInterruptionBegin:
HandleInterruptionBegin();
@@ -525,8 +511,7 @@
HandleValidRouteChange();
break;
case kMessageTypeCanPlayOrRecordChange: {
- rtc::TypedMessageData<bool>* data =
- static_cast<rtc::TypedMessageData<bool>*>(msg->pdata);
+ rtc::TypedMessageData<bool>* data = static_cast<rtc::TypedMessageData<bool>*>(msg->pdata);
HandleCanPlayOrRecordChange(data->data());
delete data;
break;
@@ -542,10 +527,8 @@
void AudioDeviceIOS::HandleInterruptionBegin() {
RTC_DCHECK_RUN_ON(&thread_checker_);
- RTCLog(@"Interruption begin. IsInterrupted changed from %d to 1.",
- is_interrupted_);
- if (audio_unit_ &&
- audio_unit_->GetState() == VoiceProcessingAudioUnit::kStarted) {
+ RTCLog(@"Interruption begin. IsInterrupted changed from %d to 1.", is_interrupted_);
+ if (audio_unit_ && audio_unit_->GetState() == VoiceProcessingAudioUnit::kStarted) {
RTCLog(@"Stopping the audio unit due to interruption begin.");
if (!audio_unit_->Stop()) {
RTCLogError(@"Failed to stop the audio unit for interruption begin.");
@@ -566,7 +549,8 @@
void AudioDeviceIOS::HandleInterruptionEnd() {
RTC_DCHECK_RUN_ON(&thread_checker_);
RTCLog(@"Interruption ended. IsInterrupted changed from %d to 0. "
- "Updating audio unit state.", is_interrupted_);
+ "Updating audio unit state.",
+ is_interrupted_);
is_interrupted_ = false;
UpdateAudioUnit([RTCAudioSession sharedInstance].canPlayOrRecord);
}
@@ -589,15 +573,13 @@
// Don't do anything if we're interrupted.
if (is_interrupted_) {
- RTCLog(@"Ignoring sample rate change to %f due to interruption.",
- sample_rate);
+ RTCLog(@"Ignoring sample rate change to %f due to interruption.", sample_rate);
return;
}
// If we don't have an audio unit yet, or the audio unit is uninitialized,
// there is no work to do.
- if (!audio_unit_ ||
- audio_unit_->GetState() < VoiceProcessingAudioUnit::kInitialized) {
+ if (!audio_unit_ || audio_unit_->GetState() < VoiceProcessingAudioUnit::kInitialized) {
return;
}
@@ -609,8 +591,7 @@
const size_t session_frames_per_buffer =
static_cast<size_t>(session_sample_rate * session_buffer_duration + .5);
const double current_sample_rate = playout_parameters_.sample_rate();
- const size_t current_frames_per_buffer =
- playout_parameters_.frames_per_buffer();
+ const size_t current_frames_per_buffer = playout_parameters_.frames_per_buffer();
RTCLog(@"Handling playout sample rate change to: %f\n"
" Session sample rate: %f frames_per_buffer: %lu\n"
" ADM sample rate: %f frames_per_buffer: %lu",
@@ -652,15 +633,13 @@
// Initialize the audio unit again with the new sample rate.
RTC_DCHECK_EQ(playout_parameters_.sample_rate(), session_sample_rate);
if (!audio_unit_->Initialize(session_sample_rate)) {
- RTCLogError(@"Failed to initialize the audio unit with sample rate: %f",
- session_sample_rate);
+ RTCLogError(@"Failed to initialize the audio unit with sample rate: %f", session_sample_rate);
return;
}
// Restart the audio unit if it was already running.
if (restart_audio_unit && !audio_unit_->Start()) {
- RTCLogError(@"Failed to start audio unit with sample rate: %f",
- session_sample_rate);
+ RTCLogError(@"Failed to start audio unit with sample rate: %f", session_sample_rate);
return;
}
RTCLog(@"Successfully handled sample rate change.");
@@ -682,8 +661,7 @@
return;
}
num_detected_playout_glitches_++;
- RTCLog(@"Number of detected playout glitches: %lld",
- num_detected_playout_glitches_);
+ RTCLog(@"Number of detected playout glitches: %lld", num_detected_playout_glitches_);
int64_t glitch_count = num_detected_playout_glitches_;
dispatch_async(dispatch_get_main_queue(), ^{
@@ -712,8 +690,7 @@
// Inform the audio device buffer (ADB) about the new audio format.
audio_device_buffer_->SetPlayoutSampleRate(playout_parameters_.sample_rate());
audio_device_buffer_->SetPlayoutChannels(playout_parameters_.channels());
- audio_device_buffer_->SetRecordingSampleRate(
- record_parameters_.sample_rate());
+ audio_device_buffer_->SetRecordingSampleRate(record_parameters_.sample_rate());
audio_device_buffer_->SetRecordingChannels(record_parameters_.channels());
}
@@ -729,8 +706,7 @@
// hardware sample rate but continue and use the non-ideal sample rate after
// reinitializing the audio parameters. Most BT headsets only support 8kHz or
// 16kHz.
- RTCAudioSessionConfiguration* webRTCConfig =
- [RTCAudioSessionConfiguration webRTCConfiguration];
+ RTCAudioSessionConfiguration* webRTCConfig = [RTCAudioSessionConfiguration webRTCConfiguration];
if (sample_rate != webRTCConfig.sampleRate) {
LOG(LS_WARNING) << "Unable to set the preferred sample rate";
}
@@ -740,18 +716,13 @@
// number of audio frames.
// Example: IO buffer size = 0.008 seconds <=> 128 audio frames at 16kHz.
// Hence, 128 is the size we expect to see in upcoming render callbacks.
- playout_parameters_.reset(sample_rate, playout_parameters_.channels(),
- io_buffer_duration);
+ playout_parameters_.reset(sample_rate, playout_parameters_.channels(), io_buffer_duration);
RTC_DCHECK(playout_parameters_.is_complete());
- record_parameters_.reset(sample_rate, record_parameters_.channels(),
- io_buffer_duration);
+ record_parameters_.reset(sample_rate, record_parameters_.channels(), io_buffer_duration);
RTC_DCHECK(record_parameters_.is_complete());
- LOG(LS_INFO) << " frames per I/O buffer: "
- << playout_parameters_.frames_per_buffer();
- LOG(LS_INFO) << " bytes per I/O buffer: "
- << playout_parameters_.GetBytesPerBuffer();
- RTC_DCHECK_EQ(playout_parameters_.GetBytesPerBuffer(),
- record_parameters_.GetBytesPerBuffer());
+ LOG(LS_INFO) << " frames per I/O buffer: " << playout_parameters_.frames_per_buffer();
+ LOG(LS_INFO) << " bytes per I/O buffer: " << playout_parameters_.GetBytesPerBuffer();
+ RTC_DCHECK_EQ(playout_parameters_.GetBytesPerBuffer(), record_parameters_.GetBytesPerBuffer());
// Update the ADB parameters since the sample rate might have changed.
UpdateAudioDeviceBuffer();
@@ -781,7 +752,8 @@
void AudioDeviceIOS::UpdateAudioUnit(bool can_play_or_record) {
RTC_DCHECK_RUN_ON(&thread_checker_);
RTCLog(@"Updating audio unit state. CanPlayOrRecord=%d IsInterrupted=%d",
- can_play_or_record, is_interrupted_);
+ can_play_or_record,
+ is_interrupted_);
if (is_interrupted_) {
RTCLog(@"Ignoring audio unit update due to interruption.");
@@ -790,8 +762,7 @@
// If we're not initialized we don't need to do anything. Audio unit will
// be initialized on initialization.
- if (!audio_is_initialized_)
- return;
+ if (!audio_is_initialized_) return;
// If we're initialized, we must have an audio unit.
RTC_DCHECK(audio_unit_);
@@ -809,13 +780,11 @@
case VoiceProcessingAudioUnit::kUninitialized:
RTCLog(@"VPAU state: Uninitialized");
should_initialize_audio_unit = can_play_or_record;
- should_start_audio_unit = should_initialize_audio_unit &&
- (playing_ || recording_);
+ should_start_audio_unit = should_initialize_audio_unit && (playing_ || recording_);
break;
case VoiceProcessingAudioUnit::kInitialized:
RTCLog(@"VPAU state: Initialized");
- should_start_audio_unit =
- can_play_or_record && (playing_ || recording_);
+ should_start_audio_unit = can_play_or_record && (playing_ || recording_);
should_uninitialize_audio_unit = !can_play_or_record;
break;
case VoiceProcessingAudioUnit::kStarted:
@@ -916,8 +885,7 @@
NSError* error = nil;
if (![session beginWebRTCSession:&error]) {
[session unlockForConfiguration];
- RTCLogError(@"Failed to begin WebRTC session: %@",
- error.localizedDescription);
+ RTCLogError(@"Failed to begin WebRTC session: %@", error.localizedDescription);
return false;
}
diff --git a/modules/audio_device/ios/audio_device_not_implemented_ios.mm b/modules/audio_device/ios/audio_device_not_implemented_ios.mm
index 6dfc02b..4de2653 100644
--- a/modules/audio_device/ios/audio_device_not_implemented_ios.mm
+++ b/modules/audio_device/ios/audio_device_not_implemented_ios.mm
@@ -15,8 +15,7 @@
namespace webrtc {
-int32_t AudioDeviceIOS::ActiveAudioLayer(
- AudioDeviceModule::AudioLayer& audioLayer) const {
+int32_t AudioDeviceIOS::ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const {
audioLayer = AudioDeviceModule::kPlatformDefaultAudio;
return 0;
}
@@ -199,8 +198,7 @@
return 0;
}
-int32_t AudioDeviceIOS::SetRecordingDevice(
- AudioDeviceModule::WindowsDeviceType) {
+int32_t AudioDeviceIOS::SetRecordingDevice(AudioDeviceModule::WindowsDeviceType) {
RTC_NOTREACHED() << "Not implemented";
return -1;
}
diff --git a/modules/audio_device/linux/audio_device_alsa_linux.cc b/modules/audio_device/linux/audio_device_alsa_linux.cc
index 0a98efd..0cc484f 100644
--- a/modules/audio_device/linux/audio_device_alsa_linux.cc
+++ b/modules/audio_device/linux/audio_device_alsa_linux.cc
@@ -26,119 +26,115 @@
// Redefine these here to be able to do late-binding
#undef snd_ctl_card_info_alloca
-#define snd_ctl_card_info_alloca(ptr) \
- do { *ptr = (snd_ctl_card_info_t *) \
- __builtin_alloca (LATE(snd_ctl_card_info_sizeof)()); \
- memset(*ptr, 0, LATE(snd_ctl_card_info_sizeof)()); } while (0)
+#define snd_ctl_card_info_alloca(ptr) \
+ do { \
+ *ptr = (snd_ctl_card_info_t*)__builtin_alloca( \
+ LATE(snd_ctl_card_info_sizeof)()); \
+ memset(*ptr, 0, LATE(snd_ctl_card_info_sizeof)()); \
+ } while (0)
#undef snd_pcm_info_alloca
-#define snd_pcm_info_alloca(pInfo) \
- do { *pInfo = (snd_pcm_info_t *) \
- __builtin_alloca (LATE(snd_pcm_info_sizeof)()); \
- memset(*pInfo, 0, LATE(snd_pcm_info_sizeof)()); } while (0)
+#define snd_pcm_info_alloca(pInfo) \
+ do { \
+ *pInfo = (snd_pcm_info_t*)__builtin_alloca(LATE(snd_pcm_info_sizeof)()); \
+ memset(*pInfo, 0, LATE(snd_pcm_info_sizeof)()); \
+ } while (0)
// snd_lib_error_handler_t
-void WebrtcAlsaErrorHandler(const char *file,
- int line,
- const char *function,
- int err,
- const char *fmt,...){};
+void WebrtcAlsaErrorHandler(const char* file,
+ int line,
+ const char* function,
+ int err,
+ const char* fmt,
+ ...){};
-namespace webrtc
-{
+namespace webrtc {
static const unsigned int ALSA_PLAYOUT_FREQ = 48000;
static const unsigned int ALSA_PLAYOUT_CH = 2;
-static const unsigned int ALSA_PLAYOUT_LATENCY = 40*1000; // in us
+static const unsigned int ALSA_PLAYOUT_LATENCY = 40 * 1000; // in us
static const unsigned int ALSA_CAPTURE_FREQ = 48000;
static const unsigned int ALSA_CAPTURE_CH = 2;
-static const unsigned int ALSA_CAPTURE_LATENCY = 40*1000; // in us
-static const unsigned int ALSA_CAPTURE_WAIT_TIMEOUT = 5; // in ms
+static const unsigned int ALSA_CAPTURE_LATENCY = 40 * 1000; // in us
+static const unsigned int ALSA_CAPTURE_WAIT_TIMEOUT = 5; // in ms
#define FUNC_GET_NUM_OF_DEVICE 0
#define FUNC_GET_DEVICE_NAME 1
#define FUNC_GET_DEVICE_NAME_FOR_AN_ENUM 2
-AudioDeviceLinuxALSA::AudioDeviceLinuxALSA() :
- _ptrAudioBuffer(NULL),
- _inputDeviceIndex(0),
- _outputDeviceIndex(0),
- _inputDeviceIsSpecified(false),
- _outputDeviceIsSpecified(false),
- _handleRecord(NULL),
- _handlePlayout(NULL),
- _recordingBuffersizeInFrame(0),
- _recordingPeriodSizeInFrame(0),
- _playoutBufferSizeInFrame(0),
- _playoutPeriodSizeInFrame(0),
- _recordingBufferSizeIn10MS(0),
- _playoutBufferSizeIn10MS(0),
- _recordingFramesIn10MS(0),
- _playoutFramesIn10MS(0),
- _recordingFreq(ALSA_CAPTURE_FREQ),
- _playoutFreq(ALSA_PLAYOUT_FREQ),
- _recChannels(ALSA_CAPTURE_CH),
- _playChannels(ALSA_PLAYOUT_CH),
- _recordingBuffer(NULL),
- _playoutBuffer(NULL),
- _recordingFramesLeft(0),
- _playoutFramesLeft(0),
- _initialized(false),
- _recording(false),
- _playing(false),
- _recIsInitialized(false),
- _playIsInitialized(false),
- _AGC(false),
- _recordingDelay(0),
- _playoutDelay(0)
-{
- memset(_oldKeyState, 0, sizeof(_oldKeyState));
- LOG(LS_INFO) << __FUNCTION__ << " created";
+AudioDeviceLinuxALSA::AudioDeviceLinuxALSA()
+ : _ptrAudioBuffer(NULL),
+ _inputDeviceIndex(0),
+ _outputDeviceIndex(0),
+ _inputDeviceIsSpecified(false),
+ _outputDeviceIsSpecified(false),
+ _handleRecord(NULL),
+ _handlePlayout(NULL),
+ _recordingBuffersizeInFrame(0),
+ _recordingPeriodSizeInFrame(0),
+ _playoutBufferSizeInFrame(0),
+ _playoutPeriodSizeInFrame(0),
+ _recordingBufferSizeIn10MS(0),
+ _playoutBufferSizeIn10MS(0),
+ _recordingFramesIn10MS(0),
+ _playoutFramesIn10MS(0),
+ _recordingFreq(ALSA_CAPTURE_FREQ),
+ _playoutFreq(ALSA_PLAYOUT_FREQ),
+ _recChannels(ALSA_CAPTURE_CH),
+ _playChannels(ALSA_PLAYOUT_CH),
+ _recordingBuffer(NULL),
+ _playoutBuffer(NULL),
+ _recordingFramesLeft(0),
+ _playoutFramesLeft(0),
+ _initialized(false),
+ _recording(false),
+ _playing(false),
+ _recIsInitialized(false),
+ _playIsInitialized(false),
+ _AGC(false),
+ _recordingDelay(0),
+ _playoutDelay(0) {
+ memset(_oldKeyState, 0, sizeof(_oldKeyState));
+ LOG(LS_INFO) << __FUNCTION__ << " created";
}
// ----------------------------------------------------------------------------
// AudioDeviceLinuxALSA - dtor
// ----------------------------------------------------------------------------
-AudioDeviceLinuxALSA::~AudioDeviceLinuxALSA()
-{
- LOG(LS_INFO) << __FUNCTION__ << " destroyed";
+AudioDeviceLinuxALSA::~AudioDeviceLinuxALSA() {
+ LOG(LS_INFO) << __FUNCTION__ << " destroyed";
- Terminate();
+ Terminate();
- // Clean up the recording buffer and playout buffer.
- if (_recordingBuffer)
- {
- delete [] _recordingBuffer;
- _recordingBuffer = NULL;
- }
- if (_playoutBuffer)
- {
- delete [] _playoutBuffer;
- _playoutBuffer = NULL;
- }
+ // Clean up the recording buffer and playout buffer.
+ if (_recordingBuffer) {
+ delete[] _recordingBuffer;
+ _recordingBuffer = NULL;
+ }
+ if (_playoutBuffer) {
+ delete[] _playoutBuffer;
+ _playoutBuffer = NULL;
+ }
}
-void AudioDeviceLinuxALSA::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer)
-{
+void AudioDeviceLinuxALSA::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
+ rtc::CritScope lock(&_critSect);
- rtc::CritScope lock(&_critSect);
+ _ptrAudioBuffer = audioBuffer;
- _ptrAudioBuffer = audioBuffer;
-
- // Inform the AudioBuffer about default settings for this implementation.
- // Set all values to zero here since the actual settings will be done by
- // InitPlayout and InitRecording later.
- _ptrAudioBuffer->SetRecordingSampleRate(0);
- _ptrAudioBuffer->SetPlayoutSampleRate(0);
- _ptrAudioBuffer->SetRecordingChannels(0);
- _ptrAudioBuffer->SetPlayoutChannels(0);
+ // Inform the AudioBuffer about default settings for this implementation.
+ // Set all values to zero here since the actual settings will be done by
+ // InitPlayout and InitRecording later.
+ _ptrAudioBuffer->SetRecordingSampleRate(0);
+ _ptrAudioBuffer->SetPlayoutSampleRate(0);
+ _ptrAudioBuffer->SetRecordingChannels(0);
+ _ptrAudioBuffer->SetPlayoutChannels(0);
}
int32_t AudioDeviceLinuxALSA::ActiveAudioLayer(
- AudioDeviceModule::AudioLayer& audioLayer) const
-{
- audioLayer = AudioDeviceModule::kLinuxAlsaAudio;
- return 0;
+ AudioDeviceModule::AudioLayer& audioLayer) const {
+ audioLayer = AudioDeviceModule::kLinuxAlsaAudio;
+ return 0;
}
AudioDeviceGeneric::InitStatus AudioDeviceLinuxALSA::Init() {
@@ -155,654 +151,541 @@
return InitStatus::OK;
}
#if defined(USE_X11)
- //Get X display handle for typing detection
- _XDisplay = XOpenDisplay(NULL);
- if (!_XDisplay) {
- LOG(LS_WARNING)
- << "failed to open X display, typing detection will not work";
- }
+ // Get X display handle for typing detection
+ _XDisplay = XOpenDisplay(NULL);
+ if (!_XDisplay) {
+ LOG(LS_WARNING)
+ << "failed to open X display, typing detection will not work";
+ }
#endif
- _initialized = true;
+ _initialized = true;
- return InitStatus::OK;
+ return InitStatus::OK;
}
-int32_t AudioDeviceLinuxALSA::Terminate()
-{
- if (!_initialized)
- {
- return 0;
- }
+int32_t AudioDeviceLinuxALSA::Terminate() {
+ if (!_initialized) {
+ return 0;
+ }
- rtc::CritScope lock(&_critSect);
+ rtc::CritScope lock(&_critSect);
- _mixerManager.Close();
+ _mixerManager.Close();
- // RECORDING
- if (_ptrThreadRec)
- {
- rtc::PlatformThread* tmpThread = _ptrThreadRec.release();
- _critSect.Leave();
+ // RECORDING
+ if (_ptrThreadRec) {
+ rtc::PlatformThread* tmpThread = _ptrThreadRec.release();
+ _critSect.Leave();
- tmpThread->Stop();
- delete tmpThread;
+ tmpThread->Stop();
+ delete tmpThread;
- _critSect.Enter();
- }
+ _critSect.Enter();
+ }
- // PLAYOUT
- if (_ptrThreadPlay)
- {
- rtc::PlatformThread* tmpThread = _ptrThreadPlay.release();
- _critSect.Leave();
+ // PLAYOUT
+ if (_ptrThreadPlay) {
+ rtc::PlatformThread* tmpThread = _ptrThreadPlay.release();
+ _critSect.Leave();
- tmpThread->Stop();
- delete tmpThread;
+ tmpThread->Stop();
+ delete tmpThread;
- _critSect.Enter();
- }
+ _critSect.Enter();
+ }
#if defined(USE_X11)
- if (_XDisplay)
- {
- XCloseDisplay(_XDisplay);
- _XDisplay = NULL;
- }
+ if (_XDisplay) {
+ XCloseDisplay(_XDisplay);
+ _XDisplay = NULL;
+ }
#endif
- _initialized = false;
- _outputDeviceIsSpecified = false;
- _inputDeviceIsSpecified = false;
+ _initialized = false;
+ _outputDeviceIsSpecified = false;
+ _inputDeviceIsSpecified = false;
+ return 0;
+}
+
+bool AudioDeviceLinuxALSA::Initialized() const {
+ return (_initialized);
+}
+
+int32_t AudioDeviceLinuxALSA::InitSpeaker() {
+ rtc::CritScope lock(&_critSect);
+
+ if (_playing) {
+ return -1;
+ }
+
+ char devName[kAdmMaxDeviceNameSize] = {0};
+ GetDevicesInfo(2, true, _outputDeviceIndex, devName, kAdmMaxDeviceNameSize);
+ return _mixerManager.OpenSpeaker(devName);
+}
+
+int32_t AudioDeviceLinuxALSA::InitMicrophone() {
+ rtc::CritScope lock(&_critSect);
+
+ if (_recording) {
+ return -1;
+ }
+
+ char devName[kAdmMaxDeviceNameSize] = {0};
+ GetDevicesInfo(2, false, _inputDeviceIndex, devName, kAdmMaxDeviceNameSize);
+ return _mixerManager.OpenMicrophone(devName);
+}
+
+bool AudioDeviceLinuxALSA::SpeakerIsInitialized() const {
+ return (_mixerManager.SpeakerIsInitialized());
+}
+
+bool AudioDeviceLinuxALSA::MicrophoneIsInitialized() const {
+ return (_mixerManager.MicrophoneIsInitialized());
+}
+
+int32_t AudioDeviceLinuxALSA::SpeakerVolumeIsAvailable(bool& available) {
+ bool wasInitialized = _mixerManager.SpeakerIsInitialized();
+
+ // Make an attempt to open up the
+ // output mixer corresponding to the currently selected output device.
+ if (!wasInitialized && InitSpeaker() == -1) {
+ // If we end up here it means that the selected speaker has no volume
+ // control.
+ available = false;
return 0;
+ }
+
+ // Given that InitSpeaker was successful, we know that a volume control
+ // exists
+ available = true;
+
+ // Close the initialized output mixer
+ if (!wasInitialized) {
+ _mixerManager.CloseSpeaker();
+ }
+
+ return 0;
}
-bool AudioDeviceLinuxALSA::Initialized() const
-{
- return (_initialized);
+int32_t AudioDeviceLinuxALSA::SetSpeakerVolume(uint32_t volume) {
+ return (_mixerManager.SetSpeakerVolume(volume));
}
-int32_t AudioDeviceLinuxALSA::InitSpeaker()
-{
+int32_t AudioDeviceLinuxALSA::SpeakerVolume(uint32_t& volume) const {
+ uint32_t level(0);
- rtc::CritScope lock(&_critSect);
+ if (_mixerManager.SpeakerVolume(level) == -1) {
+ return -1;
+ }
- if (_playing)
- {
- return -1;
- }
+ volume = level;
- char devName[kAdmMaxDeviceNameSize] = {0};
- GetDevicesInfo(2, true, _outputDeviceIndex, devName, kAdmMaxDeviceNameSize);
- return _mixerManager.OpenSpeaker(devName);
+ return 0;
}
-int32_t AudioDeviceLinuxALSA::InitMicrophone()
-{
+int32_t AudioDeviceLinuxALSA::MaxSpeakerVolume(uint32_t& maxVolume) const {
+ uint32_t maxVol(0);
- rtc::CritScope lock(&_critSect);
+ if (_mixerManager.MaxSpeakerVolume(maxVol) == -1) {
+ return -1;
+ }
- if (_recording)
- {
- return -1;
- }
+ maxVolume = maxVol;
- char devName[kAdmMaxDeviceNameSize] = {0};
- GetDevicesInfo(2, false, _inputDeviceIndex, devName, kAdmMaxDeviceNameSize);
- return _mixerManager.OpenMicrophone(devName);
+ return 0;
}
-bool AudioDeviceLinuxALSA::SpeakerIsInitialized() const
-{
- return (_mixerManager.SpeakerIsInitialized());
+int32_t AudioDeviceLinuxALSA::MinSpeakerVolume(uint32_t& minVolume) const {
+ uint32_t minVol(0);
+
+ if (_mixerManager.MinSpeakerVolume(minVol) == -1) {
+ return -1;
+ }
+
+ minVolume = minVol;
+
+ return 0;
}
-bool AudioDeviceLinuxALSA::MicrophoneIsInitialized() const
-{
- return (_mixerManager.MicrophoneIsInitialized());
-}
+int32_t AudioDeviceLinuxALSA::SpeakerMuteIsAvailable(bool& available) {
+ bool isAvailable(false);
+ bool wasInitialized = _mixerManager.SpeakerIsInitialized();
-int32_t AudioDeviceLinuxALSA::SpeakerVolumeIsAvailable(bool& available)
-{
-
- bool wasInitialized = _mixerManager.SpeakerIsInitialized();
-
- // Make an attempt to open up the
- // output mixer corresponding to the currently selected output device.
- if (!wasInitialized && InitSpeaker() == -1)
- {
- // If we end up here it means that the selected speaker has no volume
- // control.
- available = false;
- return 0;
- }
-
- // Given that InitSpeaker was successful, we know that a volume control
- // exists
- available = true;
-
- // Close the initialized output mixer
- if (!wasInitialized)
- {
- _mixerManager.CloseSpeaker();
- }
-
+ // Make an attempt to open up the
+ // output mixer corresponding to the currently selected output device.
+ //
+ if (!wasInitialized && InitSpeaker() == -1) {
+ // If we end up here it means that the selected speaker has no volume
+ // control, hence it is safe to state that there is no mute control
+ // already at this stage.
+ available = false;
return 0;
+ }
+
+ // Check if the selected speaker has a mute control
+ _mixerManager.SpeakerMuteIsAvailable(isAvailable);
+
+ available = isAvailable;
+
+ // Close the initialized output mixer
+ if (!wasInitialized) {
+ _mixerManager.CloseSpeaker();
+ }
+
+ return 0;
}
-int32_t AudioDeviceLinuxALSA::SetSpeakerVolume(uint32_t volume)
-{
-
- return (_mixerManager.SetSpeakerVolume(volume));
+int32_t AudioDeviceLinuxALSA::SetSpeakerMute(bool enable) {
+ return (_mixerManager.SetSpeakerMute(enable));
}
-int32_t AudioDeviceLinuxALSA::SpeakerVolume(uint32_t& volume) const
-{
+int32_t AudioDeviceLinuxALSA::SpeakerMute(bool& enabled) const {
+ bool muted(0);
- uint32_t level(0);
+ if (_mixerManager.SpeakerMute(muted) == -1) {
+ return -1;
+ }
- if (_mixerManager.SpeakerVolume(level) == -1)
- {
- return -1;
- }
+ enabled = muted;
- volume = level;
+ return 0;
+}
+int32_t AudioDeviceLinuxALSA::MicrophoneMuteIsAvailable(bool& available) {
+ bool isAvailable(false);
+ bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
+
+ // Make an attempt to open up the
+ // input mixer corresponding to the currently selected input device.
+ //
+ if (!wasInitialized && InitMicrophone() == -1) {
+ // If we end up here it means that the selected microphone has no volume
+ // control, hence it is safe to state that there is no mute control
+ // already at this stage.
+ available = false;
return 0;
+ }
+
+ // Check if the selected microphone has a mute control
+ //
+ _mixerManager.MicrophoneMuteIsAvailable(isAvailable);
+ available = isAvailable;
+
+ // Close the initialized input mixer
+ //
+ if (!wasInitialized) {
+ _mixerManager.CloseMicrophone();
+ }
+
+ return 0;
}
-int32_t AudioDeviceLinuxALSA::MaxSpeakerVolume(
- uint32_t& maxVolume) const
-{
-
- uint32_t maxVol(0);
-
- if (_mixerManager.MaxSpeakerVolume(maxVol) == -1)
- {
- return -1;
- }
-
- maxVolume = maxVol;
-
- return 0;
-}
-
-int32_t AudioDeviceLinuxALSA::MinSpeakerVolume(
- uint32_t& minVolume) const
-{
-
- uint32_t minVol(0);
-
- if (_mixerManager.MinSpeakerVolume(minVol) == -1)
- {
- return -1;
- }
-
- minVolume = minVol;
-
- return 0;
-}
-
-int32_t AudioDeviceLinuxALSA::SpeakerMuteIsAvailable(bool& available)
-{
-
- bool isAvailable(false);
- bool wasInitialized = _mixerManager.SpeakerIsInitialized();
-
- // Make an attempt to open up the
- // output mixer corresponding to the currently selected output device.
- //
- if (!wasInitialized && InitSpeaker() == -1)
- {
- // If we end up here it means that the selected speaker has no volume
- // control, hence it is safe to state that there is no mute control
- // already at this stage.
- available = false;
- return 0;
- }
-
- // Check if the selected speaker has a mute control
- _mixerManager.SpeakerMuteIsAvailable(isAvailable);
-
- available = isAvailable;
-
- // Close the initialized output mixer
- if (!wasInitialized)
- {
- _mixerManager.CloseSpeaker();
- }
-
- return 0;
-}
-
-int32_t AudioDeviceLinuxALSA::SetSpeakerMute(bool enable)
-{
- return (_mixerManager.SetSpeakerMute(enable));
-}
-
-int32_t AudioDeviceLinuxALSA::SpeakerMute(bool& enabled) const
-{
-
- bool muted(0);
-
- if (_mixerManager.SpeakerMute(muted) == -1)
- {
- return -1;
- }
-
- enabled = muted;
-
- return 0;
-}
-
-int32_t AudioDeviceLinuxALSA::MicrophoneMuteIsAvailable(bool& available)
-{
-
- bool isAvailable(false);
- bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
-
- // Make an attempt to open up the
- // input mixer corresponding to the currently selected input device.
- //
- if (!wasInitialized && InitMicrophone() == -1)
- {
- // If we end up here it means that the selected microphone has no volume
- // control, hence it is safe to state that there is no mute control
- // already at this stage.
- available = false;
- return 0;
- }
-
- // Check if the selected microphone has a mute control
- //
- _mixerManager.MicrophoneMuteIsAvailable(isAvailable);
- available = isAvailable;
-
- // Close the initialized input mixer
- //
- if (!wasInitialized)
- {
- _mixerManager.CloseMicrophone();
- }
-
- return 0;
-}
-
-int32_t AudioDeviceLinuxALSA::SetMicrophoneMute(bool enable)
-{
- return (_mixerManager.SetMicrophoneMute(enable));
+int32_t AudioDeviceLinuxALSA::SetMicrophoneMute(bool enable) {
+ return (_mixerManager.SetMicrophoneMute(enable));
}
// ----------------------------------------------------------------------------
// MicrophoneMute
// ----------------------------------------------------------------------------
-int32_t AudioDeviceLinuxALSA::MicrophoneMute(bool& enabled) const
-{
+int32_t AudioDeviceLinuxALSA::MicrophoneMute(bool& enabled) const {
+ bool muted(0);
- bool muted(0);
+ if (_mixerManager.MicrophoneMute(muted) == -1) {
+ return -1;
+ }
- if (_mixerManager.MicrophoneMute(muted) == -1)
- {
- return -1;
- }
-
- enabled = muted;
- return 0;
+ enabled = muted;
+ return 0;
}
-int32_t AudioDeviceLinuxALSA::StereoRecordingIsAvailable(bool& available)
-{
+int32_t AudioDeviceLinuxALSA::StereoRecordingIsAvailable(bool& available) {
+ rtc::CritScope lock(&_critSect);
- rtc::CritScope lock(&_critSect);
-
- // If we already have initialized in stereo it's obviously available
- if (_recIsInitialized && (2 == _recChannels))
- {
- available = true;
- return 0;
- }
-
- // Save rec states and the number of rec channels
- bool recIsInitialized = _recIsInitialized;
- bool recording = _recording;
- int recChannels = _recChannels;
-
- available = false;
-
- // Stop/uninitialize recording if initialized (and possibly started)
- if (_recIsInitialized)
- {
- StopRecording();
- }
-
- // Try init in stereo;
- _recChannels = 2;
- if (InitRecording() == 0)
- {
- available = true;
- }
-
- // Stop/uninitialize recording
- StopRecording();
-
- // Recover previous states
- _recChannels = recChannels;
- if (recIsInitialized)
- {
- InitRecording();
- }
- if (recording)
- {
- StartRecording();
- }
-
- return 0;
-}
-
-int32_t AudioDeviceLinuxALSA::SetStereoRecording(bool enable)
-{
-
- if (enable)
- _recChannels = 2;
- else
- _recChannels = 1;
-
- return 0;
-}
-
-int32_t AudioDeviceLinuxALSA::StereoRecording(bool& enabled) const
-{
-
- if (_recChannels == 2)
- enabled = true;
- else
- enabled = false;
-
- return 0;
-}
-
-int32_t AudioDeviceLinuxALSA::StereoPlayoutIsAvailable(bool& available)
-{
-
- rtc::CritScope lock(&_critSect);
-
- // If we already have initialized in stereo it's obviously available
- if (_playIsInitialized && (2 == _playChannels))
- {
- available = true;
- return 0;
- }
-
- // Save rec states and the number of rec channels
- bool playIsInitialized = _playIsInitialized;
- bool playing = _playing;
- int playChannels = _playChannels;
-
- available = false;
-
- // Stop/uninitialize recording if initialized (and possibly started)
- if (_playIsInitialized)
- {
- StopPlayout();
- }
-
- // Try init in stereo;
- _playChannels = 2;
- if (InitPlayout() == 0)
- {
- available = true;
- }
-
- // Stop/uninitialize recording
- StopPlayout();
-
- // Recover previous states
- _playChannels = playChannels;
- if (playIsInitialized)
- {
- InitPlayout();
- }
- if (playing)
- {
- StartPlayout();
- }
-
- return 0;
-}
-
-int32_t AudioDeviceLinuxALSA::SetStereoPlayout(bool enable)
-{
-
- if (enable)
- _playChannels = 2;
- else
- _playChannels = 1;
-
- return 0;
-}
-
-int32_t AudioDeviceLinuxALSA::StereoPlayout(bool& enabled) const
-{
-
- if (_playChannels == 2)
- enabled = true;
- else
- enabled = false;
-
- return 0;
-}
-
-int32_t AudioDeviceLinuxALSA::SetAGC(bool enable)
-{
-
- _AGC = enable;
-
- return 0;
-}
-
-bool AudioDeviceLinuxALSA::AGC() const
-{
-
- return _AGC;
-}
-
-int32_t AudioDeviceLinuxALSA::MicrophoneVolumeIsAvailable(bool& available)
-{
-
- bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
-
- // Make an attempt to open up the
- // input mixer corresponding to the currently selected output device.
- if (!wasInitialized && InitMicrophone() == -1)
- {
- // If we end up here it means that the selected microphone has no volume
- // control.
- available = false;
- return 0;
- }
-
- // Given that InitMicrophone was successful, we know that a volume control
- // exists
+ // If we already have initialized in stereo it's obviously available
+ if (_recIsInitialized && (2 == _recChannels)) {
available = true;
-
- // Close the initialized input mixer
- if (!wasInitialized)
- {
- _mixerManager.CloseMicrophone();
- }
-
return 0;
+ }
+
+ // Save rec states and the number of rec channels
+ bool recIsInitialized = _recIsInitialized;
+ bool recording = _recording;
+ int recChannels = _recChannels;
+
+ available = false;
+
+ // Stop/uninitialize recording if initialized (and possibly started)
+ if (_recIsInitialized) {
+ StopRecording();
+ }
+
+ // Try init in stereo;
+ _recChannels = 2;
+ if (InitRecording() == 0) {
+ available = true;
+ }
+
+ // Stop/uninitialize recording
+ StopRecording();
+
+ // Recover previous states
+ _recChannels = recChannels;
+ if (recIsInitialized) {
+ InitRecording();
+ }
+ if (recording) {
+ StartRecording();
+ }
+
+ return 0;
}
-int32_t AudioDeviceLinuxALSA::SetMicrophoneVolume(uint32_t volume)
-{
+int32_t AudioDeviceLinuxALSA::SetStereoRecording(bool enable) {
+ if (enable)
+ _recChannels = 2;
+ else
+ _recChannels = 1;
- return (_mixerManager.SetMicrophoneVolume(volume));
-
- return 0;
+ return 0;
}
-int32_t AudioDeviceLinuxALSA::MicrophoneVolume(uint32_t& volume) const
-{
+int32_t AudioDeviceLinuxALSA::StereoRecording(bool& enabled) const {
+ if (_recChannels == 2)
+ enabled = true;
+ else
+ enabled = false;
- uint32_t level(0);
-
- if (_mixerManager.MicrophoneVolume(level) == -1)
- {
- LOG(LS_WARNING) << "failed to retrive current microphone level";
- return -1;
- }
-
- volume = level;
-
- return 0;
+ return 0;
}
-int32_t AudioDeviceLinuxALSA::MaxMicrophoneVolume(
- uint32_t& maxVolume) const
-{
+int32_t AudioDeviceLinuxALSA::StereoPlayoutIsAvailable(bool& available) {
+ rtc::CritScope lock(&_critSect);
- uint32_t maxVol(0);
-
- if (_mixerManager.MaxMicrophoneVolume(maxVol) == -1)
- {
- return -1;
- }
-
- maxVolume = maxVol;
-
+ // If we already have initialized in stereo it's obviously available
+ if (_playIsInitialized && (2 == _playChannels)) {
+ available = true;
return 0;
+ }
+
+ // Save rec states and the number of rec channels
+ bool playIsInitialized = _playIsInitialized;
+ bool playing = _playing;
+ int playChannels = _playChannels;
+
+ available = false;
+
+ // Stop/uninitialize recording if initialized (and possibly started)
+ if (_playIsInitialized) {
+ StopPlayout();
+ }
+
+ // Try init in stereo;
+ _playChannels = 2;
+ if (InitPlayout() == 0) {
+ available = true;
+ }
+
+ // Stop/uninitialize recording
+ StopPlayout();
+
+ // Recover previous states
+ _playChannels = playChannels;
+ if (playIsInitialized) {
+ InitPlayout();
+ }
+ if (playing) {
+ StartPlayout();
+ }
+
+ return 0;
}
-int32_t AudioDeviceLinuxALSA::MinMicrophoneVolume(
- uint32_t& minVolume) const
-{
+int32_t AudioDeviceLinuxALSA::SetStereoPlayout(bool enable) {
+ if (enable)
+ _playChannels = 2;
+ else
+ _playChannels = 1;
- uint32_t minVol(0);
-
- if (_mixerManager.MinMicrophoneVolume(minVol) == -1)
- {
- return -1;
- }
-
- minVolume = minVol;
-
- return 0;
+ return 0;
}
-int16_t AudioDeviceLinuxALSA::PlayoutDevices()
-{
+int32_t AudioDeviceLinuxALSA::StereoPlayout(bool& enabled) const {
+ if (_playChannels == 2)
+ enabled = true;
+ else
+ enabled = false;
- return (int16_t)GetDevicesInfo(0, true);
+ return 0;
}
-int32_t AudioDeviceLinuxALSA::SetPlayoutDevice(uint16_t index)
-{
+int32_t AudioDeviceLinuxALSA::SetAGC(bool enable) {
+ _AGC = enable;
- if (_playIsInitialized)
- {
- return -1;
- }
+ return 0;
+}
- uint32_t nDevices = GetDevicesInfo(0, true);
- LOG(LS_VERBOSE) << "number of available audio output devices is "
- << nDevices;
+bool AudioDeviceLinuxALSA::AGC() const {
+ return _AGC;
+}
- if (index > (nDevices-1))
- {
- LOG(LS_ERROR) << "device index is out of range [0," << (nDevices-1)
- << "]";
- return -1;
- }
+int32_t AudioDeviceLinuxALSA::MicrophoneVolumeIsAvailable(bool& available) {
+ bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
- _outputDeviceIndex = index;
- _outputDeviceIsSpecified = true;
-
+ // Make an attempt to open up the
+ // input mixer corresponding to the currently selected output device.
+ if (!wasInitialized && InitMicrophone() == -1) {
+ // If we end up here it means that the selected microphone has no volume
+ // control.
+ available = false;
return 0;
+ }
+
+ // Given that InitMicrophone was successful, we know that a volume control
+ // exists
+ available = true;
+
+ // Close the initialized input mixer
+ if (!wasInitialized) {
+ _mixerManager.CloseMicrophone();
+ }
+
+ return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::SetMicrophoneVolume(uint32_t volume) {
+ return (_mixerManager.SetMicrophoneVolume(volume));
+
+ return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::MicrophoneVolume(uint32_t& volume) const {
+ uint32_t level(0);
+
+ if (_mixerManager.MicrophoneVolume(level) == -1) {
+ LOG(LS_WARNING) << "failed to retrive current microphone level";
+ return -1;
+ }
+
+ volume = level;
+
+ return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::MaxMicrophoneVolume(uint32_t& maxVolume) const {
+ uint32_t maxVol(0);
+
+ if (_mixerManager.MaxMicrophoneVolume(maxVol) == -1) {
+ return -1;
+ }
+
+ maxVolume = maxVol;
+
+ return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::MinMicrophoneVolume(uint32_t& minVolume) const {
+ uint32_t minVol(0);
+
+ if (_mixerManager.MinMicrophoneVolume(minVol) == -1) {
+ return -1;
+ }
+
+ minVolume = minVol;
+
+ return 0;
+}
+
+int16_t AudioDeviceLinuxALSA::PlayoutDevices() {
+ return (int16_t)GetDevicesInfo(0, true);
+}
+
+int32_t AudioDeviceLinuxALSA::SetPlayoutDevice(uint16_t index) {
+ if (_playIsInitialized) {
+ return -1;
+ }
+
+ uint32_t nDevices = GetDevicesInfo(0, true);
+ LOG(LS_VERBOSE) << "number of available audio output devices is " << nDevices;
+
+ if (index > (nDevices - 1)) {
+ LOG(LS_ERROR) << "device index is out of range [0," << (nDevices - 1)
+ << "]";
+ return -1;
+ }
+
+ _outputDeviceIndex = index;
+ _outputDeviceIsSpecified = true;
+
+ return 0;
}
int32_t AudioDeviceLinuxALSA::SetPlayoutDevice(
- AudioDeviceModule::WindowsDeviceType /*device*/)
-{
- LOG(LS_ERROR) << "WindowsDeviceType not supported";
- return -1;
+ AudioDeviceModule::WindowsDeviceType /*device*/) {
+ LOG(LS_ERROR) << "WindowsDeviceType not supported";
+ return -1;
}
int32_t AudioDeviceLinuxALSA::PlayoutDeviceName(
uint16_t index,
char name[kAdmMaxDeviceNameSize],
- char guid[kAdmMaxGuidSize])
-{
+ char guid[kAdmMaxGuidSize]) {
+ const uint16_t nDevices(PlayoutDevices());
- const uint16_t nDevices(PlayoutDevices());
+ if ((index > (nDevices - 1)) || (name == NULL)) {
+ return -1;
+ }
- if ((index > (nDevices-1)) || (name == NULL))
- {
- return -1;
- }
+ memset(name, 0, kAdmMaxDeviceNameSize);
- memset(name, 0, kAdmMaxDeviceNameSize);
+ if (guid != NULL) {
+ memset(guid, 0, kAdmMaxGuidSize);
+ }
- if (guid != NULL)
- {
- memset(guid, 0, kAdmMaxGuidSize);
- }
-
- return GetDevicesInfo(1, true, index, name, kAdmMaxDeviceNameSize);
+ return GetDevicesInfo(1, true, index, name, kAdmMaxDeviceNameSize);
}
int32_t AudioDeviceLinuxALSA::RecordingDeviceName(
uint16_t index,
char name[kAdmMaxDeviceNameSize],
- char guid[kAdmMaxGuidSize])
-{
+ char guid[kAdmMaxGuidSize]) {
+ const uint16_t nDevices(RecordingDevices());
- const uint16_t nDevices(RecordingDevices());
+ if ((index > (nDevices - 1)) || (name == NULL)) {
+ return -1;
+ }
- if ((index > (nDevices-1)) || (name == NULL))
- {
- return -1;
- }
+ memset(name, 0, kAdmMaxDeviceNameSize);
- memset(name, 0, kAdmMaxDeviceNameSize);
+ if (guid != NULL) {
+ memset(guid, 0, kAdmMaxGuidSize);
+ }
- if (guid != NULL)
- {
- memset(guid, 0, kAdmMaxGuidSize);
- }
-
- return GetDevicesInfo(1, false, index, name, kAdmMaxDeviceNameSize);
+ return GetDevicesInfo(1, false, index, name, kAdmMaxDeviceNameSize);
}
-int16_t AudioDeviceLinuxALSA::RecordingDevices()
-{
-
- return (int16_t)GetDevicesInfo(0, false);
+int16_t AudioDeviceLinuxALSA::RecordingDevices() {
+ return (int16_t)GetDevicesInfo(0, false);
}
-int32_t AudioDeviceLinuxALSA::SetRecordingDevice(uint16_t index)
-{
+int32_t AudioDeviceLinuxALSA::SetRecordingDevice(uint16_t index) {
+ if (_recIsInitialized) {
+ return -1;
+ }
- if (_recIsInitialized)
- {
- return -1;
- }
+ uint32_t nDevices = GetDevicesInfo(0, false);
+ LOG(LS_VERBOSE) << "number of availiable audio input devices is " << nDevices;
- uint32_t nDevices = GetDevicesInfo(0, false);
- LOG(LS_VERBOSE) << "number of availiable audio input devices is "
- << nDevices;
+ if (index > (nDevices - 1)) {
+ LOG(LS_ERROR) << "device index is out of range [0," << (nDevices - 1)
+ << "]";
+ return -1;
+ }
- if (index > (nDevices-1))
- {
- LOG(LS_ERROR) << "device index is out of range [0," << (nDevices-1)
- << "]";
- return -1;
- }
+ _inputDeviceIndex = index;
+ _inputDeviceIsSpecified = true;
- _inputDeviceIndex = index;
- _inputDeviceIsSpecified = true;
-
- return 0;
+ return 0;
}
// ----------------------------------------------------------------------------
@@ -810,1124 +693,966 @@
// ----------------------------------------------------------------------------
int32_t AudioDeviceLinuxALSA::SetRecordingDevice(
- AudioDeviceModule::WindowsDeviceType /*device*/)
-{
- LOG(LS_ERROR) << "WindowsDeviceType not supported";
+ AudioDeviceModule::WindowsDeviceType /*device*/) {
+ LOG(LS_ERROR) << "WindowsDeviceType not supported";
+ return -1;
+}
+
+int32_t AudioDeviceLinuxALSA::PlayoutIsAvailable(bool& available) {
+ available = false;
+
+ // Try to initialize the playout side with mono
+ // Assumes that user set num channels after calling this function
+ _playChannels = 1;
+ int32_t res = InitPlayout();
+
+ // Cancel effect of initialization
+ StopPlayout();
+
+ if (res != -1) {
+ available = true;
+ } else {
+ // It may be possible to play out in stereo
+ res = StereoPlayoutIsAvailable(available);
+ if (available) {
+ // Then set channels to 2 so InitPlayout doesn't fail
+ _playChannels = 2;
+ }
+ }
+
+ return res;
+}
+
+int32_t AudioDeviceLinuxALSA::RecordingIsAvailable(bool& available) {
+ available = false;
+
+ // Try to initialize the recording side with mono
+ // Assumes that user set num channels after calling this function
+ _recChannels = 1;
+ int32_t res = InitRecording();
+
+ // Cancel effect of initialization
+ StopRecording();
+
+ if (res != -1) {
+ available = true;
+ } else {
+ // It may be possible to record in stereo
+ res = StereoRecordingIsAvailable(available);
+ if (available) {
+ // Then set channels to 2 so InitPlayout doesn't fail
+ _recChannels = 2;
+ }
+ }
+
+ return res;
+}
+
+int32_t AudioDeviceLinuxALSA::InitPlayout() {
+ int errVal = 0;
+
+ rtc::CritScope lock(&_critSect);
+ if (_playing) {
return -1;
-}
+ }
-int32_t AudioDeviceLinuxALSA::PlayoutIsAvailable(bool& available)
-{
+ if (!_outputDeviceIsSpecified) {
+ return -1;
+ }
- available = false;
-
- // Try to initialize the playout side with mono
- // Assumes that user set num channels after calling this function
- _playChannels = 1;
- int32_t res = InitPlayout();
-
- // Cancel effect of initialization
- StopPlayout();
-
- if (res != -1)
- {
- available = true;
- }
- else
- {
- // It may be possible to play out in stereo
- res = StereoPlayoutIsAvailable(available);
- if (available)
- {
- // Then set channels to 2 so InitPlayout doesn't fail
- _playChannels = 2;
- }
- }
-
- return res;
-}
-
-int32_t AudioDeviceLinuxALSA::RecordingIsAvailable(bool& available)
-{
-
- available = false;
-
- // Try to initialize the recording side with mono
- // Assumes that user set num channels after calling this function
- _recChannels = 1;
- int32_t res = InitRecording();
-
- // Cancel effect of initialization
- StopRecording();
-
- if (res != -1)
- {
- available = true;
- }
- else
- {
- // It may be possible to record in stereo
- res = StereoRecordingIsAvailable(available);
- if (available)
- {
- // Then set channels to 2 so InitPlayout doesn't fail
- _recChannels = 2;
- }
- }
-
- return res;
-}
-
-int32_t AudioDeviceLinuxALSA::InitPlayout()
-{
-
- int errVal = 0;
-
- rtc::CritScope lock(&_critSect);
- if (_playing)
- {
- return -1;
- }
-
- if (!_outputDeviceIsSpecified)
- {
- return -1;
- }
-
- if (_playIsInitialized)
- {
- return 0;
- }
- // Initialize the speaker (devices might have been added or removed)
- if (InitSpeaker() == -1)
- {
- LOG(LS_WARNING) << "InitSpeaker() failed";
- }
-
- // Start by closing any existing wave-output devices
- //
- if (_handlePlayout != NULL)
- {
- LATE(snd_pcm_close)(_handlePlayout);
- _handlePlayout = NULL;
- _playIsInitialized = false;
- if (errVal < 0)
- {
- LOG(LS_ERROR)
- << "Error closing current playout sound device, error: "
- << LATE(snd_strerror)(errVal);
- }
- }
-
- // Open PCM device for playout
- char deviceName[kAdmMaxDeviceNameSize] = {0};
- GetDevicesInfo(2, true, _outputDeviceIndex, deviceName,
- kAdmMaxDeviceNameSize);
-
- LOG(LS_VERBOSE) << "InitPlayout open (" << deviceName << ")";
-
- errVal = LATE(snd_pcm_open)
- (&_handlePlayout,
- deviceName,
- SND_PCM_STREAM_PLAYBACK,
- SND_PCM_NONBLOCK);
-
- if (errVal == -EBUSY) // Device busy - try some more!
- {
- for (int i=0; i < 5; i++)
- {
- SleepMs(1000);
- errVal = LATE(snd_pcm_open)
- (&_handlePlayout,
- deviceName,
- SND_PCM_STREAM_PLAYBACK,
- SND_PCM_NONBLOCK);
- if (errVal == 0)
- {
- break;
- }
- }
- }
- if (errVal < 0)
- {
- LOG(LS_ERROR) << "unable to open playback device: "
- << LATE(snd_strerror)(errVal) << " (" << errVal << ")";
- _handlePlayout = NULL;
- return -1;
- }
-
- _playoutFramesIn10MS = _playoutFreq/100;
- if ((errVal = LATE(snd_pcm_set_params)( _handlePlayout,
-#if defined(WEBRTC_ARCH_BIG_ENDIAN)
- SND_PCM_FORMAT_S16_BE,
-#else
- SND_PCM_FORMAT_S16_LE, //format
-#endif
- SND_PCM_ACCESS_RW_INTERLEAVED, //access
- _playChannels, //channels
- _playoutFreq, //rate
- 1, //soft_resample
- ALSA_PLAYOUT_LATENCY //40*1000 //latency required overall latency in us
- )) < 0)
- { /* 0.5sec */
- _playoutFramesIn10MS = 0;
- LOG(LS_ERROR) << "unable to set playback device: "
- << LATE(snd_strerror)(errVal) << " (" << errVal << ")";
- ErrorRecovery(errVal, _handlePlayout);
- errVal = LATE(snd_pcm_close)(_handlePlayout);
- _handlePlayout = NULL;
- return -1;
- }
-
- errVal = LATE(snd_pcm_get_params)(_handlePlayout,
- &_playoutBufferSizeInFrame, &_playoutPeriodSizeInFrame);
- if (errVal < 0)
- {
- LOG(LS_ERROR) << "snd_pcm_get_params: " << LATE(snd_strerror)(errVal)
- << " (" << errVal << ")";
- _playoutBufferSizeInFrame = 0;
- _playoutPeriodSizeInFrame = 0;
- }
- else {
- LOG(LS_VERBOSE) << "playout snd_pcm_get_params buffer_size:"
- << _playoutBufferSizeInFrame << " period_size :"
- << _playoutPeriodSizeInFrame;
- }
-
- if (_ptrAudioBuffer)
- {
- // Update webrtc audio buffer with the selected parameters
- _ptrAudioBuffer->SetPlayoutSampleRate(_playoutFreq);
- _ptrAudioBuffer->SetPlayoutChannels(_playChannels);
- }
-
- // Set play buffer size
- _playoutBufferSizeIn10MS = LATE(snd_pcm_frames_to_bytes)(
- _handlePlayout, _playoutFramesIn10MS);
-
- // Init varaibles used for play
-
- if (_handlePlayout != NULL)
- {
- _playIsInitialized = true;
- return 0;
- }
- else
- {
- return -1;
- }
-
+ if (_playIsInitialized) {
return 0;
-}
+ }
+ // Initialize the speaker (devices might have been added or removed)
+ if (InitSpeaker() == -1) {
+ LOG(LS_WARNING) << "InitSpeaker() failed";
+ }
-int32_t AudioDeviceLinuxALSA::InitRecording()
-{
-
- int errVal = 0;
-
- rtc::CritScope lock(&_critSect);
-
- if (_recording)
- {
- return -1;
+ // Start by closing any existing wave-output devices
+ //
+ if (_handlePlayout != NULL) {
+ LATE(snd_pcm_close)(_handlePlayout);
+ _handlePlayout = NULL;
+ _playIsInitialized = false;
+ if (errVal < 0) {
+ LOG(LS_ERROR) << "Error closing current playout sound device, error: "
+ << LATE(snd_strerror)(errVal);
}
+ }
- if (!_inputDeviceIsSpecified)
- {
- return -1;
- }
+ // Open PCM device for playout
+ char deviceName[kAdmMaxDeviceNameSize] = {0};
+ GetDevicesInfo(2, true, _outputDeviceIndex, deviceName,
+ kAdmMaxDeviceNameSize);
- if (_recIsInitialized)
- {
- return 0;
- }
+ LOG(LS_VERBOSE) << "InitPlayout open (" << deviceName << ")";
- // Initialize the microphone (devices might have been added or removed)
- if (InitMicrophone() == -1)
- {
- LOG(LS_WARNING) << "InitMicrophone() failed";
- }
+ errVal = LATE(snd_pcm_open)(&_handlePlayout, deviceName,
+ SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
- // Start by closing any existing pcm-input devices
- //
- if (_handleRecord != NULL)
- {
- int errVal = LATE(snd_pcm_close)(_handleRecord);
- _handleRecord = NULL;
- _recIsInitialized = false;
- if (errVal < 0)
- {
- LOG(LS_ERROR)
- << "Error closing current recording sound device, error: "
- << LATE(snd_strerror)(errVal);
- }
- }
-
- // Open PCM device for recording
- // The corresponding settings for playout are made after the record settings
- char deviceName[kAdmMaxDeviceNameSize] = {0};
- GetDevicesInfo(2, false, _inputDeviceIndex, deviceName,
- kAdmMaxDeviceNameSize);
-
- LOG(LS_VERBOSE) << "InitRecording open (" << deviceName << ")";
- errVal = LATE(snd_pcm_open)
- (&_handleRecord,
- deviceName,
- SND_PCM_STREAM_CAPTURE,
- SND_PCM_NONBLOCK);
-
- // Available modes: 0 = blocking, SND_PCM_NONBLOCK, SND_PCM_ASYNC
- if (errVal == -EBUSY) // Device busy - try some more!
- {
- for (int i=0; i < 5; i++)
- {
- SleepMs(1000);
- errVal = LATE(snd_pcm_open)
- (&_handleRecord,
- deviceName,
- SND_PCM_STREAM_CAPTURE,
- SND_PCM_NONBLOCK);
- if (errVal == 0)
- {
- break;
- }
- }
- }
- if (errVal < 0)
- {
- LOG(LS_ERROR) << "unable to open record device: "
- << LATE(snd_strerror)(errVal);
- _handleRecord = NULL;
- return -1;
- }
-
- _recordingFramesIn10MS = _recordingFreq/100;
- if ((errVal = LATE(snd_pcm_set_params)(_handleRecord,
-#if defined(WEBRTC_ARCH_BIG_ENDIAN)
- SND_PCM_FORMAT_S16_BE, //format
-#else
- SND_PCM_FORMAT_S16_LE, //format
-#endif
- SND_PCM_ACCESS_RW_INTERLEAVED, //access
- _recChannels, //channels
- _recordingFreq, //rate
- 1, //soft_resample
- ALSA_CAPTURE_LATENCY //latency in us
- )) < 0)
- {
- // Fall back to another mode then.
- if (_recChannels == 1)
- _recChannels = 2;
- else
- _recChannels = 1;
-
- if ((errVal = LATE(snd_pcm_set_params)(_handleRecord,
-#if defined(WEBRTC_ARCH_BIG_ENDIAN)
- SND_PCM_FORMAT_S16_BE, //format
-#else
- SND_PCM_FORMAT_S16_LE, //format
-#endif
- SND_PCM_ACCESS_RW_INTERLEAVED, //access
- _recChannels, //channels
- _recordingFreq, //rate
- 1, //soft_resample
- ALSA_CAPTURE_LATENCY //latency in us
- )) < 0)
- {
- _recordingFramesIn10MS = 0;
- LOG(LS_ERROR) << "unable to set record settings: "
- << LATE(snd_strerror)(errVal) << " (" << errVal
- << ")";
- ErrorRecovery(errVal, _handleRecord);
- errVal = LATE(snd_pcm_close)(_handleRecord);
- _handleRecord = NULL;
- return -1;
- }
- }
-
- errVal = LATE(snd_pcm_get_params)(_handleRecord,
- &_recordingBuffersizeInFrame, &_recordingPeriodSizeInFrame);
- if (errVal < 0)
- {
- LOG(LS_ERROR) << "snd_pcm_get_params " << LATE(snd_strerror)(errVal)
- << " (" << errVal << ")";
- _recordingBuffersizeInFrame = 0;
- _recordingPeriodSizeInFrame = 0;
- }
- else {
- LOG(LS_VERBOSE) << "capture snd_pcm_get_params, buffer_size:"
- << _recordingBuffersizeInFrame << ", period_size:"
- << _recordingPeriodSizeInFrame;
- }
-
- if (_ptrAudioBuffer)
- {
- // Update webrtc audio buffer with the selected parameters
- _ptrAudioBuffer->SetRecordingSampleRate(_recordingFreq);
- _ptrAudioBuffer->SetRecordingChannels(_recChannels);
- }
-
- // Set rec buffer size and create buffer
- _recordingBufferSizeIn10MS = LATE(snd_pcm_frames_to_bytes)(
- _handleRecord, _recordingFramesIn10MS);
-
- if (_handleRecord != NULL)
- {
- // Mark recording side as initialized
- _recIsInitialized = true;
- return 0;
- }
- else
- {
- return -1;
- }
-
- return 0;
-}
-
-int32_t AudioDeviceLinuxALSA::StartRecording()
-{
-
- if (!_recIsInitialized)
- {
- return -1;
- }
-
- if (_recording)
- {
- return 0;
- }
-
- _recording = true;
-
- int errVal = 0;
- _recordingFramesLeft = _recordingFramesIn10MS;
-
- // Make sure we only create the buffer once.
- if (!_recordingBuffer)
- _recordingBuffer = new int8_t[_recordingBufferSizeIn10MS];
- if (!_recordingBuffer)
- {
- LOG(LS_ERROR) << "failed to alloc recording buffer";
- _recording = false;
- return -1;
- }
- // RECORDING
- _ptrThreadRec.reset(new rtc::PlatformThread(
- RecThreadFunc, this, "webrtc_audio_module_capture_thread"));
-
- _ptrThreadRec->Start();
- _ptrThreadRec->SetPriority(rtc::kRealtimePriority);
-
- errVal = LATE(snd_pcm_prepare)(_handleRecord);
- if (errVal < 0)
- {
- LOG(LS_ERROR) << "capture snd_pcm_prepare failed ("
- << LATE(snd_strerror)(errVal) << ")\n";
- // just log error
- // if snd_pcm_open fails will return -1
- }
-
- errVal = LATE(snd_pcm_start)(_handleRecord);
- if (errVal < 0)
- {
- LOG(LS_ERROR) << "capture snd_pcm_start err: "
- << LATE(snd_strerror)(errVal);
- errVal = LATE(snd_pcm_start)(_handleRecord);
- if (errVal < 0)
- {
- LOG(LS_ERROR) << "capture snd_pcm_start 2nd try err: "
- << LATE(snd_strerror)(errVal);
- StopRecording();
- return -1;
- }
- }
-
- return 0;
-}
-
-int32_t AudioDeviceLinuxALSA::StopRecording()
-{
-
- {
- rtc::CritScope lock(&_critSect);
-
- if (!_recIsInitialized)
- {
- return 0;
+ if (errVal == -EBUSY) // Device busy - try some more!
+ {
+ for (int i = 0; i < 5; i++) {
+ SleepMs(1000);
+ errVal = LATE(snd_pcm_open)(&_handlePlayout, deviceName,
+ SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
+ if (errVal == 0) {
+ break;
}
-
- if (_handleRecord == NULL)
- {
- return -1;
- }
-
- // Make sure we don't start recording (it's asynchronous).
- _recIsInitialized = false;
- _recording = false;
}
+ }
+ if (errVal < 0) {
+ LOG(LS_ERROR) << "unable to open playback device: "
+ << LATE(snd_strerror)(errVal) << " (" << errVal << ")";
+ _handlePlayout = NULL;
+ return -1;
+ }
- if (_ptrThreadRec)
- {
- _ptrThreadRec->Stop();
- _ptrThreadRec.reset();
- }
+ _playoutFramesIn10MS = _playoutFreq / 100;
+ if ((errVal = LATE(snd_pcm_set_params)(
+ _handlePlayout,
+#if defined(WEBRTC_ARCH_BIG_ENDIAN)
+ SND_PCM_FORMAT_S16_BE,
+#else
+ SND_PCM_FORMAT_S16_LE, // format
+#endif
+ SND_PCM_ACCESS_RW_INTERLEAVED, // access
+ _playChannels, // channels
+ _playoutFreq, // rate
+ 1, // soft_resample
+ ALSA_PLAYOUT_LATENCY // 40*1000 //latency required overall latency
+ // in us
+ )) < 0) { /* 0.5sec */
+ _playoutFramesIn10MS = 0;
+ LOG(LS_ERROR) << "unable to set playback device: "
+ << LATE(snd_strerror)(errVal) << " (" << errVal << ")";
+ ErrorRecovery(errVal, _handlePlayout);
+ errVal = LATE(snd_pcm_close)(_handlePlayout);
+ _handlePlayout = NULL;
+ return -1;
+ }
- rtc::CritScope lock(&_critSect);
- _recordingFramesLeft = 0;
- if (_recordingBuffer)
- {
- delete [] _recordingBuffer;
- _recordingBuffer = NULL;
- }
+ errVal = LATE(snd_pcm_get_params)(_handlePlayout, &_playoutBufferSizeInFrame,
+ &_playoutPeriodSizeInFrame);
+ if (errVal < 0) {
+ LOG(LS_ERROR) << "snd_pcm_get_params: " << LATE(snd_strerror)(errVal)
+ << " (" << errVal << ")";
+ _playoutBufferSizeInFrame = 0;
+ _playoutPeriodSizeInFrame = 0;
+ } else {
+ LOG(LS_VERBOSE) << "playout snd_pcm_get_params buffer_size:"
+ << _playoutBufferSizeInFrame
+ << " period_size :" << _playoutPeriodSizeInFrame;
+ }
- // Stop and close pcm recording device.
- int errVal = LATE(snd_pcm_drop)(_handleRecord);
- if (errVal < 0)
- {
- LOG(LS_ERROR) << "Error stop recording: " << LATE(snd_strerror)(errVal);
- return -1;
- }
+ if (_ptrAudioBuffer) {
+ // Update webrtc audio buffer with the selected parameters
+ _ptrAudioBuffer->SetPlayoutSampleRate(_playoutFreq);
+ _ptrAudioBuffer->SetPlayoutChannels(_playChannels);
+ }
- errVal = LATE(snd_pcm_close)(_handleRecord);
- if (errVal < 0)
- {
- LOG(LS_ERROR) << "Error closing record sound device, error: "
- << LATE(snd_strerror)(errVal);
- return -1;
- }
+ // Set play buffer size
+ _playoutBufferSizeIn10MS =
+ LATE(snd_pcm_frames_to_bytes)(_handlePlayout, _playoutFramesIn10MS);
- // Check if we have muted and unmute if so.
- bool muteEnabled = false;
- MicrophoneMute(muteEnabled);
- if (muteEnabled)
- {
- SetMicrophoneMute(false);
- }
+ // Init varaibles used for play
- // set the pcm input handle to NULL
+ if (_handlePlayout != NULL) {
+ _playIsInitialized = true;
+ return 0;
+ } else {
+ return -1;
+ }
+
+ return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::InitRecording() {
+ int errVal = 0;
+
+ rtc::CritScope lock(&_critSect);
+
+ if (_recording) {
+ return -1;
+ }
+
+ if (!_inputDeviceIsSpecified) {
+ return -1;
+ }
+
+ if (_recIsInitialized) {
+ return 0;
+ }
+
+ // Initialize the microphone (devices might have been added or removed)
+ if (InitMicrophone() == -1) {
+ LOG(LS_WARNING) << "InitMicrophone() failed";
+ }
+
+ // Start by closing any existing pcm-input devices
+ //
+ if (_handleRecord != NULL) {
+ int errVal = LATE(snd_pcm_close)(_handleRecord);
_handleRecord = NULL;
+ _recIsInitialized = false;
+ if (errVal < 0) {
+ LOG(LS_ERROR) << "Error closing current recording sound device, error: "
+ << LATE(snd_strerror)(errVal);
+ }
+ }
+
+ // Open PCM device for recording
+ // The corresponding settings for playout are made after the record settings
+ char deviceName[kAdmMaxDeviceNameSize] = {0};
+ GetDevicesInfo(2, false, _inputDeviceIndex, deviceName,
+ kAdmMaxDeviceNameSize);
+
+ LOG(LS_VERBOSE) << "InitRecording open (" << deviceName << ")";
+ errVal = LATE(snd_pcm_open)(&_handleRecord, deviceName,
+ SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK);
+
+ // Available modes: 0 = blocking, SND_PCM_NONBLOCK, SND_PCM_ASYNC
+ if (errVal == -EBUSY) // Device busy - try some more!
+ {
+ for (int i = 0; i < 5; i++) {
+ SleepMs(1000);
+ errVal = LATE(snd_pcm_open)(&_handleRecord, deviceName,
+ SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK);
+ if (errVal == 0) {
+ break;
+ }
+ }
+ }
+ if (errVal < 0) {
+ LOG(LS_ERROR) << "unable to open record device: "
+ << LATE(snd_strerror)(errVal);
+ _handleRecord = NULL;
+ return -1;
+ }
+
+ _recordingFramesIn10MS = _recordingFreq / 100;
+ if ((errVal =
+ LATE(snd_pcm_set_params)(_handleRecord,
+#if defined(WEBRTC_ARCH_BIG_ENDIAN)
+ SND_PCM_FORMAT_S16_BE, // format
+#else
+ SND_PCM_FORMAT_S16_LE, // format
+#endif
+ SND_PCM_ACCESS_RW_INTERLEAVED, // access
+ _recChannels, // channels
+ _recordingFreq, // rate
+ 1, // soft_resample
+ ALSA_CAPTURE_LATENCY // latency in us
+ )) < 0) {
+ // Fall back to another mode then.
+ if (_recChannels == 1)
+ _recChannels = 2;
+ else
+ _recChannels = 1;
+
+ if ((errVal =
+ LATE(snd_pcm_set_params)(_handleRecord,
+#if defined(WEBRTC_ARCH_BIG_ENDIAN)
+ SND_PCM_FORMAT_S16_BE, // format
+#else
+ SND_PCM_FORMAT_S16_LE, // format
+#endif
+ SND_PCM_ACCESS_RW_INTERLEAVED, // access
+ _recChannels, // channels
+ _recordingFreq, // rate
+ 1, // soft_resample
+ ALSA_CAPTURE_LATENCY // latency in us
+ )) < 0) {
+ _recordingFramesIn10MS = 0;
+ LOG(LS_ERROR) << "unable to set record settings: "
+ << LATE(snd_strerror)(errVal) << " (" << errVal << ")";
+ ErrorRecovery(errVal, _handleRecord);
+ errVal = LATE(snd_pcm_close)(_handleRecord);
+ _handleRecord = NULL;
+ return -1;
+ }
+ }
+
+ errVal = LATE(snd_pcm_get_params)(_handleRecord, &_recordingBuffersizeInFrame,
+ &_recordingPeriodSizeInFrame);
+ if (errVal < 0) {
+ LOG(LS_ERROR) << "snd_pcm_get_params " << LATE(snd_strerror)(errVal) << " ("
+ << errVal << ")";
+ _recordingBuffersizeInFrame = 0;
+ _recordingPeriodSizeInFrame = 0;
+ } else {
+ LOG(LS_VERBOSE) << "capture snd_pcm_get_params, buffer_size:"
+ << _recordingBuffersizeInFrame
+ << ", period_size:" << _recordingPeriodSizeInFrame;
+ }
+
+ if (_ptrAudioBuffer) {
+ // Update webrtc audio buffer with the selected parameters
+ _ptrAudioBuffer->SetRecordingSampleRate(_recordingFreq);
+ _ptrAudioBuffer->SetRecordingChannels(_recChannels);
+ }
+
+ // Set rec buffer size and create buffer
+ _recordingBufferSizeIn10MS =
+ LATE(snd_pcm_frames_to_bytes)(_handleRecord, _recordingFramesIn10MS);
+
+ if (_handleRecord != NULL) {
+ // Mark recording side as initialized
+ _recIsInitialized = true;
return 0;
+ } else {
+ return -1;
+ }
+
+ return 0;
}
-bool AudioDeviceLinuxALSA::RecordingIsInitialized() const
-{
- return (_recIsInitialized);
+int32_t AudioDeviceLinuxALSA::StartRecording() {
+ if (!_recIsInitialized) {
+ return -1;
+ }
+
+ if (_recording) {
+ return 0;
+ }
+
+ _recording = true;
+
+ int errVal = 0;
+ _recordingFramesLeft = _recordingFramesIn10MS;
+
+ // Make sure we only create the buffer once.
+ if (!_recordingBuffer)
+ _recordingBuffer = new int8_t[_recordingBufferSizeIn10MS];
+ if (!_recordingBuffer) {
+ LOG(LS_ERROR) << "failed to alloc recording buffer";
+ _recording = false;
+ return -1;
+ }
+ // RECORDING
+ _ptrThreadRec.reset(new rtc::PlatformThread(
+ RecThreadFunc, this, "webrtc_audio_module_capture_thread"));
+
+ _ptrThreadRec->Start();
+ _ptrThreadRec->SetPriority(rtc::kRealtimePriority);
+
+ errVal = LATE(snd_pcm_prepare)(_handleRecord);
+ if (errVal < 0) {
+ LOG(LS_ERROR) << "capture snd_pcm_prepare failed ("
+ << LATE(snd_strerror)(errVal) << ")\n";
+ // just log error
+ // if snd_pcm_open fails will return -1
+ }
+
+ errVal = LATE(snd_pcm_start)(_handleRecord);
+ if (errVal < 0) {
+ LOG(LS_ERROR) << "capture snd_pcm_start err: "
+ << LATE(snd_strerror)(errVal);
+ errVal = LATE(snd_pcm_start)(_handleRecord);
+ if (errVal < 0) {
+ LOG(LS_ERROR) << "capture snd_pcm_start 2nd try err: "
+ << LATE(snd_strerror)(errVal);
+ StopRecording();
+ return -1;
+ }
+ }
+
+ return 0;
}
-bool AudioDeviceLinuxALSA::Recording() const
-{
- return (_recording);
-}
+int32_t AudioDeviceLinuxALSA::StopRecording() {
+ {
+ rtc::CritScope lock(&_critSect);
-bool AudioDeviceLinuxALSA::PlayoutIsInitialized() const
-{
- return (_playIsInitialized);
-}
-
-int32_t AudioDeviceLinuxALSA::StartPlayout()
-{
- if (!_playIsInitialized)
- {
- return -1;
+ if (!_recIsInitialized) {
+ return 0;
}
- if (_playing)
- {
- return 0;
- }
-
- _playing = true;
-
- _playoutFramesLeft = 0;
- if (!_playoutBuffer)
- _playoutBuffer = new int8_t[_playoutBufferSizeIn10MS];
- if (!_playoutBuffer)
- {
- LOG(LS_ERROR) << "failed to alloc playout buf";
- _playing = false;
+ if (_handleRecord == NULL) {
return -1;
}
- // PLAYOUT
- _ptrThreadPlay.reset(new rtc::PlatformThread(
- PlayThreadFunc, this, "webrtc_audio_module_play_thread"));
- _ptrThreadPlay->Start();
- _ptrThreadPlay->SetPriority(rtc::kRealtimePriority);
+ // Make sure we don't start recording (it's asynchronous).
+ _recIsInitialized = false;
+ _recording = false;
+ }
- int errVal = LATE(snd_pcm_prepare)(_handlePlayout);
- if (errVal < 0)
- {
- LOG(LS_ERROR) << "playout snd_pcm_prepare failed ("
- << LATE(snd_strerror)(errVal) << ")\n";
- // just log error
- // if snd_pcm_open fails will return -1
- }
+ if (_ptrThreadRec) {
+ _ptrThreadRec->Stop();
+ _ptrThreadRec.reset();
+ }
- return 0;
+ rtc::CritScope lock(&_critSect);
+ _recordingFramesLeft = 0;
+ if (_recordingBuffer) {
+ delete[] _recordingBuffer;
+ _recordingBuffer = NULL;
+ }
+
+ // Stop and close pcm recording device.
+ int errVal = LATE(snd_pcm_drop)(_handleRecord);
+ if (errVal < 0) {
+ LOG(LS_ERROR) << "Error stop recording: " << LATE(snd_strerror)(errVal);
+ return -1;
+ }
+
+ errVal = LATE(snd_pcm_close)(_handleRecord);
+ if (errVal < 0) {
+ LOG(LS_ERROR) << "Error closing record sound device, error: "
+ << LATE(snd_strerror)(errVal);
+ return -1;
+ }
+
+ // Check if we have muted and unmute if so.
+ bool muteEnabled = false;
+ MicrophoneMute(muteEnabled);
+ if (muteEnabled) {
+ SetMicrophoneMute(false);
+ }
+
+ // set the pcm input handle to NULL
+ _handleRecord = NULL;
+ return 0;
}
-int32_t AudioDeviceLinuxALSA::StopPlayout()
-{
+bool AudioDeviceLinuxALSA::RecordingIsInitialized() const {
+ return (_recIsInitialized);
+}
- {
- rtc::CritScope lock(&_critSect);
+bool AudioDeviceLinuxALSA::Recording() const {
+ return (_recording);
+}
- if (!_playIsInitialized)
- {
- return 0;
- }
+bool AudioDeviceLinuxALSA::PlayoutIsInitialized() const {
+ return (_playIsInitialized);
+}
- if (_handlePlayout == NULL)
- {
- return -1;
- }
+int32_t AudioDeviceLinuxALSA::StartPlayout() {
+ if (!_playIsInitialized) {
+ return -1;
+ }
- _playing = false;
- }
+ if (_playing) {
+ return 0;
+ }
- // stop playout thread first
- if (_ptrThreadPlay)
- {
- _ptrThreadPlay->Stop();
- _ptrThreadPlay.reset();
- }
+ _playing = true;
+ _playoutFramesLeft = 0;
+ if (!_playoutBuffer)
+ _playoutBuffer = new int8_t[_playoutBufferSizeIn10MS];
+ if (!_playoutBuffer) {
+ LOG(LS_ERROR) << "failed to alloc playout buf";
+ _playing = false;
+ return -1;
+ }
+
+ // PLAYOUT
+ _ptrThreadPlay.reset(new rtc::PlatformThread(
+ PlayThreadFunc, this, "webrtc_audio_module_play_thread"));
+ _ptrThreadPlay->Start();
+ _ptrThreadPlay->SetPriority(rtc::kRealtimePriority);
+
+ int errVal = LATE(snd_pcm_prepare)(_handlePlayout);
+ if (errVal < 0) {
+ LOG(LS_ERROR) << "playout snd_pcm_prepare failed ("
+ << LATE(snd_strerror)(errVal) << ")\n";
+ // just log error
+ // if snd_pcm_open fails will return -1
+ }
+
+ return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::StopPlayout() {
+ {
rtc::CritScope lock(&_critSect);
- _playoutFramesLeft = 0;
- delete [] _playoutBuffer;
- _playoutBuffer = NULL;
-
- // stop and close pcm playout device
- int errVal = LATE(snd_pcm_drop)(_handlePlayout);
- if (errVal < 0)
- {
- LOG(LS_ERROR) << "Error stop playing: " << LATE(snd_strerror)(errVal);
+ if (!_playIsInitialized) {
+ return 0;
}
- errVal = LATE(snd_pcm_close)(_handlePlayout);
- if (errVal < 0)
- LOG(LS_ERROR) << "Error closing playout sound device, error: "
- << LATE(snd_strerror)(errVal);
+ if (_handlePlayout == NULL) {
+ return -1;
+ }
- // set the pcm input handle to NULL
- _playIsInitialized = false;
- _handlePlayout = NULL;
- LOG(LS_VERBOSE) << "handle_playout is now set to NULL";
+ _playing = false;
+ }
- return 0;
+ // stop playout thread first
+ if (_ptrThreadPlay) {
+ _ptrThreadPlay->Stop();
+ _ptrThreadPlay.reset();
+ }
+
+ rtc::CritScope lock(&_critSect);
+
+ _playoutFramesLeft = 0;
+ delete[] _playoutBuffer;
+ _playoutBuffer = NULL;
+
+ // stop and close pcm playout device
+ int errVal = LATE(snd_pcm_drop)(_handlePlayout);
+ if (errVal < 0) {
+ LOG(LS_ERROR) << "Error stop playing: " << LATE(snd_strerror)(errVal);
+ }
+
+ errVal = LATE(snd_pcm_close)(_handlePlayout);
+ if (errVal < 0)
+ LOG(LS_ERROR) << "Error closing playout sound device, error: "
+ << LATE(snd_strerror)(errVal);
+
+ // set the pcm input handle to NULL
+ _playIsInitialized = false;
+ _handlePlayout = NULL;
+ LOG(LS_VERBOSE) << "handle_playout is now set to NULL";
+
+ return 0;
}
-int32_t AudioDeviceLinuxALSA::PlayoutDelay(uint16_t& delayMS) const
-{
- delayMS = (uint16_t)_playoutDelay * 1000 / _playoutFreq;
- return 0;
+int32_t AudioDeviceLinuxALSA::PlayoutDelay(uint16_t& delayMS) const {
+ delayMS = (uint16_t)_playoutDelay * 1000 / _playoutFreq;
+ return 0;
}
-bool AudioDeviceLinuxALSA::Playing() const
-{
- return (_playing);
+bool AudioDeviceLinuxALSA::Playing() const {
+ return (_playing);
}
// ============================================================================
// Private Methods
// ============================================================================
-int32_t AudioDeviceLinuxALSA::GetDevicesInfo(
- const int32_t function,
- const bool playback,
- const int32_t enumDeviceNo,
- char* enumDeviceName,
- const int32_t ednLen) const
-{
+int32_t AudioDeviceLinuxALSA::GetDevicesInfo(const int32_t function,
+ const bool playback,
+ const int32_t enumDeviceNo,
+ char* enumDeviceName,
+ const int32_t ednLen) const {
+ // Device enumeration based on libjingle implementation
+ // by Tristan Schmelcher at Google Inc.
- // Device enumeration based on libjingle implementation
- // by Tristan Schmelcher at Google Inc.
+ const char* type = playback ? "Output" : "Input";
+ // dmix and dsnoop are only for playback and capture, respectively, but ALSA
+ // stupidly includes them in both lists.
+ const char* ignorePrefix = playback ? "dsnoop:" : "dmix:";
+ // (ALSA lists many more "devices" of questionable interest, but we show them
+ // just in case the weird devices may actually be desirable for some
+ // users/systems.)
- const char *type = playback ? "Output" : "Input";
- // dmix and dsnoop are only for playback and capture, respectively, but ALSA
- // stupidly includes them in both lists.
- const char *ignorePrefix = playback ? "dsnoop:" : "dmix:" ;
- // (ALSA lists many more "devices" of questionable interest, but we show them
- // just in case the weird devices may actually be desirable for some
- // users/systems.)
+ int err;
+ int enumCount(0);
+ bool keepSearching(true);
- int err;
- int enumCount(0);
- bool keepSearching(true);
-
- // From Chromium issue 95797
- // Loop through the sound cards to get Alsa device hints.
- // Don't use snd_device_name_hint(-1,..) since there is a access violation
- // inside this ALSA API with libasound.so.2.0.0.
- int card = -1;
- while (!(LATE(snd_card_next)(&card)) && (card >= 0) && keepSearching) {
- void **hints;
- err = LATE(snd_device_name_hint)(card, "pcm", &hints);
- if (err != 0)
- {
- LOG(LS_ERROR) << "GetDevicesInfo - device name hint error: "
- << LATE(snd_strerror)(err);
- return -1;
- }
-
- enumCount++; // default is 0
- if ((function == FUNC_GET_DEVICE_NAME ||
- function == FUNC_GET_DEVICE_NAME_FOR_AN_ENUM) && enumDeviceNo == 0)
- {
- strcpy(enumDeviceName, "default");
-
- err = LATE(snd_device_name_free_hint)(hints);
- if (err != 0)
- {
- LOG(LS_ERROR)
- << "GetDevicesInfo - device name free hint error: "
+ // From Chromium issue 95797
+ // Loop through the sound cards to get Alsa device hints.
+ // Don't use snd_device_name_hint(-1,..) since there is a access violation
+ // inside this ALSA API with libasound.so.2.0.0.
+ int card = -1;
+ while (!(LATE(snd_card_next)(&card)) && (card >= 0) && keepSearching) {
+ void** hints;
+ err = LATE(snd_device_name_hint)(card, "pcm", &hints);
+ if (err != 0) {
+ LOG(LS_ERROR) << "GetDevicesInfo - device name hint error: "
<< LATE(snd_strerror)(err);
- }
-
- return 0;
- }
-
- for (void **list = hints; *list != NULL; ++list)
- {
- char *actualType = LATE(snd_device_name_get_hint)(*list, "IOID");
- if (actualType)
- { // NULL means it's both.
- bool wrongType = (strcmp(actualType, type) != 0);
- free(actualType);
- if (wrongType)
- {
- // Wrong type of device (i.e., input vs. output).
- continue;
- }
- }
-
- char *name = LATE(snd_device_name_get_hint)(*list, "NAME");
- if (!name)
- {
- LOG(LS_ERROR) << "Device has no name";
- // Skip it.
- continue;
- }
-
- // Now check if we actually want to show this device.
- if (strcmp(name, "default") != 0 &&
- strcmp(name, "null") != 0 &&
- strcmp(name, "pulse") != 0 &&
- strncmp(name, ignorePrefix, strlen(ignorePrefix)) != 0)
- {
- // Yes, we do.
- char *desc = LATE(snd_device_name_get_hint)(*list, "DESC");
- if (!desc)
- {
- // Virtual devices don't necessarily have descriptions.
- // Use their names instead.
- desc = name;
- }
-
- if (FUNC_GET_NUM_OF_DEVICE == function)
- {
- LOG(LS_VERBOSE) << "Enum device " << enumCount << " - "
- << name;
-
- }
- if ((FUNC_GET_DEVICE_NAME == function) &&
- (enumDeviceNo == enumCount))
- {
- // We have found the enum device, copy the name to buffer.
- strncpy(enumDeviceName, desc, ednLen);
- enumDeviceName[ednLen-1] = '\0';
- keepSearching = false;
- // Replace '\n' with '-'.
- char * pret = strchr(enumDeviceName, '\n'/*0xa*/); //LF
- if (pret)
- *pret = '-';
- }
- if ((FUNC_GET_DEVICE_NAME_FOR_AN_ENUM == function) &&
- (enumDeviceNo == enumCount))
- {
- // We have found the enum device, copy the name to buffer.
- strncpy(enumDeviceName, name, ednLen);
- enumDeviceName[ednLen-1] = '\0';
- keepSearching = false;
- }
-
- if (keepSearching)
- ++enumCount;
-
- if (desc != name)
- free(desc);
- }
-
- free(name);
-
- if (!keepSearching)
- break;
- }
-
- err = LATE(snd_device_name_free_hint)(hints);
- if (err != 0)
- {
- LOG(LS_ERROR) << "GetDevicesInfo - device name free hint error: "
- << LATE(snd_strerror)(err);
- // Continue and return true anyway, since we did get the whole list.
- }
+ return -1;
}
- if (FUNC_GET_NUM_OF_DEVICE == function)
- {
- if (enumCount == 1) // only default?
- enumCount = 0;
- return enumCount; // Normal return point for function 0
+ enumCount++; // default is 0
+ if ((function == FUNC_GET_DEVICE_NAME ||
+ function == FUNC_GET_DEVICE_NAME_FOR_AN_ENUM) &&
+ enumDeviceNo == 0) {
+ strcpy(enumDeviceName, "default");
+
+ err = LATE(snd_device_name_free_hint)(hints);
+ if (err != 0) {
+ LOG(LS_ERROR) << "GetDevicesInfo - device name free hint error: "
+ << LATE(snd_strerror)(err);
+ }
+
+ return 0;
}
- if (keepSearching)
- {
- // If we get here for function 1 and 2, we didn't find the specified
- // enum device.
- LOG(LS_ERROR)
- << "GetDevicesInfo - Could not find device name or numbers";
- return -1;
+ for (void** list = hints; *list != NULL; ++list) {
+ char* actualType = LATE(snd_device_name_get_hint)(*list, "IOID");
+ if (actualType) { // NULL means it's both.
+ bool wrongType = (strcmp(actualType, type) != 0);
+ free(actualType);
+ if (wrongType) {
+ // Wrong type of device (i.e., input vs. output).
+ continue;
+ }
+ }
+
+ char* name = LATE(snd_device_name_get_hint)(*list, "NAME");
+ if (!name) {
+ LOG(LS_ERROR) << "Device has no name";
+ // Skip it.
+ continue;
+ }
+
+ // Now check if we actually want to show this device.
+ if (strcmp(name, "default") != 0 && strcmp(name, "null") != 0 &&
+ strcmp(name, "pulse") != 0 &&
+ strncmp(name, ignorePrefix, strlen(ignorePrefix)) != 0) {
+ // Yes, we do.
+ char* desc = LATE(snd_device_name_get_hint)(*list, "DESC");
+ if (!desc) {
+ // Virtual devices don't necessarily have descriptions.
+ // Use their names instead.
+ desc = name;
+ }
+
+ if (FUNC_GET_NUM_OF_DEVICE == function) {
+ LOG(LS_VERBOSE) << "Enum device " << enumCount << " - " << name;
+ }
+ if ((FUNC_GET_DEVICE_NAME == function) && (enumDeviceNo == enumCount)) {
+ // We have found the enum device, copy the name to buffer.
+ strncpy(enumDeviceName, desc, ednLen);
+ enumDeviceName[ednLen - 1] = '\0';
+ keepSearching = false;
+ // Replace '\n' with '-'.
+ char* pret = strchr(enumDeviceName, '\n' /*0xa*/); // LF
+ if (pret)
+ *pret = '-';
+ }
+ if ((FUNC_GET_DEVICE_NAME_FOR_AN_ENUM == function) &&
+ (enumDeviceNo == enumCount)) {
+ // We have found the enum device, copy the name to buffer.
+ strncpy(enumDeviceName, name, ednLen);
+ enumDeviceName[ednLen - 1] = '\0';
+ keepSearching = false;
+ }
+
+ if (keepSearching)
+ ++enumCount;
+
+ if (desc != name)
+ free(desc);
+ }
+
+ free(name);
+
+ if (!keepSearching)
+ break;
}
- return 0;
+ err = LATE(snd_device_name_free_hint)(hints);
+ if (err != 0) {
+ LOG(LS_ERROR) << "GetDevicesInfo - device name free hint error: "
+ << LATE(snd_strerror)(err);
+ // Continue and return true anyway, since we did get the whole list.
+ }
+ }
+
+ if (FUNC_GET_NUM_OF_DEVICE == function) {
+ if (enumCount == 1) // only default?
+ enumCount = 0;
+ return enumCount; // Normal return point for function 0
+ }
+
+ if (keepSearching) {
+ // If we get here for function 1 and 2, we didn't find the specified
+ // enum device.
+ LOG(LS_ERROR) << "GetDevicesInfo - Could not find device name or numbers";
+ return -1;
+ }
+
+ return 0;
}
-int32_t AudioDeviceLinuxALSA::InputSanityCheckAfterUnlockedPeriod() const
-{
- if (_handleRecord == NULL)
- {
- LOG(LS_ERROR) << "input state has been modified during unlocked period";
- return -1;
- }
- return 0;
+int32_t AudioDeviceLinuxALSA::InputSanityCheckAfterUnlockedPeriod() const {
+ if (_handleRecord == NULL) {
+ LOG(LS_ERROR) << "input state has been modified during unlocked period";
+ return -1;
+ }
+ return 0;
}
-int32_t AudioDeviceLinuxALSA::OutputSanityCheckAfterUnlockedPeriod() const
-{
- if (_handlePlayout == NULL)
- {
- LOG(LS_ERROR)
- << "output state has been modified during unlocked period";
- return -1;
- }
- return 0;
+int32_t AudioDeviceLinuxALSA::OutputSanityCheckAfterUnlockedPeriod() const {
+ if (_handlePlayout == NULL) {
+ LOG(LS_ERROR) << "output state has been modified during unlocked period";
+ return -1;
+ }
+ return 0;
}
int32_t AudioDeviceLinuxALSA::ErrorRecovery(int32_t error,
- snd_pcm_t* deviceHandle)
-{
- int st = LATE(snd_pcm_state)(deviceHandle);
- LOG(LS_VERBOSE) << "Trying to recover from "
- << ((LATE(snd_pcm_stream)(deviceHandle) == SND_PCM_STREAM_CAPTURE)
- ? "capture" : "playout") << " error: " << LATE(snd_strerror)(error)
- << " (" << error << ") (state " << st << ")";
+ snd_pcm_t* deviceHandle) {
+ int st = LATE(snd_pcm_state)(deviceHandle);
+ LOG(LS_VERBOSE) << "Trying to recover from "
+ << ((LATE(snd_pcm_stream)(deviceHandle) ==
+ SND_PCM_STREAM_CAPTURE)
+ ? "capture"
+ : "playout")
+ << " error: " << LATE(snd_strerror)(error) << " (" << error
+ << ") (state " << st << ")";
- // It is recommended to use snd_pcm_recover for all errors. If that function
- // cannot handle the error, the input error code will be returned, otherwise
- // 0 is returned. From snd_pcm_recover API doc: "This functions handles
- // -EINTR (4) (interrupted system call), -EPIPE (32) (playout overrun or
- // capture underrun) and -ESTRPIPE (86) (stream is suspended) error codes
- // trying to prepare given stream for next I/O."
+ // It is recommended to use snd_pcm_recover for all errors. If that function
+ // cannot handle the error, the input error code will be returned, otherwise
+ // 0 is returned. From snd_pcm_recover API doc: "This functions handles
+ // -EINTR (4) (interrupted system call), -EPIPE (32) (playout overrun or
+ // capture underrun) and -ESTRPIPE (86) (stream is suspended) error codes
+ // trying to prepare given stream for next I/O."
- /** Open */
- // SND_PCM_STATE_OPEN = 0,
- /** Setup installed */
- // SND_PCM_STATE_SETUP,
- /** Ready to start */
- // SND_PCM_STATE_PREPARED,
- /** Running */
- // SND_PCM_STATE_RUNNING,
- /** Stopped: underrun (playback) or overrun (capture) detected */
- // SND_PCM_STATE_XRUN,= 4
- /** Draining: running (playback) or stopped (capture) */
- // SND_PCM_STATE_DRAINING,
- /** Paused */
- // SND_PCM_STATE_PAUSED,
- /** Hardware is suspended */
- // SND_PCM_STATE_SUSPENDED,
- // ** Hardware is disconnected */
- // SND_PCM_STATE_DISCONNECTED,
- // SND_PCM_STATE_LAST = SND_PCM_STATE_DISCONNECTED
+ /** Open */
+ // SND_PCM_STATE_OPEN = 0,
+ /** Setup installed */
+ // SND_PCM_STATE_SETUP,
+ /** Ready to start */
+ // SND_PCM_STATE_PREPARED,
+ /** Running */
+ // SND_PCM_STATE_RUNNING,
+ /** Stopped: underrun (playback) or overrun (capture) detected */
+ // SND_PCM_STATE_XRUN,= 4
+ /** Draining: running (playback) or stopped (capture) */
+ // SND_PCM_STATE_DRAINING,
+ /** Paused */
+ // SND_PCM_STATE_PAUSED,
+ /** Hardware is suspended */
+ // SND_PCM_STATE_SUSPENDED,
+ // ** Hardware is disconnected */
+ // SND_PCM_STATE_DISCONNECTED,
+ // SND_PCM_STATE_LAST = SND_PCM_STATE_DISCONNECTED
- // snd_pcm_recover isn't available in older alsa, e.g. on the FC4 machine
- // in Sthlm lab.
+ // snd_pcm_recover isn't available in older alsa, e.g. on the FC4 machine
+ // in Sthlm lab.
- int res = LATE(snd_pcm_recover)(deviceHandle, error, 1);
- if (0 == res)
- {
- LOG(LS_VERBOSE) << "Recovery - snd_pcm_recover OK";
+ int res = LATE(snd_pcm_recover)(deviceHandle, error, 1);
+ if (0 == res) {
+ LOG(LS_VERBOSE) << "Recovery - snd_pcm_recover OK";
- if ((error == -EPIPE || error == -ESTRPIPE) && // Buf underrun/overrun.
- _recording &&
- LATE(snd_pcm_stream)(deviceHandle) == SND_PCM_STREAM_CAPTURE)
- {
- // For capture streams we also have to repeat the explicit start()
- // to get data flowing again.
- int err = LATE(snd_pcm_start)(deviceHandle);
- if (err != 0)
- {
- LOG(LS_ERROR) << "Recovery - snd_pcm_start error: " << err;
- return -1;
- }
- }
-
- if ((error == -EPIPE || error == -ESTRPIPE) && // Buf underrun/overrun.
- _playing &&
- LATE(snd_pcm_stream)(deviceHandle) == SND_PCM_STREAM_PLAYBACK)
- {
- // For capture streams we also have to repeat the explicit start() to get
- // data flowing again.
- int err = LATE(snd_pcm_start)(deviceHandle);
- if (err != 0)
- {
- LOG(LS_ERROR) << "Recovery - snd_pcm_start error: "
- << LATE(snd_strerror)(err);
- return -1;
- }
- }
-
- return -EPIPE == error ? 1 : 0;
- }
- else {
- LOG(LS_ERROR) << "Unrecoverable alsa stream error: " << res;
+ if ((error == -EPIPE || error == -ESTRPIPE) && // Buf underrun/overrun.
+ _recording &&
+ LATE(snd_pcm_stream)(deviceHandle) == SND_PCM_STREAM_CAPTURE) {
+ // For capture streams we also have to repeat the explicit start()
+ // to get data flowing again.
+ int err = LATE(snd_pcm_start)(deviceHandle);
+ if (err != 0) {
+ LOG(LS_ERROR) << "Recovery - snd_pcm_start error: " << err;
+ return -1;
+ }
}
- return res;
+ if ((error == -EPIPE || error == -ESTRPIPE) && // Buf underrun/overrun.
+ _playing &&
+ LATE(snd_pcm_stream)(deviceHandle) == SND_PCM_STREAM_PLAYBACK) {
+ // For capture streams we also have to repeat the explicit start() to get
+ // data flowing again.
+ int err = LATE(snd_pcm_start)(deviceHandle);
+ if (err != 0) {
+ LOG(LS_ERROR) << "Recovery - snd_pcm_start error: "
+ << LATE(snd_strerror)(err);
+ return -1;
+ }
+ }
+
+ return -EPIPE == error ? 1 : 0;
+ } else {
+ LOG(LS_ERROR) << "Unrecoverable alsa stream error: " << res;
+ }
+
+ return res;
}
// ============================================================================
// Thread Methods
// ============================================================================
-bool AudioDeviceLinuxALSA::PlayThreadFunc(void* pThis)
-{
- return (static_cast<AudioDeviceLinuxALSA*>(pThis)->PlayThreadProcess());
+bool AudioDeviceLinuxALSA::PlayThreadFunc(void* pThis) {
+ return (static_cast<AudioDeviceLinuxALSA*>(pThis)->PlayThreadProcess());
}
-bool AudioDeviceLinuxALSA::RecThreadFunc(void* pThis)
-{
- return (static_cast<AudioDeviceLinuxALSA*>(pThis)->RecThreadProcess());
+bool AudioDeviceLinuxALSA::RecThreadFunc(void* pThis) {
+ return (static_cast<AudioDeviceLinuxALSA*>(pThis)->RecThreadProcess());
}
-bool AudioDeviceLinuxALSA::PlayThreadProcess()
-{
- if(!_playing)
- return false;
+bool AudioDeviceLinuxALSA::PlayThreadProcess() {
+ if (!_playing)
+ return false;
- int err;
- snd_pcm_sframes_t frames;
- snd_pcm_sframes_t avail_frames;
+ int err;
+ snd_pcm_sframes_t frames;
+ snd_pcm_sframes_t avail_frames;
- Lock();
- //return a positive number of frames ready otherwise a negative error code
- avail_frames = LATE(snd_pcm_avail_update)(_handlePlayout);
- if (avail_frames < 0)
- {
- LOG(LS_ERROR) << "playout snd_pcm_avail_update error: "
- << LATE(snd_strerror)(avail_frames);
- ErrorRecovery(avail_frames, _handlePlayout);
- UnLock();
- return true;
- }
- else if (avail_frames == 0)
- {
- UnLock();
-
- //maximum tixe in milliseconds to wait, a negative value means infinity
- err = LATE(snd_pcm_wait)(_handlePlayout, 2);
- if (err == 0)
- { //timeout occured
- LOG(LS_VERBOSE) << "playout snd_pcm_wait timeout";
- }
-
- return true;
- }
-
- if (_playoutFramesLeft <= 0)
- {
- UnLock();
- _ptrAudioBuffer->RequestPlayoutData(_playoutFramesIn10MS);
- Lock();
-
- _playoutFramesLeft = _ptrAudioBuffer->GetPlayoutData(_playoutBuffer);
- assert(_playoutFramesLeft == _playoutFramesIn10MS);
- }
-
- if (static_cast<uint32_t>(avail_frames) > _playoutFramesLeft)
- avail_frames = _playoutFramesLeft;
-
- int size = LATE(snd_pcm_frames_to_bytes)(_handlePlayout,
- _playoutFramesLeft);
- frames = LATE(snd_pcm_writei)(
- _handlePlayout,
- &_playoutBuffer[_playoutBufferSizeIn10MS - size],
- avail_frames);
-
- if (frames < 0)
- {
- LOG(LS_VERBOSE) << "playout snd_pcm_writei error: "
- << LATE(snd_strerror)(frames);
- _playoutFramesLeft = 0;
- ErrorRecovery(frames, _handlePlayout);
- UnLock();
- return true;
- }
- else {
- assert(frames==avail_frames);
- _playoutFramesLeft -= frames;
- }
-
+ Lock();
+ // return a positive number of frames ready otherwise a negative error code
+ avail_frames = LATE(snd_pcm_avail_update)(_handlePlayout);
+ if (avail_frames < 0) {
+ LOG(LS_ERROR) << "playout snd_pcm_avail_update error: "
+ << LATE(snd_strerror)(avail_frames);
+ ErrorRecovery(avail_frames, _handlePlayout);
UnLock();
return true;
-}
+ } else if (avail_frames == 0) {
+ UnLock();
-bool AudioDeviceLinuxALSA::RecThreadProcess()
-{
- if (!_recording)
- return false;
+ // maximum tixe in milliseconds to wait, a negative value means infinity
+ err = LATE(snd_pcm_wait)(_handlePlayout, 2);
+ if (err == 0) { // timeout occured
+ LOG(LS_VERBOSE) << "playout snd_pcm_wait timeout";
+ }
- int err;
- snd_pcm_sframes_t frames;
- snd_pcm_sframes_t avail_frames;
- int8_t buffer[_recordingBufferSizeIn10MS];
+ return true;
+ }
+ if (_playoutFramesLeft <= 0) {
+ UnLock();
+ _ptrAudioBuffer->RequestPlayoutData(_playoutFramesIn10MS);
Lock();
- //return a positive number of frames ready otherwise a negative error code
- avail_frames = LATE(snd_pcm_avail_update)(_handleRecord);
- if (avail_frames < 0)
- {
- LOG(LS_ERROR) << "capture snd_pcm_avail_update error: "
- << LATE(snd_strerror)(avail_frames);
- ErrorRecovery(avail_frames, _handleRecord);
- UnLock();
- return true;
- }
- else if (avail_frames == 0)
- { // no frame is available now
- UnLock();
+ _playoutFramesLeft = _ptrAudioBuffer->GetPlayoutData(_playoutBuffer);
+ assert(_playoutFramesLeft == _playoutFramesIn10MS);
+ }
- //maximum time in milliseconds to wait, a negative value means infinity
- err = LATE(snd_pcm_wait)(_handleRecord,
- ALSA_CAPTURE_WAIT_TIMEOUT);
- if (err == 0) //timeout occured
- LOG(LS_VERBOSE) << "capture snd_pcm_wait timeout";
+ if (static_cast<uint32_t>(avail_frames) > _playoutFramesLeft)
+ avail_frames = _playoutFramesLeft;
- return true;
- }
+ int size = LATE(snd_pcm_frames_to_bytes)(_handlePlayout, _playoutFramesLeft);
+ frames = LATE(snd_pcm_writei)(
+ _handlePlayout, &_playoutBuffer[_playoutBufferSizeIn10MS - size],
+ avail_frames);
- if (static_cast<uint32_t>(avail_frames) > _recordingFramesLeft)
- avail_frames = _recordingFramesLeft;
-
- frames = LATE(snd_pcm_readi)(_handleRecord,
- buffer, avail_frames); // frames to be written
- if (frames < 0)
- {
- LOG(LS_ERROR) << "capture snd_pcm_readi error: "
- << LATE(snd_strerror)(frames);
- ErrorRecovery(frames, _handleRecord);
- UnLock();
- return true;
- }
- else if (frames > 0)
- {
- assert(frames == avail_frames);
-
- int left_size = LATE(snd_pcm_frames_to_bytes)(_handleRecord,
- _recordingFramesLeft);
- int size = LATE(snd_pcm_frames_to_bytes)(_handleRecord, frames);
-
- memcpy(&_recordingBuffer[_recordingBufferSizeIn10MS - left_size],
- buffer, size);
- _recordingFramesLeft -= frames;
-
- if (!_recordingFramesLeft)
- { // buf is full
- _recordingFramesLeft = _recordingFramesIn10MS;
-
- // store the recorded buffer (no action will be taken if the
- // #recorded samples is not a full buffer)
- _ptrAudioBuffer->SetRecordedBuffer(_recordingBuffer,
- _recordingFramesIn10MS);
-
- uint32_t currentMicLevel = 0;
- uint32_t newMicLevel = 0;
-
- if (AGC())
- {
- // store current mic level in the audio buffer if AGC is enabled
- if (MicrophoneVolume(currentMicLevel) == 0)
- {
- if (currentMicLevel == 0xffffffff)
- currentMicLevel = 100;
- // this call does not affect the actual microphone volume
- _ptrAudioBuffer->SetCurrentMicLevel(currentMicLevel);
- }
- }
-
- // calculate delay
- _playoutDelay = 0;
- _recordingDelay = 0;
- if (_handlePlayout)
- {
- err = LATE(snd_pcm_delay)(_handlePlayout,
- &_playoutDelay); // returned delay in frames
- if (err < 0)
- {
- // TODO(xians): Shall we call ErrorRecovery() here?
- _playoutDelay = 0;
- LOG(LS_ERROR) << "playout snd_pcm_delay: "
- << LATE(snd_strerror)(err);
- }
- }
-
- err = LATE(snd_pcm_delay)(_handleRecord,
- &_recordingDelay); // returned delay in frames
- if (err < 0)
- {
- // TODO(xians): Shall we call ErrorRecovery() here?
- _recordingDelay = 0;
- LOG(LS_ERROR) << "capture snd_pcm_delay: "
- << LATE(snd_strerror)(err);
- }
-
- // TODO(xians): Shall we add 10ms buffer delay to the record delay?
- _ptrAudioBuffer->SetVQEData(
- _playoutDelay * 1000 / _playoutFreq,
- _recordingDelay * 1000 / _recordingFreq, 0);
-
- _ptrAudioBuffer->SetTypingStatus(KeyPressed());
-
- // Deliver recorded samples at specified sample rate, mic level etc.
- // to the observer using callback.
- UnLock();
- _ptrAudioBuffer->DeliverRecordedData();
- Lock();
-
- if (AGC())
- {
- newMicLevel = _ptrAudioBuffer->NewMicLevel();
- if (newMicLevel != 0)
- {
- // The VQE will only deliver non-zero microphone levels when a
- // change is needed. Set this new mic level (received from the
- // observer as return value in the callback).
- if (SetMicrophoneVolume(newMicLevel) == -1)
- LOG(LS_WARNING)
- << "the required modification of the microphone volume failed";
- }
- }
- }
- }
-
+ if (frames < 0) {
+ LOG(LS_VERBOSE) << "playout snd_pcm_writei error: "
+ << LATE(snd_strerror)(frames);
+ _playoutFramesLeft = 0;
+ ErrorRecovery(frames, _handlePlayout);
UnLock();
return true;
+ } else {
+ assert(frames == avail_frames);
+ _playoutFramesLeft -= frames;
+ }
+
+ UnLock();
+ return true;
}
+bool AudioDeviceLinuxALSA::RecThreadProcess() {
+ if (!_recording)
+ return false;
-bool AudioDeviceLinuxALSA::KeyPressed() const{
+ int err;
+ snd_pcm_sframes_t frames;
+ snd_pcm_sframes_t avail_frames;
+ int8_t buffer[_recordingBufferSizeIn10MS];
+
+ Lock();
+
+ // return a positive number of frames ready otherwise a negative error code
+ avail_frames = LATE(snd_pcm_avail_update)(_handleRecord);
+ if (avail_frames < 0) {
+ LOG(LS_ERROR) << "capture snd_pcm_avail_update error: "
+ << LATE(snd_strerror)(avail_frames);
+ ErrorRecovery(avail_frames, _handleRecord);
+ UnLock();
+ return true;
+ } else if (avail_frames == 0) { // no frame is available now
+ UnLock();
+
+ // maximum time in milliseconds to wait, a negative value means infinity
+ err = LATE(snd_pcm_wait)(_handleRecord, ALSA_CAPTURE_WAIT_TIMEOUT);
+ if (err == 0) // timeout occured
+ LOG(LS_VERBOSE) << "capture snd_pcm_wait timeout";
+
+ return true;
+ }
+
+ if (static_cast<uint32_t>(avail_frames) > _recordingFramesLeft)
+ avail_frames = _recordingFramesLeft;
+
+ frames = LATE(snd_pcm_readi)(_handleRecord, buffer,
+ avail_frames); // frames to be written
+ if (frames < 0) {
+ LOG(LS_ERROR) << "capture snd_pcm_readi error: "
+ << LATE(snd_strerror)(frames);
+ ErrorRecovery(frames, _handleRecord);
+ UnLock();
+ return true;
+ } else if (frames > 0) {
+ assert(frames == avail_frames);
+
+ int left_size =
+ LATE(snd_pcm_frames_to_bytes)(_handleRecord, _recordingFramesLeft);
+ int size = LATE(snd_pcm_frames_to_bytes)(_handleRecord, frames);
+
+ memcpy(&_recordingBuffer[_recordingBufferSizeIn10MS - left_size], buffer,
+ size);
+ _recordingFramesLeft -= frames;
+
+ if (!_recordingFramesLeft) { // buf is full
+ _recordingFramesLeft = _recordingFramesIn10MS;
+
+ // store the recorded buffer (no action will be taken if the
+ // #recorded samples is not a full buffer)
+ _ptrAudioBuffer->SetRecordedBuffer(_recordingBuffer,
+ _recordingFramesIn10MS);
+
+ uint32_t currentMicLevel = 0;
+ uint32_t newMicLevel = 0;
+
+ if (AGC()) {
+ // store current mic level in the audio buffer if AGC is enabled
+ if (MicrophoneVolume(currentMicLevel) == 0) {
+ if (currentMicLevel == 0xffffffff)
+ currentMicLevel = 100;
+ // this call does not affect the actual microphone volume
+ _ptrAudioBuffer->SetCurrentMicLevel(currentMicLevel);
+ }
+ }
+
+ // calculate delay
+ _playoutDelay = 0;
+ _recordingDelay = 0;
+ if (_handlePlayout) {
+ err = LATE(snd_pcm_delay)(_handlePlayout,
+ &_playoutDelay); // returned delay in frames
+ if (err < 0) {
+ // TODO(xians): Shall we call ErrorRecovery() here?
+ _playoutDelay = 0;
+ LOG(LS_ERROR) << "playout snd_pcm_delay: " << LATE(snd_strerror)(err);
+ }
+ }
+
+ err = LATE(snd_pcm_delay)(_handleRecord,
+ &_recordingDelay); // returned delay in frames
+ if (err < 0) {
+ // TODO(xians): Shall we call ErrorRecovery() here?
+ _recordingDelay = 0;
+ LOG(LS_ERROR) << "capture snd_pcm_delay: " << LATE(snd_strerror)(err);
+ }
+
+ // TODO(xians): Shall we add 10ms buffer delay to the record delay?
+ _ptrAudioBuffer->SetVQEData(_playoutDelay * 1000 / _playoutFreq,
+ _recordingDelay * 1000 / _recordingFreq, 0);
+
+ _ptrAudioBuffer->SetTypingStatus(KeyPressed());
+
+ // Deliver recorded samples at specified sample rate, mic level etc.
+ // to the observer using callback.
+ UnLock();
+ _ptrAudioBuffer->DeliverRecordedData();
+ Lock();
+
+ if (AGC()) {
+ newMicLevel = _ptrAudioBuffer->NewMicLevel();
+ if (newMicLevel != 0) {
+ // The VQE will only deliver non-zero microphone levels when a
+ // change is needed. Set this new mic level (received from the
+ // observer as return value in the callback).
+ if (SetMicrophoneVolume(newMicLevel) == -1)
+ LOG(LS_WARNING)
+ << "the required modification of the microphone volume failed";
+ }
+ }
+ }
+ }
+
+ UnLock();
+ return true;
+}
+
+bool AudioDeviceLinuxALSA::KeyPressed() const {
#if defined(USE_X11)
char szKey[32];
unsigned int i = 0;
diff --git a/modules/audio_device/linux/audio_device_pulse_linux.cc b/modules/audio_device/linux/audio_device_pulse_linux.cc
index f1eddd6..b9614bf 100644
--- a/modules/audio_device/linux/audio_device_pulse_linux.cc
+++ b/modules/audio_device/linux/audio_device_pulse_linux.cc
@@ -2136,8 +2136,7 @@
NULL, (int64_t)0, PA_SEEK_RELATIVE) != PA_OK) {
_writeErrors++;
if (_writeErrors > 10) {
- LOG(LS_ERROR) << "Playout error: _writeErrors="
- << _writeErrors
+ LOG(LS_ERROR) << "Playout error: _writeErrors=" << _writeErrors
<< ", error=" << LATE(pa_context_errno)(_paContext);
_writeErrors = 0;
}
@@ -2180,8 +2179,7 @@
NULL, (int64_t)0, PA_SEEK_RELATIVE) != PA_OK) {
_writeErrors++;
if (_writeErrors > 10) {
- LOG(LS_ERROR) << "Playout error: _writeErrors="
- << _writeErrors
+ LOG(LS_ERROR) << "Playout error: _writeErrors=" << _writeErrors
<< ", error=" << LATE(pa_context_errno)(_paContext);
_writeErrors = 0;
}
diff --git a/modules/audio_device/linux/audio_mixer_manager_alsa_linux.cc b/modules/audio_device/linux/audio_mixer_manager_alsa_linux.cc
index 02d9cf9..aabf388 100644
--- a/modules/audio_device/linux/audio_mixer_manager_alsa_linux.cc
+++ b/modules/audio_device/linux/audio_mixer_manager_alsa_linux.cc
@@ -21,433 +21,368 @@
#define LATE(sym) \
LATESYM_GET(webrtc::adm_linux_alsa::AlsaSymbolTable, &AlsaSymbolTable, sym)
-namespace webrtc
-{
+namespace webrtc {
-AudioMixerManagerLinuxALSA::AudioMixerManagerLinuxALSA() :
- _outputMixerHandle(NULL),
- _inputMixerHandle(NULL),
- _outputMixerElement(NULL),
- _inputMixerElement(NULL)
-{
- LOG(LS_INFO) << __FUNCTION__ << " created";
+AudioMixerManagerLinuxALSA::AudioMixerManagerLinuxALSA()
+ : _outputMixerHandle(NULL),
+ _inputMixerHandle(NULL),
+ _outputMixerElement(NULL),
+ _inputMixerElement(NULL) {
+ LOG(LS_INFO) << __FUNCTION__ << " created";
- memset(_outputMixerStr, 0, kAdmMaxDeviceNameSize);
- memset(_inputMixerStr, 0, kAdmMaxDeviceNameSize);
+ memset(_outputMixerStr, 0, kAdmMaxDeviceNameSize);
+ memset(_inputMixerStr, 0, kAdmMaxDeviceNameSize);
}
-AudioMixerManagerLinuxALSA::~AudioMixerManagerLinuxALSA()
-{
- LOG(LS_INFO) << __FUNCTION__ << " destroyed";
- Close();
+AudioMixerManagerLinuxALSA::~AudioMixerManagerLinuxALSA() {
+ LOG(LS_INFO) << __FUNCTION__ << " destroyed";
+ Close();
}
// ============================================================================
// PUBLIC METHODS
// ============================================================================
-int32_t AudioMixerManagerLinuxALSA::Close()
-{
- LOG(LS_VERBOSE) << __FUNCTION__;
+int32_t AudioMixerManagerLinuxALSA::Close() {
+ LOG(LS_VERBOSE) << __FUNCTION__;
- rtc::CritScope lock(&_critSect);
+ rtc::CritScope lock(&_critSect);
- CloseSpeaker();
- CloseMicrophone();
+ CloseSpeaker();
+ CloseMicrophone();
- return 0;
-
+ return 0;
}
-int32_t AudioMixerManagerLinuxALSA::CloseSpeaker()
-{
- LOG(LS_VERBOSE) << __FUNCTION__;
+int32_t AudioMixerManagerLinuxALSA::CloseSpeaker() {
+ LOG(LS_VERBOSE) << __FUNCTION__;
- rtc::CritScope lock(&_critSect);
+ rtc::CritScope lock(&_critSect);
- int errVal = 0;
+ int errVal = 0;
- if (_outputMixerHandle != NULL)
- {
- LOG(LS_VERBOSE) << "Closing playout mixer";
- LATE(snd_mixer_free)(_outputMixerHandle);
- if (errVal < 0)
- {
- LOG(LS_ERROR) << "Error freeing playout mixer: "
- << LATE(snd_strerror)(errVal);
- }
- errVal = LATE(snd_mixer_detach)(_outputMixerHandle, _outputMixerStr);
- if (errVal < 0)
- {
- LOG(LS_ERROR) << "Error detaching playout mixer: "
- << LATE(snd_strerror)(errVal);
- }
- errVal = LATE(snd_mixer_close)(_outputMixerHandle);
- if (errVal < 0)
- {
- LOG(LS_ERROR) << "Error snd_mixer_close(handleMixer) errVal="
- << errVal;
- }
- _outputMixerHandle = NULL;
- _outputMixerElement = NULL;
+ if (_outputMixerHandle != NULL) {
+ LOG(LS_VERBOSE) << "Closing playout mixer";
+ LATE(snd_mixer_free)(_outputMixerHandle);
+ if (errVal < 0) {
+ LOG(LS_ERROR) << "Error freeing playout mixer: "
+ << LATE(snd_strerror)(errVal);
}
- memset(_outputMixerStr, 0, kAdmMaxDeviceNameSize);
-
- return 0;
-}
-
-int32_t AudioMixerManagerLinuxALSA::CloseMicrophone()
-{
- LOG(LS_VERBOSE) << __FUNCTION__;
-
- rtc::CritScope lock(&_critSect);
-
- int errVal = 0;
-
- if (_inputMixerHandle != NULL)
- {
- LOG(LS_VERBOSE) << "Closing record mixer";
-
- LATE(snd_mixer_free)(_inputMixerHandle);
- if (errVal < 0)
- {
- LOG(LS_ERROR) << "Error freeing record mixer: "
- << LATE(snd_strerror)(errVal);
- }
- LOG(LS_VERBOSE) << "Closing record mixer 2";
-
- errVal = LATE(snd_mixer_detach)(_inputMixerHandle, _inputMixerStr);
- if (errVal < 0)
- {
- LOG(LS_ERROR) << "Error detaching record mixer: "
- << LATE(snd_strerror)(errVal);
- }
- LOG(LS_VERBOSE) << "Closing record mixer 3";
-
- errVal = LATE(snd_mixer_close)(_inputMixerHandle);
- if (errVal < 0)
- {
- LOG(LS_ERROR) << "Error snd_mixer_close(handleMixer) errVal="
- << errVal;
- }
-
- LOG(LS_VERBOSE) << "Closing record mixer 4";
- _inputMixerHandle = NULL;
- _inputMixerElement = NULL;
+ errVal = LATE(snd_mixer_detach)(_outputMixerHandle, _outputMixerStr);
+ if (errVal < 0) {
+ LOG(LS_ERROR) << "Error detaching playout mixer: "
+ << LATE(snd_strerror)(errVal);
}
- memset(_inputMixerStr, 0, kAdmMaxDeviceNameSize);
-
- return 0;
-}
-
-int32_t AudioMixerManagerLinuxALSA::OpenSpeaker(char* deviceName)
-{
- LOG(LS_VERBOSE) << "AudioMixerManagerLinuxALSA::OpenSpeaker(name="
- << deviceName << ")";
-
- rtc::CritScope lock(&_critSect);
-
- int errVal = 0;
-
- // Close any existing output mixer handle
- //
- if (_outputMixerHandle != NULL)
- {
- LOG(LS_VERBOSE) << "Closing playout mixer";
-
- LATE(snd_mixer_free)(_outputMixerHandle);
- if (errVal < 0)
- {
- LOG(LS_ERROR) << "Error freeing playout mixer: "
- << LATE(snd_strerror)(errVal);
- }
- errVal = LATE(snd_mixer_detach)(_outputMixerHandle, _outputMixerStr);
- if (errVal < 0)
- {
- LOG(LS_ERROR) << "Error detaching playout mixer: "
- << LATE(snd_strerror)(errVal);
- }
- errVal = LATE(snd_mixer_close)(_outputMixerHandle);
- if (errVal < 0)
- {
- LOG(LS_ERROR) << "Error snd_mixer_close(handleMixer) errVal="
- << errVal;
- }
+ errVal = LATE(snd_mixer_close)(_outputMixerHandle);
+ if (errVal < 0) {
+ LOG(LS_ERROR) << "Error snd_mixer_close(handleMixer) errVal=" << errVal;
}
_outputMixerHandle = NULL;
_outputMixerElement = NULL;
+ }
+ memset(_outputMixerStr, 0, kAdmMaxDeviceNameSize);
- errVal = LATE(snd_mixer_open)(&_outputMixerHandle, 0);
- if (errVal < 0)
- {
- LOG(LS_ERROR) << "snd_mixer_open(&_outputMixerHandle, 0) - error";
- return -1;
- }
-
- char controlName[kAdmMaxDeviceNameSize] = { 0 };
- GetControlName(controlName, deviceName);
-
- LOG(LS_VERBOSE) << "snd_mixer_attach(_outputMixerHandle, " << controlName
- << ")";
-
- errVal = LATE(snd_mixer_attach)(_outputMixerHandle, controlName);
- if (errVal < 0)
- {
- LOG(LS_ERROR) << "snd_mixer_attach(_outputMixerHandle, " << controlName
- << ") error: " << LATE(snd_strerror)(errVal);
- _outputMixerHandle = NULL;
- return -1;
- }
- strcpy(_outputMixerStr, controlName);
-
- errVal = LATE(snd_mixer_selem_register)(_outputMixerHandle, NULL, NULL);
- if (errVal < 0)
- {
- LOG(LS_ERROR)
- << "snd_mixer_selem_register(_outputMixerHandle, NULL, NULL), "
- << "error: " << LATE(snd_strerror)(errVal);
- _outputMixerHandle = NULL;
- return -1;
- }
-
- // Load and find the proper mixer element
- if (LoadSpeakerMixerElement() < 0)
- {
- return -1;
- }
-
- if (_outputMixerHandle != NULL)
- {
- LOG(LS_VERBOSE) << "the output mixer device is now open ("
- << _outputMixerHandle << ")";
- }
-
- return 0;
+ return 0;
}
-int32_t AudioMixerManagerLinuxALSA::OpenMicrophone(char *deviceName)
-{
- LOG(LS_VERBOSE) << "AudioMixerManagerLinuxALSA::OpenMicrophone(name="
- << deviceName << ")";
+int32_t AudioMixerManagerLinuxALSA::CloseMicrophone() {
+ LOG(LS_VERBOSE) << __FUNCTION__;
- rtc::CritScope lock(&_critSect);
+ rtc::CritScope lock(&_critSect);
- int errVal = 0;
+ int errVal = 0;
- // Close any existing input mixer handle
- //
- if (_inputMixerHandle != NULL)
- {
- LOG(LS_VERBOSE) << "Closing record mixer";
+ if (_inputMixerHandle != NULL) {
+ LOG(LS_VERBOSE) << "Closing record mixer";
- LATE(snd_mixer_free)(_inputMixerHandle);
- if (errVal < 0)
- {
- LOG(LS_ERROR) << "Error freeing record mixer: "
- << LATE(snd_strerror)(errVal);
- }
- LOG(LS_VERBOSE) << "Closing record mixer";
-
- errVal = LATE(snd_mixer_detach)(_inputMixerHandle, _inputMixerStr);
- if (errVal < 0)
- {
- LOG(LS_ERROR) << "Error detaching record mixer: "
- << LATE(snd_strerror)(errVal);
- }
- LOG(LS_VERBOSE) << "Closing record mixer";
-
- errVal = LATE(snd_mixer_close)(_inputMixerHandle);
- if (errVal < 0)
- {
- LOG(LS_ERROR) << "Error snd_mixer_close(handleMixer) errVal="
- << errVal;
- }
- LOG(LS_VERBOSE) << "Closing record mixer";
+ LATE(snd_mixer_free)(_inputMixerHandle);
+ if (errVal < 0) {
+ LOG(LS_ERROR) << "Error freeing record mixer: "
+ << LATE(snd_strerror)(errVal);
}
+ LOG(LS_VERBOSE) << "Closing record mixer 2";
+
+ errVal = LATE(snd_mixer_detach)(_inputMixerHandle, _inputMixerStr);
+ if (errVal < 0) {
+ LOG(LS_ERROR) << "Error detaching record mixer: "
+ << LATE(snd_strerror)(errVal);
+ }
+ LOG(LS_VERBOSE) << "Closing record mixer 3";
+
+ errVal = LATE(snd_mixer_close)(_inputMixerHandle);
+ if (errVal < 0) {
+ LOG(LS_ERROR) << "Error snd_mixer_close(handleMixer) errVal=" << errVal;
+ }
+
+ LOG(LS_VERBOSE) << "Closing record mixer 4";
_inputMixerHandle = NULL;
_inputMixerElement = NULL;
+ }
+ memset(_inputMixerStr, 0, kAdmMaxDeviceNameSize);
- errVal = LATE(snd_mixer_open)(&_inputMixerHandle, 0);
- if (errVal < 0)
- {
- LOG(LS_ERROR) << "snd_mixer_open(&_inputMixerHandle, 0) - error";
- return -1;
- }
-
- char controlName[kAdmMaxDeviceNameSize] = { 0 };
- GetControlName(controlName, deviceName);
-
- LOG(LS_VERBOSE) << "snd_mixer_attach(_inputMixerHandle, " << controlName
- << ")";
-
- errVal = LATE(snd_mixer_attach)(_inputMixerHandle, controlName);
- if (errVal < 0)
- {
- LOG(LS_ERROR) << "snd_mixer_attach(_inputMixerHandle, " << controlName
- << ") error: " << LATE(snd_strerror)(errVal);
-
- _inputMixerHandle = NULL;
- return -1;
- }
- strcpy(_inputMixerStr, controlName);
-
- errVal = LATE(snd_mixer_selem_register)(_inputMixerHandle, NULL, NULL);
- if (errVal < 0)
- {
- LOG(LS_ERROR)
- << "snd_mixer_selem_register(_inputMixerHandle, NULL, NULL), "
- << "error: " << LATE(snd_strerror)(errVal);
-
- _inputMixerHandle = NULL;
- return -1;
- }
- // Load and find the proper mixer element
- if (LoadMicMixerElement() < 0)
- {
- return -1;
- }
-
- if (_inputMixerHandle != NULL)
- {
- LOG(LS_VERBOSE) << "the input mixer device is now open ("
- << _inputMixerHandle << ")";
- }
-
- return 0;
+ return 0;
}
-bool AudioMixerManagerLinuxALSA::SpeakerIsInitialized() const
-{
- LOG(LS_INFO) << __FUNCTION__;
+int32_t AudioMixerManagerLinuxALSA::OpenSpeaker(char* deviceName) {
+ LOG(LS_VERBOSE) << "AudioMixerManagerLinuxALSA::OpenSpeaker(name="
+ << deviceName << ")";
- return (_outputMixerHandle != NULL);
+ rtc::CritScope lock(&_critSect);
+
+ int errVal = 0;
+
+ // Close any existing output mixer handle
+ //
+ if (_outputMixerHandle != NULL) {
+ LOG(LS_VERBOSE) << "Closing playout mixer";
+
+ LATE(snd_mixer_free)(_outputMixerHandle);
+ if (errVal < 0) {
+ LOG(LS_ERROR) << "Error freeing playout mixer: "
+ << LATE(snd_strerror)(errVal);
+ }
+ errVal = LATE(snd_mixer_detach)(_outputMixerHandle, _outputMixerStr);
+ if (errVal < 0) {
+ LOG(LS_ERROR) << "Error detaching playout mixer: "
+ << LATE(snd_strerror)(errVal);
+ }
+ errVal = LATE(snd_mixer_close)(_outputMixerHandle);
+ if (errVal < 0) {
+ LOG(LS_ERROR) << "Error snd_mixer_close(handleMixer) errVal=" << errVal;
+ }
+ }
+ _outputMixerHandle = NULL;
+ _outputMixerElement = NULL;
+
+ errVal = LATE(snd_mixer_open)(&_outputMixerHandle, 0);
+ if (errVal < 0) {
+ LOG(LS_ERROR) << "snd_mixer_open(&_outputMixerHandle, 0) - error";
+ return -1;
+ }
+
+ char controlName[kAdmMaxDeviceNameSize] = {0};
+ GetControlName(controlName, deviceName);
+
+ LOG(LS_VERBOSE) << "snd_mixer_attach(_outputMixerHandle, " << controlName
+ << ")";
+
+ errVal = LATE(snd_mixer_attach)(_outputMixerHandle, controlName);
+ if (errVal < 0) {
+ LOG(LS_ERROR) << "snd_mixer_attach(_outputMixerHandle, " << controlName
+ << ") error: " << LATE(snd_strerror)(errVal);
+ _outputMixerHandle = NULL;
+ return -1;
+ }
+ strcpy(_outputMixerStr, controlName);
+
+ errVal = LATE(snd_mixer_selem_register)(_outputMixerHandle, NULL, NULL);
+ if (errVal < 0) {
+ LOG(LS_ERROR)
+ << "snd_mixer_selem_register(_outputMixerHandle, NULL, NULL), "
+ << "error: " << LATE(snd_strerror)(errVal);
+ _outputMixerHandle = NULL;
+ return -1;
+ }
+
+ // Load and find the proper mixer element
+ if (LoadSpeakerMixerElement() < 0) {
+ return -1;
+ }
+
+ if (_outputMixerHandle != NULL) {
+ LOG(LS_VERBOSE) << "the output mixer device is now open ("
+ << _outputMixerHandle << ")";
+ }
+
+ return 0;
}
-bool AudioMixerManagerLinuxALSA::MicrophoneIsInitialized() const
-{
- LOG(LS_INFO) << __FUNCTION__;
+int32_t AudioMixerManagerLinuxALSA::OpenMicrophone(char* deviceName) {
+ LOG(LS_VERBOSE) << "AudioMixerManagerLinuxALSA::OpenMicrophone(name="
+ << deviceName << ")";
- return (_inputMixerHandle != NULL);
+ rtc::CritScope lock(&_critSect);
+
+ int errVal = 0;
+
+ // Close any existing input mixer handle
+ //
+ if (_inputMixerHandle != NULL) {
+ LOG(LS_VERBOSE) << "Closing record mixer";
+
+ LATE(snd_mixer_free)(_inputMixerHandle);
+ if (errVal < 0) {
+ LOG(LS_ERROR) << "Error freeing record mixer: "
+ << LATE(snd_strerror)(errVal);
+ }
+ LOG(LS_VERBOSE) << "Closing record mixer";
+
+ errVal = LATE(snd_mixer_detach)(_inputMixerHandle, _inputMixerStr);
+ if (errVal < 0) {
+ LOG(LS_ERROR) << "Error detaching record mixer: "
+ << LATE(snd_strerror)(errVal);
+ }
+ LOG(LS_VERBOSE) << "Closing record mixer";
+
+ errVal = LATE(snd_mixer_close)(_inputMixerHandle);
+ if (errVal < 0) {
+ LOG(LS_ERROR) << "Error snd_mixer_close(handleMixer) errVal=" << errVal;
+ }
+ LOG(LS_VERBOSE) << "Closing record mixer";
+ }
+ _inputMixerHandle = NULL;
+ _inputMixerElement = NULL;
+
+ errVal = LATE(snd_mixer_open)(&_inputMixerHandle, 0);
+ if (errVal < 0) {
+ LOG(LS_ERROR) << "snd_mixer_open(&_inputMixerHandle, 0) - error";
+ return -1;
+ }
+
+ char controlName[kAdmMaxDeviceNameSize] = {0};
+ GetControlName(controlName, deviceName);
+
+ LOG(LS_VERBOSE) << "snd_mixer_attach(_inputMixerHandle, " << controlName
+ << ")";
+
+ errVal = LATE(snd_mixer_attach)(_inputMixerHandle, controlName);
+ if (errVal < 0) {
+ LOG(LS_ERROR) << "snd_mixer_attach(_inputMixerHandle, " << controlName
+ << ") error: " << LATE(snd_strerror)(errVal);
+
+ _inputMixerHandle = NULL;
+ return -1;
+ }
+ strcpy(_inputMixerStr, controlName);
+
+ errVal = LATE(snd_mixer_selem_register)(_inputMixerHandle, NULL, NULL);
+ if (errVal < 0) {
+ LOG(LS_ERROR) << "snd_mixer_selem_register(_inputMixerHandle, NULL, NULL), "
+ << "error: " << LATE(snd_strerror)(errVal);
+
+ _inputMixerHandle = NULL;
+ return -1;
+ }
+ // Load and find the proper mixer element
+ if (LoadMicMixerElement() < 0) {
+ return -1;
+ }
+
+ if (_inputMixerHandle != NULL) {
+ LOG(LS_VERBOSE) << "the input mixer device is now open ("
+ << _inputMixerHandle << ")";
+ }
+
+ return 0;
}
-int32_t AudioMixerManagerLinuxALSA::SetSpeakerVolume(
- uint32_t volume)
-{
- LOG(LS_VERBOSE) << "AudioMixerManagerLinuxALSA::SetSpeakerVolume(volume="
- << volume << ")";
+bool AudioMixerManagerLinuxALSA::SpeakerIsInitialized() const {
+ LOG(LS_INFO) << __FUNCTION__;
- rtc::CritScope lock(&_critSect);
-
- if (_outputMixerElement == NULL)
- {
- LOG(LS_WARNING) << "no avaliable output mixer element exists";
- return -1;
- }
-
- int errVal =
- LATE(snd_mixer_selem_set_playback_volume_all)(_outputMixerElement,
- volume);
- if (errVal < 0)
- {
- LOG(LS_ERROR) << "Error changing master volume: "
- << LATE(snd_strerror)(errVal);
- return -1;
- }
-
- return (0);
+ return (_outputMixerHandle != NULL);
}
-int32_t AudioMixerManagerLinuxALSA::SpeakerVolume(
- uint32_t& volume) const
-{
+bool AudioMixerManagerLinuxALSA::MicrophoneIsInitialized() const {
+ LOG(LS_INFO) << __FUNCTION__;
- if (_outputMixerElement == NULL)
- {
- LOG(LS_WARNING) << "no avaliable output mixer element exists";
- return -1;
- }
+ return (_inputMixerHandle != NULL);
+}
- long int vol(0);
+int32_t AudioMixerManagerLinuxALSA::SetSpeakerVolume(uint32_t volume) {
+ LOG(LS_VERBOSE) << "AudioMixerManagerLinuxALSA::SetSpeakerVolume(volume="
+ << volume << ")";
- int
- errVal = LATE(snd_mixer_selem_get_playback_volume)(
- _outputMixerElement,
- (snd_mixer_selem_channel_id_t) 0,
- &vol);
- if (errVal < 0)
- {
- LOG(LS_ERROR) << "Error getting outputvolume: "
- << LATE(snd_strerror)(errVal);
- return -1;
- }
- LOG(LS_VERBOSE) << "AudioMixerManagerLinuxALSA::SpeakerVolume() => vol="
- << vol;
+ rtc::CritScope lock(&_critSect);
- volume = static_cast<uint32_t> (vol);
+ if (_outputMixerElement == NULL) {
+ LOG(LS_WARNING) << "no avaliable output mixer element exists";
+ return -1;
+ }
- return 0;
+ int errVal = LATE(snd_mixer_selem_set_playback_volume_all)(
+ _outputMixerElement, volume);
+ if (errVal < 0) {
+ LOG(LS_ERROR) << "Error changing master volume: "
+ << LATE(snd_strerror)(errVal);
+ return -1;
+ }
+
+ return (0);
+}
+
+int32_t AudioMixerManagerLinuxALSA::SpeakerVolume(uint32_t& volume) const {
+ if (_outputMixerElement == NULL) {
+ LOG(LS_WARNING) << "no avaliable output mixer element exists";
+ return -1;
+ }
+
+ long int vol(0);
+
+ int errVal = LATE(snd_mixer_selem_get_playback_volume)(
+ _outputMixerElement, (snd_mixer_selem_channel_id_t)0, &vol);
+ if (errVal < 0) {
+ LOG(LS_ERROR) << "Error getting outputvolume: "
+ << LATE(snd_strerror)(errVal);
+ return -1;
+ }
+ LOG(LS_VERBOSE) << "AudioMixerManagerLinuxALSA::SpeakerVolume() => vol="
+ << vol;
+
+ volume = static_cast<uint32_t>(vol);
+
+ return 0;
}
int32_t AudioMixerManagerLinuxALSA::MaxSpeakerVolume(
- uint32_t& maxVolume) const
-{
+ uint32_t& maxVolume) const {
+ if (_outputMixerElement == NULL) {
+ LOG(LS_WARNING) << "no avilable output mixer element exists";
+ return -1;
+ }
- if (_outputMixerElement == NULL)
- {
- LOG(LS_WARNING) << "no avilable output mixer element exists";
- return -1;
- }
+ long int minVol(0);
+ long int maxVol(0);
- long int minVol(0);
- long int maxVol(0);
+ int errVal = LATE(snd_mixer_selem_get_playback_volume_range)(
+ _outputMixerElement, &minVol, &maxVol);
- int errVal =
- LATE(snd_mixer_selem_get_playback_volume_range)(_outputMixerElement,
- &minVol, &maxVol);
+ LOG(LS_VERBOSE) << "Playout hardware volume range, min: " << minVol
+ << ", max: " << maxVol;
- LOG(LS_VERBOSE) << "Playout hardware volume range, min: " << minVol
- << ", max: " << maxVol;
+ if (maxVol <= minVol) {
+ LOG(LS_ERROR) << "Error getting get_playback_volume_range: "
+ << LATE(snd_strerror)(errVal);
+ }
- if (maxVol <= minVol)
- {
- LOG(LS_ERROR) << "Error getting get_playback_volume_range: "
- << LATE(snd_strerror)(errVal);
- }
+ maxVolume = static_cast<uint32_t>(maxVol);
- maxVolume = static_cast<uint32_t> (maxVol);
-
- return 0;
+ return 0;
}
int32_t AudioMixerManagerLinuxALSA::MinSpeakerVolume(
- uint32_t& minVolume) const
-{
+ uint32_t& minVolume) const {
+ if (_outputMixerElement == NULL) {
+ LOG(LS_WARNING) << "no avaliable output mixer element exists";
+ return -1;
+ }
- if (_outputMixerElement == NULL)
- {
- LOG(LS_WARNING) << "no avaliable output mixer element exists";
- return -1;
- }
+ long int minVol(0);
+ long int maxVol(0);
- long int minVol(0);
- long int maxVol(0);
+ int errVal = LATE(snd_mixer_selem_get_playback_volume_range)(
+ _outputMixerElement, &minVol, &maxVol);
- int errVal =
- LATE(snd_mixer_selem_get_playback_volume_range)(_outputMixerElement,
- &minVol, &maxVol);
+ LOG(LS_VERBOSE) << "Playout hardware volume range, min: " << minVol
+ << ", max: " << maxVol;
- LOG(LS_VERBOSE) << "Playout hardware volume range, min: " << minVol
- << ", max: " << maxVol;
+ if (maxVol <= minVol) {
+ LOG(LS_ERROR) << "Error getting get_playback_volume_range: "
+ << LATE(snd_strerror)(errVal);
+ }
- if (maxVol <= minVol)
- {
- LOG(LS_ERROR) << "Error getting get_playback_volume_range: "
- << LATE(snd_strerror)(errVal);
- }
+ minVolume = static_cast<uint32_t>(minVol);
- minVolume = static_cast<uint32_t> (minVol);
-
- return 0;
+ return 0;
}
// TL: Have done testnig with these but they don't seem reliable and
@@ -534,239 +469,195 @@
}
*/
-int32_t AudioMixerManagerLinuxALSA::SpeakerVolumeIsAvailable(
- bool& available)
-{
- if (_outputMixerElement == NULL)
- {
- LOG(LS_WARNING) << "no avaliable output mixer element exists";
- return -1;
- }
+int32_t AudioMixerManagerLinuxALSA::SpeakerVolumeIsAvailable(bool& available) {
+ if (_outputMixerElement == NULL) {
+ LOG(LS_WARNING) << "no avaliable output mixer element exists";
+ return -1;
+ }
- available = LATE(snd_mixer_selem_has_playback_volume)(_outputMixerElement);
+ available = LATE(snd_mixer_selem_has_playback_volume)(_outputMixerElement);
- return 0;
+ return 0;
}
-int32_t AudioMixerManagerLinuxALSA::SpeakerMuteIsAvailable(
- bool& available)
-{
- if (_outputMixerElement == NULL)
- {
- LOG(LS_WARNING) << "no avaliable output mixer element exists";
- return -1;
- }
+int32_t AudioMixerManagerLinuxALSA::SpeakerMuteIsAvailable(bool& available) {
+ if (_outputMixerElement == NULL) {
+ LOG(LS_WARNING) << "no avaliable output mixer element exists";
+ return -1;
+ }
- available = LATE(snd_mixer_selem_has_playback_switch)(_outputMixerElement);
+ available = LATE(snd_mixer_selem_has_playback_switch)(_outputMixerElement);
- return 0;
+ return 0;
}
-int32_t AudioMixerManagerLinuxALSA::SetSpeakerMute(bool enable)
-{
- LOG(LS_VERBOSE) << "AudioMixerManagerLinuxALSA::SetSpeakerMute(enable="
- << enable << ")";
+int32_t AudioMixerManagerLinuxALSA::SetSpeakerMute(bool enable) {
+ LOG(LS_VERBOSE) << "AudioMixerManagerLinuxALSA::SetSpeakerMute(enable="
+ << enable << ")";
- rtc::CritScope lock(&_critSect);
+ rtc::CritScope lock(&_critSect);
- if (_outputMixerElement == NULL)
- {
- LOG(LS_WARNING) << "no avaliable output mixer element exists";
- return -1;
- }
+ if (_outputMixerElement == NULL) {
+ LOG(LS_WARNING) << "no avaliable output mixer element exists";
+ return -1;
+ }
- // Ensure that the selected speaker destination has a valid mute control.
- bool available(false);
- SpeakerMuteIsAvailable(available);
- if (!available)
- {
- LOG(LS_WARNING) << "it is not possible to mute the speaker";
- return -1;
- }
+ // Ensure that the selected speaker destination has a valid mute control.
+ bool available(false);
+ SpeakerMuteIsAvailable(available);
+ if (!available) {
+ LOG(LS_WARNING) << "it is not possible to mute the speaker";
+ return -1;
+ }
- // Note value = 0 (off) means muted
- int errVal =
- LATE(snd_mixer_selem_set_playback_switch_all)(_outputMixerElement,
- !enable);
- if (errVal < 0)
- {
- LOG(LS_ERROR) << "Error setting playback switch: "
- << LATE(snd_strerror)(errVal);
- return -1;
- }
+ // Note value = 0 (off) means muted
+ int errVal = LATE(snd_mixer_selem_set_playback_switch_all)(
+ _outputMixerElement, !enable);
+ if (errVal < 0) {
+ LOG(LS_ERROR) << "Error setting playback switch: "
+ << LATE(snd_strerror)(errVal);
+ return -1;
+ }
- return (0);
+ return (0);
}
-int32_t AudioMixerManagerLinuxALSA::SpeakerMute(bool& enabled) const
-{
+int32_t AudioMixerManagerLinuxALSA::SpeakerMute(bool& enabled) const {
+ if (_outputMixerElement == NULL) {
+ LOG(LS_WARNING) << "no avaliable output mixer exists";
+ return -1;
+ }
- if (_outputMixerElement == NULL)
- {
- LOG(LS_WARNING) << "no avaliable output mixer exists";
- return -1;
- }
+ // Ensure that the selected speaker destination has a valid mute control.
+ bool available =
+ LATE(snd_mixer_selem_has_playback_switch)(_outputMixerElement);
+ if (!available) {
+ LOG(LS_WARNING) << "it is not possible to mute the speaker";
+ return -1;
+ }
- // Ensure that the selected speaker destination has a valid mute control.
- bool available =
- LATE(snd_mixer_selem_has_playback_switch)(_outputMixerElement);
- if (!available)
- {
- LOG(LS_WARNING) << "it is not possible to mute the speaker";
- return -1;
- }
+ int value(false);
- int value(false);
+ // Retrieve one boolean control value for a specified mute-control
+ //
+ int errVal = LATE(snd_mixer_selem_get_playback_switch)(
+ _outputMixerElement, (snd_mixer_selem_channel_id_t)0, &value);
+ if (errVal < 0) {
+ LOG(LS_ERROR) << "Error getting playback switch: "
+ << LATE(snd_strerror)(errVal);
+ return -1;
+ }
- // Retrieve one boolean control value for a specified mute-control
- //
- int
- errVal = LATE(snd_mixer_selem_get_playback_switch)(
- _outputMixerElement,
- (snd_mixer_selem_channel_id_t) 0,
- &value);
- if (errVal < 0)
- {
- LOG(LS_ERROR) << "Error getting playback switch: "
- << LATE(snd_strerror)(errVal);
- return -1;
- }
+ // Note value = 0 (off) means muted
+ enabled = (bool)!value;
- // Note value = 0 (off) means muted
- enabled = (bool) !value;
-
- return 0;
+ return 0;
}
-int32_t AudioMixerManagerLinuxALSA::MicrophoneMuteIsAvailable(
- bool& available)
-{
- if (_inputMixerElement == NULL)
- {
- LOG(LS_WARNING) << "no avaliable input mixer element exists";
- return -1;
- }
+int32_t AudioMixerManagerLinuxALSA::MicrophoneMuteIsAvailable(bool& available) {
+ if (_inputMixerElement == NULL) {
+ LOG(LS_WARNING) << "no avaliable input mixer element exists";
+ return -1;
+ }
- available = LATE(snd_mixer_selem_has_capture_switch)(_inputMixerElement);
- return 0;
+ available = LATE(snd_mixer_selem_has_capture_switch)(_inputMixerElement);
+ return 0;
}
-int32_t AudioMixerManagerLinuxALSA::SetMicrophoneMute(bool enable)
-{
- LOG(LS_VERBOSE) << "AudioMixerManagerLinuxALSA::SetMicrophoneMute(enable="
- << enable << ")";
+int32_t AudioMixerManagerLinuxALSA::SetMicrophoneMute(bool enable) {
+ LOG(LS_VERBOSE) << "AudioMixerManagerLinuxALSA::SetMicrophoneMute(enable="
+ << enable << ")";
- rtc::CritScope lock(&_critSect);
+ rtc::CritScope lock(&_critSect);
- if (_inputMixerElement == NULL)
- {
- LOG(LS_WARNING) << "no avaliable input mixer element exists";
- return -1;
- }
+ if (_inputMixerElement == NULL) {
+ LOG(LS_WARNING) << "no avaliable input mixer element exists";
+ return -1;
+ }
- // Ensure that the selected microphone destination has a valid mute control.
- bool available(false);
- MicrophoneMuteIsAvailable(available);
- if (!available)
- {
- LOG(LS_WARNING) << "it is not possible to mute the microphone";
- return -1;
- }
+ // Ensure that the selected microphone destination has a valid mute control.
+ bool available(false);
+ MicrophoneMuteIsAvailable(available);
+ if (!available) {
+ LOG(LS_WARNING) << "it is not possible to mute the microphone";
+ return -1;
+ }
- // Note value = 0 (off) means muted
- int errVal =
- LATE(snd_mixer_selem_set_capture_switch_all)(_inputMixerElement,
- !enable);
- if (errVal < 0)
- {
- LOG(LS_ERROR) << "Error setting capture switch: "
- << LATE(snd_strerror)(errVal);
- return -1;
- }
+ // Note value = 0 (off) means muted
+ int errVal =
+ LATE(snd_mixer_selem_set_capture_switch_all)(_inputMixerElement, !enable);
+ if (errVal < 0) {
+ LOG(LS_ERROR) << "Error setting capture switch: "
+ << LATE(snd_strerror)(errVal);
+ return -1;
+ }
- return (0);
+ return (0);
}
-int32_t AudioMixerManagerLinuxALSA::MicrophoneMute(bool& enabled) const
-{
+int32_t AudioMixerManagerLinuxALSA::MicrophoneMute(bool& enabled) const {
+ if (_inputMixerElement == NULL) {
+ LOG(LS_WARNING) << "no avaliable input mixer exists";
+ return -1;
+ }
- if (_inputMixerElement == NULL)
- {
- LOG(LS_WARNING) << "no avaliable input mixer exists";
- return -1;
- }
+ // Ensure that the selected microphone destination has a valid mute control.
+ bool available = LATE(snd_mixer_selem_has_capture_switch)(_inputMixerElement);
+ if (!available) {
+ LOG(LS_WARNING) << "it is not possible to mute the microphone";
+ return -1;
+ }
- // Ensure that the selected microphone destination has a valid mute control.
- bool available =
- LATE(snd_mixer_selem_has_capture_switch)(_inputMixerElement);
- if (!available)
- {
- LOG(LS_WARNING) << "it is not possible to mute the microphone";
- return -1;
- }
+ int value(false);
- int value(false);
+ // Retrieve one boolean control value for a specified mute-control
+ //
+ int errVal = LATE(snd_mixer_selem_get_capture_switch)(
+ _inputMixerElement, (snd_mixer_selem_channel_id_t)0, &value);
+ if (errVal < 0) {
+ LOG(LS_ERROR) << "Error getting capture switch: "
+ << LATE(snd_strerror)(errVal);
+ return -1;
+ }
- // Retrieve one boolean control value for a specified mute-control
- //
- int
- errVal = LATE(snd_mixer_selem_get_capture_switch)(
- _inputMixerElement,
- (snd_mixer_selem_channel_id_t) 0,
- &value);
- if (errVal < 0)
- {
- LOG(LS_ERROR) << "Error getting capture switch: "
- << LATE(snd_strerror)(errVal);
- return -1;
- }
+ // Note value = 0 (off) means muted
+ enabled = (bool)!value;
- // Note value = 0 (off) means muted
- enabled = (bool) !value;
-
- return 0;
+ return 0;
}
int32_t AudioMixerManagerLinuxALSA::MicrophoneVolumeIsAvailable(
- bool& available)
-{
- if (_inputMixerElement == NULL)
- {
- LOG(LS_WARNING) << "no avaliable input mixer element exists";
- return -1;
- }
+ bool& available) {
+ if (_inputMixerElement == NULL) {
+ LOG(LS_WARNING) << "no avaliable input mixer element exists";
+ return -1;
+ }
- available = LATE(snd_mixer_selem_has_capture_volume)(_inputMixerElement);
+ available = LATE(snd_mixer_selem_has_capture_volume)(_inputMixerElement);
- return 0;
+ return 0;
}
-int32_t AudioMixerManagerLinuxALSA::SetMicrophoneVolume(
- uint32_t volume)
-{
- LOG(LS_VERBOSE) << "AudioMixerManagerLinuxALSA::SetMicrophoneVolume(volume="
- << volume << ")";
+int32_t AudioMixerManagerLinuxALSA::SetMicrophoneVolume(uint32_t volume) {
+ LOG(LS_VERBOSE) << "AudioMixerManagerLinuxALSA::SetMicrophoneVolume(volume="
+ << volume << ")";
- rtc::CritScope lock(&_critSect);
+ rtc::CritScope lock(&_critSect);
- if (_inputMixerElement == NULL)
- {
- LOG(LS_WARNING) << "no avaliable input mixer element exists";
- return -1;
- }
+ if (_inputMixerElement == NULL) {
+ LOG(LS_WARNING) << "no avaliable input mixer element exists";
+ return -1;
+ }
- int
- errVal =
- LATE(snd_mixer_selem_set_capture_volume_all)(_inputMixerElement,
- volume);
- if (errVal < 0)
- {
- LOG(LS_ERROR) << "Error changing microphone volume: "
- << LATE(snd_strerror)(errVal);
- return -1;
- }
+ int errVal =
+ LATE(snd_mixer_selem_set_capture_volume_all)(_inputMixerElement, volume);
+ if (errVal < 0) {
+ LOG(LS_ERROR) << "Error changing microphone volume: "
+ << LATE(snd_strerror)(errVal);
+ return -1;
+ }
- return (0);
+ return (0);
}
// TL: Have done testnig with these but they don't seem reliable and
@@ -799,8 +690,8 @@
maxVol = (long int)maxVolume;
printf("min %d max %d", minVol, maxVol);
- errVal = snd_mixer_selem_set_capture_volume_range(_inputMixerElement, minVol, maxVol);
- LOG(LS_VERBOSE) << "Capture hardware volume range, min: " << minVol
+ errVal = snd_mixer_selem_set_capture_volume_range(_inputMixerElement, minVol,
+ maxVol); LOG(LS_VERBOSE) << "Capture hardware volume range, min: " << minVol
<< ", max: " << maxVol;
if (errVal != 0)
{
@@ -855,263 +746,220 @@
}
*/
-int32_t AudioMixerManagerLinuxALSA::MicrophoneVolume(
- uint32_t& volume) const
-{
+int32_t AudioMixerManagerLinuxALSA::MicrophoneVolume(uint32_t& volume) const {
+ if (_inputMixerElement == NULL) {
+ LOG(LS_WARNING) << "no avaliable input mixer element exists";
+ return -1;
+ }
- if (_inputMixerElement == NULL)
- {
- LOG(LS_WARNING) << "no avaliable input mixer element exists";
- return -1;
- }
+ long int vol(0);
- long int vol(0);
+ int errVal = LATE(snd_mixer_selem_get_capture_volume)(
+ _inputMixerElement, (snd_mixer_selem_channel_id_t)0, &vol);
+ if (errVal < 0) {
+ LOG(LS_ERROR) << "Error getting inputvolume: "
+ << LATE(snd_strerror)(errVal);
+ return -1;
+ }
+ LOG(LS_VERBOSE) << "AudioMixerManagerLinuxALSA::MicrophoneVolume() => vol="
+ << vol;
- int
- errVal =
- LATE(snd_mixer_selem_get_capture_volume)(
- _inputMixerElement,
- (snd_mixer_selem_channel_id_t) 0,
- &vol);
- if (errVal < 0)
- {
- LOG(LS_ERROR) << "Error getting inputvolume: "
- << LATE(snd_strerror)(errVal);
- return -1;
- }
- LOG(LS_VERBOSE) << "AudioMixerManagerLinuxALSA::MicrophoneVolume() => vol="
- << vol;
+ volume = static_cast<uint32_t>(vol);
- volume = static_cast<uint32_t> (vol);
-
- return 0;
+ return 0;
}
int32_t AudioMixerManagerLinuxALSA::MaxMicrophoneVolume(
- uint32_t& maxVolume) const
-{
+ uint32_t& maxVolume) const {
+ if (_inputMixerElement == NULL) {
+ LOG(LS_WARNING) << "no avaliable input mixer element exists";
+ return -1;
+ }
- if (_inputMixerElement == NULL)
- {
- LOG(LS_WARNING) << "no avaliable input mixer element exists";
- return -1;
- }
+ long int minVol(0);
+ long int maxVol(0);
- long int minVol(0);
- long int maxVol(0);
+ // check if we have mic volume at all
+ if (!LATE(snd_mixer_selem_has_capture_volume)(_inputMixerElement)) {
+ LOG(LS_ERROR) << "No microphone volume available";
+ return -1;
+ }
- // check if we have mic volume at all
- if (!LATE(snd_mixer_selem_has_capture_volume)(_inputMixerElement))
- {
- LOG(LS_ERROR) << "No microphone volume available";
- return -1;
- }
+ int errVal = LATE(snd_mixer_selem_get_capture_volume_range)(
+ _inputMixerElement, &minVol, &maxVol);
- int errVal =
- LATE(snd_mixer_selem_get_capture_volume_range)(_inputMixerElement,
- &minVol, &maxVol);
+ LOG(LS_VERBOSE) << "Microphone hardware volume range, min: " << minVol
+ << ", max: " << maxVol;
+ if (maxVol <= minVol) {
+ LOG(LS_ERROR) << "Error getting microphone volume range: "
+ << LATE(snd_strerror)(errVal);
+ }
- LOG(LS_VERBOSE) << "Microphone hardware volume range, min: " << minVol
- << ", max: " << maxVol;
- if (maxVol <= minVol)
- {
- LOG(LS_ERROR) << "Error getting microphone volume range: "
- << LATE(snd_strerror)(errVal);
- }
+ maxVolume = static_cast<uint32_t>(maxVol);
- maxVolume = static_cast<uint32_t> (maxVol);
-
- return 0;
+ return 0;
}
int32_t AudioMixerManagerLinuxALSA::MinMicrophoneVolume(
- uint32_t& minVolume) const
-{
+ uint32_t& minVolume) const {
+ if (_inputMixerElement == NULL) {
+ LOG(LS_WARNING) << "no avaliable input mixer element exists";
+ return -1;
+ }
- if (_inputMixerElement == NULL)
- {
- LOG(LS_WARNING) << "no avaliable input mixer element exists";
- return -1;
- }
+ long int minVol(0);
+ long int maxVol(0);
- long int minVol(0);
- long int maxVol(0);
+ int errVal = LATE(snd_mixer_selem_get_capture_volume_range)(
+ _inputMixerElement, &minVol, &maxVol);
- int errVal =
- LATE(snd_mixer_selem_get_capture_volume_range)(_inputMixerElement,
- &minVol, &maxVol);
+ LOG(LS_VERBOSE) << "Microphone hardware volume range, min: " << minVol
+ << ", max: " << maxVol;
+ if (maxVol <= minVol) {
+ LOG(LS_ERROR) << "Error getting microphone volume range: "
+ << LATE(snd_strerror)(errVal);
+ }
- LOG(LS_VERBOSE) << "Microphone hardware volume range, min: " << minVol
- << ", max: " << maxVol;
- if (maxVol <= minVol)
- {
- LOG(LS_ERROR) << "Error getting microphone volume range: "
- << LATE(snd_strerror)(errVal);
- }
+ minVolume = static_cast<uint32_t>(minVol);
- minVolume = static_cast<uint32_t> (minVol);
-
- return 0;
+ return 0;
}
// ============================================================================
// Private Methods
// ============================================================================
-int32_t AudioMixerManagerLinuxALSA::LoadMicMixerElement() const
-{
- int errVal = LATE(snd_mixer_load)(_inputMixerHandle);
- if (errVal < 0)
- {
- LOG(LS_ERROR) << "snd_mixer_load(_inputMixerHandle), error: "
- << LATE(snd_strerror)(errVal);
- _inputMixerHandle = NULL;
- return -1;
+int32_t AudioMixerManagerLinuxALSA::LoadMicMixerElement() const {
+ int errVal = LATE(snd_mixer_load)(_inputMixerHandle);
+ if (errVal < 0) {
+ LOG(LS_ERROR) << "snd_mixer_load(_inputMixerHandle), error: "
+ << LATE(snd_strerror)(errVal);
+ _inputMixerHandle = NULL;
+ return -1;
+ }
+
+ snd_mixer_elem_t* elem = NULL;
+ snd_mixer_elem_t* micElem = NULL;
+ unsigned mixerIdx = 0;
+ const char* selemName = NULL;
+
+ // Find and store handles to the right mixer elements
+ for (elem = LATE(snd_mixer_first_elem)(_inputMixerHandle); elem;
+ elem = LATE(snd_mixer_elem_next)(elem), mixerIdx++) {
+ if (LATE(snd_mixer_selem_is_active)(elem)) {
+ selemName = LATE(snd_mixer_selem_get_name)(elem);
+ if (strcmp(selemName, "Capture") == 0) // "Capture", "Mic"
+ {
+ _inputMixerElement = elem;
+ LOG(LS_VERBOSE) << "Capture element set";
+ } else if (strcmp(selemName, "Mic") == 0) {
+ micElem = elem;
+ LOG(LS_VERBOSE) << "Mic element found";
+ }
}
- snd_mixer_elem_t *elem = NULL;
- snd_mixer_elem_t *micElem = NULL;
- unsigned mixerIdx = 0;
- const char *selemName = NULL;
-
- // Find and store handles to the right mixer elements
- for (elem = LATE(snd_mixer_first_elem)(_inputMixerHandle); elem; elem
- = LATE(snd_mixer_elem_next)(elem), mixerIdx++)
- {
- if (LATE(snd_mixer_selem_is_active)(elem))
- {
- selemName = LATE(snd_mixer_selem_get_name)(elem);
- if (strcmp(selemName, "Capture") == 0) // "Capture", "Mic"
- {
- _inputMixerElement = elem;
- LOG(LS_VERBOSE) << "Capture element set";
- } else if (strcmp(selemName, "Mic") == 0)
- {
- micElem = elem;
- LOG(LS_VERBOSE) << "Mic element found";
- }
- }
-
- if (_inputMixerElement)
- {
- // Use the first Capture element that is found
- // The second one may not work
- break;
- }
+ if (_inputMixerElement) {
+ // Use the first Capture element that is found
+ // The second one may not work
+ break;
}
+ }
- if (_inputMixerElement == NULL)
- {
- // We didn't find a Capture handle, use Mic.
- if (micElem != NULL)
- {
- _inputMixerElement = micElem;
- LOG(LS_VERBOSE) << "Using Mic as capture volume.";
- } else
- {
- _inputMixerElement = NULL;
- LOG(LS_ERROR) << "Could not find capture volume on the mixer.";
+ if (_inputMixerElement == NULL) {
+ // We didn't find a Capture handle, use Mic.
+ if (micElem != NULL) {
+ _inputMixerElement = micElem;
+ LOG(LS_VERBOSE) << "Using Mic as capture volume.";
+ } else {
+ _inputMixerElement = NULL;
+ LOG(LS_ERROR) << "Could not find capture volume on the mixer.";
- return -1;
- }
+ return -1;
}
+ }
- return 0;
+ return 0;
}
-int32_t AudioMixerManagerLinuxALSA::LoadSpeakerMixerElement() const
-{
- int errVal = LATE(snd_mixer_load)(_outputMixerHandle);
- if (errVal < 0)
- {
- LOG(LS_ERROR) << "snd_mixer_load(_outputMixerHandle), error: "
- << LATE(snd_strerror)(errVal);
- _outputMixerHandle = NULL;
- return -1;
+int32_t AudioMixerManagerLinuxALSA::LoadSpeakerMixerElement() const {
+ int errVal = LATE(snd_mixer_load)(_outputMixerHandle);
+ if (errVal < 0) {
+ LOG(LS_ERROR) << "snd_mixer_load(_outputMixerHandle), error: "
+ << LATE(snd_strerror)(errVal);
+ _outputMixerHandle = NULL;
+ return -1;
+ }
+
+ snd_mixer_elem_t* elem = NULL;
+ snd_mixer_elem_t* masterElem = NULL;
+ snd_mixer_elem_t* speakerElem = NULL;
+ unsigned mixerIdx = 0;
+ const char* selemName = NULL;
+
+ // Find and store handles to the right mixer elements
+ for (elem = LATE(snd_mixer_first_elem)(_outputMixerHandle); elem;
+ elem = LATE(snd_mixer_elem_next)(elem), mixerIdx++) {
+ if (LATE(snd_mixer_selem_is_active)(elem)) {
+ selemName = LATE(snd_mixer_selem_get_name)(elem);
+ LOG(LS_VERBOSE) << "snd_mixer_selem_get_name " << mixerIdx << ": "
+ << selemName << " =" << elem;
+
+ // "Master", "PCM", "Wave", "Master Mono", "PC Speaker", "PCM", "Wave"
+ if (strcmp(selemName, "PCM") == 0) {
+ _outputMixerElement = elem;
+ LOG(LS_VERBOSE) << "PCM element set";
+ } else if (strcmp(selemName, "Master") == 0) {
+ masterElem = elem;
+ LOG(LS_VERBOSE) << "Master element found";
+ } else if (strcmp(selemName, "Speaker") == 0) {
+ speakerElem = elem;
+ LOG(LS_VERBOSE) << "Speaker element found";
+ }
}
- snd_mixer_elem_t *elem = NULL;
- snd_mixer_elem_t *masterElem = NULL;
- snd_mixer_elem_t *speakerElem = NULL;
- unsigned mixerIdx = 0;
- const char *selemName = NULL;
-
- // Find and store handles to the right mixer elements
- for (elem = LATE(snd_mixer_first_elem)(_outputMixerHandle); elem; elem
- = LATE(snd_mixer_elem_next)(elem), mixerIdx++)
- {
- if (LATE(snd_mixer_selem_is_active)(elem))
- {
- selemName = LATE(snd_mixer_selem_get_name)(elem);
- LOG(LS_VERBOSE) << "snd_mixer_selem_get_name " << mixerIdx << ": "
- << selemName << " =" << elem;
-
- // "Master", "PCM", "Wave", "Master Mono", "PC Speaker", "PCM", "Wave"
- if (strcmp(selemName, "PCM") == 0)
- {
- _outputMixerElement = elem;
- LOG(LS_VERBOSE) << "PCM element set";
- } else if (strcmp(selemName, "Master") == 0)
- {
- masterElem = elem;
- LOG(LS_VERBOSE) << "Master element found";
- } else if (strcmp(selemName, "Speaker") == 0)
- {
- speakerElem = elem;
- LOG(LS_VERBOSE) << "Speaker element found";
- }
- }
-
- if (_outputMixerElement)
- {
- // We have found the element we want
- break;
- }
+ if (_outputMixerElement) {
+ // We have found the element we want
+ break;
}
+ }
- // If we didn't find a PCM Handle, use Master or Speaker
- if (_outputMixerElement == NULL)
- {
- if (masterElem != NULL)
- {
- _outputMixerElement = masterElem;
- LOG(LS_VERBOSE) << "Using Master as output volume.";
- } else if (speakerElem != NULL)
- {
- _outputMixerElement = speakerElem;
- LOG(LS_VERBOSE) << "Using Speaker as output volume.";
- } else
- {
- _outputMixerElement = NULL;
- LOG(LS_ERROR) << "Could not find output volume in the mixer.";
- return -1;
- }
+ // If we didn't find a PCM Handle, use Master or Speaker
+ if (_outputMixerElement == NULL) {
+ if (masterElem != NULL) {
+ _outputMixerElement = masterElem;
+ LOG(LS_VERBOSE) << "Using Master as output volume.";
+ } else if (speakerElem != NULL) {
+ _outputMixerElement = speakerElem;
+ LOG(LS_VERBOSE) << "Using Speaker as output volume.";
+ } else {
+ _outputMixerElement = NULL;
+ LOG(LS_ERROR) << "Could not find output volume in the mixer.";
+ return -1;
}
+ }
- return 0;
+ return 0;
}
void AudioMixerManagerLinuxALSA::GetControlName(char* controlName,
- char* deviceName) const
-{
- // Example
- // deviceName: "front:CARD=Intel,DEV=0"
- // controlName: "hw:CARD=Intel"
- char* pos1 = strchr(deviceName, ':');
- char* pos2 = strchr(deviceName, ',');
- if (!pos2) {
- // Can also be default:CARD=Intel
- pos2 = &deviceName[strlen(deviceName)];
- }
- if (pos1 && pos2) {
- strcpy(controlName, "hw");
- int nChar = (int) (pos2 - pos1);
- strncpy(&controlName[2], pos1, nChar);
- controlName[2 + nChar] = '\0';
- } else {
- strcpy(controlName, deviceName);
- }
-
+ char* deviceName) const {
+ // Example
+ // deviceName: "front:CARD=Intel,DEV=0"
+ // controlName: "hw:CARD=Intel"
+ char* pos1 = strchr(deviceName, ':');
+ char* pos2 = strchr(deviceName, ',');
+ if (!pos2) {
+ // Can also be default:CARD=Intel
+ pos2 = &deviceName[strlen(deviceName)];
+ }
+ if (pos1 && pos2) {
+ strcpy(controlName, "hw");
+ int nChar = (int)(pos2 - pos1);
+ strncpy(&controlName[2], pos1, nChar);
+ controlName[2 + nChar] = '\0';
+ } else {
+ strcpy(controlName, deviceName);
+ }
}
-}
+} // namespace webrtc
diff --git a/modules/audio_device/linux/audio_mixer_manager_pulse_linux.cc b/modules/audio_device/linux/audio_mixer_manager_pulse_linux.cc
index 21f7fd0..80896c9 100644
--- a/modules/audio_device/linux/audio_mixer_manager_pulse_linux.cc
+++ b/modules/audio_device/linux/audio_mixer_manager_pulse_linux.cc
@@ -23,8 +23,7 @@
LATESYM_GET(webrtc::adm_linux_pulse::PulseAudioSymbolTable, &PaSymbolTable, \
sym)
-namespace webrtc
-{
+namespace webrtc {
class AutoPulseLock {
public:
@@ -33,38 +32,34 @@
LATE(pa_threaded_mainloop_lock)(pa_mainloop_);
}
- ~AutoPulseLock() {
- LATE(pa_threaded_mainloop_unlock)(pa_mainloop_);
- }
+ ~AutoPulseLock() { LATE(pa_threaded_mainloop_unlock)(pa_mainloop_); }
private:
pa_threaded_mainloop* const pa_mainloop_;
};
-AudioMixerManagerLinuxPulse::AudioMixerManagerLinuxPulse() :
- _paOutputDeviceIndex(-1),
- _paInputDeviceIndex(-1),
- _paPlayStream(NULL),
- _paRecStream(NULL),
- _paMainloop(NULL),
- _paContext(NULL),
- _paVolume(0),
- _paMute(0),
- _paVolSteps(0),
- _paSpeakerMute(false),
- _paSpeakerVolume(PA_VOLUME_NORM),
- _paChannels(0),
- _paObjectsSet(false)
-{
- LOG(LS_INFO) << __FUNCTION__ << " created";
+AudioMixerManagerLinuxPulse::AudioMixerManagerLinuxPulse()
+ : _paOutputDeviceIndex(-1),
+ _paInputDeviceIndex(-1),
+ _paPlayStream(NULL),
+ _paRecStream(NULL),
+ _paMainloop(NULL),
+ _paContext(NULL),
+ _paVolume(0),
+ _paMute(0),
+ _paVolSteps(0),
+ _paSpeakerMute(false),
+ _paSpeakerVolume(PA_VOLUME_NORM),
+ _paChannels(0),
+ _paObjectsSet(false) {
+ LOG(LS_INFO) << __FUNCTION__ << " created";
}
-AudioMixerManagerLinuxPulse::~AudioMixerManagerLinuxPulse()
-{
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
- LOG(LS_INFO) << __FUNCTION__ << " destroyed";
+AudioMixerManagerLinuxPulse::~AudioMixerManagerLinuxPulse() {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ LOG(LS_INFO) << __FUNCTION__ << " destroyed";
- Close();
+ Close();
}
// ===========================================================================
@@ -73,866 +68,734 @@
int32_t AudioMixerManagerLinuxPulse::SetPulseAudioObjects(
pa_threaded_mainloop* mainloop,
- pa_context* context)
-{
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
- LOG(LS_VERBOSE) << __FUNCTION__;
+ pa_context* context) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ LOG(LS_VERBOSE) << __FUNCTION__;
- if (!mainloop || !context)
- {
- LOG(LS_ERROR) << "could not set PulseAudio objects for mixer";
- return -1;
- }
+ if (!mainloop || !context) {
+ LOG(LS_ERROR) << "could not set PulseAudio objects for mixer";
+ return -1;
+ }
- _paMainloop = mainloop;
- _paContext = context;
- _paObjectsSet = true;
+ _paMainloop = mainloop;
+ _paContext = context;
+ _paObjectsSet = true;
- LOG(LS_VERBOSE) << "the PulseAudio objects for the mixer has been set";
+ LOG(LS_VERBOSE) << "the PulseAudio objects for the mixer has been set";
- return 0;
+ return 0;
}
-int32_t AudioMixerManagerLinuxPulse::Close()
-{
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
- LOG(LS_VERBOSE) << __FUNCTION__;
+int32_t AudioMixerManagerLinuxPulse::Close() {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ LOG(LS_VERBOSE) << __FUNCTION__;
- CloseSpeaker();
- CloseMicrophone();
+ CloseSpeaker();
+ CloseMicrophone();
- _paMainloop = NULL;
- _paContext = NULL;
- _paObjectsSet = false;
+ _paMainloop = NULL;
+ _paContext = NULL;
+ _paObjectsSet = false;
- return 0;
-
+ return 0;
}
-int32_t AudioMixerManagerLinuxPulse::CloseSpeaker()
-{
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
- LOG(LS_VERBOSE) << __FUNCTION__;
+int32_t AudioMixerManagerLinuxPulse::CloseSpeaker() {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ LOG(LS_VERBOSE) << __FUNCTION__;
- // Reset the index to -1
- _paOutputDeviceIndex = -1;
- _paPlayStream = NULL;
+ // Reset the index to -1
+ _paOutputDeviceIndex = -1;
+ _paPlayStream = NULL;
- return 0;
+ return 0;
}
-int32_t AudioMixerManagerLinuxPulse::CloseMicrophone()
-{
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
- LOG(LS_VERBOSE) << __FUNCTION__;
+int32_t AudioMixerManagerLinuxPulse::CloseMicrophone() {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ LOG(LS_VERBOSE) << __FUNCTION__;
- // Reset the index to -1
- _paInputDeviceIndex = -1;
- _paRecStream = NULL;
+ // Reset the index to -1
+ _paInputDeviceIndex = -1;
+ _paRecStream = NULL;
- return 0;
+ return 0;
}
-int32_t AudioMixerManagerLinuxPulse::SetPlayStream(pa_stream* playStream)
-{
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
- LOG(LS_VERBOSE) << "AudioMixerManagerLinuxPulse::SetPlayStream(playStream)";
+int32_t AudioMixerManagerLinuxPulse::SetPlayStream(pa_stream* playStream) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ LOG(LS_VERBOSE) << "AudioMixerManagerLinuxPulse::SetPlayStream(playStream)";
- _paPlayStream = playStream;
- return 0;
+ _paPlayStream = playStream;
+ return 0;
}
-int32_t AudioMixerManagerLinuxPulse::SetRecStream(pa_stream* recStream)
-{
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
- LOG(LS_VERBOSE) << "AudioMixerManagerLinuxPulse::SetRecStream(recStream)";
+int32_t AudioMixerManagerLinuxPulse::SetRecStream(pa_stream* recStream) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ LOG(LS_VERBOSE) << "AudioMixerManagerLinuxPulse::SetRecStream(recStream)";
- _paRecStream = recStream;
- return 0;
+ _paRecStream = recStream;
+ return 0;
}
-int32_t AudioMixerManagerLinuxPulse::OpenSpeaker(
- uint16_t deviceIndex)
-{
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
- LOG(LS_VERBOSE) << "AudioMixerManagerLinuxPulse::OpenSpeaker(deviceIndex="
- << deviceIndex << ")";
+int32_t AudioMixerManagerLinuxPulse::OpenSpeaker(uint16_t deviceIndex) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ LOG(LS_VERBOSE) << "AudioMixerManagerLinuxPulse::OpenSpeaker(deviceIndex="
+ << deviceIndex << ")";
- // No point in opening the speaker
- // if PA objects have not been set
- if (!_paObjectsSet)
- {
- LOG(LS_ERROR) << "PulseAudio objects has not been set";
- return -1;
- }
+ // No point in opening the speaker
+ // if PA objects have not been set
+ if (!_paObjectsSet) {
+ LOG(LS_ERROR) << "PulseAudio objects has not been set";
+ return -1;
+ }
- // Set the index for the PulseAudio
- // output device to control
- _paOutputDeviceIndex = deviceIndex;
+ // Set the index for the PulseAudio
+ // output device to control
+ _paOutputDeviceIndex = deviceIndex;
- LOG(LS_VERBOSE) << "the output mixer device is now open";
+ LOG(LS_VERBOSE) << "the output mixer device is now open";
- return 0;
+ return 0;
}
-int32_t AudioMixerManagerLinuxPulse::OpenMicrophone(
- uint16_t deviceIndex)
-{
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
- LOG(LS_VERBOSE)
- << "AudioMixerManagerLinuxPulse::OpenMicrophone(deviceIndex="
- << deviceIndex << ")";
+int32_t AudioMixerManagerLinuxPulse::OpenMicrophone(uint16_t deviceIndex) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ LOG(LS_VERBOSE) << "AudioMixerManagerLinuxPulse::OpenMicrophone(deviceIndex="
+ << deviceIndex << ")";
- // No point in opening the microphone
- // if PA objects have not been set
- if (!_paObjectsSet)
- {
- LOG(LS_ERROR) << "PulseAudio objects have not been set";
- return -1;
- }
+ // No point in opening the microphone
+ // if PA objects have not been set
+ if (!_paObjectsSet) {
+ LOG(LS_ERROR) << "PulseAudio objects have not been set";
+ return -1;
+ }
- // Set the index for the PulseAudio
- // input device to control
- _paInputDeviceIndex = deviceIndex;
+ // Set the index for the PulseAudio
+ // input device to control
+ _paInputDeviceIndex = deviceIndex;
- LOG(LS_VERBOSE) << "the input mixer device is now open";
+ LOG(LS_VERBOSE) << "the input mixer device is now open";
- return 0;
+ return 0;
}
-bool AudioMixerManagerLinuxPulse::SpeakerIsInitialized() const
-{
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
- LOG(LS_INFO) << __FUNCTION__;
+bool AudioMixerManagerLinuxPulse::SpeakerIsInitialized() const {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ LOG(LS_INFO) << __FUNCTION__;
- return (_paOutputDeviceIndex != -1);
+ return (_paOutputDeviceIndex != -1);
}
-bool AudioMixerManagerLinuxPulse::MicrophoneIsInitialized() const
-{
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
- LOG(LS_INFO) << __FUNCTION__;
+bool AudioMixerManagerLinuxPulse::MicrophoneIsInitialized() const {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ LOG(LS_INFO) << __FUNCTION__;
- return (_paInputDeviceIndex != -1);
+ return (_paInputDeviceIndex != -1);
}
-int32_t AudioMixerManagerLinuxPulse::SetSpeakerVolume(
- uint32_t volume)
-{
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
- LOG(LS_VERBOSE) << "AudioMixerManagerLinuxPulse::SetSpeakerVolume(volume="
- << volume << ")";
+int32_t AudioMixerManagerLinuxPulse::SetSpeakerVolume(uint32_t volume) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ LOG(LS_VERBOSE) << "AudioMixerManagerLinuxPulse::SetSpeakerVolume(volume="
+ << volume << ")";
- if (_paOutputDeviceIndex == -1)
- {
- LOG(LS_WARNING) << "output device index has not been set";
- return -1;
+ if (_paOutputDeviceIndex == -1) {
+ LOG(LS_WARNING) << "output device index has not been set";
+ return -1;
+ }
+
+ bool setFailed(false);
+
+ if (_paPlayStream &&
+ (LATE(pa_stream_get_state)(_paPlayStream) != PA_STREAM_UNCONNECTED)) {
+ // We can only really set the volume if we have a connected stream
+ AutoPulseLock auto_lock(_paMainloop);
+
+ // Get the number of channels from the sample specification
+ const pa_sample_spec* spec = LATE(pa_stream_get_sample_spec)(_paPlayStream);
+ if (!spec) {
+ LOG(LS_ERROR) << "could not get sample specification";
+ return -1;
}
- bool setFailed(false);
+ // Set the same volume for all channels
+ pa_cvolume cVolumes;
+ LATE(pa_cvolume_set)(&cVolumes, spec->channels, volume);
- if (_paPlayStream && (LATE(pa_stream_get_state)(_paPlayStream)
- != PA_STREAM_UNCONNECTED))
- {
- // We can only really set the volume if we have a connected stream
- AutoPulseLock auto_lock(_paMainloop);
-
- // Get the number of channels from the sample specification
- const pa_sample_spec *spec =
- LATE(pa_stream_get_sample_spec)(_paPlayStream);
- if (!spec)
- {
- LOG(LS_ERROR) << "could not get sample specification";
- return -1;
- }
-
- // Set the same volume for all channels
- pa_cvolume cVolumes;
- LATE(pa_cvolume_set)(&cVolumes, spec->channels, volume);
-
- pa_operation* paOperation = NULL;
- paOperation = LATE(pa_context_set_sink_input_volume)(
- _paContext,
- LATE(pa_stream_get_index)(_paPlayStream),
- &cVolumes,
- PaSetVolumeCallback, NULL);
- if (!paOperation)
- {
- setFailed = true;
- }
-
- // Don't need to wait for the completion
- LATE(pa_operation_unref)(paOperation);
- } else
- {
- // We have not created a stream or it's not connected to the sink
- // Save the volume to be set at connection
- _paSpeakerVolume = volume;
+ pa_operation* paOperation = NULL;
+ paOperation = LATE(pa_context_set_sink_input_volume)(
+ _paContext, LATE(pa_stream_get_index)(_paPlayStream), &cVolumes,
+ PaSetVolumeCallback, NULL);
+ if (!paOperation) {
+ setFailed = true;
}
- if (setFailed)
- {
- LOG(LS_WARNING) << "could not set speaker volume, error="
- << LATE(pa_context_errno)(_paContext);
+ // Don't need to wait for the completion
+ LATE(pa_operation_unref)(paOperation);
+ } else {
+ // We have not created a stream or it's not connected to the sink
+ // Save the volume to be set at connection
+ _paSpeakerVolume = volume;
+ }
- return -1;
- }
+ if (setFailed) {
+ LOG(LS_WARNING) << "could not set speaker volume, error="
+ << LATE(pa_context_errno)(_paContext);
- return 0;
+ return -1;
+ }
+
+ return 0;
}
-int32_t
-AudioMixerManagerLinuxPulse::SpeakerVolume(uint32_t& volume) const
-{
- if (_paOutputDeviceIndex == -1)
- {
- LOG(LS_WARNING) << "output device index has not been set";
- return -1;
- }
+int32_t AudioMixerManagerLinuxPulse::SpeakerVolume(uint32_t& volume) const {
+ if (_paOutputDeviceIndex == -1) {
+ LOG(LS_WARNING) << "output device index has not been set";
+ return -1;
+ }
- if (_paPlayStream && (LATE(pa_stream_get_state)(_paPlayStream)
- != PA_STREAM_UNCONNECTED))
- {
- // We can only get the volume if we have a connected stream
- if (!GetSinkInputInfo())
- return -1;
-
- AutoPulseLock auto_lock(_paMainloop);
- volume = static_cast<uint32_t> (_paVolume);
- } else
- {
- AutoPulseLock auto_lock(_paMainloop);
- volume = _paSpeakerVolume;
- }
-
- LOG(LS_VERBOSE) << "AudioMixerManagerLinuxPulse::SpeakerVolume() => vol="
- << volume;
-
- return 0;
-}
-
-int32_t
-AudioMixerManagerLinuxPulse::MaxSpeakerVolume(uint32_t& maxVolume) const
-{
-
- if (_paOutputDeviceIndex == -1)
- {
- LOG(LS_WARNING) << "output device index has not been set";
- return -1;
- }
-
- // PA_VOLUME_NORM corresponds to 100% (0db)
- // but PA allows up to 150 db amplification
- maxVolume = static_cast<uint32_t> (PA_VOLUME_NORM);
-
- return 0;
-}
-
-int32_t
-AudioMixerManagerLinuxPulse::MinSpeakerVolume(uint32_t& minVolume) const
-{
-
- if (_paOutputDeviceIndex == -1)
- {
- LOG(LS_WARNING) << "output device index has not been set";
- return -1;
- }
-
- minVolume = static_cast<uint32_t> (PA_VOLUME_MUTED);
-
- return 0;
-}
-
-int32_t
-AudioMixerManagerLinuxPulse::SpeakerVolumeIsAvailable(bool& available)
-{
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
- if (_paOutputDeviceIndex == -1)
- {
- LOG(LS_WARNING) << "output device index has not been set";
- return -1;
- }
-
- // Always available in Pulse Audio
- available = true;
-
- return 0;
-}
-
-int32_t
-AudioMixerManagerLinuxPulse::SpeakerMuteIsAvailable(bool& available)
-{
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
- if (_paOutputDeviceIndex == -1)
- {
- LOG(LS_WARNING) << "output device index has not been set";
- return -1;
- }
-
- // Always available in Pulse Audio
- available = true;
-
- return 0;
-}
-
-int32_t AudioMixerManagerLinuxPulse::SetSpeakerMute(bool enable)
-{
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
- LOG(LS_VERBOSE) << "AudioMixerManagerLinuxPulse::SetSpeakerMute(enable="
- << enable << ")";
-
- if (_paOutputDeviceIndex == -1)
- {
- LOG(LS_WARNING) << "output device index has not been set";
- return -1;
- }
-
- bool setFailed(false);
-
- if (_paPlayStream && (LATE(pa_stream_get_state)(_paPlayStream)
- != PA_STREAM_UNCONNECTED))
- {
- // We can only really mute if we have a connected stream
- AutoPulseLock auto_lock(_paMainloop);
-
- pa_operation* paOperation = NULL;
- paOperation = LATE(pa_context_set_sink_input_mute)(
- _paContext,
- LATE(pa_stream_get_index)(_paPlayStream),
- (int) enable,
- PaSetVolumeCallback,
- NULL);
- if (!paOperation)
- {
- setFailed = true;
- }
-
- // Don't need to wait for the completion
- LATE(pa_operation_unref)(paOperation);
- } else
- {
- // We have not created a stream or it's not connected to the sink
- // Save the mute status to be set at connection
- _paSpeakerMute = enable;
- }
-
- if (setFailed)
- {
- LOG(LS_WARNING) << "could not mute speaker, error="
- << LATE(pa_context_errno)(_paContext);
- return -1;
- }
-
- return 0;
-}
-
-int32_t AudioMixerManagerLinuxPulse::SpeakerMute(bool& enabled) const
-{
-
- if (_paOutputDeviceIndex == -1)
- {
- LOG(LS_WARNING) << "output device index has not been set";
- return -1;
- }
-
- if (_paPlayStream && (LATE(pa_stream_get_state)(_paPlayStream)
- != PA_STREAM_UNCONNECTED))
- {
- // We can only get the mute status if we have a connected stream
- if (!GetSinkInputInfo())
- return -1;
-
- enabled = static_cast<bool> (_paMute);
- } else
- {
- enabled = _paSpeakerMute;
- }
- LOG(LS_VERBOSE) << "AudioMixerManagerLinuxPulse::SpeakerMute() => enabled="
- << enabled;
-
- return 0;
-}
-
-int32_t
-AudioMixerManagerLinuxPulse::StereoPlayoutIsAvailable(bool& available)
-{
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
- if (_paOutputDeviceIndex == -1)
- {
- LOG(LS_WARNING) << "output device index has not been set";
- return -1;
- }
-
- uint32_t deviceIndex = (uint32_t) _paOutputDeviceIndex;
-
- {
- AutoPulseLock auto_lock(_paMainloop);
-
- // Get the actual stream device index if we have a connected stream
- // The device used by the stream can be changed
- // during the call
- if (_paPlayStream && (LATE(pa_stream_get_state)(_paPlayStream)
- != PA_STREAM_UNCONNECTED))
- {
- deviceIndex = LATE(pa_stream_get_device_index)(_paPlayStream);
- }
- }
-
- if (!GetSinkInfoByIndex(deviceIndex))
+ if (_paPlayStream &&
+ (LATE(pa_stream_get_state)(_paPlayStream) != PA_STREAM_UNCONNECTED)) {
+ // We can only get the volume if we have a connected stream
+ if (!GetSinkInputInfo())
return -1;
- available = static_cast<bool> (_paChannels == 2);
+ AutoPulseLock auto_lock(_paMainloop);
+ volume = static_cast<uint32_t>(_paVolume);
+ } else {
+ AutoPulseLock auto_lock(_paMainloop);
+ volume = _paSpeakerVolume;
+ }
- return 0;
+ LOG(LS_VERBOSE) << "AudioMixerManagerLinuxPulse::SpeakerVolume() => vol="
+ << volume;
+
+ return 0;
}
-int32_t
-AudioMixerManagerLinuxPulse::StereoRecordingIsAvailable(bool& available)
-{
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
- if (_paInputDeviceIndex == -1)
- {
- LOG(LS_WARNING) << "input device index has not been set";
- return -1;
+int32_t AudioMixerManagerLinuxPulse::MaxSpeakerVolume(
+ uint32_t& maxVolume) const {
+ if (_paOutputDeviceIndex == -1) {
+ LOG(LS_WARNING) << "output device index has not been set";
+ return -1;
+ }
+
+ // PA_VOLUME_NORM corresponds to 100% (0db)
+ // but PA allows up to 150 db amplification
+ maxVolume = static_cast<uint32_t>(PA_VOLUME_NORM);
+
+ return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::MinSpeakerVolume(
+ uint32_t& minVolume) const {
+ if (_paOutputDeviceIndex == -1) {
+ LOG(LS_WARNING) << "output device index has not been set";
+ return -1;
+ }
+
+ minVolume = static_cast<uint32_t>(PA_VOLUME_MUTED);
+
+ return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::SpeakerVolumeIsAvailable(bool& available) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ if (_paOutputDeviceIndex == -1) {
+ LOG(LS_WARNING) << "output device index has not been set";
+ return -1;
+ }
+
+ // Always available in Pulse Audio
+ available = true;
+
+ return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::SpeakerMuteIsAvailable(bool& available) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ if (_paOutputDeviceIndex == -1) {
+ LOG(LS_WARNING) << "output device index has not been set";
+ return -1;
+ }
+
+ // Always available in Pulse Audio
+ available = true;
+
+ return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::SetSpeakerMute(bool enable) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ LOG(LS_VERBOSE) << "AudioMixerManagerLinuxPulse::SetSpeakerMute(enable="
+ << enable << ")";
+
+ if (_paOutputDeviceIndex == -1) {
+ LOG(LS_WARNING) << "output device index has not been set";
+ return -1;
+ }
+
+ bool setFailed(false);
+
+ if (_paPlayStream &&
+ (LATE(pa_stream_get_state)(_paPlayStream) != PA_STREAM_UNCONNECTED)) {
+ // We can only really mute if we have a connected stream
+ AutoPulseLock auto_lock(_paMainloop);
+
+ pa_operation* paOperation = NULL;
+ paOperation = LATE(pa_context_set_sink_input_mute)(
+ _paContext, LATE(pa_stream_get_index)(_paPlayStream), (int)enable,
+ PaSetVolumeCallback, NULL);
+ if (!paOperation) {
+ setFailed = true;
}
- uint32_t deviceIndex = (uint32_t) _paInputDeviceIndex;
+ // Don't need to wait for the completion
+ LATE(pa_operation_unref)(paOperation);
+ } else {
+ // We have not created a stream or it's not connected to the sink
+ // Save the mute status to be set at connection
+ _paSpeakerMute = enable;
+ }
+ if (setFailed) {
+ LOG(LS_WARNING) << "could not mute speaker, error="
+ << LATE(pa_context_errno)(_paContext);
+ return -1;
+ }
+
+ return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::SpeakerMute(bool& enabled) const {
+ if (_paOutputDeviceIndex == -1) {
+ LOG(LS_WARNING) << "output device index has not been set";
+ return -1;
+ }
+
+ if (_paPlayStream &&
+ (LATE(pa_stream_get_state)(_paPlayStream) != PA_STREAM_UNCONNECTED)) {
+ // We can only get the mute status if we have a connected stream
+ if (!GetSinkInputInfo())
+ return -1;
+
+ enabled = static_cast<bool>(_paMute);
+ } else {
+ enabled = _paSpeakerMute;
+ }
+ LOG(LS_VERBOSE) << "AudioMixerManagerLinuxPulse::SpeakerMute() => enabled="
+ << enabled;
+
+ return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::StereoPlayoutIsAvailable(bool& available) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ if (_paOutputDeviceIndex == -1) {
+ LOG(LS_WARNING) << "output device index has not been set";
+ return -1;
+ }
+
+ uint32_t deviceIndex = (uint32_t)_paOutputDeviceIndex;
+
+ {
AutoPulseLock auto_lock(_paMainloop);
// Get the actual stream device index if we have a connected stream
// The device used by the stream can be changed
// during the call
- if (_paRecStream && (LATE(pa_stream_get_state)(_paRecStream)
- != PA_STREAM_UNCONNECTED))
- {
- deviceIndex = LATE(pa_stream_get_device_index)(_paRecStream);
+ if (_paPlayStream &&
+ (LATE(pa_stream_get_state)(_paPlayStream) != PA_STREAM_UNCONNECTED)) {
+ deviceIndex = LATE(pa_stream_get_device_index)(_paPlayStream);
}
+ }
- pa_operation* paOperation = NULL;
+ if (!GetSinkInfoByIndex(deviceIndex))
+ return -1;
- // Get info for this source
- // We want to know if the actual device can record in stereo
- paOperation = LATE(pa_context_get_source_info_by_index)(
- _paContext, deviceIndex,
- PaSourceInfoCallback,
- (void*) this);
+ available = static_cast<bool>(_paChannels == 2);
- WaitForOperationCompletion(paOperation);
+ return 0;
+}
- available = static_cast<bool> (_paChannels == 2);
+int32_t AudioMixerManagerLinuxPulse::StereoRecordingIsAvailable(
+ bool& available) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ if (_paInputDeviceIndex == -1) {
+ LOG(LS_WARNING) << "input device index has not been set";
+ return -1;
+ }
- LOG(LS_VERBOSE)
- << "AudioMixerManagerLinuxPulse::StereoRecordingIsAvailable()"
- << " => available=" << available;
+ uint32_t deviceIndex = (uint32_t)_paInputDeviceIndex;
- return 0;
+ AutoPulseLock auto_lock(_paMainloop);
+
+ // Get the actual stream device index if we have a connected stream
+ // The device used by the stream can be changed
+ // during the call
+ if (_paRecStream &&
+ (LATE(pa_stream_get_state)(_paRecStream) != PA_STREAM_UNCONNECTED)) {
+ deviceIndex = LATE(pa_stream_get_device_index)(_paRecStream);
+ }
+
+ pa_operation* paOperation = NULL;
+
+ // Get info for this source
+ // We want to know if the actual device can record in stereo
+ paOperation = LATE(pa_context_get_source_info_by_index)(
+ _paContext, deviceIndex, PaSourceInfoCallback, (void*)this);
+
+ WaitForOperationCompletion(paOperation);
+
+ available = static_cast<bool>(_paChannels == 2);
+
+ LOG(LS_VERBOSE) << "AudioMixerManagerLinuxPulse::StereoRecordingIsAvailable()"
+ << " => available=" << available;
+
+ return 0;
}
int32_t AudioMixerManagerLinuxPulse::MicrophoneMuteIsAvailable(
- bool& available)
-{
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
- if (_paInputDeviceIndex == -1)
- {
- LOG(LS_WARNING) << "input device index has not been set";
- return -1;
- }
+ bool& available) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ if (_paInputDeviceIndex == -1) {
+ LOG(LS_WARNING) << "input device index has not been set";
+ return -1;
+ }
- // Always available in Pulse Audio
- available = true;
+ // Always available in Pulse Audio
+ available = true;
- return 0;
+ return 0;
}
-int32_t AudioMixerManagerLinuxPulse::SetMicrophoneMute(bool enable)
-{
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
- LOG(LS_VERBOSE) << "AudioMixerManagerLinuxPulse::SetMicrophoneMute(enable="
- << enable << ")";
+int32_t AudioMixerManagerLinuxPulse::SetMicrophoneMute(bool enable) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ LOG(LS_VERBOSE) << "AudioMixerManagerLinuxPulse::SetMicrophoneMute(enable="
+ << enable << ")";
- if (_paInputDeviceIndex == -1)
- {
- LOG(LS_WARNING) << "input device index has not been set";
- return -1;
- }
+ if (_paInputDeviceIndex == -1) {
+ LOG(LS_WARNING) << "input device index has not been set";
+ return -1;
+ }
- bool setFailed(false);
- pa_operation* paOperation = NULL;
+ bool setFailed(false);
+ pa_operation* paOperation = NULL;
- uint32_t deviceIndex = (uint32_t) _paInputDeviceIndex;
+ uint32_t deviceIndex = (uint32_t)_paInputDeviceIndex;
+ AutoPulseLock auto_lock(_paMainloop);
+
+ // Get the actual stream device index if we have a connected stream
+ // The device used by the stream can be changed
+ // during the call
+ if (_paRecStream &&
+ (LATE(pa_stream_get_state)(_paRecStream) != PA_STREAM_UNCONNECTED)) {
+ deviceIndex = LATE(pa_stream_get_device_index)(_paRecStream);
+ }
+
+ // Set mute switch for the source
+ paOperation = LATE(pa_context_set_source_mute_by_index)(
+ _paContext, deviceIndex, enable, PaSetVolumeCallback, NULL);
+
+ if (!paOperation) {
+ setFailed = true;
+ }
+
+ // Don't need to wait for this to complete.
+ LATE(pa_operation_unref)(paOperation);
+
+ if (setFailed) {
+ LOG(LS_WARNING) << "could not mute microphone, error="
+ << LATE(pa_context_errno)(_paContext);
+ return -1;
+ }
+
+ return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::MicrophoneMute(bool& enabled) const {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ if (_paInputDeviceIndex == -1) {
+ LOG(LS_WARNING) << "input device index has not been set";
+ return -1;
+ }
+
+ uint32_t deviceIndex = (uint32_t)_paInputDeviceIndex;
+
+ {
AutoPulseLock auto_lock(_paMainloop);
-
// Get the actual stream device index if we have a connected stream
// The device used by the stream can be changed
// during the call
- if (_paRecStream && (LATE(pa_stream_get_state)(_paRecStream)
- != PA_STREAM_UNCONNECTED))
- {
- deviceIndex = LATE(pa_stream_get_device_index)(_paRecStream);
+ if (_paRecStream &&
+ (LATE(pa_stream_get_state)(_paRecStream) != PA_STREAM_UNCONNECTED)) {
+ deviceIndex = LATE(pa_stream_get_device_index)(_paRecStream);
}
+ }
- // Set mute switch for the source
- paOperation = LATE(pa_context_set_source_mute_by_index)(
- _paContext, deviceIndex,
- enable,
- PaSetVolumeCallback, NULL);
+ if (!GetSourceInfoByIndex(deviceIndex))
+ return -1;
- if (!paOperation)
- {
- setFailed = true;
- }
+ enabled = static_cast<bool>(_paMute);
- // Don't need to wait for this to complete.
- LATE(pa_operation_unref)(paOperation);
+ LOG(LS_VERBOSE) << "AudioMixerManagerLinuxPulse::MicrophoneMute() => enabled="
+ << enabled;
- if (setFailed)
- {
- LOG(LS_WARNING) << "could not mute microphone, error="
- << LATE(pa_context_errno)(_paContext);
- return -1;
- }
-
- return 0;
-}
-
-int32_t AudioMixerManagerLinuxPulse::MicrophoneMute(bool& enabled) const
-{
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
- if (_paInputDeviceIndex == -1)
- {
- LOG(LS_WARNING) << "input device index has not been set";
- return -1;
- }
-
- uint32_t deviceIndex = (uint32_t) _paInputDeviceIndex;
-
- {
- AutoPulseLock auto_lock(_paMainloop);
- // Get the actual stream device index if we have a connected stream
- // The device used by the stream can be changed
- // during the call
- if (_paRecStream && (LATE(pa_stream_get_state)(_paRecStream)
- != PA_STREAM_UNCONNECTED))
- {
- deviceIndex = LATE(pa_stream_get_device_index)(_paRecStream);
- }
- }
-
- if (!GetSourceInfoByIndex(deviceIndex))
- return -1;
-
- enabled = static_cast<bool> (_paMute);
-
- LOG(LS_VERBOSE)
- << "AudioMixerManagerLinuxPulse::MicrophoneMute() => enabled="
- << enabled;
-
- return 0;
+ return 0;
}
int32_t AudioMixerManagerLinuxPulse::MicrophoneVolumeIsAvailable(
- bool& available)
-{
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
- if (_paInputDeviceIndex == -1)
- {
- LOG(LS_WARNING) << "input device index has not been set";
- return -1;
- }
+ bool& available) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ if (_paInputDeviceIndex == -1) {
+ LOG(LS_WARNING) << "input device index has not been set";
+ return -1;
+ }
- // Always available in Pulse Audio
- available = true;
+ // Always available in Pulse Audio
+ available = true;
- return 0;
+ return 0;
}
-int32_t
-AudioMixerManagerLinuxPulse::SetMicrophoneVolume(uint32_t volume)
-{
- LOG(LS_VERBOSE)
- << "AudioMixerManagerLinuxPulse::SetMicrophoneVolume(volume=" << volume
- << ")";
+int32_t AudioMixerManagerLinuxPulse::SetMicrophoneVolume(uint32_t volume) {
+ LOG(LS_VERBOSE) << "AudioMixerManagerLinuxPulse::SetMicrophoneVolume(volume="
+ << volume << ")";
- if (_paInputDeviceIndex == -1)
- {
- LOG(LS_WARNING) << "input device index has not been set";
- return -1;
- }
+ if (_paInputDeviceIndex == -1) {
+ LOG(LS_WARNING) << "input device index has not been set";
+ return -1;
+ }
- // Unlike output streams, input streams have no concept of a stream
- // volume, only a device volume. So we have to change the volume of the
- // device itself.
+ // Unlike output streams, input streams have no concept of a stream
+ // volume, only a device volume. So we have to change the volume of the
+ // device itself.
- // The device may have a different number of channels than the stream and
- // their mapping may be different, so we don't want to use the channel
- // count from our sample spec. We could use PA_CHANNELS_MAX to cover our
- // bases, and the server allows that even if the device's channel count
- // is lower, but some buggy PA clients don't like that (the pavucontrol
- // on Hardy dies in an assert if the channel count is different). So
- // instead we look up the actual number of channels that the device has.
+ // The device may have a different number of channels than the stream and
+ // their mapping may be different, so we don't want to use the channel
+ // count from our sample spec. We could use PA_CHANNELS_MAX to cover our
+ // bases, and the server allows that even if the device's channel count
+ // is lower, but some buggy PA clients don't like that (the pavucontrol
+ // on Hardy dies in an assert if the channel count is different). So
+ // instead we look up the actual number of channels that the device has.
+ AutoPulseLock auto_lock(_paMainloop);
+ uint32_t deviceIndex = (uint32_t)_paInputDeviceIndex;
+
+ // Get the actual stream device index if we have a connected stream
+ // The device used by the stream can be changed
+ // during the call
+ if (_paRecStream &&
+ (LATE(pa_stream_get_state)(_paRecStream) != PA_STREAM_UNCONNECTED)) {
+ deviceIndex = LATE(pa_stream_get_device_index)(_paRecStream);
+ }
+
+ bool setFailed(false);
+ pa_operation* paOperation = NULL;
+
+ // Get the number of channels for this source
+ paOperation = LATE(pa_context_get_source_info_by_index)(
+ _paContext, deviceIndex, PaSourceInfoCallback, (void*)this);
+
+ WaitForOperationCompletion(paOperation);
+
+ uint8_t channels = _paChannels;
+ pa_cvolume cVolumes;
+ LATE(pa_cvolume_set)(&cVolumes, channels, volume);
+
+ // Set the volume for the source
+ paOperation = LATE(pa_context_set_source_volume_by_index)(
+ _paContext, deviceIndex, &cVolumes, PaSetVolumeCallback, NULL);
+
+ if (!paOperation) {
+ setFailed = true;
+ }
+
+ // Don't need to wait for this to complete.
+ LATE(pa_operation_unref)(paOperation);
+
+ if (setFailed) {
+ LOG(LS_WARNING) << "could not set microphone volume, error="
+ << LATE(pa_context_errno)(_paContext);
+ return -1;
+ }
+
+ return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::MicrophoneVolume(uint32_t& volume) const {
+ if (_paInputDeviceIndex == -1) {
+ LOG(LS_WARNING) << "input device index has not been set";
+ return -1;
+ }
+
+ uint32_t deviceIndex = (uint32_t)_paInputDeviceIndex;
+
+ {
AutoPulseLock auto_lock(_paMainloop);
- uint32_t deviceIndex = (uint32_t) _paInputDeviceIndex;
-
- // Get the actual stream device index if we have a connected stream
- // The device used by the stream can be changed
- // during the call
- if (_paRecStream && (LATE(pa_stream_get_state)(_paRecStream)
- != PA_STREAM_UNCONNECTED))
- {
- deviceIndex = LATE(pa_stream_get_device_index)(_paRecStream);
+ // Get the actual stream device index if we have a connected stream.
+ // The device used by the stream can be changed during the call.
+ if (_paRecStream &&
+ (LATE(pa_stream_get_state)(_paRecStream) != PA_STREAM_UNCONNECTED)) {
+ deviceIndex = LATE(pa_stream_get_device_index)(_paRecStream);
}
+ }
- bool setFailed(false);
- pa_operation* paOperation = NULL;
+ if (!GetSourceInfoByIndex(deviceIndex))
+ return -1;
- // Get the number of channels for this source
- paOperation
- = LATE(pa_context_get_source_info_by_index)(_paContext, deviceIndex,
- PaSourceInfoCallback,
- (void*) this);
+ {
+ AutoPulseLock auto_lock(_paMainloop);
+ volume = static_cast<uint32_t>(_paVolume);
+ }
- WaitForOperationCompletion(paOperation);
+ LOG(LS_VERBOSE) << "AudioMixerManagerLinuxPulse::MicrophoneVolume() => vol="
+ << volume;
- uint8_t channels = _paChannels;
- pa_cvolume cVolumes;
- LATE(pa_cvolume_set)(&cVolumes, channels, volume);
-
- // Set the volume for the source
- paOperation
- = LATE(pa_context_set_source_volume_by_index)(_paContext, deviceIndex,
- &cVolumes,
- PaSetVolumeCallback,
- NULL);
-
- if (!paOperation)
- {
- setFailed = true;
- }
-
- // Don't need to wait for this to complete.
- LATE(pa_operation_unref)(paOperation);
-
- if (setFailed)
- {
- LOG(LS_WARNING) << "could not set microphone volume, error="
- << LATE(pa_context_errno)(_paContext);
- return -1;
- }
-
- return 0;
+ return 0;
}
-int32_t
-AudioMixerManagerLinuxPulse::MicrophoneVolume(uint32_t& volume) const
-{
+int32_t AudioMixerManagerLinuxPulse::MaxMicrophoneVolume(
+ uint32_t& maxVolume) const {
+ if (_paInputDeviceIndex == -1) {
+ LOG(LS_WARNING) << "input device index has not been set";
+ return -1;
+ }
- if (_paInputDeviceIndex == -1)
- {
- LOG(LS_WARNING) << "input device index has not been set";
- return -1;
- }
+ // PA_VOLUME_NORM corresponds to 100% (0db)
+ // PA allows up to 150 db amplification (PA_VOLUME_MAX)
+ // but that doesn't work well for all sound cards
+ maxVolume = static_cast<uint32_t>(PA_VOLUME_NORM);
- uint32_t deviceIndex = (uint32_t) _paInputDeviceIndex;
-
- {
- AutoPulseLock auto_lock(_paMainloop);
- // Get the actual stream device index if we have a connected stream.
- // The device used by the stream can be changed during the call.
- if (_paRecStream && (LATE(pa_stream_get_state)(_paRecStream)
- != PA_STREAM_UNCONNECTED))
- {
- deviceIndex = LATE(pa_stream_get_device_index)(_paRecStream);
- }
- }
-
- if (!GetSourceInfoByIndex(deviceIndex))
- return -1;
-
- {
- AutoPulseLock auto_lock(_paMainloop);
- volume = static_cast<uint32_t> (_paVolume);
- }
-
- LOG(LS_VERBOSE)
- << "AudioMixerManagerLinuxPulse::MicrophoneVolume() => vol="
- << volume;
-
- return 0;
+ return 0;
}
-int32_t
-AudioMixerManagerLinuxPulse::MaxMicrophoneVolume(uint32_t& maxVolume) const
-{
+int32_t AudioMixerManagerLinuxPulse::MinMicrophoneVolume(
+ uint32_t& minVolume) const {
+ if (_paInputDeviceIndex == -1) {
+ LOG(LS_WARNING) << "input device index has not been set";
+ return -1;
+ }
- if (_paInputDeviceIndex == -1)
- {
- LOG(LS_WARNING) << "input device index has not been set";
- return -1;
- }
+ minVolume = static_cast<uint32_t>(PA_VOLUME_MUTED);
- // PA_VOLUME_NORM corresponds to 100% (0db)
- // PA allows up to 150 db amplification (PA_VOLUME_MAX)
- // but that doesn't work well for all sound cards
- maxVolume = static_cast<uint32_t> (PA_VOLUME_NORM);
-
- return 0;
-}
-
-int32_t
-AudioMixerManagerLinuxPulse::MinMicrophoneVolume(uint32_t& minVolume) const
-{
-
- if (_paInputDeviceIndex == -1)
- {
- LOG(LS_WARNING) << "input device index has not been set";
- return -1;
- }
-
- minVolume = static_cast<uint32_t> (PA_VOLUME_MUTED);
-
- return 0;
+ return 0;
}
// ===========================================================================
// Private Methods
// ===========================================================================
-void
-AudioMixerManagerLinuxPulse::PaSinkInfoCallback(pa_context */*c*/,
- const pa_sink_info *i,
- int eol,
- void *pThis)
-{
- static_cast<AudioMixerManagerLinuxPulse*> (pThis)->
- PaSinkInfoCallbackHandler(i, eol);
+void AudioMixerManagerLinuxPulse::PaSinkInfoCallback(pa_context* /*c*/,
+ const pa_sink_info* i,
+ int eol,
+ void* pThis) {
+ static_cast<AudioMixerManagerLinuxPulse*>(pThis)->PaSinkInfoCallbackHandler(
+ i, eol);
}
-void
-AudioMixerManagerLinuxPulse::PaSinkInputInfoCallback(
- pa_context */*c*/,
- const pa_sink_input_info *i,
+void AudioMixerManagerLinuxPulse::PaSinkInputInfoCallback(
+ pa_context* /*c*/,
+ const pa_sink_input_info* i,
int eol,
- void *pThis)
-{
- static_cast<AudioMixerManagerLinuxPulse*> (pThis)->
- PaSinkInputInfoCallbackHandler(i, eol);
+ void* pThis) {
+ static_cast<AudioMixerManagerLinuxPulse*>(pThis)
+ ->PaSinkInputInfoCallbackHandler(i, eol);
}
-
-void
-AudioMixerManagerLinuxPulse::PaSourceInfoCallback(pa_context */*c*/,
- const pa_source_info *i,
- int eol,
- void *pThis)
-{
- static_cast<AudioMixerManagerLinuxPulse*> (pThis)->
- PaSourceInfoCallbackHandler(i, eol);
+void AudioMixerManagerLinuxPulse::PaSourceInfoCallback(pa_context* /*c*/,
+ const pa_source_info* i,
+ int eol,
+ void* pThis) {
+ static_cast<AudioMixerManagerLinuxPulse*>(pThis)->PaSourceInfoCallbackHandler(
+ i, eol);
}
-void
-AudioMixerManagerLinuxPulse::PaSetVolumeCallback(pa_context * c,
- int success,
- void */*pThis*/)
-{
- if (!success)
- {
- LOG(LS_ERROR) << "failed to set volume";
- }
+void AudioMixerManagerLinuxPulse::PaSetVolumeCallback(pa_context* c,
+ int success,
+ void* /*pThis*/) {
+ if (!success) {
+ LOG(LS_ERROR) << "failed to set volume";
+ }
}
void AudioMixerManagerLinuxPulse::PaSinkInfoCallbackHandler(
- const pa_sink_info *i,
- int eol)
-{
- if (eol)
- {
- // Signal that we are done
- LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
- return;
- }
+ const pa_sink_info* i,
+ int eol) {
+ if (eol) {
+ // Signal that we are done
+ LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
+ return;
+ }
- _paChannels = i->channel_map.channels; // Get number of channels
- pa_volume_t paVolume = PA_VOLUME_MUTED; // Minimum possible value.
- for (int j = 0; j < _paChannels; ++j)
- {
- if (paVolume < i->volume.values[j])
- {
- paVolume = i->volume.values[j];
- }
+ _paChannels = i->channel_map.channels; // Get number of channels
+ pa_volume_t paVolume = PA_VOLUME_MUTED; // Minimum possible value.
+ for (int j = 0; j < _paChannels; ++j) {
+ if (paVolume < i->volume.values[j]) {
+ paVolume = i->volume.values[j];
}
- _paVolume = paVolume; // get the max volume for any channel
- _paMute = i->mute; // get mute status
+ }
+ _paVolume = paVolume; // get the max volume for any channel
+ _paMute = i->mute; // get mute status
- // supported since PA 0.9.15
- //_paVolSteps = i->n_volume_steps; // get the number of volume steps
- // default value is PA_VOLUME_NORM+1
- _paVolSteps = PA_VOLUME_NORM + 1;
+ // supported since PA 0.9.15
+ //_paVolSteps = i->n_volume_steps; // get the number of volume steps
+ // default value is PA_VOLUME_NORM+1
+ _paVolSteps = PA_VOLUME_NORM + 1;
}
void AudioMixerManagerLinuxPulse::PaSinkInputInfoCallbackHandler(
- const pa_sink_input_info *i,
- int eol)
-{
- if (eol)
- {
- // Signal that we are done
- LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
- return;
- }
+ const pa_sink_input_info* i,
+ int eol) {
+ if (eol) {
+ // Signal that we are done
+ LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
+ return;
+ }
- _paChannels = i->channel_map.channels; // Get number of channels
- pa_volume_t paVolume = PA_VOLUME_MUTED; // Minimum possible value.
- for (int j = 0; j < _paChannels; ++j)
- {
- if (paVolume < i->volume.values[j])
- {
- paVolume = i->volume.values[j];
- }
+ _paChannels = i->channel_map.channels; // Get number of channels
+ pa_volume_t paVolume = PA_VOLUME_MUTED; // Minimum possible value.
+ for (int j = 0; j < _paChannels; ++j) {
+ if (paVolume < i->volume.values[j]) {
+ paVolume = i->volume.values[j];
}
- _paVolume = paVolume; // Get the max volume for any channel
- _paMute = i->mute; // Get mute status
+ }
+ _paVolume = paVolume; // Get the max volume for any channel
+ _paMute = i->mute; // Get mute status
}
void AudioMixerManagerLinuxPulse::PaSourceInfoCallbackHandler(
- const pa_source_info *i,
- int eol)
-{
- if (eol)
- {
- // Signal that we are done
- LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
- return;
- }
+ const pa_source_info* i,
+ int eol) {
+ if (eol) {
+ // Signal that we are done
+ LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
+ return;
+ }
- _paChannels = i->channel_map.channels; // Get number of channels
- pa_volume_t paVolume = PA_VOLUME_MUTED; // Minimum possible value.
- for (int j = 0; j < _paChannels; ++j)
- {
- if (paVolume < i->volume.values[j])
- {
- paVolume = i->volume.values[j];
- }
+ _paChannels = i->channel_map.channels; // Get number of channels
+ pa_volume_t paVolume = PA_VOLUME_MUTED; // Minimum possible value.
+ for (int j = 0; j < _paChannels; ++j) {
+ if (paVolume < i->volume.values[j]) {
+ paVolume = i->volume.values[j];
}
- _paVolume = paVolume; // Get the max volume for any channel
- _paMute = i->mute; // Get mute status
+ }
+ _paVolume = paVolume; // Get the max volume for any channel
+ _paMute = i->mute; // Get mute status
- // supported since PA 0.9.15
- //_paVolSteps = i->n_volume_steps; // Get the number of volume steps
- // default value is PA_VOLUME_NORM+1
- _paVolSteps = PA_VOLUME_NORM + 1;
+ // supported since PA 0.9.15
+ //_paVolSteps = i->n_volume_steps; // Get the number of volume steps
+ // default value is PA_VOLUME_NORM+1
+ _paVolSteps = PA_VOLUME_NORM + 1;
}
void AudioMixerManagerLinuxPulse::WaitForOperationCompletion(
- pa_operation* paOperation) const
-{
- while (LATE(pa_operation_get_state)(paOperation) == PA_OPERATION_RUNNING)
- {
- LATE(pa_threaded_mainloop_wait)(_paMainloop);
- }
+ pa_operation* paOperation) const {
+ while (LATE(pa_operation_get_state)(paOperation) == PA_OPERATION_RUNNING) {
+ LATE(pa_threaded_mainloop_wait)(_paMainloop);
+ }
- LATE(pa_operation_unref)(paOperation);
+ LATE(pa_operation_unref)(paOperation);
}
bool AudioMixerManagerLinuxPulse::GetSinkInputInfo() const {
@@ -941,37 +804,33 @@
AutoPulseLock auto_lock(_paMainloop);
// Get info for this stream (sink input).
paOperation = LATE(pa_context_get_sink_input_info)(
- _paContext,
- LATE(pa_stream_get_index)(_paPlayStream),
- PaSinkInputInfoCallback,
- (void*) this);
+ _paContext, LATE(pa_stream_get_index)(_paPlayStream),
+ PaSinkInputInfoCallback, (void*)this);
WaitForOperationCompletion(paOperation);
return true;
}
-bool AudioMixerManagerLinuxPulse::GetSinkInfoByIndex(
- int device_index) const {
+bool AudioMixerManagerLinuxPulse::GetSinkInfoByIndex(int device_index) const {
pa_operation* paOperation = NULL;
AutoPulseLock auto_lock(_paMainloop);
- paOperation = LATE(pa_context_get_sink_info_by_index)(_paContext,
- device_index, PaSinkInfoCallback, (void*) this);
+ paOperation = LATE(pa_context_get_sink_info_by_index)(
+ _paContext, device_index, PaSinkInfoCallback, (void*)this);
WaitForOperationCompletion(paOperation);
return true;
}
-bool AudioMixerManagerLinuxPulse::GetSourceInfoByIndex(
- int device_index) const {
+bool AudioMixerManagerLinuxPulse::GetSourceInfoByIndex(int device_index) const {
pa_operation* paOperation = NULL;
AutoPulseLock auto_lock(_paMainloop);
- paOperation = LATE(pa_context_get_source_info_by_index)(
- _paContext, device_index, PaSourceInfoCallback, (void*) this);
+ paOperation = LATE(pa_context_get_source_info_by_index)(
+ _paContext, device_index, PaSourceInfoCallback, (void*)this);
WaitForOperationCompletion(paOperation);
return true;
}
-}
+} // namespace webrtc
diff --git a/modules/audio_device/linux/latebindingsymboltable_linux.cc b/modules/audio_device/linux/latebindingsymboltable_linux.cc
index 35f53fa..7a66c34 100644
--- a/modules/audio_device/linux/latebindingsymboltable_linux.cc
+++ b/modules/audio_device/linux/latebindingsymboltable_linux.cc
@@ -19,9 +19,9 @@
namespace webrtc {
namespace adm_linux {
-inline static const char *GetDllError() {
+inline static const char* GetDllError() {
#ifdef WEBRTC_LINUX
- char *err = dlerror();
+ char* err = dlerror();
if (err) {
return err;
} else {
@@ -64,11 +64,11 @@
}
static bool LoadSymbol(DllHandle handle,
- const char *symbol_name,
- void **symbol) {
+ const char* symbol_name,
+ void** symbol) {
#ifdef WEBRTC_LINUX
*symbol = dlsym(handle, symbol_name);
- char *err = dlerror();
+ char* err = dlerror();
if (err) {
LOG(LS_ERROR) << "Error loading symbol " << symbol_name << " : " << err;
return false;
@@ -87,8 +87,8 @@
// caller may later interpret as a valid address.
bool InternalLoadSymbols(DllHandle handle,
int num_symbols,
- const char *const symbol_names[],
- void *symbols[]) {
+ const char* const symbol_names[],
+ void* symbols[]) {
#ifdef WEBRTC_LINUX
// Clear any old errors.
dlerror();
diff --git a/modules/audio_device/mac/audio_device_mac.cc b/modules/audio_device/mac/audio_device_mac.cc
index 27d1cc4..635bd0d 100644
--- a/modules/audio_device/mac/audio_device_mac.cc
+++ b/modules/audio_device/mac/audio_device_mac.cc
@@ -23,34 +23,31 @@
namespace webrtc {
-#define WEBRTC_CA_RETURN_ON_ERR(expr) \
+#define WEBRTC_CA_RETURN_ON_ERR(expr) \
+ do { \
+ err = expr; \
+ if (err != noErr) { \
+ logCAMsg(rtc::LS_ERROR, "Error in " #expr, (const char*)&err); \
+ return -1; \
+ } \
+ } while (0)
+
+#define WEBRTC_CA_LOG_ERR(expr) \
+ do { \
+ err = expr; \
+ if (err != noErr) { \
+ logCAMsg(rtc::LS_ERROR, "Error in " #expr, (const char*)&err); \
+ } \
+ } while (0)
+
+#define WEBRTC_CA_LOG_WARN(expr) \
do { \
err = expr; \
if (err != noErr) { \
- logCAMsg(rtc::LS_ERROR, "Error in " #expr, \
- (const char*) & err); \
- return -1; \
+ logCAMsg(rtc::LS_WARNING, "Error in " #expr, (const char*)&err); \
} \
} while (0)
-#define WEBRTC_CA_LOG_ERR(expr) \
- do { \
- err = expr; \
- if (err != noErr) { \
- logCAMsg(rtc::LS_ERROR, "Error in " #expr, \
- (const char*) & err); \
- } \
- } while (0)
-
-#define WEBRTC_CA_LOG_WARN(expr) \
- do { \
- err = expr; \
- if (err != noErr) { \
- logCAMsg(rtc::LS_WARNING, "Error in " #expr, \
- (const char*) & err); \
- } \
- } while (0)
-
enum { MaxNumberDevices = 64 };
void AudioDeviceMac::AtomicSet32(int32_t* theValue, int32_t newValue) {
@@ -94,7 +91,7 @@
}
#else
// We need to flip the characters in this case.
- switch (sev) {
+ switch (sev) {
case rtc::LS_ERROR:
LOG(LS_ERROR) << msg << ": " << err[3] << err[2] << err[1] << err[0];
break;
@@ -373,8 +370,8 @@
err = AudioHardwareUnload();
if (err != noErr) {
- logCAMsg(rtc::LS_ERROR,
- "Error in AudioHardwareUnload()", (const char*)&err);
+ logCAMsg(rtc::LS_ERROR, "Error in AudioHardwareUnload()",
+ (const char*)&err);
retVal = -1;
}
@@ -1038,8 +1035,7 @@
_outputDeviceID, &propertyAddress, 0, NULL, &size, &_outStreamFormat));
if (_outStreamFormat.mFormatID != kAudioFormatLinearPCM) {
- logCAMsg(rtc::LS_ERROR,
- "Unacceptable output stream format -> mFormatID",
+ logCAMsg(rtc::LS_ERROR, "Unacceptable output stream format -> mFormatID",
(const char*)&_outStreamFormat.mFormatID);
return -1;
}
@@ -1146,8 +1142,7 @@
_inputDeviceID, &propertyAddress, 0, NULL, &size, &_inStreamFormat));
if (_inStreamFormat.mFormatID != kAudioFormatLinearPCM) {
- logCAMsg(rtc::LS_ERROR,
- "Unacceptable input stream format -> mFormatID",
+ logCAMsg(rtc::LS_ERROR, "Unacceptable input stream format -> mFormatID",
(const char*)&_inStreamFormat.mFormatID);
return -1;
}
@@ -1348,12 +1343,11 @@
_critSect.Leave(); // Cannot be under lock, risk of deadlock
if (kEventTimeout == _stopEventRec.Wait(2000)) {
rtc::CritScope critScoped(&_critSect);
- LOG(LS_WARNING)
- << "Timed out stopping the capture IOProc."
- << "We may have failed to detect a device removal.";
+ LOG(LS_WARNING) << "Timed out stopping the capture IOProc."
+ << "We may have failed to detect a device removal.";
WEBRTC_CA_LOG_WARN(AudioDeviceStop(_inputDeviceID, _inDeviceIOProcID));
WEBRTC_CA_LOG_WARN(
- AudioDeviceDestroyIOProcID(_inputDeviceID, _inDeviceIOProcID));
+ AudioDeviceDestroyIOProcID(_inputDeviceID, _inDeviceIOProcID));
}
_critSect.Enter();
_doStopRec = false;
@@ -1377,9 +1371,8 @@
_critSect.Leave(); // Cannot be under lock, risk of deadlock
if (kEventTimeout == _stopEvent.Wait(2000)) {
rtc::CritScope critScoped(&_critSect);
- LOG(LS_WARNING)
- << "Timed out stopping the shared IOProc."
- << "We may have failed to detect a device removal.";
+ LOG(LS_WARNING) << "Timed out stopping the shared IOProc."
+ << "We may have failed to detect a device removal.";
// We assume rendering on a shared device has stopped as well if
// the IOProc times out.
WEBRTC_CA_LOG_WARN(AudioDeviceStop(_outputDeviceID, _deviceIOProcID));
@@ -1391,7 +1384,7 @@
LOG(LS_INFO) << "Recording stopped (shared device)";
} else if (_recIsInitialized && !_playing && !_playIsInitialized) {
WEBRTC_CA_LOG_WARN(
- AudioDeviceDestroyIOProcID(_outputDeviceID, _deviceIOProcID));
+ AudioDeviceDestroyIOProcID(_outputDeviceID, _deviceIOProcID));
LOG(LS_INFO) << "Recording uninitialized (shared device)";
}
}
@@ -1486,9 +1479,8 @@
_critSect.Leave(); // Cannot be under lock, risk of deadlock
if (kEventTimeout == _stopEvent.Wait(2000)) {
rtc::CritScope critScoped(&_critSect);
- LOG(LS_WARNING)
- << "Timed out stopping the render IOProc."
- << "We may have failed to detect a device removal.";
+ LOG(LS_WARNING) << "Timed out stopping the render IOProc."
+ << "We may have failed to detect a device removal.";
// We assume capturing on a shared device has stopped as well if the
// IOProc times out.
@@ -1501,11 +1493,11 @@
LOG(LS_INFO) << "Playout stopped";
} else if (_twoDevices && _playIsInitialized) {
WEBRTC_CA_LOG_WARN(
- AudioDeviceDestroyIOProcID(_outputDeviceID, _deviceIOProcID));
+ AudioDeviceDestroyIOProcID(_outputDeviceID, _deviceIOProcID));
LOG(LS_INFO) << "Playout uninitialized (output device)";
} else if (!_twoDevices && _playIsInitialized && !_recIsInitialized) {
WEBRTC_CA_LOG_WARN(
- AudioDeviceDestroyIOProcID(_outputDeviceID, _deviceIOProcID));
+ AudioDeviceDestroyIOProcID(_outputDeviceID, _deviceIOProcID));
LOG(LS_INFO) << "Playout uninitialized (shared device)";
}
@@ -1829,8 +1821,8 @@
_ptrAudioBuffer->SetPlayoutChannels((uint8_t)_playChannels);
}
- _renderDelayOffsetSamples = _renderBufSizeSamples -
- N_BUFFERS_OUT * ENGINE_PLAY_BUF_SIZE_IN_SAMPLES *
+ _renderDelayOffsetSamples =
+ _renderBufSizeSamples - N_BUFFERS_OUT * ENGINE_PLAY_BUF_SIZE_IN_SAMPLES *
_outDesiredFormat.mChannelsPerFrame;
_outDesiredFormat.mBytesPerPacket =
@@ -1909,9 +1901,9 @@
static_cast<uint32_t>((1.0e6 * latency) / _outStreamFormat.mSampleRate);
LOG(LS_VERBOSE) << "initial playout status: _renderDelayOffsetSamples="
- << _renderDelayOffsetSamples << ", _renderDelayUs="
- << _renderDelayUs << ", _renderLatencyUs="
- << _renderLatencyUs;
+ << _renderDelayOffsetSamples
+ << ", _renderDelayUs=" << _renderDelayUs
+ << ", _renderLatencyUs=" << _renderLatencyUs;
return 0;
}
@@ -1970,8 +1962,8 @@
AtomicSet32(&_captureDeviceIsAlive, 0);
_mixerManager.CloseMicrophone();
} else if (err != noErr) {
- logCAMsg(rtc::LS_ERROR,
- "Error in AudioDeviceGetProperty()", (const char*)&err);
+ logCAMsg(rtc::LS_ERROR, "Error in AudioDeviceGetProperty()",
+ (const char*)&err);
return -1;
}
}
@@ -1989,8 +1981,8 @@
AtomicSet32(&_renderDeviceIsAlive, 0);
_mixerManager.CloseSpeaker();
} else if (err != noErr) {
- logCAMsg(rtc::LS_ERROR,
- "Error in AudioDeviceGetProperty()", (const char*)&err);
+ logCAMsg(rtc::LS_ERROR, "Error in AudioDeviceGetProperty()",
+ (const char*)&err);
return -1;
}
}
@@ -2016,8 +2008,7 @@
objectId, &propertyAddress, 0, NULL, &size, &streamFormat));
if (streamFormat.mFormatID != kAudioFormatLinearPCM) {
- logCAMsg(rtc::LS_ERROR,
- "Unacceptable input stream format -> mFormatID",
+ logCAMsg(rtc::LS_ERROR, "Unacceptable input stream format -> mFormatID",
(const char*)&streamFormat.mFormatID);
return -1;
}
@@ -2042,8 +2033,7 @@
LOG(LS_VERBOSE) << "mBytesPerFrame = " << streamFormat.mBytesPerFrame
<< ", mBitsPerChannel = " << streamFormat.mBitsPerChannel;
LOG(LS_VERBOSE) << "mFormatFlags = " << streamFormat.mFormatFlags;
- logCAMsg(rtc::LS_VERBOSE, "mFormatID",
- (const char*)&streamFormat.mFormatID);
+ logCAMsg(rtc::LS_VERBOSE, "mFormatID", (const char*)&streamFormat.mFormatID);
if (propertyAddress.mScope == kAudioDevicePropertyScopeInput) {
const int io_block_size_samples = streamFormat.mChannelsPerFrame *
@@ -2247,8 +2237,8 @@
LOG(LS_ERROR) << "Error in AudioConverterFillComplexBuffer()";
return 1;
} else {
- logCAMsg(rtc::LS_ERROR,
- "Error in AudioConverterFillComplexBuffer()", (const char*)&err);
+ logCAMsg(rtc::LS_ERROR, "Error in AudioConverterFillComplexBuffer()",
+ (const char*)&err);
return 1;
}
}
@@ -2485,8 +2475,8 @@
// This is our own error.
return false;
} else {
- logCAMsg(rtc::LS_ERROR,
- "Error in AudioConverterFillComplexBuffer()", (const char*)&err);
+ logCAMsg(rtc::LS_ERROR, "Error in AudioConverterFillComplexBuffer()",
+ (const char*)&err);
return false;
}
}
diff --git a/modules/audio_device/mac/audio_mixer_manager_mac.cc b/modules/audio_device/mac/audio_mixer_manager_mac.cc
index 14d3f98..928fae7 100644
--- a/modules/audio_device/mac/audio_mixer_manager_mac.cc
+++ b/modules/audio_device/mac/audio_mixer_manager_mac.cc
@@ -14,34 +14,31 @@
namespace webrtc {
-#define WEBRTC_CA_RETURN_ON_ERR(expr) \
+#define WEBRTC_CA_RETURN_ON_ERR(expr) \
+ do { \
+ err = expr; \
+ if (err != noErr) { \
+ logCAMsg(rtc::LS_ERROR, "Error in " #expr, (const char*)&err); \
+ return -1; \
+ } \
+ } while (0)
+
+#define WEBRTC_CA_LOG_ERR(expr) \
+ do { \
+ err = expr; \
+ if (err != noErr) { \
+ logCAMsg(rtc::LS_ERROR, "Error in " #expr, (const char*)&err); \
+ } \
+ } while (0)
+
+#define WEBRTC_CA_LOG_WARN(expr) \
do { \
err = expr; \
if (err != noErr) { \
- logCAMsg(rtc::LS_ERROR, "Error in " #expr, \
- (const char*) & err); \
- return -1; \
+ logCAMsg(rtc::LS_WARNING, "Error in " #expr, (const char*)&err); \
} \
} while (0)
-#define WEBRTC_CA_LOG_ERR(expr) \
- do { \
- err = expr; \
- if (err != noErr) { \
- logCAMsg(rtc::LS_ERROR, "Error in " #expr, \
- (const char*) & err); \
- } \
- } while (0)
-
-#define WEBRTC_CA_LOG_WARN(expr) \
- do { \
- err = expr; \
- if (err != noErr) { \
- logCAMsg(rtc::LS_WARNING, "Error in " #expr, \
- (const char*) & err); \
- } \
- } while (0)
-
AudioMixerManagerMac::AudioMixerManagerMac()
: _inputDeviceID(kAudioObjectUnknown),
_outputDeviceID(kAudioObjectUnknown),
@@ -876,8 +873,8 @@
// CoreAudio errors are best interpreted as four character strings.
void AudioMixerManagerMac::logCAMsg(const rtc::LoggingSeverity sev,
- const char* msg,
- const char* err) {
+ const char* msg,
+ const char* err) {
RTC_DCHECK(msg != NULL);
RTC_DCHECK(err != NULL);
RTC_DCHECK(sev == rtc::LS_ERROR || sev == rtc::LS_WARNING);
@@ -895,7 +892,7 @@
}
#else
// We need to flip the characters in this case.
- switch (sev) {
+ switch (sev) {
case rtc::LS_ERROR:
LOG(LS_ERROR) << msg << ": " << err[3] << err[2] << err[1] << err[0];
break;
diff --git a/modules/audio_device/win/audio_device_core_win.cc b/modules/audio_device/win/audio_device_core_win.cc
index 6fcbb6e..32c3f94 100644
--- a/modules/audio_device/win/audio_device_core_win.cc
+++ b/modules/audio_device/win/audio_device_core_win.cc
@@ -8,13 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#pragma warning(disable: 4995) // name was marked as #pragma deprecated
+#pragma warning(disable : 4995) // name was marked as #pragma deprecated
#if (_MSC_VER >= 1310) && (_MSC_VER < 1400)
// Reports the major and minor versions of the compiler.
-// For example, 1310 for Microsoft Visual C++ .NET 2003. 1310 represents version 13 and a 1.0 point release.
-// The Visual C++ 2005 compiler version is 1400.
-// Type cl /? at the command line to see the major and minor versions of your compiler along with the build number.
+// For example, 1310 for Microsoft Visual C++ .NET 2003. 1310 represents version
+// 13 and a 1.0 point release. The Visual C++ 2005 compiler version is 1400.
+// Type cl /? at the command line to see the major and minor versions of your
+// compiler along with the build number.
#pragma message(">> INFO: Windows Core Audio is not supported in VS 2003")
#endif
@@ -27,13 +28,13 @@
#include <assert.h>
#include <string.h>
-#include <windows.h>
+#include <Functiondiscoverykeys_devpkey.h>
#include <comdef.h>
#include <dmo.h>
-#include <Functiondiscoverykeys_devpkey.h>
#include <mmsystem.h>
#include <strsafe.h>
#include <uuids.h>
+#include <windows.h>
#include <iomanip>
@@ -42,25 +43,38 @@
#include "system_wrappers/include/sleep.h"
// Macro that calls a COM method returning HRESULT value.
-#define EXIT_ON_ERROR(hres) do { if (FAILED(hres)) goto Exit; } while(0)
+#define EXIT_ON_ERROR(hres) \
+ do { \
+ if (FAILED(hres)) \
+ goto Exit; \
+ } while (0)
// Macro that continues to a COM error.
-#define CONTINUE_ON_ERROR(hres) do { if (FAILED(hres)) goto Next; } while(0)
+#define CONTINUE_ON_ERROR(hres) \
+ do { \
+ if (FAILED(hres)) \
+ goto Next; \
+ } while (0)
// Macro that releases a COM object if not NULL.
-#define SAFE_RELEASE(p) do { if ((p)) { (p)->Release(); (p) = NULL; } } while(0)
+#define SAFE_RELEASE(p) \
+ do { \
+ if ((p)) { \
+ (p)->Release(); \
+ (p) = NULL; \
+ } \
+ } while (0)
-#define ROUND(x) ((x) >=0 ? (int)((x) + 0.5) : (int)((x) - 0.5))
+#define ROUND(x) ((x) >= 0 ? (int)((x) + 0.5) : (int)((x)-0.5))
// REFERENCE_TIME time units per millisecond
-#define REFTIMES_PER_MILLISEC 10000
+#define REFTIMES_PER_MILLISEC 10000
-typedef struct tagTHREADNAME_INFO
-{
- DWORD dwType; // must be 0x1000
- LPCSTR szName; // pointer to name (in user addr space)
- DWORD dwThreadID; // thread ID (-1=caller thread)
- DWORD dwFlags; // reserved for future use, must be zero
+typedef struct tagTHREADNAME_INFO {
+ DWORD dwType; // must be 0x1000
+ LPCSTR szName; // pointer to name (in user addr space)
+ DWORD dwThreadID; // thread ID (-1=caller thread)
+ DWORD dwFlags; // reserved for future use, must be zero
} THREADNAME_INFO;
namespace webrtc {
@@ -68,11 +82,7 @@
enum { COM_THREADING_MODEL = COINIT_MULTITHREADED };
-enum
-{
- kAecCaptureStreamIndex = 0,
- kAecRenderStreamIndex = 1
-};
+enum { kAecCaptureStreamIndex = 0, kAecRenderStreamIndex = 1 };
// An implementation of IMediaBuffer, as required for
// IMediaObject::ProcessOutput(). After consuming data provided by
@@ -80,95 +90,75 @@
//
// Example implementation:
// http://msdn.microsoft.com/en-us/library/dd376684(v=vs.85).aspx
-class MediaBufferImpl : public IMediaBuffer
-{
-public:
- explicit MediaBufferImpl(DWORD maxLength)
- : _data(new BYTE[maxLength]),
- _length(0),
- _maxLength(maxLength),
- _refCount(0)
- {}
+class MediaBufferImpl : public IMediaBuffer {
+ public:
+ explicit MediaBufferImpl(DWORD maxLength)
+ : _data(new BYTE[maxLength]),
+ _length(0),
+ _maxLength(maxLength),
+ _refCount(0) {}
- // IMediaBuffer methods.
- STDMETHOD(GetBufferAndLength(BYTE** ppBuffer, DWORD* pcbLength))
- {
- if (!ppBuffer || !pcbLength)
- {
- return E_POINTER;
- }
-
- *ppBuffer = _data;
- *pcbLength = _length;
-
- return S_OK;
+ // IMediaBuffer methods.
+ STDMETHOD(GetBufferAndLength(BYTE** ppBuffer, DWORD* pcbLength)) {
+ if (!ppBuffer || !pcbLength) {
+ return E_POINTER;
}
- STDMETHOD(GetMaxLength(DWORD* pcbMaxLength))
- {
- if (!pcbMaxLength)
- {
- return E_POINTER;
- }
+ *ppBuffer = _data;
+ *pcbLength = _length;
- *pcbMaxLength = _maxLength;
- return S_OK;
+ return S_OK;
+ }
+
+ STDMETHOD(GetMaxLength(DWORD* pcbMaxLength)) {
+ if (!pcbMaxLength) {
+ return E_POINTER;
}
- STDMETHOD(SetLength(DWORD cbLength))
- {
- if (cbLength > _maxLength)
- {
- return E_INVALIDARG;
- }
+ *pcbMaxLength = _maxLength;
+ return S_OK;
+ }
- _length = cbLength;
- return S_OK;
+ STDMETHOD(SetLength(DWORD cbLength)) {
+ if (cbLength > _maxLength) {
+ return E_INVALIDARG;
}
- // IUnknown methods.
- STDMETHOD_(ULONG, AddRef())
- {
- return InterlockedIncrement(&_refCount);
+ _length = cbLength;
+ return S_OK;
+ }
+
+ // IUnknown methods.
+ STDMETHOD_(ULONG, AddRef()) { return InterlockedIncrement(&_refCount); }
+
+ STDMETHOD(QueryInterface(REFIID riid, void** ppv)) {
+ if (!ppv) {
+ return E_POINTER;
+ } else if (riid != IID_IMediaBuffer && riid != IID_IUnknown) {
+ return E_NOINTERFACE;
}
- STDMETHOD(QueryInterface(REFIID riid, void** ppv))
- {
- if (!ppv)
- {
- return E_POINTER;
- }
- else if (riid != IID_IMediaBuffer && riid != IID_IUnknown)
- {
- return E_NOINTERFACE;
- }
+ *ppv = static_cast<IMediaBuffer*>(this);
+ AddRef();
+ return S_OK;
+ }
- *ppv = static_cast<IMediaBuffer*>(this);
- AddRef();
- return S_OK;
+ STDMETHOD_(ULONG, Release()) {
+ LONG refCount = InterlockedDecrement(&_refCount);
+ if (refCount == 0) {
+ delete this;
}
- STDMETHOD_(ULONG, Release())
- {
- LONG refCount = InterlockedDecrement(&_refCount);
- if (refCount == 0)
- {
- delete this;
- }
+ return refCount;
+ }
- return refCount;
- }
+ private:
+ ~MediaBufferImpl() { delete[] _data; }
-private:
- ~MediaBufferImpl()
- {
- delete [] _data;
- }
-
- BYTE* _data;
- DWORD _length;
- const DWORD _maxLength;
- LONG _refCount;
+ BYTE* _data;
+ DWORD _length;
+ const DWORD _maxLength;
+ LONG _refCount;
};
} // namespace
@@ -180,228 +170,204 @@
// CoreAudioIsSupported
// ----------------------------------------------------------------------------
-bool AudioDeviceWindowsCore::CoreAudioIsSupported()
-{
- LOG(LS_VERBOSE) << __FUNCTION__;
+bool AudioDeviceWindowsCore::CoreAudioIsSupported() {
+ LOG(LS_VERBOSE) << __FUNCTION__;
- bool MMDeviceIsAvailable(false);
- bool coreAudioIsSupported(false);
+ bool MMDeviceIsAvailable(false);
+ bool coreAudioIsSupported(false);
- HRESULT hr(S_OK);
- TCHAR buf[MAXERRORLENGTH];
- TCHAR errorText[MAXERRORLENGTH];
+ HRESULT hr(S_OK);
+ TCHAR buf[MAXERRORLENGTH];
+ TCHAR errorText[MAXERRORLENGTH];
- // 1) Check if Windows version is Vista SP1 or later.
- //
- // CoreAudio is only available on Vista SP1 and later.
- //
- OSVERSIONINFOEX osvi;
- DWORDLONG dwlConditionMask = 0;
- int op = VER_LESS_EQUAL;
+ // 1) Check if Windows version is Vista SP1 or later.
+ //
+ // CoreAudio is only available on Vista SP1 and later.
+ //
+ OSVERSIONINFOEX osvi;
+ DWORDLONG dwlConditionMask = 0;
+ int op = VER_LESS_EQUAL;
- // Initialize the OSVERSIONINFOEX structure.
- ZeroMemory(&osvi, sizeof(OSVERSIONINFOEX));
- osvi.dwOSVersionInfoSize = sizeof(OSVERSIONINFOEX);
- osvi.dwMajorVersion = 6;
- osvi.dwMinorVersion = 0;
- osvi.wServicePackMajor = 0;
- osvi.wServicePackMinor = 0;
- osvi.wProductType = VER_NT_WORKSTATION;
+ // Initialize the OSVERSIONINFOEX structure.
+ ZeroMemory(&osvi, sizeof(OSVERSIONINFOEX));
+ osvi.dwOSVersionInfoSize = sizeof(OSVERSIONINFOEX);
+ osvi.dwMajorVersion = 6;
+ osvi.dwMinorVersion = 0;
+ osvi.wServicePackMajor = 0;
+ osvi.wServicePackMinor = 0;
+ osvi.wProductType = VER_NT_WORKSTATION;
- // Initialize the condition mask.
- VER_SET_CONDITION(dwlConditionMask, VER_MAJORVERSION, op);
- VER_SET_CONDITION(dwlConditionMask, VER_MINORVERSION, op);
- VER_SET_CONDITION(dwlConditionMask, VER_SERVICEPACKMAJOR, op);
- VER_SET_CONDITION(dwlConditionMask, VER_SERVICEPACKMINOR, op);
- VER_SET_CONDITION(dwlConditionMask, VER_PRODUCT_TYPE, VER_EQUAL);
+ // Initialize the condition mask.
+ VER_SET_CONDITION(dwlConditionMask, VER_MAJORVERSION, op);
+ VER_SET_CONDITION(dwlConditionMask, VER_MINORVERSION, op);
+ VER_SET_CONDITION(dwlConditionMask, VER_SERVICEPACKMAJOR, op);
+ VER_SET_CONDITION(dwlConditionMask, VER_SERVICEPACKMINOR, op);
+ VER_SET_CONDITION(dwlConditionMask, VER_PRODUCT_TYPE, VER_EQUAL);
- DWORD dwTypeMask = VER_MAJORVERSION | VER_MINORVERSION |
- VER_SERVICEPACKMAJOR | VER_SERVICEPACKMINOR |
- VER_PRODUCT_TYPE;
+ DWORD dwTypeMask = VER_MAJORVERSION | VER_MINORVERSION |
+ VER_SERVICEPACKMAJOR | VER_SERVICEPACKMINOR |
+ VER_PRODUCT_TYPE;
- // Perform the test.
- BOOL isVistaRTMorXP = VerifyVersionInfo(&osvi, dwTypeMask,
- dwlConditionMask);
- if (isVistaRTMorXP != 0)
- {
- LOG(LS_VERBOSE)
- << "*** Windows Core Audio is only supported on Vista SP1 or later"
- << " => will revert to the Wave API ***";
- return false;
+ // Perform the test.
+ BOOL isVistaRTMorXP = VerifyVersionInfo(&osvi, dwTypeMask, dwlConditionMask);
+ if (isVistaRTMorXP != 0) {
+ LOG(LS_VERBOSE)
+ << "*** Windows Core Audio is only supported on Vista SP1 or later"
+ << " => will revert to the Wave API ***";
+ return false;
+ }
+
+ // 2) Initializes the COM library for use by the calling thread.
+
+ // The COM init wrapper sets the thread's concurrency model to MTA,
+ // and creates a new apartment for the thread if one is required. The
+ // wrapper also ensures that each call to CoInitializeEx is balanced
+ // by a corresponding call to CoUninitialize.
+ //
+ ScopedCOMInitializer comInit(ScopedCOMInitializer::kMTA);
+ if (!comInit.succeeded()) {
+ // Things will work even if an STA thread is calling this method but we
+ // want to ensure that MTA is used and therefore return false here.
+ return false;
+ }
+
+ // 3) Check if the MMDevice API is available.
+ //
+ // The Windows Multimedia Device (MMDevice) API enables audio clients to
+ // discover audio endpoint devices, determine their capabilities, and create
+ // driver instances for those devices.
+ // Header file Mmdeviceapi.h defines the interfaces in the MMDevice API.
+ // The MMDevice API consists of several interfaces. The first of these is the
+ // IMMDeviceEnumerator interface. To access the interfaces in the MMDevice
+ // API, a client obtains a reference to the IMMDeviceEnumerator interface of a
+ // device-enumerator object by calling the CoCreateInstance function.
+ //
+ // Through the IMMDeviceEnumerator interface, the client can obtain references
+ // to the other interfaces in the MMDevice API. The MMDevice API implements
+ // the following interfaces:
+ //
+ // IMMDevice Represents an audio device.
+ // IMMDeviceCollection Represents a collection of audio devices.
+ // IMMDeviceEnumerator Provides methods for enumerating audio devices.
+ // IMMEndpoint Represents an audio endpoint device.
+ //
+ IMMDeviceEnumerator* pIMMD(NULL);
+ const CLSID CLSID_MMDeviceEnumerator = __uuidof(MMDeviceEnumerator);
+ const IID IID_IMMDeviceEnumerator = __uuidof(IMMDeviceEnumerator);
+
+ hr = CoCreateInstance(
+ CLSID_MMDeviceEnumerator, // GUID value of MMDeviceEnumerator coclass
+ NULL, CLSCTX_ALL,
+ IID_IMMDeviceEnumerator, // GUID value of the IMMDeviceEnumerator
+ // interface
+ (void**)&pIMMD);
+
+ if (FAILED(hr)) {
+ LOG(LS_ERROR) << "AudioDeviceWindowsCore::CoreAudioIsSupported()"
+ << " Failed to create the required COM object (hr=" << hr
+ << ")";
+ LOG(LS_VERBOSE) << "AudioDeviceWindowsCore::CoreAudioIsSupported()"
+ << " CoCreateInstance(MMDeviceEnumerator) failed (hr=" << hr
+ << ")";
+
+ const DWORD dwFlags =
+ FORMAT_MESSAGE_FROM_SYSTEM | FORMAT_MESSAGE_IGNORE_INSERTS;
+ const DWORD dwLangID = MAKELANGID(LANG_ENGLISH, SUBLANG_ENGLISH_US);
+
+ // Gets the system's human readable message string for this HRESULT.
+ // All error message in English by default.
+ DWORD messageLength = ::FormatMessageW(dwFlags, 0, hr, dwLangID, errorText,
+ MAXERRORLENGTH, NULL);
+
+ assert(messageLength <= MAXERRORLENGTH);
+
+ // Trims tailing white space (FormatMessage() leaves a trailing cr-lf.).
+ for (; messageLength && ::isspace(errorText[messageLength - 1]);
+ --messageLength) {
+ errorText[messageLength - 1] = '\0';
}
- // 2) Initializes the COM library for use by the calling thread.
+ StringCchPrintf(buf, MAXERRORLENGTH, TEXT("Error details: "));
+ StringCchCat(buf, MAXERRORLENGTH, errorText);
+ LOG(LS_VERBOSE) << buf;
+ } else {
+ MMDeviceIsAvailable = true;
+ LOG(LS_VERBOSE) << "AudioDeviceWindowsCore::CoreAudioIsSupported()"
+ << " CoCreateInstance(MMDeviceEnumerator) succeeded (hr="
+ << hr << ")";
+ SAFE_RELEASE(pIMMD);
+ }
- // The COM init wrapper sets the thread's concurrency model to MTA,
- // and creates a new apartment for the thread if one is required. The
- // wrapper also ensures that each call to CoInitializeEx is balanced
- // by a corresponding call to CoUninitialize.
- //
- ScopedCOMInitializer comInit(ScopedCOMInitializer::kMTA);
- if (!comInit.succeeded()) {
- // Things will work even if an STA thread is calling this method but we
- // want to ensure that MTA is used and therefore return false here.
+ // 4) Verify that we can create and initialize our Core Audio class.
+ //
+ // Also, perform a limited "API test" to ensure that Core Audio is supported
+ // for all devices.
+ //
+ if (MMDeviceIsAvailable) {
+ coreAudioIsSupported = false;
+
+ AudioDeviceWindowsCore* p = new AudioDeviceWindowsCore();
+ if (p == NULL) {
return false;
}
- // 3) Check if the MMDevice API is available.
- //
- // The Windows Multimedia Device (MMDevice) API enables audio clients to
- // discover audio endpoint devices, determine their capabilities, and create
- // driver instances for those devices.
- // Header file Mmdeviceapi.h defines the interfaces in the MMDevice API.
- // The MMDevice API consists of several interfaces. The first of these is the
- // IMMDeviceEnumerator interface. To access the interfaces in the MMDevice API,
- // a client obtains a reference to the IMMDeviceEnumerator interface of a
- // device-enumerator object by calling the CoCreateInstance function.
- //
- // Through the IMMDeviceEnumerator interface, the client can obtain references
- // to the other interfaces in the MMDevice API. The MMDevice API implements
- // the following interfaces:
- //
- // IMMDevice Represents an audio device.
- // IMMDeviceCollection Represents a collection of audio devices.
- // IMMDeviceEnumerator Provides methods for enumerating audio devices.
- // IMMEndpoint Represents an audio endpoint device.
- //
- IMMDeviceEnumerator* pIMMD(NULL);
- const CLSID CLSID_MMDeviceEnumerator = __uuidof(MMDeviceEnumerator);
- const IID IID_IMMDeviceEnumerator = __uuidof(IMMDeviceEnumerator);
+ int ok(0);
+ int temp_ok(0);
+ bool available(false);
- hr = CoCreateInstance(
- CLSID_MMDeviceEnumerator, // GUID value of MMDeviceEnumerator coclass
- NULL,
- CLSCTX_ALL,
- IID_IMMDeviceEnumerator, // GUID value of the IMMDeviceEnumerator interface
- (void**)&pIMMD );
-
- if (FAILED(hr))
- {
- LOG(LS_ERROR) << "AudioDeviceWindowsCore::CoreAudioIsSupported()"
- << " Failed to create the required COM object (hr="
- << hr << ")";
- LOG(LS_VERBOSE) << "AudioDeviceWindowsCore::CoreAudioIsSupported()"
- << " CoCreateInstance(MMDeviceEnumerator) failed (hr="
- << hr << ")";
-
- const DWORD dwFlags = FORMAT_MESSAGE_FROM_SYSTEM |
- FORMAT_MESSAGE_IGNORE_INSERTS;
- const DWORD dwLangID = MAKELANGID(LANG_ENGLISH, SUBLANG_ENGLISH_US);
-
- // Gets the system's human readable message string for this HRESULT.
- // All error message in English by default.
- DWORD messageLength = ::FormatMessageW(dwFlags,
- 0,
- hr,
- dwLangID,
- errorText,
- MAXERRORLENGTH,
- NULL);
-
- assert(messageLength <= MAXERRORLENGTH);
-
- // Trims tailing white space (FormatMessage() leaves a trailing cr-lf.).
- for (; messageLength && ::isspace(errorText[messageLength - 1]);
- --messageLength)
- {
- errorText[messageLength - 1] = '\0';
- }
-
- StringCchPrintf(buf, MAXERRORLENGTH, TEXT("Error details: "));
- StringCchCat(buf, MAXERRORLENGTH, errorText);
- LOG(LS_VERBOSE) << buf;
- }
- else
- {
- MMDeviceIsAvailable = true;
- LOG(LS_VERBOSE) << "AudioDeviceWindowsCore::CoreAudioIsSupported()"
- << " CoCreateInstance(MMDeviceEnumerator) succeeded (hr=" << hr
- << ")";
- SAFE_RELEASE(pIMMD);
+ if (p->Init() != InitStatus::OK) {
+ ok |= -1;
}
- // 4) Verify that we can create and initialize our Core Audio class.
- //
- // Also, perform a limited "API test" to ensure that Core Audio is supported for all devices.
- //
- if (MMDeviceIsAvailable)
- {
- coreAudioIsSupported = false;
-
- AudioDeviceWindowsCore* p = new AudioDeviceWindowsCore();
- if (p == NULL)
- {
- return false;
- }
-
- int ok(0);
- int temp_ok(0);
- bool available(false);
-
- if (p->Init() != InitStatus::OK) {
- ok |= -1;
- }
-
- int16_t numDevsRec = p->RecordingDevices();
- for (uint16_t i = 0; i < numDevsRec; i++)
- {
- ok |= p->SetRecordingDevice(i);
- temp_ok = p->RecordingIsAvailable(available);
- ok |= temp_ok;
- ok |= (available == false);
- if (available)
- {
- ok |= p->InitMicrophone();
- }
- if (ok)
- {
- LOG(LS_WARNING)
- << "AudioDeviceWindowsCore::CoreAudioIsSupported()"
- << " Failed to use Core Audio Recording for device id="
- << i;
- }
- }
-
- int16_t numDevsPlay = p->PlayoutDevices();
- for (uint16_t i = 0; i < numDevsPlay; i++)
- {
- ok |= p->SetPlayoutDevice(i);
- temp_ok = p->PlayoutIsAvailable(available);
- ok |= temp_ok;
- ok |= (available == false);
- if (available)
- {
- ok |= p->InitSpeaker();
- }
- if (ok)
- {
- LOG(LS_WARNING)
- << "AudioDeviceWindowsCore::CoreAudioIsSupported()"
- << " Failed to use Core Audio Playout for device id=" << i;
- }
- }
-
- ok |= p->Terminate();
-
- if (ok == 0)
- {
- coreAudioIsSupported = true;
- }
-
- delete p;
+ int16_t numDevsRec = p->RecordingDevices();
+ for (uint16_t i = 0; i < numDevsRec; i++) {
+ ok |= p->SetRecordingDevice(i);
+ temp_ok = p->RecordingIsAvailable(available);
+ ok |= temp_ok;
+ ok |= (available == false);
+ if (available) {
+ ok |= p->InitMicrophone();
+ }
+ if (ok) {
+ LOG(LS_WARNING) << "AudioDeviceWindowsCore::CoreAudioIsSupported()"
+ << " Failed to use Core Audio Recording for device id="
+ << i;
+ }
}
- if (coreAudioIsSupported)
- {
- LOG(LS_VERBOSE) << "*** Windows Core Audio is supported ***";
- }
- else
- {
- LOG(LS_VERBOSE) << "*** Windows Core Audio is NOT supported"
- << " => will revert to the Wave API ***";
+ int16_t numDevsPlay = p->PlayoutDevices();
+ for (uint16_t i = 0; i < numDevsPlay; i++) {
+ ok |= p->SetPlayoutDevice(i);
+ temp_ok = p->PlayoutIsAvailable(available);
+ ok |= temp_ok;
+ ok |= (available == false);
+ if (available) {
+ ok |= p->InitSpeaker();
+ }
+ if (ok) {
+ LOG(LS_WARNING) << "AudioDeviceWindowsCore::CoreAudioIsSupported()"
+ << " Failed to use Core Audio Playout for device id="
+ << i;
+ }
}
- return (coreAudioIsSupported);
+ ok |= p->Terminate();
+
+ if (ok == 0) {
+ coreAudioIsSupported = true;
+ }
+
+ delete p;
+ }
+
+ if (coreAudioIsSupported) {
+ LOG(LS_VERBOSE) << "*** Windows Core Audio is supported ***";
+ } else {
+ LOG(LS_VERBOSE) << "*** Windows Core Audio is NOT supported"
+ << " => will revert to the Wave API ***";
+ }
+
+ return (coreAudioIsSupported);
}
// ============================================================================
@@ -566,76 +532,62 @@
// AudioDeviceWindowsCore() - dtor
// ----------------------------------------------------------------------------
-AudioDeviceWindowsCore::~AudioDeviceWindowsCore()
-{
- LOG(LS_INFO) << __FUNCTION__ << " destroyed";
+AudioDeviceWindowsCore::~AudioDeviceWindowsCore() {
+ LOG(LS_INFO) << __FUNCTION__ << " destroyed";
- Terminate();
+ Terminate();
- // The IMMDeviceEnumerator is created during construction. Must release
- // it here and not in Terminate() since we don't recreate it in Init().
- SAFE_RELEASE(_ptrEnumerator);
+ // The IMMDeviceEnumerator is created during construction. Must release
+ // it here and not in Terminate() since we don't recreate it in Init().
+ SAFE_RELEASE(_ptrEnumerator);
- _ptrAudioBuffer = NULL;
+ _ptrAudioBuffer = NULL;
- if (NULL != _hRenderSamplesReadyEvent)
- {
- CloseHandle(_hRenderSamplesReadyEvent);
- _hRenderSamplesReadyEvent = NULL;
+ if (NULL != _hRenderSamplesReadyEvent) {
+ CloseHandle(_hRenderSamplesReadyEvent);
+ _hRenderSamplesReadyEvent = NULL;
+ }
+
+ if (NULL != _hCaptureSamplesReadyEvent) {
+ CloseHandle(_hCaptureSamplesReadyEvent);
+ _hCaptureSamplesReadyEvent = NULL;
+ }
+
+ if (NULL != _hRenderStartedEvent) {
+ CloseHandle(_hRenderStartedEvent);
+ _hRenderStartedEvent = NULL;
+ }
+
+ if (NULL != _hCaptureStartedEvent) {
+ CloseHandle(_hCaptureStartedEvent);
+ _hCaptureStartedEvent = NULL;
+ }
+
+ if (NULL != _hShutdownRenderEvent) {
+ CloseHandle(_hShutdownRenderEvent);
+ _hShutdownRenderEvent = NULL;
+ }
+
+ if (NULL != _hShutdownCaptureEvent) {
+ CloseHandle(_hShutdownCaptureEvent);
+ _hShutdownCaptureEvent = NULL;
+ }
+
+ if (NULL != _hSetCaptureVolumeEvent) {
+ CloseHandle(_hSetCaptureVolumeEvent);
+ _hSetCaptureVolumeEvent = NULL;
+ }
+
+ if (_avrtLibrary) {
+ BOOL freeOK = FreeLibrary(_avrtLibrary);
+ if (!freeOK) {
+ LOG(LS_WARNING) << "AudioDeviceWindowsCore::~AudioDeviceWindowsCore()"
+ << " failed to free the loaded Avrt DLL module correctly";
+ } else {
+ LOG(LS_WARNING) << "AudioDeviceWindowsCore::~AudioDeviceWindowsCore()"
+ << " the Avrt DLL module is now unloaded";
}
-
- if (NULL != _hCaptureSamplesReadyEvent)
- {
- CloseHandle(_hCaptureSamplesReadyEvent);
- _hCaptureSamplesReadyEvent = NULL;
- }
-
- if (NULL != _hRenderStartedEvent)
- {
- CloseHandle(_hRenderStartedEvent);
- _hRenderStartedEvent = NULL;
- }
-
- if (NULL != _hCaptureStartedEvent)
- {
- CloseHandle(_hCaptureStartedEvent);
- _hCaptureStartedEvent = NULL;
- }
-
- if (NULL != _hShutdownRenderEvent)
- {
- CloseHandle(_hShutdownRenderEvent);
- _hShutdownRenderEvent = NULL;
- }
-
- if (NULL != _hShutdownCaptureEvent)
- {
- CloseHandle(_hShutdownCaptureEvent);
- _hShutdownCaptureEvent = NULL;
- }
-
- if (NULL != _hSetCaptureVolumeEvent)
- {
- CloseHandle(_hSetCaptureVolumeEvent);
- _hSetCaptureVolumeEvent = NULL;
- }
-
- if (_avrtLibrary)
- {
- BOOL freeOK = FreeLibrary(_avrtLibrary);
- if (!freeOK)
- {
- LOG(LS_WARNING)
- << "AudioDeviceWindowsCore::~AudioDeviceWindowsCore()"
- << " failed to free the loaded Avrt DLL module correctly";
- }
- else
- {
- LOG(LS_WARNING)
- << "AudioDeviceWindowsCore::~AudioDeviceWindowsCore()"
- << " the Avrt DLL module is now unloaded";
- }
- }
+ }
}
// ============================================================================
@@ -646,28 +598,26 @@
// AttachAudioBuffer
// ----------------------------------------------------------------------------
-void AudioDeviceWindowsCore::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer)
-{
+void AudioDeviceWindowsCore::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
+ _ptrAudioBuffer = audioBuffer;
- _ptrAudioBuffer = audioBuffer;
-
- // Inform the AudioBuffer about default settings for this implementation.
- // Set all values to zero here since the actual settings will be done by
- // InitPlayout and InitRecording later.
- _ptrAudioBuffer->SetRecordingSampleRate(0);
- _ptrAudioBuffer->SetPlayoutSampleRate(0);
- _ptrAudioBuffer->SetRecordingChannels(0);
- _ptrAudioBuffer->SetPlayoutChannels(0);
+ // Inform the AudioBuffer about default settings for this implementation.
+ // Set all values to zero here since the actual settings will be done by
+ // InitPlayout and InitRecording later.
+ _ptrAudioBuffer->SetRecordingSampleRate(0);
+ _ptrAudioBuffer->SetPlayoutSampleRate(0);
+ _ptrAudioBuffer->SetRecordingChannels(0);
+ _ptrAudioBuffer->SetPlayoutChannels(0);
}
// ----------------------------------------------------------------------------
// ActiveAudioLayer
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const
-{
- audioLayer = AudioDeviceModule::kWindowsCoreAudio;
- return 0;
+int32_t AudioDeviceWindowsCore::ActiveAudioLayer(
+ AudioDeviceModule::AudioLayer& audioLayer) const {
+ audioLayer = AudioDeviceModule::kWindowsCoreAudio;
+ return 0;
}
// ----------------------------------------------------------------------------
@@ -696,339 +646,297 @@
// Terminate
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::Terminate()
-{
+int32_t AudioDeviceWindowsCore::Terminate() {
+ rtc::CritScope lock(&_critSect);
- rtc::CritScope lock(&_critSect);
-
- if (!_initialized) {
- return 0;
- }
-
- _initialized = false;
- _speakerIsInitialized = false;
- _microphoneIsInitialized = false;
- _playing = false;
- _recording = false;
-
- SAFE_RELEASE(_ptrRenderCollection);
- SAFE_RELEASE(_ptrCaptureCollection);
- SAFE_RELEASE(_ptrDeviceOut);
- SAFE_RELEASE(_ptrDeviceIn);
- SAFE_RELEASE(_ptrClientOut);
- SAFE_RELEASE(_ptrClientIn);
- SAFE_RELEASE(_ptrRenderClient);
- SAFE_RELEASE(_ptrCaptureClient);
- SAFE_RELEASE(_ptrCaptureVolume);
- SAFE_RELEASE(_ptrRenderSimpleVolume);
-
+ if (!_initialized) {
return 0;
+ }
+
+ _initialized = false;
+ _speakerIsInitialized = false;
+ _microphoneIsInitialized = false;
+ _playing = false;
+ _recording = false;
+
+ SAFE_RELEASE(_ptrRenderCollection);
+ SAFE_RELEASE(_ptrCaptureCollection);
+ SAFE_RELEASE(_ptrDeviceOut);
+ SAFE_RELEASE(_ptrDeviceIn);
+ SAFE_RELEASE(_ptrClientOut);
+ SAFE_RELEASE(_ptrClientIn);
+ SAFE_RELEASE(_ptrRenderClient);
+ SAFE_RELEASE(_ptrCaptureClient);
+ SAFE_RELEASE(_ptrCaptureVolume);
+ SAFE_RELEASE(_ptrRenderSimpleVolume);
+
+ return 0;
}
// ----------------------------------------------------------------------------
// Initialized
// ----------------------------------------------------------------------------
-bool AudioDeviceWindowsCore::Initialized() const
-{
- return (_initialized);
+bool AudioDeviceWindowsCore::Initialized() const {
+ return (_initialized);
}
// ----------------------------------------------------------------------------
// InitSpeaker
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::InitSpeaker()
-{
+int32_t AudioDeviceWindowsCore::InitSpeaker() {
+ rtc::CritScope lock(&_critSect);
- rtc::CritScope lock(&_critSect);
+ if (_playing) {
+ return -1;
+ }
- if (_playing)
- {
- return -1;
+ if (_ptrDeviceOut == NULL) {
+ return -1;
+ }
+
+ if (_usingOutputDeviceIndex) {
+ int16_t nDevices = PlayoutDevices();
+ if (_outputDeviceIndex > (nDevices - 1)) {
+ LOG(LS_ERROR) << "current device selection is invalid => unable to"
+ << " initialize";
+ return -1;
}
+ }
- if (_ptrDeviceOut == NULL)
- {
- return -1;
- }
+ int32_t ret(0);
- if (_usingOutputDeviceIndex)
- {
- int16_t nDevices = PlayoutDevices();
- if (_outputDeviceIndex > (nDevices - 1))
- {
- LOG(LS_ERROR) << "current device selection is invalid => unable to"
- << " initialize";
- return -1;
- }
- }
+ SAFE_RELEASE(_ptrDeviceOut);
+ if (_usingOutputDeviceIndex) {
+ // Refresh the selected rendering endpoint device using current index
+ ret = _GetListDevice(eRender, _outputDeviceIndex, &_ptrDeviceOut);
+ } else {
+ ERole role;
+ (_outputDevice == AudioDeviceModule::kDefaultDevice)
+ ? role = eConsole
+ : role = eCommunications;
+ // Refresh the selected rendering endpoint device using role
+ ret = _GetDefaultDevice(eRender, role, &_ptrDeviceOut);
+ }
- int32_t ret(0);
-
+ if (ret != 0 || (_ptrDeviceOut == NULL)) {
+ LOG(LS_ERROR) << "failed to initialize the rendering enpoint device";
SAFE_RELEASE(_ptrDeviceOut);
- if (_usingOutputDeviceIndex)
- {
- // Refresh the selected rendering endpoint device using current index
- ret = _GetListDevice(eRender, _outputDeviceIndex, &_ptrDeviceOut);
- }
- else
- {
- ERole role;
- (_outputDevice == AudioDeviceModule::kDefaultDevice) ? role = eConsole : role = eCommunications;
- // Refresh the selected rendering endpoint device using role
- ret = _GetDefaultDevice(eRender, role, &_ptrDeviceOut);
- }
+ return -1;
+ }
- if (ret != 0 || (_ptrDeviceOut == NULL))
- {
- LOG(LS_ERROR) << "failed to initialize the rendering enpoint device";
- SAFE_RELEASE(_ptrDeviceOut);
- return -1;
- }
-
- IAudioSessionManager* pManager = NULL;
- ret = _ptrDeviceOut->Activate(__uuidof(IAudioSessionManager),
- CLSCTX_ALL,
- NULL,
- (void**)&pManager);
- if (ret != 0 || pManager == NULL)
- {
- LOG(LS_ERROR) << "failed to initialize the render manager";
- SAFE_RELEASE(pManager);
- return -1;
- }
-
- SAFE_RELEASE(_ptrRenderSimpleVolume);
- ret = pManager->GetSimpleAudioVolume(NULL, FALSE, &_ptrRenderSimpleVolume);
- if (ret != 0 || _ptrRenderSimpleVolume == NULL)
- {
- LOG(LS_ERROR) << "failed to initialize the render simple volume";
- SAFE_RELEASE(pManager);
- SAFE_RELEASE(_ptrRenderSimpleVolume);
- return -1;
- }
+ IAudioSessionManager* pManager = NULL;
+ ret = _ptrDeviceOut->Activate(__uuidof(IAudioSessionManager), CLSCTX_ALL,
+ NULL, (void**)&pManager);
+ if (ret != 0 || pManager == NULL) {
+ LOG(LS_ERROR) << "failed to initialize the render manager";
SAFE_RELEASE(pManager);
+ return -1;
+ }
- _speakerIsInitialized = true;
+ SAFE_RELEASE(_ptrRenderSimpleVolume);
+ ret = pManager->GetSimpleAudioVolume(NULL, FALSE, &_ptrRenderSimpleVolume);
+ if (ret != 0 || _ptrRenderSimpleVolume == NULL) {
+ LOG(LS_ERROR) << "failed to initialize the render simple volume";
+ SAFE_RELEASE(pManager);
+ SAFE_RELEASE(_ptrRenderSimpleVolume);
+ return -1;
+ }
+ SAFE_RELEASE(pManager);
- return 0;
+ _speakerIsInitialized = true;
+
+ return 0;
}
// ----------------------------------------------------------------------------
// InitMicrophone
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::InitMicrophone()
-{
+int32_t AudioDeviceWindowsCore::InitMicrophone() {
+ rtc::CritScope lock(&_critSect);
- rtc::CritScope lock(&_critSect);
+ if (_recording) {
+ return -1;
+ }
- if (_recording)
- {
- return -1;
+ if (_ptrDeviceIn == NULL) {
+ return -1;
+ }
+
+ if (_usingInputDeviceIndex) {
+ int16_t nDevices = RecordingDevices();
+ if (_inputDeviceIndex > (nDevices - 1)) {
+ LOG(LS_ERROR) << "current device selection is invalid => unable to"
+ << " initialize";
+ return -1;
}
+ }
- if (_ptrDeviceIn == NULL)
- {
- return -1;
- }
+ int32_t ret(0);
- if (_usingInputDeviceIndex)
- {
- int16_t nDevices = RecordingDevices();
- if (_inputDeviceIndex > (nDevices - 1))
- {
- LOG(LS_ERROR) << "current device selection is invalid => unable to"
- << " initialize";
- return -1;
- }
- }
+ SAFE_RELEASE(_ptrDeviceIn);
+ if (_usingInputDeviceIndex) {
+ // Refresh the selected capture endpoint device using current index
+ ret = _GetListDevice(eCapture, _inputDeviceIndex, &_ptrDeviceIn);
+ } else {
+ ERole role;
+ (_inputDevice == AudioDeviceModule::kDefaultDevice)
+ ? role = eConsole
+ : role = eCommunications;
+ // Refresh the selected capture endpoint device using role
+ ret = _GetDefaultDevice(eCapture, role, &_ptrDeviceIn);
+ }
- int32_t ret(0);
-
+ if (ret != 0 || (_ptrDeviceIn == NULL)) {
+ LOG(LS_ERROR) << "failed to initialize the capturing enpoint device";
SAFE_RELEASE(_ptrDeviceIn);
- if (_usingInputDeviceIndex)
- {
- // Refresh the selected capture endpoint device using current index
- ret = _GetListDevice(eCapture, _inputDeviceIndex, &_ptrDeviceIn);
- }
- else
- {
- ERole role;
- (_inputDevice == AudioDeviceModule::kDefaultDevice) ? role = eConsole : role = eCommunications;
- // Refresh the selected capture endpoint device using role
- ret = _GetDefaultDevice(eCapture, role, &_ptrDeviceIn);
- }
+ return -1;
+ }
- if (ret != 0 || (_ptrDeviceIn == NULL))
- {
- LOG(LS_ERROR) << "failed to initialize the capturing enpoint device";
- SAFE_RELEASE(_ptrDeviceIn);
- return -1;
- }
-
+ SAFE_RELEASE(_ptrCaptureVolume);
+ ret = _ptrDeviceIn->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL, NULL,
+ reinterpret_cast<void**>(&_ptrCaptureVolume));
+ if (ret != 0 || _ptrCaptureVolume == NULL) {
+ LOG(LS_ERROR) << "failed to initialize the capture volume";
SAFE_RELEASE(_ptrCaptureVolume);
- ret = _ptrDeviceIn->Activate(__uuidof(IAudioEndpointVolume),
- CLSCTX_ALL,
- NULL,
- reinterpret_cast<void **>(&_ptrCaptureVolume));
- if (ret != 0 || _ptrCaptureVolume == NULL)
- {
- LOG(LS_ERROR) << "failed to initialize the capture volume";
- SAFE_RELEASE(_ptrCaptureVolume);
- return -1;
- }
+ return -1;
+ }
- _microphoneIsInitialized = true;
+ _microphoneIsInitialized = true;
- return 0;
+ return 0;
}
// ----------------------------------------------------------------------------
// SpeakerIsInitialized
// ----------------------------------------------------------------------------
-bool AudioDeviceWindowsCore::SpeakerIsInitialized() const
-{
-
- return (_speakerIsInitialized);
+bool AudioDeviceWindowsCore::SpeakerIsInitialized() const {
+ return (_speakerIsInitialized);
}
// ----------------------------------------------------------------------------
// MicrophoneIsInitialized
// ----------------------------------------------------------------------------
-bool AudioDeviceWindowsCore::MicrophoneIsInitialized() const
-{
-
- return (_microphoneIsInitialized);
+bool AudioDeviceWindowsCore::MicrophoneIsInitialized() const {
+ return (_microphoneIsInitialized);
}
// ----------------------------------------------------------------------------
// SpeakerVolumeIsAvailable
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::SpeakerVolumeIsAvailable(bool& available)
-{
+int32_t AudioDeviceWindowsCore::SpeakerVolumeIsAvailable(bool& available) {
+ rtc::CritScope lock(&_critSect);
- rtc::CritScope lock(&_critSect);
+ if (_ptrDeviceOut == NULL) {
+ return -1;
+ }
- if (_ptrDeviceOut == NULL)
- {
- return -1;
- }
+ HRESULT hr = S_OK;
+ IAudioSessionManager* pManager = NULL;
+ ISimpleAudioVolume* pVolume = NULL;
- HRESULT hr = S_OK;
- IAudioSessionManager* pManager = NULL;
- ISimpleAudioVolume* pVolume = NULL;
+ hr = _ptrDeviceOut->Activate(__uuidof(IAudioSessionManager), CLSCTX_ALL, NULL,
+ (void**)&pManager);
+ EXIT_ON_ERROR(hr);
- hr = _ptrDeviceOut->Activate(__uuidof(IAudioSessionManager), CLSCTX_ALL, NULL, (void**)&pManager);
- EXIT_ON_ERROR(hr);
+ hr = pManager->GetSimpleAudioVolume(NULL, FALSE, &pVolume);
+ EXIT_ON_ERROR(hr);
- hr = pManager->GetSimpleAudioVolume(NULL, FALSE, &pVolume);
- EXIT_ON_ERROR(hr);
+ float volume(0.0f);
+ hr = pVolume->GetMasterVolume(&volume);
+ if (FAILED(hr)) {
+ available = false;
+ }
+ available = true;
- float volume(0.0f);
- hr = pVolume->GetMasterVolume(&volume);
- if (FAILED(hr))
- {
- available = false;
- }
- available = true;
+ SAFE_RELEASE(pManager);
+ SAFE_RELEASE(pVolume);
- SAFE_RELEASE(pManager);
- SAFE_RELEASE(pVolume);
-
- return 0;
+ return 0;
Exit:
- _TraceCOMError(hr);
- SAFE_RELEASE(pManager);
- SAFE_RELEASE(pVolume);
- return -1;
+ _TraceCOMError(hr);
+ SAFE_RELEASE(pManager);
+ SAFE_RELEASE(pVolume);
+ return -1;
}
// ----------------------------------------------------------------------------
// SetSpeakerVolume
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::SetSpeakerVolume(uint32_t volume)
-{
+int32_t AudioDeviceWindowsCore::SetSpeakerVolume(uint32_t volume) {
+ {
+ rtc::CritScope lock(&_critSect);
- {
- rtc::CritScope lock(&_critSect);
-
- if (!_speakerIsInitialized)
- {
- return -1;
- }
-
- if (_ptrDeviceOut == NULL)
- {
- return -1;
- }
+ if (!_speakerIsInitialized) {
+ return -1;
}
- if (volume < (uint32_t)MIN_CORE_SPEAKER_VOLUME ||
- volume > (uint32_t)MAX_CORE_SPEAKER_VOLUME)
- {
- return -1;
+ if (_ptrDeviceOut == NULL) {
+ return -1;
}
+ }
- HRESULT hr = S_OK;
+ if (volume < (uint32_t)MIN_CORE_SPEAKER_VOLUME ||
+ volume > (uint32_t)MAX_CORE_SPEAKER_VOLUME) {
+ return -1;
+ }
- // scale input volume to valid range (0.0 to 1.0)
- const float fLevel = (float)volume/MAX_CORE_SPEAKER_VOLUME;
- _volumeMutex.Enter();
- hr = _ptrRenderSimpleVolume->SetMasterVolume(fLevel,NULL);
- _volumeMutex.Leave();
- EXIT_ON_ERROR(hr);
+ HRESULT hr = S_OK;
- return 0;
+ // scale input volume to valid range (0.0 to 1.0)
+ const float fLevel = (float)volume / MAX_CORE_SPEAKER_VOLUME;
+ _volumeMutex.Enter();
+ hr = _ptrRenderSimpleVolume->SetMasterVolume(fLevel, NULL);
+ _volumeMutex.Leave();
+ EXIT_ON_ERROR(hr);
+
+ return 0;
Exit:
- _TraceCOMError(hr);
- return -1;
+ _TraceCOMError(hr);
+ return -1;
}
// ----------------------------------------------------------------------------
// SpeakerVolume
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::SpeakerVolume(uint32_t& volume) const
-{
+int32_t AudioDeviceWindowsCore::SpeakerVolume(uint32_t& volume) const {
+ {
+ rtc::CritScope lock(&_critSect);
- {
- rtc::CritScope lock(&_critSect);
-
- if (!_speakerIsInitialized)
- {
- return -1;
- }
-
- if (_ptrDeviceOut == NULL)
- {
- return -1;
- }
+ if (!_speakerIsInitialized) {
+ return -1;
}
- HRESULT hr = S_OK;
- float fLevel(0.0f);
+ if (_ptrDeviceOut == NULL) {
+ return -1;
+ }
+ }
- _volumeMutex.Enter();
- hr = _ptrRenderSimpleVolume->GetMasterVolume(&fLevel);
- _volumeMutex.Leave();
- EXIT_ON_ERROR(hr);
+ HRESULT hr = S_OK;
+ float fLevel(0.0f);
- // scale input volume range [0.0,1.0] to valid output range
- volume = static_cast<uint32_t> (fLevel*MAX_CORE_SPEAKER_VOLUME);
+ _volumeMutex.Enter();
+ hr = _ptrRenderSimpleVolume->GetMasterVolume(&fLevel);
+ _volumeMutex.Leave();
+ EXIT_ON_ERROR(hr);
- return 0;
+ // scale input volume range [0.0,1.0] to valid output range
+ volume = static_cast<uint32_t>(fLevel * MAX_CORE_SPEAKER_VOLUME);
+
+ return 0;
Exit:
- _TraceCOMError(hr);
- return -1;
+ _TraceCOMError(hr);
+ return -1;
}
// ----------------------------------------------------------------------------
@@ -1040,503 +948,451 @@
// how it is used today in VoE.
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::MaxSpeakerVolume(uint32_t& maxVolume) const
-{
+int32_t AudioDeviceWindowsCore::MaxSpeakerVolume(uint32_t& maxVolume) const {
+ if (!_speakerIsInitialized) {
+ return -1;
+ }
- if (!_speakerIsInitialized)
- {
- return -1;
- }
+ maxVolume = static_cast<uint32_t>(MAX_CORE_SPEAKER_VOLUME);
- maxVolume = static_cast<uint32_t> (MAX_CORE_SPEAKER_VOLUME);
-
- return 0;
+ return 0;
}
// ----------------------------------------------------------------------------
// MinSpeakerVolume
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::MinSpeakerVolume(uint32_t& minVolume) const
-{
+int32_t AudioDeviceWindowsCore::MinSpeakerVolume(uint32_t& minVolume) const {
+ if (!_speakerIsInitialized) {
+ return -1;
+ }
- if (!_speakerIsInitialized)
- {
- return -1;
- }
+ minVolume = static_cast<uint32_t>(MIN_CORE_SPEAKER_VOLUME);
- minVolume = static_cast<uint32_t> (MIN_CORE_SPEAKER_VOLUME);
-
- return 0;
+ return 0;
}
// ----------------------------------------------------------------------------
// SpeakerMuteIsAvailable
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::SpeakerMuteIsAvailable(bool& available)
-{
+int32_t AudioDeviceWindowsCore::SpeakerMuteIsAvailable(bool& available) {
+ rtc::CritScope lock(&_critSect);
- rtc::CritScope lock(&_critSect);
+ if (_ptrDeviceOut == NULL) {
+ return -1;
+ }
- if (_ptrDeviceOut == NULL)
- {
- return -1;
- }
+ HRESULT hr = S_OK;
+ IAudioEndpointVolume* pVolume = NULL;
- HRESULT hr = S_OK;
- IAudioEndpointVolume* pVolume = NULL;
+ // Query the speaker system mute state.
+ hr = _ptrDeviceOut->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL, NULL,
+ reinterpret_cast<void**>(&pVolume));
+ EXIT_ON_ERROR(hr);
- // Query the speaker system mute state.
- hr = _ptrDeviceOut->Activate(__uuidof(IAudioEndpointVolume),
- CLSCTX_ALL, NULL, reinterpret_cast<void**>(&pVolume));
- EXIT_ON_ERROR(hr);
+ BOOL mute;
+ hr = pVolume->GetMute(&mute);
+ if (FAILED(hr))
+ available = false;
+ else
+ available = true;
- BOOL mute;
- hr = pVolume->GetMute(&mute);
- if (FAILED(hr))
- available = false;
- else
- available = true;
+ SAFE_RELEASE(pVolume);
- SAFE_RELEASE(pVolume);
-
- return 0;
+ return 0;
Exit:
- _TraceCOMError(hr);
- SAFE_RELEASE(pVolume);
- return -1;
+ _TraceCOMError(hr);
+ SAFE_RELEASE(pVolume);
+ return -1;
}
// ----------------------------------------------------------------------------
// SetSpeakerMute
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::SetSpeakerMute(bool enable)
-{
+int32_t AudioDeviceWindowsCore::SetSpeakerMute(bool enable) {
+ rtc::CritScope lock(&_critSect);
- rtc::CritScope lock(&_critSect);
+ if (!_speakerIsInitialized) {
+ return -1;
+ }
- if (!_speakerIsInitialized)
- {
- return -1;
- }
+ if (_ptrDeviceOut == NULL) {
+ return -1;
+ }
- if (_ptrDeviceOut == NULL)
- {
- return -1;
- }
+ HRESULT hr = S_OK;
+ IAudioEndpointVolume* pVolume = NULL;
- HRESULT hr = S_OK;
- IAudioEndpointVolume* pVolume = NULL;
+ // Set the speaker system mute state.
+ hr = _ptrDeviceOut->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL, NULL,
+ reinterpret_cast<void**>(&pVolume));
+ EXIT_ON_ERROR(hr);
- // Set the speaker system mute state.
- hr = _ptrDeviceOut->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL, NULL, reinterpret_cast<void**>(&pVolume));
- EXIT_ON_ERROR(hr);
+ const BOOL mute(enable);
+ hr = pVolume->SetMute(mute, NULL);
+ EXIT_ON_ERROR(hr);
- const BOOL mute(enable);
- hr = pVolume->SetMute(mute, NULL);
- EXIT_ON_ERROR(hr);
+ SAFE_RELEASE(pVolume);
- SAFE_RELEASE(pVolume);
-
- return 0;
+ return 0;
Exit:
- _TraceCOMError(hr);
- SAFE_RELEASE(pVolume);
- return -1;
+ _TraceCOMError(hr);
+ SAFE_RELEASE(pVolume);
+ return -1;
}
// ----------------------------------------------------------------------------
// SpeakerMute
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::SpeakerMute(bool& enabled) const
-{
+int32_t AudioDeviceWindowsCore::SpeakerMute(bool& enabled) const {
+ if (!_speakerIsInitialized) {
+ return -1;
+ }
- if (!_speakerIsInitialized)
- {
- return -1;
- }
+ if (_ptrDeviceOut == NULL) {
+ return -1;
+ }
- if (_ptrDeviceOut == NULL)
- {
- return -1;
- }
+ HRESULT hr = S_OK;
+ IAudioEndpointVolume* pVolume = NULL;
- HRESULT hr = S_OK;
- IAudioEndpointVolume* pVolume = NULL;
+ // Query the speaker system mute state.
+ hr = _ptrDeviceOut->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL, NULL,
+ reinterpret_cast<void**>(&pVolume));
+ EXIT_ON_ERROR(hr);
- // Query the speaker system mute state.
- hr = _ptrDeviceOut->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL, NULL, reinterpret_cast<void**>(&pVolume));
- EXIT_ON_ERROR(hr);
+ BOOL mute;
+ hr = pVolume->GetMute(&mute);
+ EXIT_ON_ERROR(hr);
- BOOL mute;
- hr = pVolume->GetMute(&mute);
- EXIT_ON_ERROR(hr);
+ enabled = (mute == TRUE) ? true : false;
- enabled = (mute == TRUE) ? true : false;
+ SAFE_RELEASE(pVolume);
- SAFE_RELEASE(pVolume);
-
- return 0;
+ return 0;
Exit:
- _TraceCOMError(hr);
- SAFE_RELEASE(pVolume);
- return -1;
+ _TraceCOMError(hr);
+ SAFE_RELEASE(pVolume);
+ return -1;
}
// ----------------------------------------------------------------------------
// MicrophoneMuteIsAvailable
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::MicrophoneMuteIsAvailable(bool& available)
-{
+int32_t AudioDeviceWindowsCore::MicrophoneMuteIsAvailable(bool& available) {
+ rtc::CritScope lock(&_critSect);
- rtc::CritScope lock(&_critSect);
+ if (_ptrDeviceIn == NULL) {
+ return -1;
+ }
- if (_ptrDeviceIn == NULL)
- {
- return -1;
- }
+ HRESULT hr = S_OK;
+ IAudioEndpointVolume* pVolume = NULL;
- HRESULT hr = S_OK;
- IAudioEndpointVolume* pVolume = NULL;
+ // Query the microphone system mute state.
+ hr = _ptrDeviceIn->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL, NULL,
+ reinterpret_cast<void**>(&pVolume));
+ EXIT_ON_ERROR(hr);
- // Query the microphone system mute state.
- hr = _ptrDeviceIn->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL, NULL, reinterpret_cast<void**>(&pVolume));
- EXIT_ON_ERROR(hr);
+ BOOL mute;
+ hr = pVolume->GetMute(&mute);
+ if (FAILED(hr))
+ available = false;
+ else
+ available = true;
- BOOL mute;
- hr = pVolume->GetMute(&mute);
- if (FAILED(hr))
- available = false;
- else
- available = true;
-
- SAFE_RELEASE(pVolume);
- return 0;
+ SAFE_RELEASE(pVolume);
+ return 0;
Exit:
- _TraceCOMError(hr);
- SAFE_RELEASE(pVolume);
- return -1;
+ _TraceCOMError(hr);
+ SAFE_RELEASE(pVolume);
+ return -1;
}
// ----------------------------------------------------------------------------
// SetMicrophoneMute
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::SetMicrophoneMute(bool enable)
-{
+int32_t AudioDeviceWindowsCore::SetMicrophoneMute(bool enable) {
+ if (!_microphoneIsInitialized) {
+ return -1;
+ }
- if (!_microphoneIsInitialized)
- {
- return -1;
- }
+ if (_ptrDeviceIn == NULL) {
+ return -1;
+ }
- if (_ptrDeviceIn == NULL)
- {
- return -1;
- }
+ HRESULT hr = S_OK;
+ IAudioEndpointVolume* pVolume = NULL;
- HRESULT hr = S_OK;
- IAudioEndpointVolume* pVolume = NULL;
+ // Set the microphone system mute state.
+ hr = _ptrDeviceIn->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL, NULL,
+ reinterpret_cast<void**>(&pVolume));
+ EXIT_ON_ERROR(hr);
- // Set the microphone system mute state.
- hr = _ptrDeviceIn->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL, NULL, reinterpret_cast<void**>(&pVolume));
- EXIT_ON_ERROR(hr);
+ const BOOL mute(enable);
+ hr = pVolume->SetMute(mute, NULL);
+ EXIT_ON_ERROR(hr);
- const BOOL mute(enable);
- hr = pVolume->SetMute(mute, NULL);
- EXIT_ON_ERROR(hr);
-
- SAFE_RELEASE(pVolume);
- return 0;
+ SAFE_RELEASE(pVolume);
+ return 0;
Exit:
- _TraceCOMError(hr);
- SAFE_RELEASE(pVolume);
- return -1;
+ _TraceCOMError(hr);
+ SAFE_RELEASE(pVolume);
+ return -1;
}
// ----------------------------------------------------------------------------
// MicrophoneMute
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::MicrophoneMute(bool& enabled) const
-{
+int32_t AudioDeviceWindowsCore::MicrophoneMute(bool& enabled) const {
+ if (!_microphoneIsInitialized) {
+ return -1;
+ }
- if (!_microphoneIsInitialized)
- {
- return -1;
- }
+ HRESULT hr = S_OK;
+ IAudioEndpointVolume* pVolume = NULL;
- HRESULT hr = S_OK;
- IAudioEndpointVolume* pVolume = NULL;
+ // Query the microphone system mute state.
+ hr = _ptrDeviceIn->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL, NULL,
+ reinterpret_cast<void**>(&pVolume));
+ EXIT_ON_ERROR(hr);
- // Query the microphone system mute state.
- hr = _ptrDeviceIn->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL, NULL, reinterpret_cast<void**>(&pVolume));
- EXIT_ON_ERROR(hr);
+ BOOL mute;
+ hr = pVolume->GetMute(&mute);
+ EXIT_ON_ERROR(hr);
- BOOL mute;
- hr = pVolume->GetMute(&mute);
- EXIT_ON_ERROR(hr);
+ enabled = (mute == TRUE) ? true : false;
- enabled = (mute == TRUE) ? true : false;
-
- SAFE_RELEASE(pVolume);
- return 0;
+ SAFE_RELEASE(pVolume);
+ return 0;
Exit:
- _TraceCOMError(hr);
- SAFE_RELEASE(pVolume);
- return -1;
+ _TraceCOMError(hr);
+ SAFE_RELEASE(pVolume);
+ return -1;
}
// ----------------------------------------------------------------------------
// StereoRecordingIsAvailable
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::StereoRecordingIsAvailable(bool& available)
-{
-
- available = true;
- return 0;
+int32_t AudioDeviceWindowsCore::StereoRecordingIsAvailable(bool& available) {
+ available = true;
+ return 0;
}
// ----------------------------------------------------------------------------
// SetStereoRecording
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::SetStereoRecording(bool enable)
-{
+int32_t AudioDeviceWindowsCore::SetStereoRecording(bool enable) {
+ rtc::CritScope lock(&_critSect);
- rtc::CritScope lock(&_critSect);
+ if (enable) {
+ _recChannelsPrioList[0] = 2; // try stereo first
+ _recChannelsPrioList[1] = 1;
+ _recChannels = 2;
+ } else {
+ _recChannelsPrioList[0] = 1; // try mono first
+ _recChannelsPrioList[1] = 2;
+ _recChannels = 1;
+ }
- if (enable)
- {
- _recChannelsPrioList[0] = 2; // try stereo first
- _recChannelsPrioList[1] = 1;
- _recChannels = 2;
- }
- else
- {
- _recChannelsPrioList[0] = 1; // try mono first
- _recChannelsPrioList[1] = 2;
- _recChannels = 1;
- }
-
- return 0;
+ return 0;
}
// ----------------------------------------------------------------------------
// StereoRecording
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::StereoRecording(bool& enabled) const
-{
+int32_t AudioDeviceWindowsCore::StereoRecording(bool& enabled) const {
+ if (_recChannels == 2)
+ enabled = true;
+ else
+ enabled = false;
- if (_recChannels == 2)
- enabled = true;
- else
- enabled = false;
-
- return 0;
+ return 0;
}
// ----------------------------------------------------------------------------
// StereoPlayoutIsAvailable
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::StereoPlayoutIsAvailable(bool& available)
-{
-
- available = true;
- return 0;
+int32_t AudioDeviceWindowsCore::StereoPlayoutIsAvailable(bool& available) {
+ available = true;
+ return 0;
}
// ----------------------------------------------------------------------------
// SetStereoPlayout
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::SetStereoPlayout(bool enable)
-{
+int32_t AudioDeviceWindowsCore::SetStereoPlayout(bool enable) {
+ rtc::CritScope lock(&_critSect);
- rtc::CritScope lock(&_critSect);
+ if (enable) {
+ _playChannelsPrioList[0] = 2; // try stereo first
+ _playChannelsPrioList[1] = 1;
+ _playChannels = 2;
+ } else {
+ _playChannelsPrioList[0] = 1; // try mono first
+ _playChannelsPrioList[1] = 2;
+ _playChannels = 1;
+ }
- if (enable)
- {
- _playChannelsPrioList[0] = 2; // try stereo first
- _playChannelsPrioList[1] = 1;
- _playChannels = 2;
- }
- else
- {
- _playChannelsPrioList[0] = 1; // try mono first
- _playChannelsPrioList[1] = 2;
- _playChannels = 1;
- }
-
- return 0;
+ return 0;
}
// ----------------------------------------------------------------------------
// StereoPlayout
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::StereoPlayout(bool& enabled) const
-{
+int32_t AudioDeviceWindowsCore::StereoPlayout(bool& enabled) const {
+ if (_playChannels == 2)
+ enabled = true;
+ else
+ enabled = false;
- if (_playChannels == 2)
- enabled = true;
- else
- enabled = false;
-
- return 0;
+ return 0;
}
// ----------------------------------------------------------------------------
// SetAGC
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::SetAGC(bool enable)
-{
- rtc::CritScope lock(&_critSect);
- _AGC = enable;
- return 0;
+int32_t AudioDeviceWindowsCore::SetAGC(bool enable) {
+ rtc::CritScope lock(&_critSect);
+ _AGC = enable;
+ return 0;
}
// ----------------------------------------------------------------------------
// AGC
// ----------------------------------------------------------------------------
-bool AudioDeviceWindowsCore::AGC() const
-{
- rtc::CritScope lock(&_critSect);
- return _AGC;
+bool AudioDeviceWindowsCore::AGC() const {
+ rtc::CritScope lock(&_critSect);
+ return _AGC;
}
// ----------------------------------------------------------------------------
// MicrophoneVolumeIsAvailable
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::MicrophoneVolumeIsAvailable(bool& available)
-{
+int32_t AudioDeviceWindowsCore::MicrophoneVolumeIsAvailable(bool& available) {
+ rtc::CritScope lock(&_critSect);
- rtc::CritScope lock(&_critSect);
+ if (_ptrDeviceIn == NULL) {
+ return -1;
+ }
- if (_ptrDeviceIn == NULL)
- {
- return -1;
- }
+ HRESULT hr = S_OK;
+ IAudioEndpointVolume* pVolume = NULL;
- HRESULT hr = S_OK;
- IAudioEndpointVolume* pVolume = NULL;
+ hr = _ptrDeviceIn->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL, NULL,
+ reinterpret_cast<void**>(&pVolume));
+ EXIT_ON_ERROR(hr);
- hr = _ptrDeviceIn->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL, NULL, reinterpret_cast<void**>(&pVolume));
- EXIT_ON_ERROR(hr);
+ float volume(0.0f);
+ hr = pVolume->GetMasterVolumeLevelScalar(&volume);
+ if (FAILED(hr)) {
+ available = false;
+ }
+ available = true;
- float volume(0.0f);
- hr = pVolume->GetMasterVolumeLevelScalar(&volume);
- if (FAILED(hr))
- {
- available = false;
- }
- available = true;
-
- SAFE_RELEASE(pVolume);
- return 0;
+ SAFE_RELEASE(pVolume);
+ return 0;
Exit:
- _TraceCOMError(hr);
- SAFE_RELEASE(pVolume);
- return -1;
+ _TraceCOMError(hr);
+ SAFE_RELEASE(pVolume);
+ return -1;
}
// ----------------------------------------------------------------------------
// SetMicrophoneVolume
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::SetMicrophoneVolume(uint32_t volume)
-{
- LOG(LS_VERBOSE) << "AudioDeviceWindowsCore::SetMicrophoneVolume(volume="
- << volume << ")";
+int32_t AudioDeviceWindowsCore::SetMicrophoneVolume(uint32_t volume) {
+ LOG(LS_VERBOSE) << "AudioDeviceWindowsCore::SetMicrophoneVolume(volume="
+ << volume << ")";
- {
- rtc::CritScope lock(&_critSect);
+ {
+ rtc::CritScope lock(&_critSect);
- if (!_microphoneIsInitialized)
- {
- return -1;
- }
-
- if (_ptrDeviceIn == NULL)
- {
- return -1;
- }
+ if (!_microphoneIsInitialized) {
+ return -1;
}
- if (volume < static_cast<uint32_t>(MIN_CORE_MICROPHONE_VOLUME) ||
- volume > static_cast<uint32_t>(MAX_CORE_MICROPHONE_VOLUME))
- {
- return -1;
+ if (_ptrDeviceIn == NULL) {
+ return -1;
}
+ }
- HRESULT hr = S_OK;
- // scale input volume to valid range (0.0 to 1.0)
- const float fLevel = static_cast<float>(volume)/MAX_CORE_MICROPHONE_VOLUME;
- _volumeMutex.Enter();
- _ptrCaptureVolume->SetMasterVolumeLevelScalar(fLevel, NULL);
- _volumeMutex.Leave();
- EXIT_ON_ERROR(hr);
+ if (volume < static_cast<uint32_t>(MIN_CORE_MICROPHONE_VOLUME) ||
+ volume > static_cast<uint32_t>(MAX_CORE_MICROPHONE_VOLUME)) {
+ return -1;
+ }
- return 0;
+ HRESULT hr = S_OK;
+ // scale input volume to valid range (0.0 to 1.0)
+ const float fLevel = static_cast<float>(volume) / MAX_CORE_MICROPHONE_VOLUME;
+ _volumeMutex.Enter();
+ _ptrCaptureVolume->SetMasterVolumeLevelScalar(fLevel, NULL);
+ _volumeMutex.Leave();
+ EXIT_ON_ERROR(hr);
+
+ return 0;
Exit:
- _TraceCOMError(hr);
- return -1;
+ _TraceCOMError(hr);
+ return -1;
}
// ----------------------------------------------------------------------------
// MicrophoneVolume
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::MicrophoneVolume(uint32_t& volume) const
-{
- {
- rtc::CritScope lock(&_critSect);
+int32_t AudioDeviceWindowsCore::MicrophoneVolume(uint32_t& volume) const {
+ {
+ rtc::CritScope lock(&_critSect);
- if (!_microphoneIsInitialized)
- {
- return -1;
- }
-
- if (_ptrDeviceIn == NULL)
- {
- return -1;
- }
+ if (!_microphoneIsInitialized) {
+ return -1;
}
- HRESULT hr = S_OK;
- float fLevel(0.0f);
- volume = 0;
- _volumeMutex.Enter();
- hr = _ptrCaptureVolume->GetMasterVolumeLevelScalar(&fLevel);
- _volumeMutex.Leave();
- EXIT_ON_ERROR(hr);
+ if (_ptrDeviceIn == NULL) {
+ return -1;
+ }
+ }
- // scale input volume range [0.0,1.0] to valid output range
- volume = static_cast<uint32_t> (fLevel*MAX_CORE_MICROPHONE_VOLUME);
+ HRESULT hr = S_OK;
+ float fLevel(0.0f);
+ volume = 0;
+ _volumeMutex.Enter();
+ hr = _ptrCaptureVolume->GetMasterVolumeLevelScalar(&fLevel);
+ _volumeMutex.Leave();
+ EXIT_ON_ERROR(hr);
- return 0;
+ // scale input volume range [0.0,1.0] to valid output range
+ volume = static_cast<uint32_t>(fLevel * MAX_CORE_MICROPHONE_VOLUME);
+
+ return 0;
Exit:
- _TraceCOMError(hr);
- return -1;
+ _TraceCOMError(hr);
+ return -1;
}
// ----------------------------------------------------------------------------
@@ -1548,166 +1404,142 @@
// how it is used today in VoE.
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::MaxMicrophoneVolume(uint32_t& maxVolume) const
-{
- LOG(LS_VERBOSE) << __FUNCTION__;
+int32_t AudioDeviceWindowsCore::MaxMicrophoneVolume(uint32_t& maxVolume) const {
+ LOG(LS_VERBOSE) << __FUNCTION__;
- if (!_microphoneIsInitialized)
- {
- return -1;
- }
+ if (!_microphoneIsInitialized) {
+ return -1;
+ }
- maxVolume = static_cast<uint32_t> (MAX_CORE_MICROPHONE_VOLUME);
+ maxVolume = static_cast<uint32_t>(MAX_CORE_MICROPHONE_VOLUME);
- return 0;
+ return 0;
}
// ----------------------------------------------------------------------------
// MinMicrophoneVolume
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::MinMicrophoneVolume(uint32_t& minVolume) const
-{
+int32_t AudioDeviceWindowsCore::MinMicrophoneVolume(uint32_t& minVolume) const {
+ if (!_microphoneIsInitialized) {
+ return -1;
+ }
- if (!_microphoneIsInitialized)
- {
- return -1;
- }
+ minVolume = static_cast<uint32_t>(MIN_CORE_MICROPHONE_VOLUME);
- minVolume = static_cast<uint32_t> (MIN_CORE_MICROPHONE_VOLUME);
-
- return 0;
+ return 0;
}
// ----------------------------------------------------------------------------
// PlayoutDevices
// ----------------------------------------------------------------------------
-int16_t AudioDeviceWindowsCore::PlayoutDevices()
-{
+int16_t AudioDeviceWindowsCore::PlayoutDevices() {
+ rtc::CritScope lock(&_critSect);
- rtc::CritScope lock(&_critSect);
+ if (_RefreshDeviceList(eRender) != -1) {
+ return (_DeviceListCount(eRender));
+ }
- if (_RefreshDeviceList(eRender) != -1)
- {
- return (_DeviceListCount(eRender));
- }
-
- return -1;
+ return -1;
}
// ----------------------------------------------------------------------------
// SetPlayoutDevice I (II)
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::SetPlayoutDevice(uint16_t index)
-{
+int32_t AudioDeviceWindowsCore::SetPlayoutDevice(uint16_t index) {
+ if (_playIsInitialized) {
+ return -1;
+ }
- if (_playIsInitialized)
- {
- return -1;
- }
+ // Get current number of available rendering endpoint devices and refresh the
+ // rendering collection.
+ UINT nDevices = PlayoutDevices();
- // Get current number of available rendering endpoint devices and refresh the rendering collection.
- UINT nDevices = PlayoutDevices();
+ if (index < 0 || index > (nDevices - 1)) {
+ LOG(LS_ERROR) << "device index is out of range [0," << (nDevices - 1)
+ << "]";
+ return -1;
+ }
- if (index < 0 || index > (nDevices-1))
- {
- LOG(LS_ERROR) << "device index is out of range [0," << (nDevices-1)
- << "]";
- return -1;
- }
+ rtc::CritScope lock(&_critSect);
- rtc::CritScope lock(&_critSect);
+ HRESULT hr(S_OK);
- HRESULT hr(S_OK);
+ assert(_ptrRenderCollection != NULL);
- assert(_ptrRenderCollection != NULL);
-
- // Select an endpoint rendering device given the specified index
+ // Select an endpoint rendering device given the specified index
+ SAFE_RELEASE(_ptrDeviceOut);
+ hr = _ptrRenderCollection->Item(index, &_ptrDeviceOut);
+ if (FAILED(hr)) {
+ _TraceCOMError(hr);
SAFE_RELEASE(_ptrDeviceOut);
- hr = _ptrRenderCollection->Item(
- index,
- &_ptrDeviceOut);
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- SAFE_RELEASE(_ptrDeviceOut);
- return -1;
- }
+ return -1;
+ }
- WCHAR szDeviceName[MAX_PATH];
- const int bufferLen = sizeof(szDeviceName)/sizeof(szDeviceName)[0];
+ WCHAR szDeviceName[MAX_PATH];
+ const int bufferLen = sizeof(szDeviceName) / sizeof(szDeviceName)[0];
- // Get the endpoint device's friendly-name
- if (_GetDeviceName(_ptrDeviceOut, szDeviceName, bufferLen) == 0)
- {
- LOG(LS_VERBOSE) << "friendly name: \"" << szDeviceName << "\"";
- }
+ // Get the endpoint device's friendly-name
+ if (_GetDeviceName(_ptrDeviceOut, szDeviceName, bufferLen) == 0) {
+ LOG(LS_VERBOSE) << "friendly name: \"" << szDeviceName << "\"";
+ }
- _usingOutputDeviceIndex = true;
- _outputDeviceIndex = index;
+ _usingOutputDeviceIndex = true;
+ _outputDeviceIndex = index;
- return 0;
+ return 0;
}
// ----------------------------------------------------------------------------
// SetPlayoutDevice II (II)
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::SetPlayoutDevice(AudioDeviceModule::WindowsDeviceType device)
-{
- if (_playIsInitialized)
- {
- return -1;
- }
+int32_t AudioDeviceWindowsCore::SetPlayoutDevice(
+ AudioDeviceModule::WindowsDeviceType device) {
+ if (_playIsInitialized) {
+ return -1;
+ }
- ERole role(eCommunications);
+ ERole role(eCommunications);
- if (device == AudioDeviceModule::kDefaultDevice)
- {
- role = eConsole;
- }
- else if (device == AudioDeviceModule::kDefaultCommunicationDevice)
- {
- role = eCommunications;
- }
+ if (device == AudioDeviceModule::kDefaultDevice) {
+ role = eConsole;
+ } else if (device == AudioDeviceModule::kDefaultCommunicationDevice) {
+ role = eCommunications;
+ }
- rtc::CritScope lock(&_critSect);
+ rtc::CritScope lock(&_critSect);
- // Refresh the list of rendering endpoint devices
- _RefreshDeviceList(eRender);
+ // Refresh the list of rendering endpoint devices
+ _RefreshDeviceList(eRender);
- HRESULT hr(S_OK);
+ HRESULT hr(S_OK);
- assert(_ptrEnumerator != NULL);
+ assert(_ptrEnumerator != NULL);
- // Select an endpoint rendering device given the specified role
+ // Select an endpoint rendering device given the specified role
+ SAFE_RELEASE(_ptrDeviceOut);
+ hr = _ptrEnumerator->GetDefaultAudioEndpoint(eRender, role, &_ptrDeviceOut);
+ if (FAILED(hr)) {
+ _TraceCOMError(hr);
SAFE_RELEASE(_ptrDeviceOut);
- hr = _ptrEnumerator->GetDefaultAudioEndpoint(
- eRender,
- role,
- &_ptrDeviceOut);
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- SAFE_RELEASE(_ptrDeviceOut);
- return -1;
- }
+ return -1;
+ }
- WCHAR szDeviceName[MAX_PATH];
- const int bufferLen = sizeof(szDeviceName)/sizeof(szDeviceName)[0];
+ WCHAR szDeviceName[MAX_PATH];
+ const int bufferLen = sizeof(szDeviceName) / sizeof(szDeviceName)[0];
- // Get the endpoint device's friendly-name
- if (_GetDeviceName(_ptrDeviceOut, szDeviceName, bufferLen) == 0)
- {
- LOG(LS_VERBOSE) << "friendly name: \"" << szDeviceName << "\"";
- }
+ // Get the endpoint device's friendly-name
+ if (_GetDeviceName(_ptrDeviceOut, szDeviceName, bufferLen) == 0) {
+ LOG(LS_VERBOSE) << "friendly name: \"" << szDeviceName << "\"";
+ }
- _usingOutputDeviceIndex = false;
- _outputDevice = device;
+ _usingOutputDeviceIndex = false;
+ _outputDevice = device;
- return 0;
+ return 0;
}
// ----------------------------------------------------------------------------
@@ -1717,81 +1549,70 @@
int32_t AudioDeviceWindowsCore::PlayoutDeviceName(
uint16_t index,
char name[kAdmMaxDeviceNameSize],
- char guid[kAdmMaxGuidSize])
-{
+ char guid[kAdmMaxGuidSize]) {
+ bool defaultCommunicationDevice(false);
+ const int16_t nDevices(PlayoutDevices()); // also updates the list of devices
- bool defaultCommunicationDevice(false);
- const int16_t nDevices(PlayoutDevices()); // also updates the list of devices
+ // Special fix for the case when the user selects '-1' as index (<=> Default
+ // Communication Device)
+ if (index == (uint16_t)(-1)) {
+ defaultCommunicationDevice = true;
+ index = 0;
+ LOG(LS_VERBOSE) << "Default Communication endpoint device will be used";
+ }
- // Special fix for the case when the user selects '-1' as index (<=> Default Communication Device)
- if (index == (uint16_t)(-1))
- {
- defaultCommunicationDevice = true;
- index = 0;
- LOG(LS_VERBOSE) << "Default Communication endpoint device will be used";
+ if ((index > (nDevices - 1)) || (name == NULL)) {
+ return -1;
+ }
+
+ memset(name, 0, kAdmMaxDeviceNameSize);
+
+ if (guid != NULL) {
+ memset(guid, 0, kAdmMaxGuidSize);
+ }
+
+ rtc::CritScope lock(&_critSect);
+
+ int32_t ret(-1);
+ WCHAR szDeviceName[MAX_PATH];
+ const int bufferLen = sizeof(szDeviceName) / sizeof(szDeviceName)[0];
+
+ // Get the endpoint device's friendly-name
+ if (defaultCommunicationDevice) {
+ ret = _GetDefaultDeviceName(eRender, eCommunications, szDeviceName,
+ bufferLen);
+ } else {
+ ret = _GetListDeviceName(eRender, index, szDeviceName, bufferLen);
+ }
+
+ if (ret == 0) {
+ // Convert the endpoint device's friendly-name to UTF-8
+ if (WideCharToMultiByte(CP_UTF8, 0, szDeviceName, -1, name,
+ kAdmMaxDeviceNameSize, NULL, NULL) == 0) {
+ LOG(LS_ERROR) << "WideCharToMultiByte(CP_UTF8) failed with error code "
+ << GetLastError();
}
+ }
- if ((index > (nDevices-1)) || (name == NULL))
- {
- return -1;
+ // Get the endpoint ID string (uniquely identifies the device among all audio
+ // endpoint devices)
+ if (defaultCommunicationDevice) {
+ ret =
+ _GetDefaultDeviceID(eRender, eCommunications, szDeviceName, bufferLen);
+ } else {
+ ret = _GetListDeviceID(eRender, index, szDeviceName, bufferLen);
+ }
+
+ if (guid != NULL && ret == 0) {
+ // Convert the endpoint device's ID string to UTF-8
+ if (WideCharToMultiByte(CP_UTF8, 0, szDeviceName, -1, guid, kAdmMaxGuidSize,
+ NULL, NULL) == 0) {
+ LOG(LS_ERROR) << "WideCharToMultiByte(CP_UTF8) failed with error code "
+ << GetLastError();
}
+ }
- memset(name, 0, kAdmMaxDeviceNameSize);
-
- if (guid != NULL)
- {
- memset(guid, 0, kAdmMaxGuidSize);
- }
-
- rtc::CritScope lock(&_critSect);
-
- int32_t ret(-1);
- WCHAR szDeviceName[MAX_PATH];
- const int bufferLen = sizeof(szDeviceName)/sizeof(szDeviceName)[0];
-
- // Get the endpoint device's friendly-name
- if (defaultCommunicationDevice)
- {
- ret = _GetDefaultDeviceName(eRender, eCommunications, szDeviceName, bufferLen);
- }
- else
- {
- ret = _GetListDeviceName(eRender, index, szDeviceName, bufferLen);
- }
-
- if (ret == 0)
- {
- // Convert the endpoint device's friendly-name to UTF-8
- if (WideCharToMultiByte(CP_UTF8, 0, szDeviceName, -1, name, kAdmMaxDeviceNameSize, NULL, NULL) == 0)
- {
- LOG(LS_ERROR)
- << "WideCharToMultiByte(CP_UTF8) failed with error code "
- << GetLastError();
- }
- }
-
- // Get the endpoint ID string (uniquely identifies the device among all audio endpoint devices)
- if (defaultCommunicationDevice)
- {
- ret = _GetDefaultDeviceID(eRender, eCommunications, szDeviceName, bufferLen);
- }
- else
- {
- ret = _GetListDeviceID(eRender, index, szDeviceName, bufferLen);
- }
-
- if (guid != NULL && ret == 0)
- {
- // Convert the endpoint device's ID string to UTF-8
- if (WideCharToMultiByte(CP_UTF8, 0, szDeviceName, -1, guid, kAdmMaxGuidSize, NULL, NULL) == 0)
- {
- LOG(LS_ERROR)
- << "WideCharToMultiByte(CP_UTF8) failed with error code "
- << GetLastError();
- }
- }
-
- return ret;
+ return ret;
}
// ----------------------------------------------------------------------------
@@ -1801,1244 +1622,1081 @@
int32_t AudioDeviceWindowsCore::RecordingDeviceName(
uint16_t index,
char name[kAdmMaxDeviceNameSize],
- char guid[kAdmMaxGuidSize])
-{
+ char guid[kAdmMaxGuidSize]) {
+ bool defaultCommunicationDevice(false);
+ const int16_t nDevices(
+ RecordingDevices()); // also updates the list of devices
- bool defaultCommunicationDevice(false);
- const int16_t nDevices(RecordingDevices()); // also updates the list of devices
+ // Special fix for the case when the user selects '-1' as index (<=> Default
+ // Communication Device)
+ if (index == (uint16_t)(-1)) {
+ defaultCommunicationDevice = true;
+ index = 0;
+ LOG(LS_VERBOSE) << "Default Communication endpoint device will be used";
+ }
- // Special fix for the case when the user selects '-1' as index (<=> Default Communication Device)
- if (index == (uint16_t)(-1))
- {
- defaultCommunicationDevice = true;
- index = 0;
- LOG(LS_VERBOSE) << "Default Communication endpoint device will be used";
+ if ((index > (nDevices - 1)) || (name == NULL)) {
+ return -1;
+ }
+
+ memset(name, 0, kAdmMaxDeviceNameSize);
+
+ if (guid != NULL) {
+ memset(guid, 0, kAdmMaxGuidSize);
+ }
+
+ rtc::CritScope lock(&_critSect);
+
+ int32_t ret(-1);
+ WCHAR szDeviceName[MAX_PATH];
+ const int bufferLen = sizeof(szDeviceName) / sizeof(szDeviceName)[0];
+
+ // Get the endpoint device's friendly-name
+ if (defaultCommunicationDevice) {
+ ret = _GetDefaultDeviceName(eCapture, eCommunications, szDeviceName,
+ bufferLen);
+ } else {
+ ret = _GetListDeviceName(eCapture, index, szDeviceName, bufferLen);
+ }
+
+ if (ret == 0) {
+ // Convert the endpoint device's friendly-name to UTF-8
+ if (WideCharToMultiByte(CP_UTF8, 0, szDeviceName, -1, name,
+ kAdmMaxDeviceNameSize, NULL, NULL) == 0) {
+ LOG(LS_ERROR) << "WideCharToMultiByte(CP_UTF8) failed with error code "
+ << GetLastError();
}
+ }
- if ((index > (nDevices-1)) || (name == NULL))
- {
- return -1;
+ // Get the endpoint ID string (uniquely identifies the device among all audio
+ // endpoint devices)
+ if (defaultCommunicationDevice) {
+ ret =
+ _GetDefaultDeviceID(eCapture, eCommunications, szDeviceName, bufferLen);
+ } else {
+ ret = _GetListDeviceID(eCapture, index, szDeviceName, bufferLen);
+ }
+
+ if (guid != NULL && ret == 0) {
+ // Convert the endpoint device's ID string to UTF-8
+ if (WideCharToMultiByte(CP_UTF8, 0, szDeviceName, -1, guid, kAdmMaxGuidSize,
+ NULL, NULL) == 0) {
+ LOG(LS_ERROR) << "WideCharToMultiByte(CP_UTF8) failed with error code "
+ << GetLastError();
}
+ }
- memset(name, 0, kAdmMaxDeviceNameSize);
-
- if (guid != NULL)
- {
- memset(guid, 0, kAdmMaxGuidSize);
- }
-
- rtc::CritScope lock(&_critSect);
-
- int32_t ret(-1);
- WCHAR szDeviceName[MAX_PATH];
- const int bufferLen = sizeof(szDeviceName)/sizeof(szDeviceName)[0];
-
- // Get the endpoint device's friendly-name
- if (defaultCommunicationDevice)
- {
- ret = _GetDefaultDeviceName(eCapture, eCommunications, szDeviceName, bufferLen);
- }
- else
- {
- ret = _GetListDeviceName(eCapture, index, szDeviceName, bufferLen);
- }
-
- if (ret == 0)
- {
- // Convert the endpoint device's friendly-name to UTF-8
- if (WideCharToMultiByte(CP_UTF8, 0, szDeviceName, -1, name, kAdmMaxDeviceNameSize, NULL, NULL) == 0)
- {
- LOG(LS_ERROR)
- << "WideCharToMultiByte(CP_UTF8) failed with error code "
- << GetLastError();
- }
- }
-
- // Get the endpoint ID string (uniquely identifies the device among all audio endpoint devices)
- if (defaultCommunicationDevice)
- {
- ret = _GetDefaultDeviceID(eCapture, eCommunications, szDeviceName, bufferLen);
- }
- else
- {
- ret = _GetListDeviceID(eCapture, index, szDeviceName, bufferLen);
- }
-
- if (guid != NULL && ret == 0)
- {
- // Convert the endpoint device's ID string to UTF-8
- if (WideCharToMultiByte(CP_UTF8, 0, szDeviceName, -1, guid, kAdmMaxGuidSize, NULL, NULL) == 0)
- {
- LOG(LS_ERROR)
- << "WideCharToMultiByte(CP_UTF8) failed with error code "
- << GetLastError();
- }
- }
-
- return ret;
+ return ret;
}
// ----------------------------------------------------------------------------
// RecordingDevices
// ----------------------------------------------------------------------------
-int16_t AudioDeviceWindowsCore::RecordingDevices()
-{
+int16_t AudioDeviceWindowsCore::RecordingDevices() {
+ rtc::CritScope lock(&_critSect);
- rtc::CritScope lock(&_critSect);
+ if (_RefreshDeviceList(eCapture) != -1) {
+ return (_DeviceListCount(eCapture));
+ }
- if (_RefreshDeviceList(eCapture) != -1)
- {
- return (_DeviceListCount(eCapture));
- }
-
- return -1;
+ return -1;
}
// ----------------------------------------------------------------------------
// SetRecordingDevice I (II)
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::SetRecordingDevice(uint16_t index)
-{
+int32_t AudioDeviceWindowsCore::SetRecordingDevice(uint16_t index) {
+ if (_recIsInitialized) {
+ return -1;
+ }
- if (_recIsInitialized)
- {
- return -1;
- }
+ // Get current number of available capture endpoint devices and refresh the
+ // capture collection.
+ UINT nDevices = RecordingDevices();
- // Get current number of available capture endpoint devices and refresh the capture collection.
- UINT nDevices = RecordingDevices();
+ if (index < 0 || index > (nDevices - 1)) {
+ LOG(LS_ERROR) << "device index is out of range [0," << (nDevices - 1)
+ << "]";
+ return -1;
+ }
- if (index < 0 || index > (nDevices-1))
- {
- LOG(LS_ERROR) << "device index is out of range [0," << (nDevices-1)
- << "]";
- return -1;
- }
+ rtc::CritScope lock(&_critSect);
- rtc::CritScope lock(&_critSect);
+ HRESULT hr(S_OK);
- HRESULT hr(S_OK);
+ assert(_ptrCaptureCollection != NULL);
- assert(_ptrCaptureCollection != NULL);
-
- // Select an endpoint capture device given the specified index
+ // Select an endpoint capture device given the specified index
+ SAFE_RELEASE(_ptrDeviceIn);
+ hr = _ptrCaptureCollection->Item(index, &_ptrDeviceIn);
+ if (FAILED(hr)) {
+ _TraceCOMError(hr);
SAFE_RELEASE(_ptrDeviceIn);
- hr = _ptrCaptureCollection->Item(
- index,
- &_ptrDeviceIn);
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- SAFE_RELEASE(_ptrDeviceIn);
- return -1;
- }
+ return -1;
+ }
- WCHAR szDeviceName[MAX_PATH];
- const int bufferLen = sizeof(szDeviceName)/sizeof(szDeviceName)[0];
+ WCHAR szDeviceName[MAX_PATH];
+ const int bufferLen = sizeof(szDeviceName) / sizeof(szDeviceName)[0];
- // Get the endpoint device's friendly-name
- if (_GetDeviceName(_ptrDeviceIn, szDeviceName, bufferLen) == 0)
- {
- LOG(LS_VERBOSE) << "friendly name: \"" << szDeviceName << "\"";
- }
+ // Get the endpoint device's friendly-name
+ if (_GetDeviceName(_ptrDeviceIn, szDeviceName, bufferLen) == 0) {
+ LOG(LS_VERBOSE) << "friendly name: \"" << szDeviceName << "\"";
+ }
- _usingInputDeviceIndex = true;
- _inputDeviceIndex = index;
+ _usingInputDeviceIndex = true;
+ _inputDeviceIndex = index;
- return 0;
+ return 0;
}
// ----------------------------------------------------------------------------
// SetRecordingDevice II (II)
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::SetRecordingDevice(AudioDeviceModule::WindowsDeviceType device)
-{
- if (_recIsInitialized)
- {
- return -1;
- }
+int32_t AudioDeviceWindowsCore::SetRecordingDevice(
+ AudioDeviceModule::WindowsDeviceType device) {
+ if (_recIsInitialized) {
+ return -1;
+ }
- ERole role(eCommunications);
+ ERole role(eCommunications);
- if (device == AudioDeviceModule::kDefaultDevice)
- {
- role = eConsole;
- }
- else if (device == AudioDeviceModule::kDefaultCommunicationDevice)
- {
- role = eCommunications;
- }
+ if (device == AudioDeviceModule::kDefaultDevice) {
+ role = eConsole;
+ } else if (device == AudioDeviceModule::kDefaultCommunicationDevice) {
+ role = eCommunications;
+ }
- rtc::CritScope lock(&_critSect);
+ rtc::CritScope lock(&_critSect);
- // Refresh the list of capture endpoint devices
- _RefreshDeviceList(eCapture);
+ // Refresh the list of capture endpoint devices
+ _RefreshDeviceList(eCapture);
- HRESULT hr(S_OK);
+ HRESULT hr(S_OK);
- assert(_ptrEnumerator != NULL);
+ assert(_ptrEnumerator != NULL);
- // Select an endpoint capture device given the specified role
+ // Select an endpoint capture device given the specified role
+ SAFE_RELEASE(_ptrDeviceIn);
+ hr = _ptrEnumerator->GetDefaultAudioEndpoint(eCapture, role, &_ptrDeviceIn);
+ if (FAILED(hr)) {
+ _TraceCOMError(hr);
SAFE_RELEASE(_ptrDeviceIn);
- hr = _ptrEnumerator->GetDefaultAudioEndpoint(
- eCapture,
- role,
- &_ptrDeviceIn);
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- SAFE_RELEASE(_ptrDeviceIn);
- return -1;
- }
+ return -1;
+ }
- WCHAR szDeviceName[MAX_PATH];
- const int bufferLen = sizeof(szDeviceName)/sizeof(szDeviceName)[0];
+ WCHAR szDeviceName[MAX_PATH];
+ const int bufferLen = sizeof(szDeviceName) / sizeof(szDeviceName)[0];
- // Get the endpoint device's friendly-name
- if (_GetDeviceName(_ptrDeviceIn, szDeviceName, bufferLen) == 0)
- {
- LOG(LS_VERBOSE) << "friendly name: \"" << szDeviceName << "\"";
- }
+ // Get the endpoint device's friendly-name
+ if (_GetDeviceName(_ptrDeviceIn, szDeviceName, bufferLen) == 0) {
+ LOG(LS_VERBOSE) << "friendly name: \"" << szDeviceName << "\"";
+ }
- _usingInputDeviceIndex = false;
- _inputDevice = device;
+ _usingInputDeviceIndex = false;
+ _inputDevice = device;
- return 0;
+ return 0;
}
// ----------------------------------------------------------------------------
// PlayoutIsAvailable
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::PlayoutIsAvailable(bool& available)
-{
+int32_t AudioDeviceWindowsCore::PlayoutIsAvailable(bool& available) {
+ available = false;
- available = false;
+ // Try to initialize the playout side
+ int32_t res = InitPlayout();
- // Try to initialize the playout side
- int32_t res = InitPlayout();
+ // Cancel effect of initialization
+ StopPlayout();
- // Cancel effect of initialization
- StopPlayout();
+ if (res != -1) {
+ available = true;
+ }
- if (res != -1)
- {
- available = true;
- }
-
- return 0;
+ return 0;
}
// ----------------------------------------------------------------------------
// RecordingIsAvailable
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::RecordingIsAvailable(bool& available)
-{
+int32_t AudioDeviceWindowsCore::RecordingIsAvailable(bool& available) {
+ available = false;
- available = false;
+ // Try to initialize the recording side
+ int32_t res = InitRecording();
- // Try to initialize the recording side
- int32_t res = InitRecording();
+ // Cancel effect of initialization
+ StopRecording();
- // Cancel effect of initialization
- StopRecording();
+ if (res != -1) {
+ available = true;
+ }
- if (res != -1)
- {
- available = true;
- }
-
- return 0;
+ return 0;
}
// ----------------------------------------------------------------------------
// InitPlayout
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::InitPlayout()
-{
+int32_t AudioDeviceWindowsCore::InitPlayout() {
+ rtc::CritScope lock(&_critSect);
- rtc::CritScope lock(&_critSect);
+ if (_playing) {
+ return -1;
+ }
- if (_playing)
- {
- return -1;
- }
-
- if (_playIsInitialized)
- {
- return 0;
- }
-
- if (_ptrDeviceOut == NULL)
- {
- return -1;
- }
-
- // Initialize the speaker (devices might have been added or removed)
- if (InitSpeaker() == -1)
- {
- LOG(LS_WARNING) << "InitSpeaker() failed";
- }
-
- // Ensure that the updated rendering endpoint device is valid
- if (_ptrDeviceOut == NULL)
- {
- return -1;
- }
-
- if (_builtInAecEnabled && _recIsInitialized)
- {
- // Ensure the correct render device is configured in case
- // InitRecording() was called before InitPlayout().
- if (SetDMOProperties() == -1)
- {
- return -1;
- }
- }
-
- HRESULT hr = S_OK;
- WAVEFORMATEX* pWfxOut = NULL;
- WAVEFORMATEX Wfx = WAVEFORMATEX();
- WAVEFORMATEX* pWfxClosestMatch = NULL;
-
- // Create COM object with IAudioClient interface.
- SAFE_RELEASE(_ptrClientOut);
- hr = _ptrDeviceOut->Activate(
- __uuidof(IAudioClient),
- CLSCTX_ALL,
- NULL,
- (void**)&_ptrClientOut);
- EXIT_ON_ERROR(hr);
-
- // Retrieve the stream format that the audio engine uses for its internal
- // processing (mixing) of shared-mode streams.
- hr = _ptrClientOut->GetMixFormat(&pWfxOut);
- if (SUCCEEDED(hr))
- {
- LOG(LS_VERBOSE) << "Audio Engine's current rendering mix format:";
- // format type
- LOG(LS_VERBOSE) << "wFormatTag : 0x" << std::hex
- << pWfxOut->wFormatTag << std::dec << " ("
- << pWfxOut->wFormatTag << ")";
- // number of channels (i.e. mono, stereo...)
- LOG(LS_VERBOSE) << "nChannels : " << pWfxOut->nChannels;
- // sample rate
- LOG(LS_VERBOSE) << "nSamplesPerSec : " << pWfxOut->nSamplesPerSec;
- // for buffer estimation
- LOG(LS_VERBOSE) << "nAvgBytesPerSec: " << pWfxOut->nAvgBytesPerSec;
- // block size of data
- LOG(LS_VERBOSE) << "nBlockAlign : " << pWfxOut->nBlockAlign;
- // number of bits per sample of mono data
- LOG(LS_VERBOSE) << "wBitsPerSample : " << pWfxOut->wBitsPerSample;
- LOG(LS_VERBOSE) << "cbSize : " << pWfxOut->cbSize;
- }
-
- // Set wave format
- Wfx.wFormatTag = WAVE_FORMAT_PCM;
- Wfx.wBitsPerSample = 16;
- Wfx.cbSize = 0;
-
- const int freqs[] = {48000, 44100, 16000, 96000, 32000, 8000};
- hr = S_FALSE;
-
- // Iterate over frequencies and channels, in order of priority
- for (unsigned int freq = 0; freq < sizeof(freqs)/sizeof(freqs[0]); freq++)
- {
- for (unsigned int chan = 0; chan < sizeof(_playChannelsPrioList)/sizeof(_playChannelsPrioList[0]); chan++)
- {
- Wfx.nChannels = _playChannelsPrioList[chan];
- Wfx.nSamplesPerSec = freqs[freq];
- Wfx.nBlockAlign = Wfx.nChannels * Wfx.wBitsPerSample / 8;
- Wfx.nAvgBytesPerSec = Wfx.nSamplesPerSec * Wfx.nBlockAlign;
- // If the method succeeds and the audio endpoint device supports the specified stream format,
- // it returns S_OK. If the method succeeds and provides a closest match to the specified format,
- // it returns S_FALSE.
- hr = _ptrClientOut->IsFormatSupported(
- AUDCLNT_SHAREMODE_SHARED,
- &Wfx,
- &pWfxClosestMatch);
- if (hr == S_OK)
- {
- break;
- }
- else
- {
- if (pWfxClosestMatch)
- {
- LOG(INFO) << "nChannels=" << Wfx.nChannels <<
- ", nSamplesPerSec=" << Wfx.nSamplesPerSec <<
- " is not supported. Closest match: " <<
- "nChannels=" << pWfxClosestMatch->nChannels <<
- ", nSamplesPerSec=" << pWfxClosestMatch->nSamplesPerSec;
- CoTaskMemFree(pWfxClosestMatch);
- pWfxClosestMatch = NULL;
- }
- else
- {
- LOG(INFO) << "nChannels=" << Wfx.nChannels <<
- ", nSamplesPerSec=" << Wfx.nSamplesPerSec <<
- " is not supported. No closest match.";
- }
- }
- }
- if (hr == S_OK)
- break;
- }
-
- // TODO(andrew): what happens in the event of failure in the above loop?
- // Is _ptrClientOut->Initialize expected to fail?
- // Same in InitRecording().
- if (hr == S_OK)
- {
- _playAudioFrameSize = Wfx.nBlockAlign;
- // Block size in frames is the number of samples each channel in 10ms.
- _playBlockSizeInFrames = Wfx.nSamplesPerSec / 100;
- // Block size in samples is block size in frames times number of
- // channels.
- _playBlockSizeInSamples = _playBlockSizeInFrames * Wfx.nChannels;
- _playSampleRate = Wfx.nSamplesPerSec;
- _devicePlaySampleRate = Wfx.nSamplesPerSec; // The device itself continues to run at 44.1 kHz.
- _devicePlayBlockSize = Wfx.nSamplesPerSec/100;
- _playChannels = Wfx.nChannels;
-
- LOG(LS_VERBOSE) << "VoE selected this rendering format:";
- LOG(LS_VERBOSE) << "wFormatTag : 0x" << std::hex
- << Wfx.wFormatTag << std::dec << " (" << Wfx.wFormatTag
- << ")";
- LOG(LS_VERBOSE) << "nChannels : " << Wfx.nChannels;
- LOG(LS_VERBOSE) << "nSamplesPerSec : " << Wfx.nSamplesPerSec;
- LOG(LS_VERBOSE) << "nAvgBytesPerSec : " << Wfx.nAvgBytesPerSec;
- LOG(LS_VERBOSE) << "nBlockAlign : " << Wfx.nBlockAlign;
- LOG(LS_VERBOSE) << "wBitsPerSample : " << Wfx.wBitsPerSample;
- LOG(LS_VERBOSE) << "cbSize : " << Wfx.cbSize;
- LOG(LS_VERBOSE) << "Additional settings:";
- LOG(LS_VERBOSE) << "_playAudioFrameSize: " << _playAudioFrameSize;
- LOG(LS_VERBOSE) << "_playBlockSizeInFrames : "
- << _playBlockSizeInFrames;
- LOG(LS_VERBOSE) << "_playChannels : " << _playChannels;
- }
-
- // Create a rendering stream.
- //
- // ****************************************************************************
- // For a shared-mode stream that uses event-driven buffering, the caller must
- // set both hnsPeriodicity and hnsBufferDuration to 0. The Initialize method
- // determines how large a buffer to allocate based on the scheduling period
- // of the audio engine. Although the client's buffer processing thread is
- // event driven, the basic buffer management process, as described previously,
- // is unaltered.
- // Each time the thread awakens, it should call IAudioClient::GetCurrentPadding
- // to determine how much data to write to a rendering buffer or read from a capture
- // buffer. In contrast to the two buffers that the Initialize method allocates
- // for an exclusive-mode stream that uses event-driven buffering, a shared-mode
- // stream requires a single buffer.
- // ****************************************************************************
- //
- REFERENCE_TIME hnsBufferDuration = 0; // ask for minimum buffer size (default)
- if (_devicePlaySampleRate == 44100)
- {
- // Ask for a larger buffer size (30ms) when using 44.1kHz as render rate.
- // There seems to be a larger risk of underruns for 44.1 compared
- // with the default rate (48kHz). When using default, we set the requested
- // buffer duration to 0, which sets the buffer to the minimum size
- // required by the engine thread. The actual buffer size can then be
- // read by GetBufferSize() and it is 20ms on most machines.
- hnsBufferDuration = 30*10000;
- }
- hr = _ptrClientOut->Initialize(
- AUDCLNT_SHAREMODE_SHARED, // share Audio Engine with other applications
- AUDCLNT_STREAMFLAGS_EVENTCALLBACK, // processing of the audio buffer by the client will be event driven
- hnsBufferDuration, // requested buffer capacity as a time value (in 100-nanosecond units)
- 0, // periodicity
- &Wfx, // selected wave format
- NULL); // session GUID
-
- if (FAILED(hr))
- {
- LOG(LS_ERROR) << "IAudioClient::Initialize() failed:";
- }
- EXIT_ON_ERROR(hr);
-
- if (_ptrAudioBuffer)
- {
- // Update the audio buffer with the selected parameters
- _ptrAudioBuffer->SetPlayoutSampleRate(_playSampleRate);
- _ptrAudioBuffer->SetPlayoutChannels((uint8_t)_playChannels);
- }
- else
- {
- // We can enter this state during CoreAudioIsSupported() when no AudioDeviceImplementation
- // has been created, hence the AudioDeviceBuffer does not exist.
- // It is OK to end up here since we don't initiate any media in CoreAudioIsSupported().
- LOG(LS_VERBOSE)
- << "AudioDeviceBuffer must be attached before streaming can start";
- }
-
- // Get the actual size of the shared (endpoint buffer).
- // Typical value is 960 audio frames <=> 20ms @ 48kHz sample rate.
- UINT bufferFrameCount(0);
- hr = _ptrClientOut->GetBufferSize(
- &bufferFrameCount);
- if (SUCCEEDED(hr))
- {
- LOG(LS_VERBOSE) << "IAudioClient::GetBufferSize() => "
- << bufferFrameCount << " (<=> "
- << bufferFrameCount*_playAudioFrameSize << " bytes)";
- }
-
- // Set the event handle that the system signals when an audio buffer is ready
- // to be processed by the client.
- hr = _ptrClientOut->SetEventHandle(
- _hRenderSamplesReadyEvent);
- EXIT_ON_ERROR(hr);
-
- // Get an IAudioRenderClient interface.
- SAFE_RELEASE(_ptrRenderClient);
- hr = _ptrClientOut->GetService(
- __uuidof(IAudioRenderClient),
- (void**)&_ptrRenderClient);
- EXIT_ON_ERROR(hr);
-
- // Mark playout side as initialized
- _playIsInitialized = true;
-
- CoTaskMemFree(pWfxOut);
- CoTaskMemFree(pWfxClosestMatch);
-
- LOG(LS_VERBOSE) << "render side is now initialized";
+ if (_playIsInitialized) {
return 0;
+ }
+
+ if (_ptrDeviceOut == NULL) {
+ return -1;
+ }
+
+ // Initialize the speaker (devices might have been added or removed)
+ if (InitSpeaker() == -1) {
+ LOG(LS_WARNING) << "InitSpeaker() failed";
+ }
+
+ // Ensure that the updated rendering endpoint device is valid
+ if (_ptrDeviceOut == NULL) {
+ return -1;
+ }
+
+ if (_builtInAecEnabled && _recIsInitialized) {
+ // Ensure the correct render device is configured in case
+ // InitRecording() was called before InitPlayout().
+ if (SetDMOProperties() == -1) {
+ return -1;
+ }
+ }
+
+ HRESULT hr = S_OK;
+ WAVEFORMATEX* pWfxOut = NULL;
+ WAVEFORMATEX Wfx = WAVEFORMATEX();
+ WAVEFORMATEX* pWfxClosestMatch = NULL;
+
+ // Create COM object with IAudioClient interface.
+ SAFE_RELEASE(_ptrClientOut);
+ hr = _ptrDeviceOut->Activate(__uuidof(IAudioClient), CLSCTX_ALL, NULL,
+ (void**)&_ptrClientOut);
+ EXIT_ON_ERROR(hr);
+
+ // Retrieve the stream format that the audio engine uses for its internal
+ // processing (mixing) of shared-mode streams.
+ hr = _ptrClientOut->GetMixFormat(&pWfxOut);
+ if (SUCCEEDED(hr)) {
+ LOG(LS_VERBOSE) << "Audio Engine's current rendering mix format:";
+ // format type
+ LOG(LS_VERBOSE) << "wFormatTag : 0x" << std::hex << pWfxOut->wFormatTag
+ << std::dec << " (" << pWfxOut->wFormatTag << ")";
+ // number of channels (i.e. mono, stereo...)
+ LOG(LS_VERBOSE) << "nChannels : " << pWfxOut->nChannels;
+ // sample rate
+ LOG(LS_VERBOSE) << "nSamplesPerSec : " << pWfxOut->nSamplesPerSec;
+ // for buffer estimation
+ LOG(LS_VERBOSE) << "nAvgBytesPerSec: " << pWfxOut->nAvgBytesPerSec;
+ // block size of data
+ LOG(LS_VERBOSE) << "nBlockAlign : " << pWfxOut->nBlockAlign;
+ // number of bits per sample of mono data
+ LOG(LS_VERBOSE) << "wBitsPerSample : " << pWfxOut->wBitsPerSample;
+ LOG(LS_VERBOSE) << "cbSize : " << pWfxOut->cbSize;
+ }
+
+ // Set wave format
+ Wfx.wFormatTag = WAVE_FORMAT_PCM;
+ Wfx.wBitsPerSample = 16;
+ Wfx.cbSize = 0;
+
+ const int freqs[] = {48000, 44100, 16000, 96000, 32000, 8000};
+ hr = S_FALSE;
+
+ // Iterate over frequencies and channels, in order of priority
+ for (unsigned int freq = 0; freq < sizeof(freqs) / sizeof(freqs[0]); freq++) {
+ for (unsigned int chan = 0; chan < sizeof(_playChannelsPrioList) /
+ sizeof(_playChannelsPrioList[0]);
+ chan++) {
+ Wfx.nChannels = _playChannelsPrioList[chan];
+ Wfx.nSamplesPerSec = freqs[freq];
+ Wfx.nBlockAlign = Wfx.nChannels * Wfx.wBitsPerSample / 8;
+ Wfx.nAvgBytesPerSec = Wfx.nSamplesPerSec * Wfx.nBlockAlign;
+ // If the method succeeds and the audio endpoint device supports the
+ // specified stream format, it returns S_OK. If the method succeeds and
+ // provides a closest match to the specified format, it returns S_FALSE.
+ hr = _ptrClientOut->IsFormatSupported(AUDCLNT_SHAREMODE_SHARED, &Wfx,
+ &pWfxClosestMatch);
+ if (hr == S_OK) {
+ break;
+ } else {
+ if (pWfxClosestMatch) {
+ LOG(INFO) << "nChannels=" << Wfx.nChannels
+ << ", nSamplesPerSec=" << Wfx.nSamplesPerSec
+ << " is not supported. Closest match: "
+ << "nChannels=" << pWfxClosestMatch->nChannels
+ << ", nSamplesPerSec=" << pWfxClosestMatch->nSamplesPerSec;
+ CoTaskMemFree(pWfxClosestMatch);
+ pWfxClosestMatch = NULL;
+ } else {
+ LOG(INFO) << "nChannels=" << Wfx.nChannels
+ << ", nSamplesPerSec=" << Wfx.nSamplesPerSec
+ << " is not supported. No closest match.";
+ }
+ }
+ }
+ if (hr == S_OK)
+ break;
+ }
+
+ // TODO(andrew): what happens in the event of failure in the above loop?
+ // Is _ptrClientOut->Initialize expected to fail?
+ // Same in InitRecording().
+ if (hr == S_OK) {
+ _playAudioFrameSize = Wfx.nBlockAlign;
+ // Block size in frames is the number of samples each channel in 10ms.
+ _playBlockSizeInFrames = Wfx.nSamplesPerSec / 100;
+ // Block size in samples is block size in frames times number of
+ // channels.
+ _playBlockSizeInSamples = _playBlockSizeInFrames * Wfx.nChannels;
+ _playSampleRate = Wfx.nSamplesPerSec;
+ _devicePlaySampleRate =
+ Wfx.nSamplesPerSec; // The device itself continues to run at 44.1 kHz.
+ _devicePlayBlockSize = Wfx.nSamplesPerSec / 100;
+ _playChannels = Wfx.nChannels;
+
+ LOG(LS_VERBOSE) << "VoE selected this rendering format:";
+ LOG(LS_VERBOSE) << "wFormatTag : 0x" << std::hex << Wfx.wFormatTag
+ << std::dec << " (" << Wfx.wFormatTag << ")";
+ LOG(LS_VERBOSE) << "nChannels : " << Wfx.nChannels;
+ LOG(LS_VERBOSE) << "nSamplesPerSec : " << Wfx.nSamplesPerSec;
+ LOG(LS_VERBOSE) << "nAvgBytesPerSec : " << Wfx.nAvgBytesPerSec;
+ LOG(LS_VERBOSE) << "nBlockAlign : " << Wfx.nBlockAlign;
+ LOG(LS_VERBOSE) << "wBitsPerSample : " << Wfx.wBitsPerSample;
+ LOG(LS_VERBOSE) << "cbSize : " << Wfx.cbSize;
+ LOG(LS_VERBOSE) << "Additional settings:";
+ LOG(LS_VERBOSE) << "_playAudioFrameSize: " << _playAudioFrameSize;
+ LOG(LS_VERBOSE) << "_playBlockSizeInFrames : "
+ << _playBlockSizeInFrames;
+ LOG(LS_VERBOSE) << "_playChannels : " << _playChannels;
+ }
+
+ // Create a rendering stream.
+ //
+ // ****************************************************************************
+ // For a shared-mode stream that uses event-driven buffering, the caller must
+ // set both hnsPeriodicity and hnsBufferDuration to 0. The Initialize method
+ // determines how large a buffer to allocate based on the scheduling period
+ // of the audio engine. Although the client's buffer processing thread is
+ // event driven, the basic buffer management process, as described previously,
+ // is unaltered.
+ // Each time the thread awakens, it should call
+ // IAudioClient::GetCurrentPadding to determine how much data to write to a
+ // rendering buffer or read from a capture buffer. In contrast to the two
+ // buffers that the Initialize method allocates for an exclusive-mode stream
+ // that uses event-driven buffering, a shared-mode stream requires a single
+ // buffer.
+ // ****************************************************************************
+ //
+ REFERENCE_TIME hnsBufferDuration =
+ 0; // ask for minimum buffer size (default)
+ if (_devicePlaySampleRate == 44100) {
+ // Ask for a larger buffer size (30ms) when using 44.1kHz as render rate.
+ // There seems to be a larger risk of underruns for 44.1 compared
+ // with the default rate (48kHz). When using default, we set the requested
+ // buffer duration to 0, which sets the buffer to the minimum size
+ // required by the engine thread. The actual buffer size can then be
+ // read by GetBufferSize() and it is 20ms on most machines.
+ hnsBufferDuration = 30 * 10000;
+ }
+ hr = _ptrClientOut->Initialize(
+ AUDCLNT_SHAREMODE_SHARED, // share Audio Engine with other applications
+ AUDCLNT_STREAMFLAGS_EVENTCALLBACK, // processing of the audio buffer by
+ // the client will be event driven
+ hnsBufferDuration, // requested buffer capacity as a time value (in
+ // 100-nanosecond units)
+ 0, // periodicity
+ &Wfx, // selected wave format
+ NULL); // session GUID
+
+ if (FAILED(hr)) {
+ LOG(LS_ERROR) << "IAudioClient::Initialize() failed:";
+ }
+ EXIT_ON_ERROR(hr);
+
+ if (_ptrAudioBuffer) {
+ // Update the audio buffer with the selected parameters
+ _ptrAudioBuffer->SetPlayoutSampleRate(_playSampleRate);
+ _ptrAudioBuffer->SetPlayoutChannels((uint8_t)_playChannels);
+ } else {
+ // We can enter this state during CoreAudioIsSupported() when no
+ // AudioDeviceImplementation has been created, hence the AudioDeviceBuffer
+ // does not exist. It is OK to end up here since we don't initiate any media
+ // in CoreAudioIsSupported().
+ LOG(LS_VERBOSE)
+ << "AudioDeviceBuffer must be attached before streaming can start";
+ }
+
+ // Get the actual size of the shared (endpoint buffer).
+ // Typical value is 960 audio frames <=> 20ms @ 48kHz sample rate.
+ UINT bufferFrameCount(0);
+ hr = _ptrClientOut->GetBufferSize(&bufferFrameCount);
+ if (SUCCEEDED(hr)) {
+ LOG(LS_VERBOSE) << "IAudioClient::GetBufferSize() => " << bufferFrameCount
+ << " (<=> " << bufferFrameCount * _playAudioFrameSize
+ << " bytes)";
+ }
+
+ // Set the event handle that the system signals when an audio buffer is ready
+ // to be processed by the client.
+ hr = _ptrClientOut->SetEventHandle(_hRenderSamplesReadyEvent);
+ EXIT_ON_ERROR(hr);
+
+ // Get an IAudioRenderClient interface.
+ SAFE_RELEASE(_ptrRenderClient);
+ hr = _ptrClientOut->GetService(__uuidof(IAudioRenderClient),
+ (void**)&_ptrRenderClient);
+ EXIT_ON_ERROR(hr);
+
+ // Mark playout side as initialized
+ _playIsInitialized = true;
+
+ CoTaskMemFree(pWfxOut);
+ CoTaskMemFree(pWfxClosestMatch);
+
+ LOG(LS_VERBOSE) << "render side is now initialized";
+ return 0;
Exit:
- _TraceCOMError(hr);
- CoTaskMemFree(pWfxOut);
- CoTaskMemFree(pWfxClosestMatch);
- SAFE_RELEASE(_ptrClientOut);
- SAFE_RELEASE(_ptrRenderClient);
- return -1;
+ _TraceCOMError(hr);
+ CoTaskMemFree(pWfxOut);
+ CoTaskMemFree(pWfxClosestMatch);
+ SAFE_RELEASE(_ptrClientOut);
+ SAFE_RELEASE(_ptrRenderClient);
+ return -1;
}
// Capture initialization when the built-in AEC DirectX Media Object (DMO) is
// used. Called from InitRecording(), most of which is skipped over. The DMO
// handles device initialization itself.
// Reference: http://msdn.microsoft.com/en-us/library/ff819492(v=vs.85).aspx
-int32_t AudioDeviceWindowsCore::InitRecordingDMO()
-{
- assert(_builtInAecEnabled);
- assert(_dmo != NULL);
+int32_t AudioDeviceWindowsCore::InitRecordingDMO() {
+ assert(_builtInAecEnabled);
+ assert(_dmo != NULL);
- if (SetDMOProperties() == -1)
- {
- return -1;
- }
+ if (SetDMOProperties() == -1) {
+ return -1;
+ }
- DMO_MEDIA_TYPE mt = {0};
- HRESULT hr = MoInitMediaType(&mt, sizeof(WAVEFORMATEX));
- if (FAILED(hr))
- {
- MoFreeMediaType(&mt);
- _TraceCOMError(hr);
- return -1;
- }
- mt.majortype = MEDIATYPE_Audio;
- mt.subtype = MEDIASUBTYPE_PCM;
- mt.formattype = FORMAT_WaveFormatEx;
-
- // Supported formats
- // nChannels: 1 (in AEC-only mode)
- // nSamplesPerSec: 8000, 11025, 16000, 22050
- // wBitsPerSample: 16
- WAVEFORMATEX* ptrWav = reinterpret_cast<WAVEFORMATEX*>(mt.pbFormat);
- ptrWav->wFormatTag = WAVE_FORMAT_PCM;
- ptrWav->nChannels = 1;
- // 16000 is the highest we can support with our resampler.
- ptrWav->nSamplesPerSec = 16000;
- ptrWav->nAvgBytesPerSec = 32000;
- ptrWav->nBlockAlign = 2;
- ptrWav->wBitsPerSample = 16;
- ptrWav->cbSize = 0;
-
- // Set the VoE format equal to the AEC output format.
- _recAudioFrameSize = ptrWav->nBlockAlign;
- _recSampleRate = ptrWav->nSamplesPerSec;
- _recBlockSize = ptrWav->nSamplesPerSec / 100;
- _recChannels = ptrWav->nChannels;
-
- // Set the DMO output format parameters.
- hr = _dmo->SetOutputType(kAecCaptureStreamIndex, &mt, 0);
+ DMO_MEDIA_TYPE mt = {0};
+ HRESULT hr = MoInitMediaType(&mt, sizeof(WAVEFORMATEX));
+ if (FAILED(hr)) {
MoFreeMediaType(&mt);
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- return -1;
- }
+ _TraceCOMError(hr);
+ return -1;
+ }
+ mt.majortype = MEDIATYPE_Audio;
+ mt.subtype = MEDIASUBTYPE_PCM;
+ mt.formattype = FORMAT_WaveFormatEx;
- if (_ptrAudioBuffer)
- {
- _ptrAudioBuffer->SetRecordingSampleRate(_recSampleRate);
- _ptrAudioBuffer->SetRecordingChannels(_recChannels);
- }
- else
- {
- // Refer to InitRecording() for comments.
- LOG(LS_VERBOSE)
- << "AudioDeviceBuffer must be attached before streaming can start";
- }
+ // Supported formats
+ // nChannels: 1 (in AEC-only mode)
+ // nSamplesPerSec: 8000, 11025, 16000, 22050
+ // wBitsPerSample: 16
+ WAVEFORMATEX* ptrWav = reinterpret_cast<WAVEFORMATEX*>(mt.pbFormat);
+ ptrWav->wFormatTag = WAVE_FORMAT_PCM;
+ ptrWav->nChannels = 1;
+ // 16000 is the highest we can support with our resampler.
+ ptrWav->nSamplesPerSec = 16000;
+ ptrWav->nAvgBytesPerSec = 32000;
+ ptrWav->nBlockAlign = 2;
+ ptrWav->wBitsPerSample = 16;
+ ptrWav->cbSize = 0;
- _mediaBuffer = new MediaBufferImpl(_recBlockSize * _recAudioFrameSize);
+ // Set the VoE format equal to the AEC output format.
+ _recAudioFrameSize = ptrWav->nBlockAlign;
+ _recSampleRate = ptrWav->nSamplesPerSec;
+ _recBlockSize = ptrWav->nSamplesPerSec / 100;
+ _recChannels = ptrWav->nChannels;
- // Optional, but if called, must be after media types are set.
- hr = _dmo->AllocateStreamingResources();
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- return -1;
- }
+ // Set the DMO output format parameters.
+ hr = _dmo->SetOutputType(kAecCaptureStreamIndex, &mt, 0);
+ MoFreeMediaType(&mt);
+ if (FAILED(hr)) {
+ _TraceCOMError(hr);
+ return -1;
+ }
- _recIsInitialized = true;
- LOG(LS_VERBOSE) << "Capture side is now initialized";
+ if (_ptrAudioBuffer) {
+ _ptrAudioBuffer->SetRecordingSampleRate(_recSampleRate);
+ _ptrAudioBuffer->SetRecordingChannels(_recChannels);
+ } else {
+ // Refer to InitRecording() for comments.
+ LOG(LS_VERBOSE)
+ << "AudioDeviceBuffer must be attached before streaming can start";
+ }
- return 0;
+ _mediaBuffer = new MediaBufferImpl(_recBlockSize * _recAudioFrameSize);
+
+ // Optional, but if called, must be after media types are set.
+ hr = _dmo->AllocateStreamingResources();
+ if (FAILED(hr)) {
+ _TraceCOMError(hr);
+ return -1;
+ }
+
+ _recIsInitialized = true;
+ LOG(LS_VERBOSE) << "Capture side is now initialized";
+
+ return 0;
}
// ----------------------------------------------------------------------------
// InitRecording
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::InitRecording()
-{
+int32_t AudioDeviceWindowsCore::InitRecording() {
+ rtc::CritScope lock(&_critSect);
- rtc::CritScope lock(&_critSect);
+ if (_recording) {
+ return -1;
+ }
- if (_recording)
- {
- return -1;
- }
-
- if (_recIsInitialized)
- {
- return 0;
- }
-
- if (QueryPerformanceFrequency(&_perfCounterFreq) == 0)
- {
- return -1;
- }
- _perfCounterFactor = 10000000.0 / (double)_perfCounterFreq.QuadPart;
-
- if (_ptrDeviceIn == NULL)
- {
- return -1;
- }
-
- // Initialize the microphone (devices might have been added or removed)
- if (InitMicrophone() == -1)
- {
- LOG(LS_WARNING) << "InitMicrophone() failed";
- }
-
- // Ensure that the updated capturing endpoint device is valid
- if (_ptrDeviceIn == NULL)
- {
- return -1;
- }
-
- if (_builtInAecEnabled)
- {
- // The DMO will configure the capture device.
- return InitRecordingDMO();
- }
-
- HRESULT hr = S_OK;
- WAVEFORMATEX* pWfxIn = NULL;
- WAVEFORMATEXTENSIBLE Wfx = WAVEFORMATEXTENSIBLE();
- WAVEFORMATEX* pWfxClosestMatch = NULL;
-
- // Create COM object with IAudioClient interface.
- SAFE_RELEASE(_ptrClientIn);
- hr = _ptrDeviceIn->Activate(
- __uuidof(IAudioClient),
- CLSCTX_ALL,
- NULL,
- (void**)&_ptrClientIn);
- EXIT_ON_ERROR(hr);
-
- // Retrieve the stream format that the audio engine uses for its internal
- // processing (mixing) of shared-mode streams.
- hr = _ptrClientIn->GetMixFormat(&pWfxIn);
- if (SUCCEEDED(hr))
- {
- LOG(LS_VERBOSE) << "Audio Engine's current capturing mix format:";
- // format type
- LOG(LS_VERBOSE) << "wFormatTag : 0x" << std::hex
- << pWfxIn->wFormatTag << std::dec << " ("
- << pWfxIn->wFormatTag << ")";
- // number of channels (i.e. mono, stereo...)
- LOG(LS_VERBOSE) << "nChannels : " << pWfxIn->nChannels;
- // sample rate
- LOG(LS_VERBOSE) << "nSamplesPerSec : " << pWfxIn->nSamplesPerSec;
- // for buffer estimation
- LOG(LS_VERBOSE) << "nAvgBytesPerSec: " << pWfxIn->nAvgBytesPerSec;
- // block size of data
- LOG(LS_VERBOSE) << "nBlockAlign : " << pWfxIn->nBlockAlign;
- // number of bits per sample of mono data
- LOG(LS_VERBOSE) << "wBitsPerSample : " << pWfxIn->wBitsPerSample;
- LOG(LS_VERBOSE) << "cbSize : " << pWfxIn->cbSize;
- }
-
- // Set wave format
- Wfx.Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
- Wfx.Format.wBitsPerSample = 16;
- Wfx.Format.cbSize = 22;
- Wfx.dwChannelMask = 0;
- Wfx.Samples.wValidBitsPerSample = Wfx.Format.wBitsPerSample;
- Wfx.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
-
- const int freqs[6] = {48000, 44100, 16000, 96000, 32000, 8000};
- hr = S_FALSE;
-
- // Iterate over frequencies and channels, in order of priority
- for (unsigned int freq = 0; freq < sizeof(freqs)/sizeof(freqs[0]); freq++)
- {
- for (unsigned int chan = 0; chan < sizeof(_recChannelsPrioList)/sizeof(_recChannelsPrioList[0]); chan++)
- {
- Wfx.Format.nChannels = _recChannelsPrioList[chan];
- Wfx.Format.nSamplesPerSec = freqs[freq];
- Wfx.Format.nBlockAlign = Wfx.Format.nChannels *
- Wfx.Format.wBitsPerSample / 8;
- Wfx.Format.nAvgBytesPerSec = Wfx.Format.nSamplesPerSec *
- Wfx.Format.nBlockAlign;
- // If the method succeeds and the audio endpoint device supports the specified stream format,
- // it returns S_OK. If the method succeeds and provides a closest match to the specified format,
- // it returns S_FALSE.
- hr = _ptrClientIn->IsFormatSupported(
- AUDCLNT_SHAREMODE_SHARED,
- (WAVEFORMATEX*)&Wfx,
- &pWfxClosestMatch);
- if (hr == S_OK)
- {
- break;
- }
- else
- {
- if (pWfxClosestMatch)
- {
- LOG(INFO) << "nChannels=" << Wfx.Format.nChannels <<
- ", nSamplesPerSec=" << Wfx.Format.nSamplesPerSec <<
- " is not supported. Closest match: " <<
- "nChannels=" << pWfxClosestMatch->nChannels <<
- ", nSamplesPerSec=" << pWfxClosestMatch->nSamplesPerSec;
- CoTaskMemFree(pWfxClosestMatch);
- pWfxClosestMatch = NULL;
- }
- else
- {
- LOG(INFO) << "nChannels=" << Wfx.Format.nChannels <<
- ", nSamplesPerSec=" << Wfx.Format.nSamplesPerSec <<
- " is not supported. No closest match.";
- }
- }
- }
- if (hr == S_OK)
- break;
- }
-
- if (hr == S_OK)
- {
- _recAudioFrameSize = Wfx.Format.nBlockAlign;
- _recSampleRate = Wfx.Format.nSamplesPerSec;
- _recBlockSize = Wfx.Format.nSamplesPerSec/100;
- _recChannels = Wfx.Format.nChannels;
-
- LOG(LS_VERBOSE) << "VoE selected this capturing format:";
- LOG(LS_VERBOSE) << "wFormatTag : 0x" << std::hex
- << Wfx.Format.wFormatTag << std::dec
- << " (" << Wfx.Format.wFormatTag << ")";
- LOG(LS_VERBOSE) << "nChannels : " << Wfx.Format.nChannels;
- LOG(LS_VERBOSE) << "nSamplesPerSec : " << Wfx.Format.nSamplesPerSec;
- LOG(LS_VERBOSE) << "nAvgBytesPerSec : " << Wfx.Format.nAvgBytesPerSec;
- LOG(LS_VERBOSE) << "nBlockAlign : " << Wfx.Format.nBlockAlign;
- LOG(LS_VERBOSE) << "wBitsPerSample : " << Wfx.Format.wBitsPerSample;
- LOG(LS_VERBOSE) << "cbSize : " << Wfx.Format.cbSize;
- LOG(LS_VERBOSE) << "Additional settings:";
- LOG(LS_VERBOSE) << "_recAudioFrameSize: " << _recAudioFrameSize;
- LOG(LS_VERBOSE) << "_recBlockSize : " << _recBlockSize;
- LOG(LS_VERBOSE) << "_recChannels : " << _recChannels;
- }
-
- // Create a capturing stream.
- hr = _ptrClientIn->Initialize(
- AUDCLNT_SHAREMODE_SHARED, // share Audio Engine with other applications
- AUDCLNT_STREAMFLAGS_EVENTCALLBACK | // processing of the audio buffer by the client will be event driven
- AUDCLNT_STREAMFLAGS_NOPERSIST, // volume and mute settings for an audio session will not persist across system restarts
- 0, // required for event-driven shared mode
- 0, // periodicity
- (WAVEFORMATEX*)&Wfx, // selected wave format
- NULL); // session GUID
-
-
- if (hr != S_OK)
- {
- LOG(LS_ERROR) << "IAudioClient::Initialize() failed:";
- }
- EXIT_ON_ERROR(hr);
-
- if (_ptrAudioBuffer)
- {
- // Update the audio buffer with the selected parameters
- _ptrAudioBuffer->SetRecordingSampleRate(_recSampleRate);
- _ptrAudioBuffer->SetRecordingChannels((uint8_t)_recChannels);
- }
- else
- {
- // We can enter this state during CoreAudioIsSupported() when no AudioDeviceImplementation
- // has been created, hence the AudioDeviceBuffer does not exist.
- // It is OK to end up here since we don't initiate any media in CoreAudioIsSupported().
- LOG(LS_VERBOSE)
- << "AudioDeviceBuffer must be attached before streaming can start";
- }
-
- // Get the actual size of the shared (endpoint buffer).
- // Typical value is 960 audio frames <=> 20ms @ 48kHz sample rate.
- UINT bufferFrameCount(0);
- hr = _ptrClientIn->GetBufferSize(
- &bufferFrameCount);
- if (SUCCEEDED(hr))
- {
- LOG(LS_VERBOSE) << "IAudioClient::GetBufferSize() => "
- << bufferFrameCount << " (<=> "
- << bufferFrameCount*_recAudioFrameSize << " bytes)";
- }
-
- // Set the event handle that the system signals when an audio buffer is ready
- // to be processed by the client.
- hr = _ptrClientIn->SetEventHandle(
- _hCaptureSamplesReadyEvent);
- EXIT_ON_ERROR(hr);
-
- // Get an IAudioCaptureClient interface.
- SAFE_RELEASE(_ptrCaptureClient);
- hr = _ptrClientIn->GetService(
- __uuidof(IAudioCaptureClient),
- (void**)&_ptrCaptureClient);
- EXIT_ON_ERROR(hr);
-
- // Mark capture side as initialized
- _recIsInitialized = true;
-
- CoTaskMemFree(pWfxIn);
- CoTaskMemFree(pWfxClosestMatch);
-
- LOG(LS_VERBOSE) << "capture side is now initialized";
+ if (_recIsInitialized) {
return 0;
+ }
+
+ if (QueryPerformanceFrequency(&_perfCounterFreq) == 0) {
+ return -1;
+ }
+ _perfCounterFactor = 10000000.0 / (double)_perfCounterFreq.QuadPart;
+
+ if (_ptrDeviceIn == NULL) {
+ return -1;
+ }
+
+ // Initialize the microphone (devices might have been added or removed)
+ if (InitMicrophone() == -1) {
+ LOG(LS_WARNING) << "InitMicrophone() failed";
+ }
+
+ // Ensure that the updated capturing endpoint device is valid
+ if (_ptrDeviceIn == NULL) {
+ return -1;
+ }
+
+ if (_builtInAecEnabled) {
+ // The DMO will configure the capture device.
+ return InitRecordingDMO();
+ }
+
+ HRESULT hr = S_OK;
+ WAVEFORMATEX* pWfxIn = NULL;
+ WAVEFORMATEXTENSIBLE Wfx = WAVEFORMATEXTENSIBLE();
+ WAVEFORMATEX* pWfxClosestMatch = NULL;
+
+ // Create COM object with IAudioClient interface.
+ SAFE_RELEASE(_ptrClientIn);
+ hr = _ptrDeviceIn->Activate(__uuidof(IAudioClient), CLSCTX_ALL, NULL,
+ (void**)&_ptrClientIn);
+ EXIT_ON_ERROR(hr);
+
+ // Retrieve the stream format that the audio engine uses for its internal
+ // processing (mixing) of shared-mode streams.
+ hr = _ptrClientIn->GetMixFormat(&pWfxIn);
+ if (SUCCEEDED(hr)) {
+ LOG(LS_VERBOSE) << "Audio Engine's current capturing mix format:";
+ // format type
+ LOG(LS_VERBOSE) << "wFormatTag : 0x" << std::hex << pWfxIn->wFormatTag
+ << std::dec << " (" << pWfxIn->wFormatTag << ")";
+ // number of channels (i.e. mono, stereo...)
+ LOG(LS_VERBOSE) << "nChannels : " << pWfxIn->nChannels;
+ // sample rate
+ LOG(LS_VERBOSE) << "nSamplesPerSec : " << pWfxIn->nSamplesPerSec;
+ // for buffer estimation
+ LOG(LS_VERBOSE) << "nAvgBytesPerSec: " << pWfxIn->nAvgBytesPerSec;
+ // block size of data
+ LOG(LS_VERBOSE) << "nBlockAlign : " << pWfxIn->nBlockAlign;
+ // number of bits per sample of mono data
+ LOG(LS_VERBOSE) << "wBitsPerSample : " << pWfxIn->wBitsPerSample;
+ LOG(LS_VERBOSE) << "cbSize : " << pWfxIn->cbSize;
+ }
+
+ // Set wave format
+ Wfx.Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
+ Wfx.Format.wBitsPerSample = 16;
+ Wfx.Format.cbSize = 22;
+ Wfx.dwChannelMask = 0;
+ Wfx.Samples.wValidBitsPerSample = Wfx.Format.wBitsPerSample;
+ Wfx.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
+
+ const int freqs[6] = {48000, 44100, 16000, 96000, 32000, 8000};
+ hr = S_FALSE;
+
+ // Iterate over frequencies and channels, in order of priority
+ for (unsigned int freq = 0; freq < sizeof(freqs) / sizeof(freqs[0]); freq++) {
+ for (unsigned int chan = 0;
+ chan < sizeof(_recChannelsPrioList) / sizeof(_recChannelsPrioList[0]);
+ chan++) {
+ Wfx.Format.nChannels = _recChannelsPrioList[chan];
+ Wfx.Format.nSamplesPerSec = freqs[freq];
+ Wfx.Format.nBlockAlign =
+ Wfx.Format.nChannels * Wfx.Format.wBitsPerSample / 8;
+ Wfx.Format.nAvgBytesPerSec =
+ Wfx.Format.nSamplesPerSec * Wfx.Format.nBlockAlign;
+ // If the method succeeds and the audio endpoint device supports the
+ // specified stream format, it returns S_OK. If the method succeeds and
+ // provides a closest match to the specified format, it returns S_FALSE.
+ hr = _ptrClientIn->IsFormatSupported(
+ AUDCLNT_SHAREMODE_SHARED, (WAVEFORMATEX*)&Wfx, &pWfxClosestMatch);
+ if (hr == S_OK) {
+ break;
+ } else {
+ if (pWfxClosestMatch) {
+ LOG(INFO) << "nChannels=" << Wfx.Format.nChannels
+ << ", nSamplesPerSec=" << Wfx.Format.nSamplesPerSec
+ << " is not supported. Closest match: "
+ << "nChannels=" << pWfxClosestMatch->nChannels
+ << ", nSamplesPerSec=" << pWfxClosestMatch->nSamplesPerSec;
+ CoTaskMemFree(pWfxClosestMatch);
+ pWfxClosestMatch = NULL;
+ } else {
+ LOG(INFO) << "nChannels=" << Wfx.Format.nChannels
+ << ", nSamplesPerSec=" << Wfx.Format.nSamplesPerSec
+ << " is not supported. No closest match.";
+ }
+ }
+ }
+ if (hr == S_OK)
+ break;
+ }
+
+ if (hr == S_OK) {
+ _recAudioFrameSize = Wfx.Format.nBlockAlign;
+ _recSampleRate = Wfx.Format.nSamplesPerSec;
+ _recBlockSize = Wfx.Format.nSamplesPerSec / 100;
+ _recChannels = Wfx.Format.nChannels;
+
+ LOG(LS_VERBOSE) << "VoE selected this capturing format:";
+ LOG(LS_VERBOSE) << "wFormatTag : 0x" << std::hex
+ << Wfx.Format.wFormatTag << std::dec << " ("
+ << Wfx.Format.wFormatTag << ")";
+ LOG(LS_VERBOSE) << "nChannels : " << Wfx.Format.nChannels;
+ LOG(LS_VERBOSE) << "nSamplesPerSec : " << Wfx.Format.nSamplesPerSec;
+ LOG(LS_VERBOSE) << "nAvgBytesPerSec : " << Wfx.Format.nAvgBytesPerSec;
+ LOG(LS_VERBOSE) << "nBlockAlign : " << Wfx.Format.nBlockAlign;
+ LOG(LS_VERBOSE) << "wBitsPerSample : " << Wfx.Format.wBitsPerSample;
+ LOG(LS_VERBOSE) << "cbSize : " << Wfx.Format.cbSize;
+ LOG(LS_VERBOSE) << "Additional settings:";
+ LOG(LS_VERBOSE) << "_recAudioFrameSize: " << _recAudioFrameSize;
+ LOG(LS_VERBOSE) << "_recBlockSize : " << _recBlockSize;
+ LOG(LS_VERBOSE) << "_recChannels : " << _recChannels;
+ }
+
+ // Create a capturing stream.
+ hr = _ptrClientIn->Initialize(
+ AUDCLNT_SHAREMODE_SHARED, // share Audio Engine with other applications
+ AUDCLNT_STREAMFLAGS_EVENTCALLBACK | // processing of the audio buffer by
+ // the client will be event driven
+ AUDCLNT_STREAMFLAGS_NOPERSIST, // volume and mute settings for an
+ // audio session will not persist
+ // across system restarts
+ 0, // required for event-driven shared mode
+ 0, // periodicity
+ (WAVEFORMATEX*)&Wfx, // selected wave format
+ NULL); // session GUID
+
+ if (hr != S_OK) {
+ LOG(LS_ERROR) << "IAudioClient::Initialize() failed:";
+ }
+ EXIT_ON_ERROR(hr);
+
+ if (_ptrAudioBuffer) {
+ // Update the audio buffer with the selected parameters
+ _ptrAudioBuffer->SetRecordingSampleRate(_recSampleRate);
+ _ptrAudioBuffer->SetRecordingChannels((uint8_t)_recChannels);
+ } else {
+ // We can enter this state during CoreAudioIsSupported() when no
+ // AudioDeviceImplementation has been created, hence the AudioDeviceBuffer
+ // does not exist. It is OK to end up here since we don't initiate any media
+ // in CoreAudioIsSupported().
+ LOG(LS_VERBOSE)
+ << "AudioDeviceBuffer must be attached before streaming can start";
+ }
+
+ // Get the actual size of the shared (endpoint buffer).
+ // Typical value is 960 audio frames <=> 20ms @ 48kHz sample rate.
+ UINT bufferFrameCount(0);
+ hr = _ptrClientIn->GetBufferSize(&bufferFrameCount);
+ if (SUCCEEDED(hr)) {
+ LOG(LS_VERBOSE) << "IAudioClient::GetBufferSize() => " << bufferFrameCount
+ << " (<=> " << bufferFrameCount * _recAudioFrameSize
+ << " bytes)";
+ }
+
+ // Set the event handle that the system signals when an audio buffer is ready
+ // to be processed by the client.
+ hr = _ptrClientIn->SetEventHandle(_hCaptureSamplesReadyEvent);
+ EXIT_ON_ERROR(hr);
+
+ // Get an IAudioCaptureClient interface.
+ SAFE_RELEASE(_ptrCaptureClient);
+ hr = _ptrClientIn->GetService(__uuidof(IAudioCaptureClient),
+ (void**)&_ptrCaptureClient);
+ EXIT_ON_ERROR(hr);
+
+ // Mark capture side as initialized
+ _recIsInitialized = true;
+
+ CoTaskMemFree(pWfxIn);
+ CoTaskMemFree(pWfxClosestMatch);
+
+ LOG(LS_VERBOSE) << "capture side is now initialized";
+ return 0;
Exit:
- _TraceCOMError(hr);
- CoTaskMemFree(pWfxIn);
- CoTaskMemFree(pWfxClosestMatch);
- SAFE_RELEASE(_ptrClientIn);
- SAFE_RELEASE(_ptrCaptureClient);
- return -1;
+ _TraceCOMError(hr);
+ CoTaskMemFree(pWfxIn);
+ CoTaskMemFree(pWfxClosestMatch);
+ SAFE_RELEASE(_ptrClientIn);
+ SAFE_RELEASE(_ptrCaptureClient);
+ return -1;
}
// ----------------------------------------------------------------------------
// StartRecording
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::StartRecording()
-{
+int32_t AudioDeviceWindowsCore::StartRecording() {
+ if (!_recIsInitialized) {
+ return -1;
+ }
- if (!_recIsInitialized)
- {
- return -1;
- }
-
- if (_hRecThread != NULL)
- {
- return 0;
- }
-
- if (_recording)
- {
- return 0;
- }
-
- {
- rtc::CritScope critScoped(&_critSect);
-
- // Create thread which will drive the capturing
- LPTHREAD_START_ROUTINE lpStartAddress = WSAPICaptureThread;
- if (_builtInAecEnabled)
- {
- // Redirect to the DMO polling method.
- lpStartAddress = WSAPICaptureThreadPollDMO;
-
- if (!_playing)
- {
- // The DMO won't provide us captured output data unless we
- // give it render data to process.
- LOG(LS_ERROR)
- << "Playout must be started before recording when using"
- << " the built-in AEC";
- return -1;
- }
- }
-
- assert(_hRecThread == NULL);
- _hRecThread = CreateThread(NULL,
- 0,
- lpStartAddress,
- this,
- 0,
- NULL);
- if (_hRecThread == NULL)
- {
- LOG(LS_ERROR) << "failed to create the recording thread";
- return -1;
- }
-
- // Set thread priority to highest possible
- SetThreadPriority(_hRecThread, THREAD_PRIORITY_TIME_CRITICAL);
-
- assert(_hGetCaptureVolumeThread == NULL);
- _hGetCaptureVolumeThread = CreateThread(NULL,
- 0,
- GetCaptureVolumeThread,
- this,
- 0,
- NULL);
- if (_hGetCaptureVolumeThread == NULL)
- {
- LOG(LS_ERROR) << "failed to create the volume getter thread";
- return -1;
- }
-
- assert(_hSetCaptureVolumeThread == NULL);
- _hSetCaptureVolumeThread = CreateThread(NULL,
- 0,
- SetCaptureVolumeThread,
- this,
- 0,
- NULL);
- if (_hSetCaptureVolumeThread == NULL)
- {
- LOG(LS_ERROR) << "failed to create the volume setter thread";
- return -1;
- }
- } // critScoped
-
- DWORD ret = WaitForSingleObject(_hCaptureStartedEvent, 1000);
- if (ret != WAIT_OBJECT_0)
- {
- LOG(LS_VERBOSE) << "capturing did not start up properly";
- return -1;
- }
- LOG(LS_VERBOSE) << "capture audio stream has now started...";
-
- _recording = true;
-
+ if (_hRecThread != NULL) {
return 0;
+ }
+
+ if (_recording) {
+ return 0;
+ }
+
+ {
+ rtc::CritScope critScoped(&_critSect);
+
+ // Create thread which will drive the capturing
+ LPTHREAD_START_ROUTINE lpStartAddress = WSAPICaptureThread;
+ if (_builtInAecEnabled) {
+ // Redirect to the DMO polling method.
+ lpStartAddress = WSAPICaptureThreadPollDMO;
+
+ if (!_playing) {
+ // The DMO won't provide us captured output data unless we
+ // give it render data to process.
+ LOG(LS_ERROR) << "Playout must be started before recording when using"
+ << " the built-in AEC";
+ return -1;
+ }
+ }
+
+ assert(_hRecThread == NULL);
+ _hRecThread = CreateThread(NULL, 0, lpStartAddress, this, 0, NULL);
+ if (_hRecThread == NULL) {
+ LOG(LS_ERROR) << "failed to create the recording thread";
+ return -1;
+ }
+
+ // Set thread priority to highest possible
+ SetThreadPriority(_hRecThread, THREAD_PRIORITY_TIME_CRITICAL);
+
+ assert(_hGetCaptureVolumeThread == NULL);
+ _hGetCaptureVolumeThread =
+ CreateThread(NULL, 0, GetCaptureVolumeThread, this, 0, NULL);
+ if (_hGetCaptureVolumeThread == NULL) {
+ LOG(LS_ERROR) << "failed to create the volume getter thread";
+ return -1;
+ }
+
+ assert(_hSetCaptureVolumeThread == NULL);
+ _hSetCaptureVolumeThread =
+ CreateThread(NULL, 0, SetCaptureVolumeThread, this, 0, NULL);
+ if (_hSetCaptureVolumeThread == NULL) {
+ LOG(LS_ERROR) << "failed to create the volume setter thread";
+ return -1;
+ }
+ } // critScoped
+
+ DWORD ret = WaitForSingleObject(_hCaptureStartedEvent, 1000);
+ if (ret != WAIT_OBJECT_0) {
+ LOG(LS_VERBOSE) << "capturing did not start up properly";
+ return -1;
+ }
+ LOG(LS_VERBOSE) << "capture audio stream has now started...";
+
+ _recording = true;
+
+ return 0;
}
// ----------------------------------------------------------------------------
// StopRecording
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::StopRecording()
-{
- int32_t err = 0;
+int32_t AudioDeviceWindowsCore::StopRecording() {
+ int32_t err = 0;
- if (!_recIsInitialized)
- {
- return 0;
- }
+ if (!_recIsInitialized) {
+ return 0;
+ }
- _Lock();
+ _Lock();
- if (_hRecThread == NULL)
- {
- LOG(LS_VERBOSE)
- << "no capturing stream is active => close down WASAPI only";
- SAFE_RELEASE(_ptrClientIn);
- SAFE_RELEASE(_ptrCaptureClient);
- _recIsInitialized = false;
- _recording = false;
- _UnLock();
- return 0;
- }
-
- // Stop the driving thread...
- LOG(LS_VERBOSE) << "closing down the webrtc_core_audio_capture_thread...";
- // Manual-reset event; it will remain signalled to stop all capture threads.
- SetEvent(_hShutdownCaptureEvent);
-
- _UnLock();
- DWORD ret = WaitForSingleObject(_hRecThread, 2000);
- if (ret != WAIT_OBJECT_0)
- {
- LOG(LS_ERROR)
- << "failed to close down webrtc_core_audio_capture_thread";
- err = -1;
- }
- else
- {
- LOG(LS_VERBOSE) << "webrtc_core_audio_capture_thread is now closed";
- }
-
- ret = WaitForSingleObject(_hGetCaptureVolumeThread, 2000);
- if (ret != WAIT_OBJECT_0)
- {
- // the thread did not stop as it should
- LOG(LS_ERROR) << "failed to close down volume getter thread";
- err = -1;
- }
- else
- {
- LOG(LS_VERBOSE) << "volume getter thread is now closed";
- }
-
- ret = WaitForSingleObject(_hSetCaptureVolumeThread, 2000);
- if (ret != WAIT_OBJECT_0)
- {
- // the thread did not stop as it should
- LOG(LS_ERROR) << "failed to close down volume setter thread";
- err = -1;
- }
- else
- {
- LOG(LS_VERBOSE) << "volume setter thread is now closed";
- }
- _Lock();
-
- ResetEvent(_hShutdownCaptureEvent); // Must be manually reset.
- // Ensure that the thread has released these interfaces properly.
- assert(err == -1 || _ptrClientIn == NULL);
- assert(err == -1 || _ptrCaptureClient == NULL);
-
+ if (_hRecThread == NULL) {
+ LOG(LS_VERBOSE)
+ << "no capturing stream is active => close down WASAPI only";
+ SAFE_RELEASE(_ptrClientIn);
+ SAFE_RELEASE(_ptrCaptureClient);
_recIsInitialized = false;
_recording = false;
-
- // These will create thread leaks in the result of an error,
- // but we can at least resume the call.
- CloseHandle(_hRecThread);
- _hRecThread = NULL;
-
- CloseHandle(_hGetCaptureVolumeThread);
- _hGetCaptureVolumeThread = NULL;
-
- CloseHandle(_hSetCaptureVolumeThread);
- _hSetCaptureVolumeThread = NULL;
-
- if (_builtInAecEnabled)
- {
- assert(_dmo != NULL);
- // This is necessary. Otherwise the DMO can generate garbage render
- // audio even after rendering has stopped.
- HRESULT hr = _dmo->FreeStreamingResources();
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- err = -1;
- }
- }
-
- // Reset the recording delay value.
- _sndCardRecDelay = 0;
-
_UnLock();
+ return 0;
+ }
- return err;
+ // Stop the driving thread...
+ LOG(LS_VERBOSE) << "closing down the webrtc_core_audio_capture_thread...";
+ // Manual-reset event; it will remain signalled to stop all capture threads.
+ SetEvent(_hShutdownCaptureEvent);
+
+ _UnLock();
+ DWORD ret = WaitForSingleObject(_hRecThread, 2000);
+ if (ret != WAIT_OBJECT_0) {
+ LOG(LS_ERROR) << "failed to close down webrtc_core_audio_capture_thread";
+ err = -1;
+ } else {
+ LOG(LS_VERBOSE) << "webrtc_core_audio_capture_thread is now closed";
+ }
+
+ ret = WaitForSingleObject(_hGetCaptureVolumeThread, 2000);
+ if (ret != WAIT_OBJECT_0) {
+ // the thread did not stop as it should
+ LOG(LS_ERROR) << "failed to close down volume getter thread";
+ err = -1;
+ } else {
+ LOG(LS_VERBOSE) << "volume getter thread is now closed";
+ }
+
+ ret = WaitForSingleObject(_hSetCaptureVolumeThread, 2000);
+ if (ret != WAIT_OBJECT_0) {
+ // the thread did not stop as it should
+ LOG(LS_ERROR) << "failed to close down volume setter thread";
+ err = -1;
+ } else {
+ LOG(LS_VERBOSE) << "volume setter thread is now closed";
+ }
+ _Lock();
+
+ ResetEvent(_hShutdownCaptureEvent); // Must be manually reset.
+ // Ensure that the thread has released these interfaces properly.
+ assert(err == -1 || _ptrClientIn == NULL);
+ assert(err == -1 || _ptrCaptureClient == NULL);
+
+ _recIsInitialized = false;
+ _recording = false;
+
+ // These will create thread leaks in the result of an error,
+ // but we can at least resume the call.
+ CloseHandle(_hRecThread);
+ _hRecThread = NULL;
+
+ CloseHandle(_hGetCaptureVolumeThread);
+ _hGetCaptureVolumeThread = NULL;
+
+ CloseHandle(_hSetCaptureVolumeThread);
+ _hSetCaptureVolumeThread = NULL;
+
+ if (_builtInAecEnabled) {
+ assert(_dmo != NULL);
+ // This is necessary. Otherwise the DMO can generate garbage render
+ // audio even after rendering has stopped.
+ HRESULT hr = _dmo->FreeStreamingResources();
+ if (FAILED(hr)) {
+ _TraceCOMError(hr);
+ err = -1;
+ }
+ }
+
+ // Reset the recording delay value.
+ _sndCardRecDelay = 0;
+
+ _UnLock();
+
+ return err;
}
// ----------------------------------------------------------------------------
// RecordingIsInitialized
// ----------------------------------------------------------------------------
-bool AudioDeviceWindowsCore::RecordingIsInitialized() const
-{
- return (_recIsInitialized);
+bool AudioDeviceWindowsCore::RecordingIsInitialized() const {
+ return (_recIsInitialized);
}
// ----------------------------------------------------------------------------
// Recording
// ----------------------------------------------------------------------------
-bool AudioDeviceWindowsCore::Recording() const
-{
- return (_recording);
+bool AudioDeviceWindowsCore::Recording() const {
+ return (_recording);
}
// ----------------------------------------------------------------------------
// PlayoutIsInitialized
// ----------------------------------------------------------------------------
-bool AudioDeviceWindowsCore::PlayoutIsInitialized() const
-{
-
- return (_playIsInitialized);
+bool AudioDeviceWindowsCore::PlayoutIsInitialized() const {
+ return (_playIsInitialized);
}
// ----------------------------------------------------------------------------
// StartPlayout
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::StartPlayout()
-{
+int32_t AudioDeviceWindowsCore::StartPlayout() {
+ if (!_playIsInitialized) {
+ return -1;
+ }
- if (!_playIsInitialized)
- {
- return -1;
- }
-
- if (_hPlayThread != NULL)
- {
- return 0;
- }
-
- if (_playing)
- {
- return 0;
- }
-
- {
- rtc::CritScope critScoped(&_critSect);
-
- // Create thread which will drive the rendering.
- assert(_hPlayThread == NULL);
- _hPlayThread = CreateThread(
- NULL,
- 0,
- WSAPIRenderThread,
- this,
- 0,
- NULL);
- if (_hPlayThread == NULL)
- {
- LOG(LS_ERROR) << "failed to create the playout thread";
- return -1;
- }
-
- // Set thread priority to highest possible.
- SetThreadPriority(_hPlayThread, THREAD_PRIORITY_TIME_CRITICAL);
- } // critScoped
-
- DWORD ret = WaitForSingleObject(_hRenderStartedEvent, 1000);
- if (ret != WAIT_OBJECT_0)
- {
- LOG(LS_VERBOSE) << "rendering did not start up properly";
- return -1;
- }
-
- _playing = true;
- LOG(LS_VERBOSE) << "rendering audio stream has now started...";
-
+ if (_hPlayThread != NULL) {
return 0;
+ }
+
+ if (_playing) {
+ return 0;
+ }
+
+ {
+ rtc::CritScope critScoped(&_critSect);
+
+ // Create thread which will drive the rendering.
+ assert(_hPlayThread == NULL);
+ _hPlayThread = CreateThread(NULL, 0, WSAPIRenderThread, this, 0, NULL);
+ if (_hPlayThread == NULL) {
+ LOG(LS_ERROR) << "failed to create the playout thread";
+ return -1;
+ }
+
+ // Set thread priority to highest possible.
+ SetThreadPriority(_hPlayThread, THREAD_PRIORITY_TIME_CRITICAL);
+ } // critScoped
+
+ DWORD ret = WaitForSingleObject(_hRenderStartedEvent, 1000);
+ if (ret != WAIT_OBJECT_0) {
+ LOG(LS_VERBOSE) << "rendering did not start up properly";
+ return -1;
+ }
+
+ _playing = true;
+ LOG(LS_VERBOSE) << "rendering audio stream has now started...";
+
+ return 0;
}
// ----------------------------------------------------------------------------
// StopPlayout
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::StopPlayout()
-{
-
- if (!_playIsInitialized)
- {
- return 0;
- }
-
- {
- rtc::CritScope critScoped(&_critSect) ;
-
- if (_hPlayThread == NULL)
- {
- LOG(LS_VERBOSE)
- << "no rendering stream is active => close down WASAPI only";
- SAFE_RELEASE(_ptrClientOut);
- SAFE_RELEASE(_ptrRenderClient);
- _playIsInitialized = false;
- _playing = false;
- return 0;
- }
-
- // stop the driving thread...
- LOG(LS_VERBOSE)
- << "closing down the webrtc_core_audio_render_thread...";
- SetEvent(_hShutdownRenderEvent);
- } // critScoped
-
- DWORD ret = WaitForSingleObject(_hPlayThread, 2000);
- if (ret != WAIT_OBJECT_0)
- {
- // the thread did not stop as it should
- LOG(LS_ERROR) << "failed to close down webrtc_core_audio_render_thread";
- CloseHandle(_hPlayThread);
- _hPlayThread = NULL;
- _playIsInitialized = false;
- _playing = false;
- return -1;
- }
-
- {
- rtc::CritScope critScoped(&_critSect);
- LOG(LS_VERBOSE) << "webrtc_core_audio_render_thread is now closed";
-
- // to reset this event manually at each time we finish with it,
- // in case that the render thread has exited before StopPlayout(),
- // this event might be caught by the new render thread within same VoE instance.
- ResetEvent(_hShutdownRenderEvent);
-
- SAFE_RELEASE(_ptrClientOut);
- SAFE_RELEASE(_ptrRenderClient);
-
- _playIsInitialized = false;
- _playing = false;
-
- CloseHandle(_hPlayThread);
- _hPlayThread = NULL;
-
- if (_builtInAecEnabled && _recording)
- {
- // The DMO won't provide us captured output data unless we
- // give it render data to process.
- //
- // We still permit the playout to shutdown, and trace a warning.
- // Otherwise, VoE can get into a state which will never permit
- // playout to stop properly.
- LOG(LS_WARNING)
- << "Recording should be stopped before playout when using the"
- << " built-in AEC";
- }
-
- // Reset the playout delay value.
- _sndCardPlayDelay = 0;
- } // critScoped
-
+int32_t AudioDeviceWindowsCore::StopPlayout() {
+ if (!_playIsInitialized) {
return 0;
+ }
+
+ {
+ rtc::CritScope critScoped(&_critSect);
+
+ if (_hPlayThread == NULL) {
+ LOG(LS_VERBOSE)
+ << "no rendering stream is active => close down WASAPI only";
+ SAFE_RELEASE(_ptrClientOut);
+ SAFE_RELEASE(_ptrRenderClient);
+ _playIsInitialized = false;
+ _playing = false;
+ return 0;
+ }
+
+ // stop the driving thread...
+ LOG(LS_VERBOSE) << "closing down the webrtc_core_audio_render_thread...";
+ SetEvent(_hShutdownRenderEvent);
+ } // critScoped
+
+ DWORD ret = WaitForSingleObject(_hPlayThread, 2000);
+ if (ret != WAIT_OBJECT_0) {
+ // the thread did not stop as it should
+ LOG(LS_ERROR) << "failed to close down webrtc_core_audio_render_thread";
+ CloseHandle(_hPlayThread);
+ _hPlayThread = NULL;
+ _playIsInitialized = false;
+ _playing = false;
+ return -1;
+ }
+
+ {
+ rtc::CritScope critScoped(&_critSect);
+ LOG(LS_VERBOSE) << "webrtc_core_audio_render_thread is now closed";
+
+ // to reset this event manually at each time we finish with it,
+ // in case that the render thread has exited before StopPlayout(),
+ // this event might be caught by the new render thread within same VoE
+ // instance.
+ ResetEvent(_hShutdownRenderEvent);
+
+ SAFE_RELEASE(_ptrClientOut);
+ SAFE_RELEASE(_ptrRenderClient);
+
+ _playIsInitialized = false;
+ _playing = false;
+
+ CloseHandle(_hPlayThread);
+ _hPlayThread = NULL;
+
+ if (_builtInAecEnabled && _recording) {
+ // The DMO won't provide us captured output data unless we
+ // give it render data to process.
+ //
+ // We still permit the playout to shutdown, and trace a warning.
+ // Otherwise, VoE can get into a state which will never permit
+ // playout to stop properly.
+ LOG(LS_WARNING)
+ << "Recording should be stopped before playout when using the"
+ << " built-in AEC";
+ }
+
+ // Reset the playout delay value.
+ _sndCardPlayDelay = 0;
+ } // critScoped
+
+ return 0;
}
// ----------------------------------------------------------------------------
// PlayoutDelay
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::PlayoutDelay(uint16_t& delayMS) const
-{
- rtc::CritScope critScoped(&_critSect);
- delayMS = static_cast<uint16_t>(_sndCardPlayDelay);
- return 0;
+int32_t AudioDeviceWindowsCore::PlayoutDelay(uint16_t& delayMS) const {
+ rtc::CritScope critScoped(&_critSect);
+ delayMS = static_cast<uint16_t>(_sndCardPlayDelay);
+ return 0;
}
// ----------------------------------------------------------------------------
// Playing
// ----------------------------------------------------------------------------
-bool AudioDeviceWindowsCore::Playing() const
-{
- return (_playing);
+bool AudioDeviceWindowsCore::Playing() const {
+ return (_playing);
}
// ============================================================================
@@ -3049,1068 +2707,953 @@
// [static] WSAPIRenderThread
// ----------------------------------------------------------------------------
-DWORD WINAPI AudioDeviceWindowsCore::WSAPIRenderThread(LPVOID context)
-{
- return reinterpret_cast<AudioDeviceWindowsCore*>(context)->
- DoRenderThread();
+DWORD WINAPI AudioDeviceWindowsCore::WSAPIRenderThread(LPVOID context) {
+ return reinterpret_cast<AudioDeviceWindowsCore*>(context)->DoRenderThread();
}
// ----------------------------------------------------------------------------
// [static] WSAPICaptureThread
// ----------------------------------------------------------------------------
-DWORD WINAPI AudioDeviceWindowsCore::WSAPICaptureThread(LPVOID context)
-{
- return reinterpret_cast<AudioDeviceWindowsCore*>(context)->
- DoCaptureThread();
+DWORD WINAPI AudioDeviceWindowsCore::WSAPICaptureThread(LPVOID context) {
+ return reinterpret_cast<AudioDeviceWindowsCore*>(context)->DoCaptureThread();
}
-DWORD WINAPI AudioDeviceWindowsCore::WSAPICaptureThreadPollDMO(LPVOID context)
-{
- return reinterpret_cast<AudioDeviceWindowsCore*>(context)->
- DoCaptureThreadPollDMO();
+DWORD WINAPI AudioDeviceWindowsCore::WSAPICaptureThreadPollDMO(LPVOID context) {
+ return reinterpret_cast<AudioDeviceWindowsCore*>(context)
+ ->DoCaptureThreadPollDMO();
}
-DWORD WINAPI AudioDeviceWindowsCore::GetCaptureVolumeThread(LPVOID context)
-{
- return reinterpret_cast<AudioDeviceWindowsCore*>(context)->
- DoGetCaptureVolumeThread();
+DWORD WINAPI AudioDeviceWindowsCore::GetCaptureVolumeThread(LPVOID context) {
+ return reinterpret_cast<AudioDeviceWindowsCore*>(context)
+ ->DoGetCaptureVolumeThread();
}
-DWORD WINAPI AudioDeviceWindowsCore::SetCaptureVolumeThread(LPVOID context)
-{
- return reinterpret_cast<AudioDeviceWindowsCore*>(context)->
- DoSetCaptureVolumeThread();
+DWORD WINAPI AudioDeviceWindowsCore::SetCaptureVolumeThread(LPVOID context) {
+ return reinterpret_cast<AudioDeviceWindowsCore*>(context)
+ ->DoSetCaptureVolumeThread();
}
-DWORD AudioDeviceWindowsCore::DoGetCaptureVolumeThread()
-{
- HANDLE waitObject = _hShutdownCaptureEvent;
+DWORD AudioDeviceWindowsCore::DoGetCaptureVolumeThread() {
+ HANDLE waitObject = _hShutdownCaptureEvent;
- while (1)
- {
- if (AGC())
- {
- uint32_t currentMicLevel = 0;
- if (MicrophoneVolume(currentMicLevel) == 0)
- {
- // This doesn't set the system volume, just stores it.
- _Lock();
- if (_ptrAudioBuffer)
- {
- _ptrAudioBuffer->SetCurrentMicLevel(currentMicLevel);
- }
- _UnLock();
- }
- }
-
- DWORD waitResult = WaitForSingleObject(waitObject,
- GET_MIC_VOLUME_INTERVAL_MS);
- switch (waitResult)
- {
- case WAIT_OBJECT_0: // _hShutdownCaptureEvent
- return 0;
- case WAIT_TIMEOUT: // timeout notification
- break;
- default: // unexpected error
- LOG(LS_WARNING)
- << "unknown wait termination on get volume thread";
- return 1;
- }
- }
-}
-
-DWORD AudioDeviceWindowsCore::DoSetCaptureVolumeThread()
-{
- HANDLE waitArray[2] = {_hShutdownCaptureEvent, _hSetCaptureVolumeEvent};
-
- while (1)
- {
- DWORD waitResult = WaitForMultipleObjects(2, waitArray, FALSE, INFINITE);
- switch (waitResult)
- {
- case WAIT_OBJECT_0: // _hShutdownCaptureEvent
- return 0;
- case WAIT_OBJECT_0 + 1: // _hSetCaptureVolumeEvent
- break;
- default: // unexpected error
- LOG(LS_WARNING)
- << "unknown wait termination on set volume thread";
- return 1;
- }
-
+ while (1) {
+ if (AGC()) {
+ uint32_t currentMicLevel = 0;
+ if (MicrophoneVolume(currentMicLevel) == 0) {
+ // This doesn't set the system volume, just stores it.
_Lock();
- uint32_t newMicLevel = _newMicLevel;
- _UnLock();
-
- if (SetMicrophoneVolume(newMicLevel) == -1)
- {
- LOG(LS_WARNING)
- << "the required modification of the microphone volume failed";
+ if (_ptrAudioBuffer) {
+ _ptrAudioBuffer->SetCurrentMicLevel(currentMicLevel);
}
+ _UnLock();
+ }
}
+
+ DWORD waitResult =
+ WaitForSingleObject(waitObject, GET_MIC_VOLUME_INTERVAL_MS);
+ switch (waitResult) {
+ case WAIT_OBJECT_0: // _hShutdownCaptureEvent
+ return 0;
+ case WAIT_TIMEOUT: // timeout notification
+ break;
+ default: // unexpected error
+ LOG(LS_WARNING) << "unknown wait termination on get volume thread";
+ return 1;
+ }
+ }
+}
+
+DWORD AudioDeviceWindowsCore::DoSetCaptureVolumeThread() {
+ HANDLE waitArray[2] = {_hShutdownCaptureEvent, _hSetCaptureVolumeEvent};
+
+ while (1) {
+ DWORD waitResult = WaitForMultipleObjects(2, waitArray, FALSE, INFINITE);
+ switch (waitResult) {
+ case WAIT_OBJECT_0: // _hShutdownCaptureEvent
+ return 0;
+ case WAIT_OBJECT_0 + 1: // _hSetCaptureVolumeEvent
+ break;
+ default: // unexpected error
+ LOG(LS_WARNING) << "unknown wait termination on set volume thread";
+ return 1;
+ }
+
+ _Lock();
+ uint32_t newMicLevel = _newMicLevel;
+ _UnLock();
+
+ if (SetMicrophoneVolume(newMicLevel) == -1) {
+ LOG(LS_WARNING)
+ << "the required modification of the microphone volume failed";
+ }
+ }
}
// ----------------------------------------------------------------------------
// DoRenderThread
// ----------------------------------------------------------------------------
-DWORD AudioDeviceWindowsCore::DoRenderThread()
-{
+DWORD AudioDeviceWindowsCore::DoRenderThread() {
+ bool keepPlaying = true;
+ HANDLE waitArray[2] = {_hShutdownRenderEvent, _hRenderSamplesReadyEvent};
+ HRESULT hr = S_OK;
+ HANDLE hMmTask = NULL;
- bool keepPlaying = true;
- HANDLE waitArray[2] = {_hShutdownRenderEvent, _hRenderSamplesReadyEvent};
- HRESULT hr = S_OK;
- HANDLE hMmTask = NULL;
+ // Initialize COM as MTA in this thread.
+ ScopedCOMInitializer comInit(ScopedCOMInitializer::kMTA);
+ if (!comInit.succeeded()) {
+ LOG(LS_ERROR) << "failed to initialize COM in render thread";
+ return 1;
+ }
- // Initialize COM as MTA in this thread.
- ScopedCOMInitializer comInit(ScopedCOMInitializer::kMTA);
- if (!comInit.succeeded()) {
- LOG(LS_ERROR) << "failed to initialize COM in render thread";
- return 1;
+ rtc::SetCurrentThreadName("webrtc_core_audio_render_thread");
+
+ // Use Multimedia Class Scheduler Service (MMCSS) to boost the thread
+ // priority.
+ //
+ if (_winSupportAvrt) {
+ DWORD taskIndex(0);
+ hMmTask = _PAvSetMmThreadCharacteristicsA("Pro Audio", &taskIndex);
+ if (hMmTask) {
+ if (FALSE == _PAvSetMmThreadPriority(hMmTask, AVRT_PRIORITY_CRITICAL)) {
+ LOG(LS_WARNING) << "failed to boost play-thread using MMCSS";
+ }
+ LOG(LS_VERBOSE)
+ << "render thread is now registered with MMCSS (taskIndex="
+ << taskIndex << ")";
+ } else {
+ LOG(LS_WARNING) << "failed to enable MMCSS on render thread (err="
+ << GetLastError() << ")";
+ _TraceCOMError(GetLastError());
+ }
+ }
+
+ _Lock();
+
+ IAudioClock* clock = NULL;
+
+ // Get size of rendering buffer (length is expressed as the number of audio
+ // frames the buffer can hold). This value is fixed during the rendering
+ // session.
+ //
+ UINT32 bufferLength = 0;
+ hr = _ptrClientOut->GetBufferSize(&bufferLength);
+ EXIT_ON_ERROR(hr);
+ LOG(LS_VERBOSE) << "[REND] size of buffer : " << bufferLength;
+
+ // Get maximum latency for the current stream (will not change for the
+ // lifetime of the IAudioClient object).
+ //
+ REFERENCE_TIME latency;
+ _ptrClientOut->GetStreamLatency(&latency);
+ LOG(LS_VERBOSE) << "[REND] max stream latency : " << (DWORD)latency << " ("
+ << (double)(latency / 10000.0) << " ms)";
+
+ // Get the length of the periodic interval separating successive processing
+ // passes by the audio engine on the data in the endpoint buffer.
+ //
+ // The period between processing passes by the audio engine is fixed for a
+ // particular audio endpoint device and represents the smallest processing
+ // quantum for the audio engine. This period plus the stream latency between
+ // the buffer and endpoint device represents the minimum possible latency that
+ // an audio application can achieve. Typical value: 100000 <=> 0.01 sec =
+ // 10ms.
+ //
+ REFERENCE_TIME devPeriod = 0;
+ REFERENCE_TIME devPeriodMin = 0;
+ _ptrClientOut->GetDevicePeriod(&devPeriod, &devPeriodMin);
+ LOG(LS_VERBOSE) << "[REND] device period : " << (DWORD)devPeriod
+ << " (" << (double)(devPeriod / 10000.0) << " ms)";
+
+ // Derive initial rendering delay.
+ // Example: 10*(960/480) + 15 = 20 + 15 = 35ms
+ //
+ int playout_delay = 10 * (bufferLength / _playBlockSizeInFrames) +
+ (int)((latency + devPeriod) / 10000);
+ _sndCardPlayDelay = playout_delay;
+ _writtenSamples = 0;
+ LOG(LS_VERBOSE) << "[REND] initial delay : " << playout_delay;
+
+ double endpointBufferSizeMS =
+ 10.0 * ((double)bufferLength / (double)_devicePlayBlockSize);
+ LOG(LS_VERBOSE) << "[REND] endpointBufferSizeMS : " << endpointBufferSizeMS;
+
+ // Before starting the stream, fill the rendering buffer with silence.
+ //
+ BYTE* pData = NULL;
+ hr = _ptrRenderClient->GetBuffer(bufferLength, &pData);
+ EXIT_ON_ERROR(hr);
+
+ hr =
+ _ptrRenderClient->ReleaseBuffer(bufferLength, AUDCLNT_BUFFERFLAGS_SILENT);
+ EXIT_ON_ERROR(hr);
+
+ _writtenSamples += bufferLength;
+
+ hr = _ptrClientOut->GetService(__uuidof(IAudioClock), (void**)&clock);
+ if (FAILED(hr)) {
+ LOG(LS_WARNING)
+ << "failed to get IAudioClock interface from the IAudioClient";
+ }
+
+ // Start up the rendering audio stream.
+ hr = _ptrClientOut->Start();
+ EXIT_ON_ERROR(hr);
+
+ _UnLock();
+
+ // Set event which will ensure that the calling thread modifies the playing
+ // state to true.
+ //
+ SetEvent(_hRenderStartedEvent);
+
+ // >> ------------------ THREAD LOOP ------------------
+
+ while (keepPlaying) {
+ // Wait for a render notification event or a shutdown event
+ DWORD waitResult = WaitForMultipleObjects(2, waitArray, FALSE, 500);
+ switch (waitResult) {
+ case WAIT_OBJECT_0 + 0: // _hShutdownRenderEvent
+ keepPlaying = false;
+ break;
+ case WAIT_OBJECT_0 + 1: // _hRenderSamplesReadyEvent
+ break;
+ case WAIT_TIMEOUT: // timeout notification
+ LOG(LS_WARNING) << "render event timed out after 0.5 seconds";
+ goto Exit;
+ default: // unexpected error
+ LOG(LS_WARNING) << "unknown wait termination on render side";
+ goto Exit;
}
- rtc::SetCurrentThreadName("webrtc_core_audio_render_thread");
+ while (keepPlaying) {
+ _Lock();
- // Use Multimedia Class Scheduler Service (MMCSS) to boost the thread priority.
- //
- if (_winSupportAvrt)
- {
- DWORD taskIndex(0);
- hMmTask = _PAvSetMmThreadCharacteristicsA("Pro Audio", &taskIndex);
- if (hMmTask)
- {
- if (FALSE == _PAvSetMmThreadPriority(hMmTask, AVRT_PRIORITY_CRITICAL))
- {
- LOG(LS_WARNING) << "failed to boost play-thread using MMCSS";
- }
- LOG(LS_VERBOSE)
- << "render thread is now registered with MMCSS (taskIndex="
- << taskIndex << ")";
- }
- else
- {
- LOG(LS_WARNING) << "failed to enable MMCSS on render thread (err="
- << GetLastError() << ")";
- _TraceCOMError(GetLastError());
- }
- }
+ // Sanity check to ensure that essential states are not modified
+ // during the unlocked period.
+ if (_ptrRenderClient == NULL || _ptrClientOut == NULL) {
+ _UnLock();
+ LOG(LS_ERROR)
+ << "output state has been modified during unlocked period";
+ goto Exit;
+ }
- _Lock();
+ // Get the number of frames of padding (queued up to play) in the endpoint
+ // buffer.
+ UINT32 padding = 0;
+ hr = _ptrClientOut->GetCurrentPadding(&padding);
+ EXIT_ON_ERROR(hr);
- IAudioClock* clock = NULL;
+ // Derive the amount of available space in the output buffer
+ uint32_t framesAvailable = bufferLength - padding;
- // Get size of rendering buffer (length is expressed as the number of audio frames the buffer can hold).
- // This value is fixed during the rendering session.
- //
- UINT32 bufferLength = 0;
- hr = _ptrClientOut->GetBufferSize(&bufferLength);
- EXIT_ON_ERROR(hr);
- LOG(LS_VERBOSE) << "[REND] size of buffer : " << bufferLength;
+ // Do we have 10 ms available in the render buffer?
+ if (framesAvailable < _playBlockSizeInFrames) {
+ // Not enough space in render buffer to store next render packet.
+ _UnLock();
+ break;
+ }
- // Get maximum latency for the current stream (will not change for the lifetime of the IAudioClient object).
- //
- REFERENCE_TIME latency;
- _ptrClientOut->GetStreamLatency(&latency);
- LOG(LS_VERBOSE) << "[REND] max stream latency : " << (DWORD)latency
- << " (" << (double)(latency/10000.0) << " ms)";
+ // Write n*10ms buffers to the render buffer
+ const uint32_t n10msBuffers = (framesAvailable / _playBlockSizeInFrames);
+ for (uint32_t n = 0; n < n10msBuffers; n++) {
+ // Get pointer (i.e., grab the buffer) to next space in the shared
+ // render buffer.
+ hr = _ptrRenderClient->GetBuffer(_playBlockSizeInFrames, &pData);
+ EXIT_ON_ERROR(hr);
- // Get the length of the periodic interval separating successive processing passes by
- // the audio engine on the data in the endpoint buffer.
- //
- // The period between processing passes by the audio engine is fixed for a particular
- // audio endpoint device and represents the smallest processing quantum for the audio engine.
- // This period plus the stream latency between the buffer and endpoint device represents
- // the minimum possible latency that an audio application can achieve.
- // Typical value: 100000 <=> 0.01 sec = 10ms.
- //
- REFERENCE_TIME devPeriod = 0;
- REFERENCE_TIME devPeriodMin = 0;
- _ptrClientOut->GetDevicePeriod(&devPeriod, &devPeriodMin);
- LOG(LS_VERBOSE) << "[REND] device period : " << (DWORD)devPeriod
- << " (" << (double)(devPeriod/10000.0) << " ms)";
+ if (_ptrAudioBuffer) {
+ // Request data to be played out (#bytes =
+ // _playBlockSizeInFrames*_audioFrameSize)
+ _UnLock();
+ int32_t nSamples =
+ _ptrAudioBuffer->RequestPlayoutData(_playBlockSizeInFrames);
+ _Lock();
- // Derive initial rendering delay.
- // Example: 10*(960/480) + 15 = 20 + 15 = 35ms
- //
- int playout_delay = 10 * (bufferLength / _playBlockSizeInFrames) +
- (int)((latency + devPeriod) / 10000);
- _sndCardPlayDelay = playout_delay;
- _writtenSamples = 0;
- LOG(LS_VERBOSE) << "[REND] initial delay : " << playout_delay;
-
- double endpointBufferSizeMS = 10.0 * ((double)bufferLength / (double)_devicePlayBlockSize);
- LOG(LS_VERBOSE) << "[REND] endpointBufferSizeMS : " << endpointBufferSizeMS;
-
- // Before starting the stream, fill the rendering buffer with silence.
- //
- BYTE *pData = NULL;
- hr = _ptrRenderClient->GetBuffer(bufferLength, &pData);
- EXIT_ON_ERROR(hr);
-
- hr = _ptrRenderClient->ReleaseBuffer(bufferLength, AUDCLNT_BUFFERFLAGS_SILENT);
- EXIT_ON_ERROR(hr);
-
- _writtenSamples += bufferLength;
-
- hr = _ptrClientOut->GetService(__uuidof(IAudioClock), (void**)&clock);
- if (FAILED(hr)) {
- LOG(LS_WARNING)
- << "failed to get IAudioClock interface from the IAudioClient";
- }
-
- // Start up the rendering audio stream.
- hr = _ptrClientOut->Start();
- EXIT_ON_ERROR(hr);
-
- _UnLock();
-
- // Set event which will ensure that the calling thread modifies the playing state to true.
- //
- SetEvent(_hRenderStartedEvent);
-
- // >> ------------------ THREAD LOOP ------------------
-
- while (keepPlaying)
- {
- // Wait for a render notification event or a shutdown event
- DWORD waitResult = WaitForMultipleObjects(2, waitArray, FALSE, 500);
- switch (waitResult)
- {
- case WAIT_OBJECT_0 + 0: // _hShutdownRenderEvent
- keepPlaying = false;
- break;
- case WAIT_OBJECT_0 + 1: // _hRenderSamplesReadyEvent
- break;
- case WAIT_TIMEOUT: // timeout notification
- LOG(LS_WARNING) << "render event timed out after 0.5 seconds";
- goto Exit;
- default: // unexpected error
- LOG(LS_WARNING) << "unknown wait termination on render side";
- goto Exit;
- }
-
- while (keepPlaying)
- {
- _Lock();
-
- // Sanity check to ensure that essential states are not modified
- // during the unlocked period.
- if (_ptrRenderClient == NULL || _ptrClientOut == NULL)
- {
- _UnLock();
- LOG(LS_ERROR)
- << "output state has been modified during unlocked period";
- goto Exit;
- }
-
- // Get the number of frames of padding (queued up to play) in the endpoint buffer.
- UINT32 padding = 0;
- hr = _ptrClientOut->GetCurrentPadding(&padding);
- EXIT_ON_ERROR(hr);
-
- // Derive the amount of available space in the output buffer
- uint32_t framesAvailable = bufferLength - padding;
-
- // Do we have 10 ms available in the render buffer?
- if (framesAvailable < _playBlockSizeInFrames) {
- // Not enough space in render buffer to store next render packet.
- _UnLock();
- break;
- }
-
- // Write n*10ms buffers to the render buffer
- const uint32_t n10msBuffers =
- (framesAvailable / _playBlockSizeInFrames);
- for (uint32_t n = 0; n < n10msBuffers; n++)
- {
- // Get pointer (i.e., grab the buffer) to next space in the shared render buffer.
- hr =
- _ptrRenderClient->GetBuffer(_playBlockSizeInFrames, &pData);
- EXIT_ON_ERROR(hr);
-
- if (_ptrAudioBuffer)
- {
- // Request data to be played out (#bytes =
- // _playBlockSizeInFrames*_audioFrameSize)
- _UnLock();
- int32_t nSamples = _ptrAudioBuffer->RequestPlayoutData(
- _playBlockSizeInFrames);
- _Lock();
-
- if (nSamples == -1) {
- _UnLock();
- LOG(LS_ERROR) << "failed to read data from render client";
- goto Exit;
- }
-
- // Sanity check to ensure that essential states are not modified during the unlocked period
- if (_ptrRenderClient == NULL || _ptrClientOut == NULL)
- {
- _UnLock();
- LOG(LS_ERROR)
- << "output state has been modified during unlocked"
- << " period";
- goto Exit;
- }
- if (nSamples !=
- static_cast<int32_t>(_playBlockSizeInSamples)) {
- LOG(LS_WARNING)
- << "nSamples(" << nSamples
- << ") != _playBlockSizeInSamples("
- << _playBlockSizeInSamples << ")";
- }
-
- // Get the actual (stored) data
- nSamples = _ptrAudioBuffer->GetPlayoutData((int8_t*)pData);
- }
-
- DWORD dwFlags(0);
- hr = _ptrRenderClient->ReleaseBuffer(_playBlockSizeInFrames,
- dwFlags);
- // See http://msdn.microsoft.com/en-us/library/dd316605(VS.85).aspx
- // for more details regarding AUDCLNT_E_DEVICE_INVALIDATED.
- EXIT_ON_ERROR(hr);
-
- _writtenSamples += _playBlockSizeInFrames;
- }
-
- // Check the current delay on the playout side.
- if (clock) {
- UINT64 pos = 0;
- UINT64 freq = 1;
- clock->GetPosition(&pos, NULL);
- clock->GetFrequency(&freq);
- playout_delay = ROUND((double(_writtenSamples) /
- _devicePlaySampleRate - double(pos) / freq) * 1000.0);
- _sndCardPlayDelay = playout_delay;
- }
-
+ if (nSamples == -1) {
_UnLock();
+ LOG(LS_ERROR) << "failed to read data from render client";
+ goto Exit;
+ }
+
+ // Sanity check to ensure that essential states are not modified
+ // during the unlocked period
+ if (_ptrRenderClient == NULL || _ptrClientOut == NULL) {
+ _UnLock();
+ LOG(LS_ERROR) << "output state has been modified during unlocked"
+ << " period";
+ goto Exit;
+ }
+ if (nSamples != static_cast<int32_t>(_playBlockSizeInSamples)) {
+ LOG(LS_WARNING)
+ << "nSamples(" << nSamples << ") != _playBlockSizeInSamples("
+ << _playBlockSizeInSamples << ")";
+ }
+
+ // Get the actual (stored) data
+ nSamples = _ptrAudioBuffer->GetPlayoutData((int8_t*)pData);
}
+
+ DWORD dwFlags(0);
+ hr = _ptrRenderClient->ReleaseBuffer(_playBlockSizeInFrames, dwFlags);
+ // See http://msdn.microsoft.com/en-us/library/dd316605(VS.85).aspx
+ // for more details regarding AUDCLNT_E_DEVICE_INVALIDATED.
+ EXIT_ON_ERROR(hr);
+
+ _writtenSamples += _playBlockSizeInFrames;
+ }
+
+ // Check the current delay on the playout side.
+ if (clock) {
+ UINT64 pos = 0;
+ UINT64 freq = 1;
+ clock->GetPosition(&pos, NULL);
+ clock->GetFrequency(&freq);
+ playout_delay = ROUND((double(_writtenSamples) / _devicePlaySampleRate -
+ double(pos) / freq) *
+ 1000.0);
+ _sndCardPlayDelay = playout_delay;
+ }
+
+ _UnLock();
}
+ }
- // ------------------ THREAD LOOP ------------------ <<
+ // ------------------ THREAD LOOP ------------------ <<
- SleepMs(static_cast<DWORD>(endpointBufferSizeMS+0.5));
- hr = _ptrClientOut->Stop();
+ SleepMs(static_cast<DWORD>(endpointBufferSizeMS + 0.5));
+ hr = _ptrClientOut->Stop();
Exit:
- SAFE_RELEASE(clock);
+ SAFE_RELEASE(clock);
- if (FAILED(hr))
- {
- _ptrClientOut->Stop();
- _UnLock();
- _TraceCOMError(hr);
- }
-
- if (_winSupportAvrt)
- {
- if (NULL != hMmTask)
- {
- _PAvRevertMmThreadCharacteristics(hMmTask);
- }
- }
-
- _Lock();
-
- if (keepPlaying)
- {
- if (_ptrClientOut != NULL)
- {
- hr = _ptrClientOut->Stop();
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- }
- hr = _ptrClientOut->Reset();
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- }
- }
- LOG(LS_ERROR)
- << "Playout error: rendering thread has ended pre-maturely";
- }
- else
- {
- LOG(LS_VERBOSE) << "_Rendering thread is now terminated properly";
- }
-
+ if (FAILED(hr)) {
+ _ptrClientOut->Stop();
_UnLock();
+ _TraceCOMError(hr);
+ }
- return (DWORD)hr;
+ if (_winSupportAvrt) {
+ if (NULL != hMmTask) {
+ _PAvRevertMmThreadCharacteristics(hMmTask);
+ }
+ }
+
+ _Lock();
+
+ if (keepPlaying) {
+ if (_ptrClientOut != NULL) {
+ hr = _ptrClientOut->Stop();
+ if (FAILED(hr)) {
+ _TraceCOMError(hr);
+ }
+ hr = _ptrClientOut->Reset();
+ if (FAILED(hr)) {
+ _TraceCOMError(hr);
+ }
+ }
+ LOG(LS_ERROR) << "Playout error: rendering thread has ended pre-maturely";
+ } else {
+ LOG(LS_VERBOSE) << "_Rendering thread is now terminated properly";
+ }
+
+ _UnLock();
+
+ return (DWORD)hr;
}
-DWORD AudioDeviceWindowsCore::InitCaptureThreadPriority()
-{
- _hMmTask = NULL;
+DWORD AudioDeviceWindowsCore::InitCaptureThreadPriority() {
+ _hMmTask = NULL;
- rtc::SetCurrentThreadName("webrtc_core_audio_capture_thread");
+ rtc::SetCurrentThreadName("webrtc_core_audio_capture_thread");
- // Use Multimedia Class Scheduler Service (MMCSS) to boost the thread
- // priority.
- if (_winSupportAvrt)
- {
- DWORD taskIndex(0);
- _hMmTask = _PAvSetMmThreadCharacteristicsA("Pro Audio", &taskIndex);
- if (_hMmTask)
- {
- if (!_PAvSetMmThreadPriority(_hMmTask, AVRT_PRIORITY_CRITICAL))
- {
- LOG(LS_WARNING) << "failed to boost rec-thread using MMCSS";
- }
- LOG(LS_VERBOSE)
- << "capture thread is now registered with MMCSS (taskIndex="
- << taskIndex << ")";
- }
- else
- {
- LOG(LS_WARNING) << "failed to enable MMCSS on capture thread (err="
- << GetLastError() << ")";
- _TraceCOMError(GetLastError());
- }
+ // Use Multimedia Class Scheduler Service (MMCSS) to boost the thread
+ // priority.
+ if (_winSupportAvrt) {
+ DWORD taskIndex(0);
+ _hMmTask = _PAvSetMmThreadCharacteristicsA("Pro Audio", &taskIndex);
+ if (_hMmTask) {
+ if (!_PAvSetMmThreadPriority(_hMmTask, AVRT_PRIORITY_CRITICAL)) {
+ LOG(LS_WARNING) << "failed to boost rec-thread using MMCSS";
+ }
+ LOG(LS_VERBOSE)
+ << "capture thread is now registered with MMCSS (taskIndex="
+ << taskIndex << ")";
+ } else {
+ LOG(LS_WARNING) << "failed to enable MMCSS on capture thread (err="
+ << GetLastError() << ")";
+ _TraceCOMError(GetLastError());
}
+ }
- return S_OK;
+ return S_OK;
}
-void AudioDeviceWindowsCore::RevertCaptureThreadPriority()
-{
- if (_winSupportAvrt)
- {
- if (NULL != _hMmTask)
- {
- _PAvRevertMmThreadCharacteristics(_hMmTask);
- }
+void AudioDeviceWindowsCore::RevertCaptureThreadPriority() {
+ if (_winSupportAvrt) {
+ if (NULL != _hMmTask) {
+ _PAvRevertMmThreadCharacteristics(_hMmTask);
}
+ }
- _hMmTask = NULL;
+ _hMmTask = NULL;
}
-DWORD AudioDeviceWindowsCore::DoCaptureThreadPollDMO()
-{
- assert(_mediaBuffer != NULL);
- bool keepRecording = true;
+DWORD AudioDeviceWindowsCore::DoCaptureThreadPollDMO() {
+ assert(_mediaBuffer != NULL);
+ bool keepRecording = true;
- // Initialize COM as MTA in this thread.
- ScopedCOMInitializer comInit(ScopedCOMInitializer::kMTA);
- if (!comInit.succeeded()) {
- LOG(LS_ERROR) << "failed to initialize COM in polling DMO thread";
- return 1;
- }
+ // Initialize COM as MTA in this thread.
+ ScopedCOMInitializer comInit(ScopedCOMInitializer::kMTA);
+ if (!comInit.succeeded()) {
+ LOG(LS_ERROR) << "failed to initialize COM in polling DMO thread";
+ return 1;
+ }
- HRESULT hr = InitCaptureThreadPriority();
- if (FAILED(hr))
- {
- return hr;
- }
-
- // Set event which will ensure that the calling thread modifies the
- // recording state to true.
- SetEvent(_hCaptureStartedEvent);
-
- // >> ---------------------------- THREAD LOOP ----------------------------
- while (keepRecording)
- {
- // Poll the DMO every 5 ms.
- // (The same interval used in the Wave implementation.)
- DWORD waitResult = WaitForSingleObject(_hShutdownCaptureEvent, 5);
- switch (waitResult)
- {
- case WAIT_OBJECT_0: // _hShutdownCaptureEvent
- keepRecording = false;
- break;
- case WAIT_TIMEOUT: // timeout notification
- break;
- default: // unexpected error
- LOG(LS_WARNING) << "Unknown wait termination on capture side";
- hr = -1; // To signal an error callback.
- keepRecording = false;
- break;
- }
-
- while (keepRecording)
- {
- rtc::CritScope critScoped(&_critSect);
-
- DWORD dwStatus = 0;
- {
- DMO_OUTPUT_DATA_BUFFER dmoBuffer = {0};
- dmoBuffer.pBuffer = _mediaBuffer;
- dmoBuffer.pBuffer->AddRef();
-
- // Poll the DMO for AEC processed capture data. The DMO will
- // copy available data to |dmoBuffer|, and should only return
- // 10 ms frames. The value of |dwStatus| should be ignored.
- hr = _dmo->ProcessOutput(0, 1, &dmoBuffer, &dwStatus);
- SAFE_RELEASE(dmoBuffer.pBuffer);
- dwStatus = dmoBuffer.dwStatus;
- }
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- keepRecording = false;
- assert(false);
- break;
- }
-
- ULONG bytesProduced = 0;
- BYTE* data;
- // Get a pointer to the data buffer. This should be valid until
- // the next call to ProcessOutput.
- hr = _mediaBuffer->GetBufferAndLength(&data, &bytesProduced);
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- keepRecording = false;
- assert(false);
- break;
- }
-
- // TODO(andrew): handle AGC.
-
- if (bytesProduced > 0)
- {
- const int kSamplesProduced = bytesProduced / _recAudioFrameSize;
- // TODO(andrew): verify that this is always satisfied. It might
- // be that ProcessOutput will try to return more than 10 ms if
- // we fail to call it frequently enough.
- assert(kSamplesProduced == static_cast<int>(_recBlockSize));
- assert(sizeof(BYTE) == sizeof(int8_t));
- _ptrAudioBuffer->SetRecordedBuffer(
- reinterpret_cast<int8_t*>(data),
- kSamplesProduced);
- _ptrAudioBuffer->SetVQEData(0, 0, 0);
-
- _UnLock(); // Release lock while making the callback.
- _ptrAudioBuffer->DeliverRecordedData();
- _Lock();
- }
-
- // Reset length to indicate buffer availability.
- hr = _mediaBuffer->SetLength(0);
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- keepRecording = false;
- assert(false);
- break;
- }
-
- if (!(dwStatus & DMO_OUTPUT_DATA_BUFFERF_INCOMPLETE))
- {
- // The DMO cannot currently produce more data. This is the
- // normal case; otherwise it means the DMO had more than 10 ms
- // of data available and ProcessOutput should be called again.
- break;
- }
- }
- }
- // ---------------------------- THREAD LOOP ---------------------------- <<
-
- RevertCaptureThreadPriority();
-
- if (FAILED(hr))
- {
- LOG(LS_ERROR)
- << "Recording error: capturing thread has ended prematurely";
- }
- else
- {
- LOG(LS_VERBOSE) << "Capturing thread is now terminated properly";
- }
-
+ HRESULT hr = InitCaptureThreadPriority();
+ if (FAILED(hr)) {
return hr;
-}
+ }
+ // Set event which will ensure that the calling thread modifies the
+ // recording state to true.
+ SetEvent(_hCaptureStartedEvent);
+
+ // >> ---------------------------- THREAD LOOP ----------------------------
+ while (keepRecording) {
+ // Poll the DMO every 5 ms.
+ // (The same interval used in the Wave implementation.)
+ DWORD waitResult = WaitForSingleObject(_hShutdownCaptureEvent, 5);
+ switch (waitResult) {
+ case WAIT_OBJECT_0: // _hShutdownCaptureEvent
+ keepRecording = false;
+ break;
+ case WAIT_TIMEOUT: // timeout notification
+ break;
+ default: // unexpected error
+ LOG(LS_WARNING) << "Unknown wait termination on capture side";
+ hr = -1; // To signal an error callback.
+ keepRecording = false;
+ break;
+ }
+
+ while (keepRecording) {
+ rtc::CritScope critScoped(&_critSect);
+
+ DWORD dwStatus = 0;
+ {
+ DMO_OUTPUT_DATA_BUFFER dmoBuffer = {0};
+ dmoBuffer.pBuffer = _mediaBuffer;
+ dmoBuffer.pBuffer->AddRef();
+
+ // Poll the DMO for AEC processed capture data. The DMO will
+ // copy available data to |dmoBuffer|, and should only return
+ // 10 ms frames. The value of |dwStatus| should be ignored.
+ hr = _dmo->ProcessOutput(0, 1, &dmoBuffer, &dwStatus);
+ SAFE_RELEASE(dmoBuffer.pBuffer);
+ dwStatus = dmoBuffer.dwStatus;
+ }
+ if (FAILED(hr)) {
+ _TraceCOMError(hr);
+ keepRecording = false;
+ assert(false);
+ break;
+ }
+
+ ULONG bytesProduced = 0;
+ BYTE* data;
+ // Get a pointer to the data buffer. This should be valid until
+ // the next call to ProcessOutput.
+ hr = _mediaBuffer->GetBufferAndLength(&data, &bytesProduced);
+ if (FAILED(hr)) {
+ _TraceCOMError(hr);
+ keepRecording = false;
+ assert(false);
+ break;
+ }
+
+ // TODO(andrew): handle AGC.
+
+ if (bytesProduced > 0) {
+ const int kSamplesProduced = bytesProduced / _recAudioFrameSize;
+ // TODO(andrew): verify that this is always satisfied. It might
+ // be that ProcessOutput will try to return more than 10 ms if
+ // we fail to call it frequently enough.
+ assert(kSamplesProduced == static_cast<int>(_recBlockSize));
+ assert(sizeof(BYTE) == sizeof(int8_t));
+ _ptrAudioBuffer->SetRecordedBuffer(reinterpret_cast<int8_t*>(data),
+ kSamplesProduced);
+ _ptrAudioBuffer->SetVQEData(0, 0, 0);
+
+ _UnLock(); // Release lock while making the callback.
+ _ptrAudioBuffer->DeliverRecordedData();
+ _Lock();
+ }
+
+ // Reset length to indicate buffer availability.
+ hr = _mediaBuffer->SetLength(0);
+ if (FAILED(hr)) {
+ _TraceCOMError(hr);
+ keepRecording = false;
+ assert(false);
+ break;
+ }
+
+ if (!(dwStatus & DMO_OUTPUT_DATA_BUFFERF_INCOMPLETE)) {
+ // The DMO cannot currently produce more data. This is the
+ // normal case; otherwise it means the DMO had more than 10 ms
+ // of data available and ProcessOutput should be called again.
+ break;
+ }
+ }
+ }
+ // ---------------------------- THREAD LOOP ---------------------------- <<
+
+ RevertCaptureThreadPriority();
+
+ if (FAILED(hr)) {
+ LOG(LS_ERROR) << "Recording error: capturing thread has ended prematurely";
+ } else {
+ LOG(LS_VERBOSE) << "Capturing thread is now terminated properly";
+ }
+
+ return hr;
+}
// ----------------------------------------------------------------------------
// DoCaptureThread
// ----------------------------------------------------------------------------
-DWORD AudioDeviceWindowsCore::DoCaptureThread()
-{
+DWORD AudioDeviceWindowsCore::DoCaptureThread() {
+ bool keepRecording = true;
+ HANDLE waitArray[2] = {_hShutdownCaptureEvent, _hCaptureSamplesReadyEvent};
+ HRESULT hr = S_OK;
- bool keepRecording = true;
- HANDLE waitArray[2] = {_hShutdownCaptureEvent, _hCaptureSamplesReadyEvent};
- HRESULT hr = S_OK;
+ LARGE_INTEGER t1;
- LARGE_INTEGER t1;
+ BYTE* syncBuffer = NULL;
+ UINT32 syncBufIndex = 0;
- BYTE* syncBuffer = NULL;
- UINT32 syncBufIndex = 0;
+ _readSamples = 0;
- _readSamples = 0;
+ // Initialize COM as MTA in this thread.
+ ScopedCOMInitializer comInit(ScopedCOMInitializer::kMTA);
+ if (!comInit.succeeded()) {
+ LOG(LS_ERROR) << "failed to initialize COM in capture thread";
+ return 1;
+ }
- // Initialize COM as MTA in this thread.
- ScopedCOMInitializer comInit(ScopedCOMInitializer::kMTA);
- if (!comInit.succeeded()) {
- LOG(LS_ERROR) << "failed to initialize COM in capture thread";
- return 1;
+ hr = InitCaptureThreadPriority();
+ if (FAILED(hr)) {
+ return hr;
+ }
+
+ _Lock();
+
+ // Get size of capturing buffer (length is expressed as the number of audio
+ // frames the buffer can hold). This value is fixed during the capturing
+ // session.
+ //
+ UINT32 bufferLength = 0;
+ if (_ptrClientIn == NULL) {
+ LOG(LS_ERROR)
+ << "input state has been modified before capture loop starts.";
+ return 1;
+ }
+ hr = _ptrClientIn->GetBufferSize(&bufferLength);
+ EXIT_ON_ERROR(hr);
+ LOG(LS_VERBOSE) << "[CAPT] size of buffer : " << bufferLength;
+
+ // Allocate memory for sync buffer.
+ // It is used for compensation between native 44.1 and internal 44.0 and
+ // for cases when the capture buffer is larger than 10ms.
+ //
+ const UINT32 syncBufferSize = 2 * (bufferLength * _recAudioFrameSize);
+ syncBuffer = new BYTE[syncBufferSize];
+ if (syncBuffer == NULL) {
+ return (DWORD)E_POINTER;
+ }
+ LOG(LS_VERBOSE) << "[CAPT] size of sync buffer : " << syncBufferSize
+ << " [bytes]";
+
+ // Get maximum latency for the current stream (will not change for the
+ // lifetime of the IAudioClient object).
+ //
+ REFERENCE_TIME latency;
+ _ptrClientIn->GetStreamLatency(&latency);
+ LOG(LS_VERBOSE) << "[CAPT] max stream latency : " << (DWORD)latency << " ("
+ << (double)(latency / 10000.0) << " ms)";
+
+ // Get the length of the periodic interval separating successive processing
+ // passes by the audio engine on the data in the endpoint buffer.
+ //
+ REFERENCE_TIME devPeriod = 0;
+ REFERENCE_TIME devPeriodMin = 0;
+ _ptrClientIn->GetDevicePeriod(&devPeriod, &devPeriodMin);
+ LOG(LS_VERBOSE) << "[CAPT] device period : " << (DWORD)devPeriod
+ << " (" << (double)(devPeriod / 10000.0) << " ms)";
+
+ double extraDelayMS = (double)((latency + devPeriod) / 10000.0);
+ LOG(LS_VERBOSE) << "[CAPT] extraDelayMS : " << extraDelayMS;
+
+ double endpointBufferSizeMS =
+ 10.0 * ((double)bufferLength / (double)_recBlockSize);
+ LOG(LS_VERBOSE) << "[CAPT] endpointBufferSizeMS : " << endpointBufferSizeMS;
+
+ // Start up the capturing stream.
+ //
+ hr = _ptrClientIn->Start();
+ EXIT_ON_ERROR(hr);
+
+ _UnLock();
+
+ // Set event which will ensure that the calling thread modifies the recording
+ // state to true.
+ //
+ SetEvent(_hCaptureStartedEvent);
+
+ // >> ---------------------------- THREAD LOOP ----------------------------
+
+ while (keepRecording) {
+ // Wait for a capture notification event or a shutdown event
+ DWORD waitResult = WaitForMultipleObjects(2, waitArray, FALSE, 500);
+ switch (waitResult) {
+ case WAIT_OBJECT_0 + 0: // _hShutdownCaptureEvent
+ keepRecording = false;
+ break;
+ case WAIT_OBJECT_0 + 1: // _hCaptureSamplesReadyEvent
+ break;
+ case WAIT_TIMEOUT: // timeout notification
+ LOG(LS_WARNING) << "capture event timed out after 0.5 seconds";
+ goto Exit;
+ default: // unexpected error
+ LOG(LS_WARNING) << "unknown wait termination on capture side";
+ goto Exit;
}
- hr = InitCaptureThreadPriority();
- if (FAILED(hr))
- {
- return hr;
- }
+ while (keepRecording) {
+ BYTE* pData = 0;
+ UINT32 framesAvailable = 0;
+ DWORD flags = 0;
+ UINT64 recTime = 0;
+ UINT64 recPos = 0;
- _Lock();
+ _Lock();
- // Get size of capturing buffer (length is expressed as the number of audio frames the buffer can hold).
- // This value is fixed during the capturing session.
- //
- UINT32 bufferLength = 0;
- if (_ptrClientIn == NULL)
- {
- LOG(LS_ERROR)
- << "input state has been modified before capture loop starts.";
- return 1;
- }
- hr = _ptrClientIn->GetBufferSize(&bufferLength);
- EXIT_ON_ERROR(hr);
- LOG(LS_VERBOSE) << "[CAPT] size of buffer : " << bufferLength;
+ // Sanity check to ensure that essential states are not modified
+ // during the unlocked period.
+ if (_ptrCaptureClient == NULL || _ptrClientIn == NULL) {
+ _UnLock();
+ LOG(LS_ERROR) << "input state has been modified during unlocked period";
+ goto Exit;
+ }
- // Allocate memory for sync buffer.
- // It is used for compensation between native 44.1 and internal 44.0 and
- // for cases when the capture buffer is larger than 10ms.
- //
- const UINT32 syncBufferSize = 2*(bufferLength * _recAudioFrameSize);
- syncBuffer = new BYTE[syncBufferSize];
- if (syncBuffer == NULL)
- {
- return (DWORD)E_POINTER;
- }
- LOG(LS_VERBOSE) << "[CAPT] size of sync buffer : " << syncBufferSize
- << " [bytes]";
+ // Find out how much capture data is available
+ //
+ hr = _ptrCaptureClient->GetBuffer(
+ &pData, // packet which is ready to be read by used
+ &framesAvailable, // #frames in the captured packet (can be zero)
+ &flags, // support flags (check)
+ &recPos, // device position of first audio frame in data packet
+ &recTime); // value of performance counter at the time of recording
+ // the first audio frame
- // Get maximum latency for the current stream (will not change for the lifetime of the IAudioClient object).
- //
- REFERENCE_TIME latency;
- _ptrClientIn->GetStreamLatency(&latency);
- LOG(LS_VERBOSE) << "[CAPT] max stream latency : " << (DWORD)latency
- << " (" << (double)(latency / 10000.0) << " ms)";
-
- // Get the length of the periodic interval separating successive processing passes by
- // the audio engine on the data in the endpoint buffer.
- //
- REFERENCE_TIME devPeriod = 0;
- REFERENCE_TIME devPeriodMin = 0;
- _ptrClientIn->GetDevicePeriod(&devPeriod, &devPeriodMin);
- LOG(LS_VERBOSE) << "[CAPT] device period : " << (DWORD)devPeriod
- << " (" << (double)(devPeriod / 10000.0) << " ms)";
-
- double extraDelayMS = (double)((latency + devPeriod) / 10000.0);
- LOG(LS_VERBOSE) << "[CAPT] extraDelayMS : " << extraDelayMS;
-
- double endpointBufferSizeMS = 10.0 * ((double)bufferLength / (double)_recBlockSize);
- LOG(LS_VERBOSE) << "[CAPT] endpointBufferSizeMS : " << endpointBufferSizeMS;
-
- // Start up the capturing stream.
- //
- hr = _ptrClientIn->Start();
- EXIT_ON_ERROR(hr);
-
- _UnLock();
-
- // Set event which will ensure that the calling thread modifies the recording state to true.
- //
- SetEvent(_hCaptureStartedEvent);
-
- // >> ---------------------------- THREAD LOOP ----------------------------
-
- while (keepRecording)
- {
- // Wait for a capture notification event or a shutdown event
- DWORD waitResult = WaitForMultipleObjects(2, waitArray, FALSE, 500);
- switch (waitResult)
- {
- case WAIT_OBJECT_0 + 0: // _hShutdownCaptureEvent
- keepRecording = false;
- break;
- case WAIT_OBJECT_0 + 1: // _hCaptureSamplesReadyEvent
- break;
- case WAIT_TIMEOUT: // timeout notification
- LOG(LS_WARNING) << "capture event timed out after 0.5 seconds";
- goto Exit;
- default: // unexpected error
- LOG(LS_WARNING) << "unknown wait termination on capture side";
- goto Exit;
+ if (SUCCEEDED(hr)) {
+ if (AUDCLNT_S_BUFFER_EMPTY == hr) {
+ // Buffer was empty => start waiting for a new capture notification
+ // event
+ _UnLock();
+ break;
}
- while (keepRecording)
- {
- BYTE *pData = 0;
- UINT32 framesAvailable = 0;
- DWORD flags = 0;
- UINT64 recTime = 0;
- UINT64 recPos = 0;
+ if (flags & AUDCLNT_BUFFERFLAGS_SILENT) {
+ // Treat all of the data in the packet as silence and ignore the
+ // actual data values.
+ LOG(LS_WARNING) << "AUDCLNT_BUFFERFLAGS_SILENT";
+ pData = NULL;
+ }
- _Lock();
+ assert(framesAvailable != 0);
+
+ if (pData) {
+ CopyMemory(&syncBuffer[syncBufIndex * _recAudioFrameSize], pData,
+ framesAvailable * _recAudioFrameSize);
+ } else {
+ ZeroMemory(&syncBuffer[syncBufIndex * _recAudioFrameSize],
+ framesAvailable * _recAudioFrameSize);
+ }
+ assert(syncBufferSize >= (syncBufIndex * _recAudioFrameSize) +
+ framesAvailable * _recAudioFrameSize);
+
+ // Release the capture buffer
+ //
+ hr = _ptrCaptureClient->ReleaseBuffer(framesAvailable);
+ EXIT_ON_ERROR(hr);
+
+ _readSamples += framesAvailable;
+ syncBufIndex += framesAvailable;
+
+ QueryPerformanceCounter(&t1);
+
+ // Get the current recording and playout delay.
+ uint32_t sndCardRecDelay = (uint32_t)(
+ ((((UINT64)t1.QuadPart * _perfCounterFactor) - recTime) / 10000) +
+ (10 * syncBufIndex) / _recBlockSize - 10);
+ uint32_t sndCardPlayDelay = static_cast<uint32_t>(_sndCardPlayDelay);
+
+ _sndCardRecDelay = sndCardRecDelay;
+
+ while (syncBufIndex >= _recBlockSize) {
+ if (_ptrAudioBuffer) {
+ _ptrAudioBuffer->SetRecordedBuffer((const int8_t*)syncBuffer,
+ _recBlockSize);
+ _ptrAudioBuffer->SetVQEData(sndCardPlayDelay, sndCardRecDelay, 0);
+
+ _ptrAudioBuffer->SetTypingStatus(KeyPressed());
+
+ _UnLock(); // release lock while making the callback
+ _ptrAudioBuffer->DeliverRecordedData();
+ _Lock(); // restore the lock
// Sanity check to ensure that essential states are not modified
- // during the unlocked period.
- if (_ptrCaptureClient == NULL || _ptrClientIn == NULL)
- {
- _UnLock();
- LOG(LS_ERROR)
- << "input state has been modified during unlocked period";
- goto Exit;
+ // during the unlocked period
+ if (_ptrCaptureClient == NULL || _ptrClientIn == NULL) {
+ _UnLock();
+ LOG(LS_ERROR) << "input state has been modified during"
+ << " unlocked period";
+ goto Exit;
}
+ }
- // Find out how much capture data is available
- //
- hr = _ptrCaptureClient->GetBuffer(&pData, // packet which is ready to be read by used
- &framesAvailable, // #frames in the captured packet (can be zero)
- &flags, // support flags (check)
- &recPos, // device position of first audio frame in data packet
- &recTime); // value of performance counter at the time of recording the first audio frame
-
- if (SUCCEEDED(hr))
- {
- if (AUDCLNT_S_BUFFER_EMPTY == hr)
- {
- // Buffer was empty => start waiting for a new capture notification event
- _UnLock();
- break;
- }
-
- if (flags & AUDCLNT_BUFFERFLAGS_SILENT)
- {
- // Treat all of the data in the packet as silence and ignore the actual data values.
- LOG(LS_WARNING) << "AUDCLNT_BUFFERFLAGS_SILENT";
- pData = NULL;
- }
-
- assert(framesAvailable != 0);
-
- if (pData)
- {
- CopyMemory(&syncBuffer[syncBufIndex*_recAudioFrameSize], pData, framesAvailable*_recAudioFrameSize);
- }
- else
- {
- ZeroMemory(&syncBuffer[syncBufIndex*_recAudioFrameSize], framesAvailable*_recAudioFrameSize);
- }
- assert(syncBufferSize >= (syncBufIndex*_recAudioFrameSize)+framesAvailable*_recAudioFrameSize);
-
- // Release the capture buffer
- //
- hr = _ptrCaptureClient->ReleaseBuffer(framesAvailable);
- EXIT_ON_ERROR(hr);
-
- _readSamples += framesAvailable;
- syncBufIndex += framesAvailable;
-
- QueryPerformanceCounter(&t1);
-
- // Get the current recording and playout delay.
- uint32_t sndCardRecDelay = (uint32_t)
- (((((UINT64)t1.QuadPart * _perfCounterFactor) - recTime)
- / 10000) + (10*syncBufIndex) / _recBlockSize - 10);
- uint32_t sndCardPlayDelay =
- static_cast<uint32_t>(_sndCardPlayDelay);
-
- _sndCardRecDelay = sndCardRecDelay;
-
- while (syncBufIndex >= _recBlockSize)
- {
- if (_ptrAudioBuffer)
- {
- _ptrAudioBuffer->SetRecordedBuffer((const int8_t*)syncBuffer, _recBlockSize);
- _ptrAudioBuffer->SetVQEData(sndCardPlayDelay,
- sndCardRecDelay,
- 0);
-
- _ptrAudioBuffer->SetTypingStatus(KeyPressed());
-
- _UnLock(); // release lock while making the callback
- _ptrAudioBuffer->DeliverRecordedData();
- _Lock(); // restore the lock
-
- // Sanity check to ensure that essential states are not modified during the unlocked period
- if (_ptrCaptureClient == NULL || _ptrClientIn == NULL)
- {
- _UnLock();
- LOG(LS_ERROR)
- << "input state has been modified during"
- << " unlocked period";
- goto Exit;
- }
- }
-
- // store remaining data which was not able to deliver as 10ms segment
- MoveMemory(&syncBuffer[0], &syncBuffer[_recBlockSize*_recAudioFrameSize], (syncBufIndex-_recBlockSize)*_recAudioFrameSize);
- syncBufIndex -= _recBlockSize;
- sndCardRecDelay -= 10;
- }
-
- if (_AGC)
- {
- uint32_t newMicLevel = _ptrAudioBuffer->NewMicLevel();
- if (newMicLevel != 0)
- {
- // The VQE will only deliver non-zero microphone levels when a change is needed.
- // Set this new mic level (received from the observer as return value in the callback).
- LOG(LS_VERBOSE) << "AGC change of volume: new="
- << newMicLevel;
- // We store this outside of the audio buffer to avoid
- // having it overwritten by the getter thread.
- _newMicLevel = newMicLevel;
- SetEvent(_hSetCaptureVolumeEvent);
- }
- }
- }
- else
- {
- // If GetBuffer returns AUDCLNT_E_BUFFER_ERROR, the thread consuming the audio samples
- // must wait for the next processing pass. The client might benefit from keeping a count
- // of the failed GetBuffer calls. If GetBuffer returns this error repeatedly, the client
- // can start a new processing loop after shutting down the current client by calling
- // IAudioClient::Stop, IAudioClient::Reset, and releasing the audio client.
- LOG(LS_ERROR) << "IAudioCaptureClient::GetBuffer returned"
- << " AUDCLNT_E_BUFFER_ERROR, hr = 0x"
- << std::hex << hr << std::dec;
- goto Exit;
- }
-
- _UnLock();
+ // store remaining data which was not able to deliver as 10ms segment
+ MoveMemory(&syncBuffer[0],
+ &syncBuffer[_recBlockSize * _recAudioFrameSize],
+ (syncBufIndex - _recBlockSize) * _recAudioFrameSize);
+ syncBufIndex -= _recBlockSize;
+ sndCardRecDelay -= 10;
}
- }
- // ---------------------------- THREAD LOOP ---------------------------- <<
+ if (_AGC) {
+ uint32_t newMicLevel = _ptrAudioBuffer->NewMicLevel();
+ if (newMicLevel != 0) {
+ // The VQE will only deliver non-zero microphone levels when a
+ // change is needed. Set this new mic level (received from the
+ // observer as return value in the callback).
+ LOG(LS_VERBOSE) << "AGC change of volume: new=" << newMicLevel;
+ // We store this outside of the audio buffer to avoid
+ // having it overwritten by the getter thread.
+ _newMicLevel = newMicLevel;
+ SetEvent(_hSetCaptureVolumeEvent);
+ }
+ }
+ } else {
+ // If GetBuffer returns AUDCLNT_E_BUFFER_ERROR, the thread consuming the
+ // audio samples must wait for the next processing pass. The client
+ // might benefit from keeping a count of the failed GetBuffer calls. If
+ // GetBuffer returns this error repeatedly, the client can start a new
+ // processing loop after shutting down the current client by calling
+ // IAudioClient::Stop, IAudioClient::Reset, and releasing the audio
+ // client.
+ LOG(LS_ERROR) << "IAudioCaptureClient::GetBuffer returned"
+ << " AUDCLNT_E_BUFFER_ERROR, hr = 0x" << std::hex << hr
+ << std::dec;
+ goto Exit;
+ }
- if (_ptrClientIn)
- {
- hr = _ptrClientIn->Stop();
+ _UnLock();
}
+ }
+
+ // ---------------------------- THREAD LOOP ---------------------------- <<
+
+ if (_ptrClientIn) {
+ hr = _ptrClientIn->Stop();
+ }
Exit:
- if (FAILED(hr))
- {
- _ptrClientIn->Stop();
- _UnLock();
- _TraceCOMError(hr);
- }
-
- RevertCaptureThreadPriority();
-
- _Lock();
-
- if (keepRecording)
- {
- if (_ptrClientIn != NULL)
- {
- hr = _ptrClientIn->Stop();
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- }
- hr = _ptrClientIn->Reset();
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- }
- }
-
- LOG(LS_ERROR)
- << "Recording error: capturing thread has ended pre-maturely";
- }
- else
- {
- LOG(LS_VERBOSE) << "_Capturing thread is now terminated properly";
- }
-
- SAFE_RELEASE(_ptrClientIn);
- SAFE_RELEASE(_ptrCaptureClient);
-
+ if (FAILED(hr)) {
+ _ptrClientIn->Stop();
_UnLock();
+ _TraceCOMError(hr);
+ }
- if (syncBuffer)
- {
- delete [] syncBuffer;
+ RevertCaptureThreadPriority();
+
+ _Lock();
+
+ if (keepRecording) {
+ if (_ptrClientIn != NULL) {
+ hr = _ptrClientIn->Stop();
+ if (FAILED(hr)) {
+ _TraceCOMError(hr);
+ }
+ hr = _ptrClientIn->Reset();
+ if (FAILED(hr)) {
+ _TraceCOMError(hr);
+ }
}
- return (DWORD)hr;
+ LOG(LS_ERROR) << "Recording error: capturing thread has ended pre-maturely";
+ } else {
+ LOG(LS_VERBOSE) << "_Capturing thread is now terminated properly";
+ }
+
+ SAFE_RELEASE(_ptrClientIn);
+ SAFE_RELEASE(_ptrCaptureClient);
+
+ _UnLock();
+
+ if (syncBuffer) {
+ delete[] syncBuffer;
+ }
+
+ return (DWORD)hr;
}
-int32_t AudioDeviceWindowsCore::EnableBuiltInAEC(bool enable)
-{
+int32_t AudioDeviceWindowsCore::EnableBuiltInAEC(bool enable) {
+ if (_recIsInitialized) {
+ LOG(LS_ERROR)
+ << "Attempt to set Windows AEC with recording already initialized";
+ return -1;
+ }
- if (_recIsInitialized)
- {
- LOG(LS_ERROR)
- << "Attempt to set Windows AEC with recording already initialized";
- return -1;
- }
+ if (_dmo == NULL) {
+ LOG(LS_ERROR)
+ << "Built-in AEC DMO was not initialized properly at create time";
+ return -1;
+ }
- if (_dmo == NULL)
- {
- LOG(LS_ERROR)
- << "Built-in AEC DMO was not initialized properly at create time";
- return -1;
- }
-
- _builtInAecEnabled = enable;
- return 0;
+ _builtInAecEnabled = enable;
+ return 0;
}
-int AudioDeviceWindowsCore::SetDMOProperties()
-{
- HRESULT hr = S_OK;
- assert(_dmo != NULL);
+int AudioDeviceWindowsCore::SetDMOProperties() {
+ HRESULT hr = S_OK;
+ assert(_dmo != NULL);
- rtc::scoped_refptr<IPropertyStore> ps;
- {
- IPropertyStore* ptrPS = NULL;
- hr = _dmo->QueryInterface(IID_IPropertyStore,
- reinterpret_cast<void**>(&ptrPS));
- if (FAILED(hr) || ptrPS == NULL)
- {
- _TraceCOMError(hr);
- return -1;
- }
- ps = ptrPS;
- SAFE_RELEASE(ptrPS);
+ rtc::scoped_refptr<IPropertyStore> ps;
+ {
+ IPropertyStore* ptrPS = NULL;
+ hr = _dmo->QueryInterface(IID_IPropertyStore,
+ reinterpret_cast<void**>(&ptrPS));
+ if (FAILED(hr) || ptrPS == NULL) {
+ _TraceCOMError(hr);
+ return -1;
+ }
+ ps = ptrPS;
+ SAFE_RELEASE(ptrPS);
+ }
+
+ // Set the AEC system mode.
+ // SINGLE_CHANNEL_AEC - AEC processing only.
+ if (SetVtI4Property(ps, MFPKEY_WMAAECMA_SYSTEM_MODE, SINGLE_CHANNEL_AEC)) {
+ return -1;
+ }
+
+ // Set the AEC source mode.
+ // VARIANT_TRUE - Source mode (we poll the AEC for captured data).
+ if (SetBoolProperty(ps, MFPKEY_WMAAECMA_DMO_SOURCE_MODE, VARIANT_TRUE) ==
+ -1) {
+ return -1;
+ }
+
+ // Enable the feature mode.
+ // This lets us override all the default processing settings below.
+ if (SetBoolProperty(ps, MFPKEY_WMAAECMA_FEATURE_MODE, VARIANT_TRUE) == -1) {
+ return -1;
+ }
+
+ // Disable analog AGC (default enabled).
+ if (SetBoolProperty(ps, MFPKEY_WMAAECMA_MIC_GAIN_BOUNDER, VARIANT_FALSE) ==
+ -1) {
+ return -1;
+ }
+
+ // Disable noise suppression (default enabled).
+ // 0 - Disabled, 1 - Enabled
+ if (SetVtI4Property(ps, MFPKEY_WMAAECMA_FEATR_NS, 0) == -1) {
+ return -1;
+ }
+
+ // Relevant parameters to leave at default settings:
+ // MFPKEY_WMAAECMA_FEATR_AGC - Digital AGC (disabled).
+ // MFPKEY_WMAAECMA_FEATR_CENTER_CLIP - AEC center clipping (enabled).
+ // MFPKEY_WMAAECMA_FEATR_ECHO_LENGTH - Filter length (256 ms).
+ // TODO(andrew): investigate decresing the length to 128 ms.
+ // MFPKEY_WMAAECMA_FEATR_FRAME_SIZE - Frame size (0).
+ // 0 is automatic; defaults to 160 samples (or 10 ms frames at the
+ // selected 16 kHz) as long as mic array processing is disabled.
+ // MFPKEY_WMAAECMA_FEATR_NOISE_FILL - Comfort noise (enabled).
+ // MFPKEY_WMAAECMA_FEATR_VAD - VAD (disabled).
+
+ // Set the devices selected by VoE. If using a default device, we need to
+ // search for the device index.
+ int inDevIndex = _inputDeviceIndex;
+ int outDevIndex = _outputDeviceIndex;
+ if (!_usingInputDeviceIndex) {
+ ERole role = eCommunications;
+ if (_inputDevice == AudioDeviceModule::kDefaultDevice) {
+ role = eConsole;
}
- // Set the AEC system mode.
- // SINGLE_CHANNEL_AEC - AEC processing only.
- if (SetVtI4Property(ps,
- MFPKEY_WMAAECMA_SYSTEM_MODE,
- SINGLE_CHANNEL_AEC))
- {
- return -1;
+ if (_GetDefaultDeviceIndex(eCapture, role, &inDevIndex) == -1) {
+ return -1;
+ }
+ }
+
+ if (!_usingOutputDeviceIndex) {
+ ERole role = eCommunications;
+ if (_outputDevice == AudioDeviceModule::kDefaultDevice) {
+ role = eConsole;
}
- // Set the AEC source mode.
- // VARIANT_TRUE - Source mode (we poll the AEC for captured data).
- if (SetBoolProperty(ps,
- MFPKEY_WMAAECMA_DMO_SOURCE_MODE,
- VARIANT_TRUE) == -1)
- {
- return -1;
+ if (_GetDefaultDeviceIndex(eRender, role, &outDevIndex) == -1) {
+ return -1;
}
+ }
- // Enable the feature mode.
- // This lets us override all the default processing settings below.
- if (SetBoolProperty(ps,
- MFPKEY_WMAAECMA_FEATURE_MODE,
- VARIANT_TRUE) == -1)
- {
- return -1;
- }
+ DWORD devIndex = static_cast<uint32_t>(outDevIndex << 16) +
+ static_cast<uint32_t>(0x0000ffff & inDevIndex);
+ LOG(LS_VERBOSE) << "Capture device index: " << inDevIndex
+ << ", render device index: " << outDevIndex;
+ if (SetVtI4Property(ps, MFPKEY_WMAAECMA_DEVICE_INDEXES, devIndex) == -1) {
+ return -1;
+ }
- // Disable analog AGC (default enabled).
- if (SetBoolProperty(ps,
- MFPKEY_WMAAECMA_MIC_GAIN_BOUNDER,
- VARIANT_FALSE) == -1)
- {
- return -1;
- }
-
- // Disable noise suppression (default enabled).
- // 0 - Disabled, 1 - Enabled
- if (SetVtI4Property(ps,
- MFPKEY_WMAAECMA_FEATR_NS,
- 0) == -1)
- {
- return -1;
- }
-
- // Relevant parameters to leave at default settings:
- // MFPKEY_WMAAECMA_FEATR_AGC - Digital AGC (disabled).
- // MFPKEY_WMAAECMA_FEATR_CENTER_CLIP - AEC center clipping (enabled).
- // MFPKEY_WMAAECMA_FEATR_ECHO_LENGTH - Filter length (256 ms).
- // TODO(andrew): investigate decresing the length to 128 ms.
- // MFPKEY_WMAAECMA_FEATR_FRAME_SIZE - Frame size (0).
- // 0 is automatic; defaults to 160 samples (or 10 ms frames at the
- // selected 16 kHz) as long as mic array processing is disabled.
- // MFPKEY_WMAAECMA_FEATR_NOISE_FILL - Comfort noise (enabled).
- // MFPKEY_WMAAECMA_FEATR_VAD - VAD (disabled).
-
- // Set the devices selected by VoE. If using a default device, we need to
- // search for the device index.
- int inDevIndex = _inputDeviceIndex;
- int outDevIndex = _outputDeviceIndex;
- if (!_usingInputDeviceIndex)
- {
- ERole role = eCommunications;
- if (_inputDevice == AudioDeviceModule::kDefaultDevice)
- {
- role = eConsole;
- }
-
- if (_GetDefaultDeviceIndex(eCapture, role, &inDevIndex) == -1)
- {
- return -1;
- }
- }
-
- if (!_usingOutputDeviceIndex)
- {
- ERole role = eCommunications;
- if (_outputDevice == AudioDeviceModule::kDefaultDevice)
- {
- role = eConsole;
- }
-
- if (_GetDefaultDeviceIndex(eRender, role, &outDevIndex) == -1)
- {
- return -1;
- }
- }
-
- DWORD devIndex = static_cast<uint32_t>(outDevIndex << 16) +
- static_cast<uint32_t>(0x0000ffff & inDevIndex);
- LOG(LS_VERBOSE) << "Capture device index: " << inDevIndex
- << ", render device index: " << outDevIndex;
- if (SetVtI4Property(ps,
- MFPKEY_WMAAECMA_DEVICE_INDEXES,
- devIndex) == -1)
- {
- return -1;
- }
-
- return 0;
+ return 0;
}
int AudioDeviceWindowsCore::SetBoolProperty(IPropertyStore* ptrPS,
REFPROPERTYKEY key,
- VARIANT_BOOL value)
-{
- PROPVARIANT pv;
- PropVariantInit(&pv);
- pv.vt = VT_BOOL;
- pv.boolVal = value;
- HRESULT hr = ptrPS->SetValue(key, pv);
- PropVariantClear(&pv);
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- return -1;
- }
- return 0;
+ VARIANT_BOOL value) {
+ PROPVARIANT pv;
+ PropVariantInit(&pv);
+ pv.vt = VT_BOOL;
+ pv.boolVal = value;
+ HRESULT hr = ptrPS->SetValue(key, pv);
+ PropVariantClear(&pv);
+ if (FAILED(hr)) {
+ _TraceCOMError(hr);
+ return -1;
+ }
+ return 0;
}
int AudioDeviceWindowsCore::SetVtI4Property(IPropertyStore* ptrPS,
REFPROPERTYKEY key,
- LONG value)
-{
- PROPVARIANT pv;
- PropVariantInit(&pv);
- pv.vt = VT_I4;
- pv.lVal = value;
- HRESULT hr = ptrPS->SetValue(key, pv);
- PropVariantClear(&pv);
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- return -1;
- }
- return 0;
+ LONG value) {
+ PROPVARIANT pv;
+ PropVariantInit(&pv);
+ pv.vt = VT_I4;
+ pv.lVal = value;
+ HRESULT hr = ptrPS->SetValue(key, pv);
+ PropVariantClear(&pv);
+ if (FAILED(hr)) {
+ _TraceCOMError(hr);
+ return -1;
+ }
+ return 0;
}
// ----------------------------------------------------------------------------
@@ -4121,40 +3664,33 @@
// such devices.
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::_RefreshDeviceList(EDataFlow dir)
-{
- LOG(LS_VERBOSE) << __FUNCTION__;
+int32_t AudioDeviceWindowsCore::_RefreshDeviceList(EDataFlow dir) {
+ LOG(LS_VERBOSE) << __FUNCTION__;
- HRESULT hr = S_OK;
- IMMDeviceCollection *pCollection = NULL;
+ HRESULT hr = S_OK;
+ IMMDeviceCollection* pCollection = NULL;
- assert(dir == eRender || dir == eCapture);
- assert(_ptrEnumerator != NULL);
+ assert(dir == eRender || dir == eCapture);
+ assert(_ptrEnumerator != NULL);
- // Create a fresh list of devices using the specified direction
- hr = _ptrEnumerator->EnumAudioEndpoints(
- dir,
- DEVICE_STATE_ACTIVE,
- &pCollection);
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- SAFE_RELEASE(pCollection);
- return -1;
- }
+ // Create a fresh list of devices using the specified direction
+ hr = _ptrEnumerator->EnumAudioEndpoints(dir, DEVICE_STATE_ACTIVE,
+ &pCollection);
+ if (FAILED(hr)) {
+ _TraceCOMError(hr);
+ SAFE_RELEASE(pCollection);
+ return -1;
+ }
- if (dir == eRender)
- {
- SAFE_RELEASE(_ptrRenderCollection);
- _ptrRenderCollection = pCollection;
- }
- else
- {
- SAFE_RELEASE(_ptrCaptureCollection);
- _ptrCaptureCollection = pCollection;
- }
+ if (dir == eRender) {
+ SAFE_RELEASE(_ptrRenderCollection);
+ _ptrRenderCollection = pCollection;
+ } else {
+ SAFE_RELEASE(_ptrCaptureCollection);
+ _ptrCaptureCollection = pCollection;
+ }
- return 0;
+ return 0;
}
// ----------------------------------------------------------------------------
@@ -4164,31 +3700,26 @@
// current list of such devices.
// ----------------------------------------------------------------------------
-int16_t AudioDeviceWindowsCore::_DeviceListCount(EDataFlow dir)
-{
- LOG(LS_VERBOSE) << __FUNCTION__;
+int16_t AudioDeviceWindowsCore::_DeviceListCount(EDataFlow dir) {
+ LOG(LS_VERBOSE) << __FUNCTION__;
- HRESULT hr = S_OK;
- UINT count = 0;
+ HRESULT hr = S_OK;
+ UINT count = 0;
- assert(eRender == dir || eCapture == dir);
+ assert(eRender == dir || eCapture == dir);
- if (eRender == dir && NULL != _ptrRenderCollection)
- {
- hr = _ptrRenderCollection->GetCount(&count);
- }
- else if (NULL != _ptrCaptureCollection)
- {
- hr = _ptrCaptureCollection->GetCount(&count);
- }
+ if (eRender == dir && NULL != _ptrRenderCollection) {
+ hr = _ptrRenderCollection->GetCount(&count);
+ } else if (NULL != _ptrCaptureCollection) {
+ hr = _ptrCaptureCollection->GetCount(&count);
+ }
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- return -1;
- }
+ if (FAILED(hr)) {
+ _TraceCOMError(hr);
+ return -1;
+ }
- return static_cast<int16_t> (count);
+ return static_cast<int16_t>(count);
}
// ----------------------------------------------------------------------------
@@ -4202,34 +3733,32 @@
// in _RefreshDeviceList().
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::_GetListDeviceName(EDataFlow dir, int index, LPWSTR szBuffer, int bufferLen)
-{
- LOG(LS_VERBOSE) << __FUNCTION__;
+int32_t AudioDeviceWindowsCore::_GetListDeviceName(EDataFlow dir,
+ int index,
+ LPWSTR szBuffer,
+ int bufferLen) {
+ LOG(LS_VERBOSE) << __FUNCTION__;
- HRESULT hr = S_OK;
- IMMDevice *pDevice = NULL;
+ HRESULT hr = S_OK;
+ IMMDevice* pDevice = NULL;
- assert(dir == eRender || dir == eCapture);
+ assert(dir == eRender || dir == eCapture);
- if (eRender == dir && NULL != _ptrRenderCollection)
- {
- hr = _ptrRenderCollection->Item(index, &pDevice);
- }
- else if (NULL != _ptrCaptureCollection)
- {
- hr = _ptrCaptureCollection->Item(index, &pDevice);
- }
+ if (eRender == dir && NULL != _ptrRenderCollection) {
+ hr = _ptrRenderCollection->Item(index, &pDevice);
+ } else if (NULL != _ptrCaptureCollection) {
+ hr = _ptrCaptureCollection->Item(index, &pDevice);
+ }
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- SAFE_RELEASE(pDevice);
- return -1;
- }
-
- int32_t res = _GetDeviceName(pDevice, szBuffer, bufferLen);
+ if (FAILED(hr)) {
+ _TraceCOMError(hr);
SAFE_RELEASE(pDevice);
- return res;
+ return -1;
+ }
+
+ int32_t res = _GetDeviceName(pDevice, szBuffer, bufferLen);
+ SAFE_RELEASE(pDevice);
+ return res;
}
// ----------------------------------------------------------------------------
@@ -4241,32 +3770,30 @@
// Uses: _ptrEnumerator
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::_GetDefaultDeviceName(EDataFlow dir, ERole role, LPWSTR szBuffer, int bufferLen)
-{
- LOG(LS_VERBOSE) << __FUNCTION__;
+int32_t AudioDeviceWindowsCore::_GetDefaultDeviceName(EDataFlow dir,
+ ERole role,
+ LPWSTR szBuffer,
+ int bufferLen) {
+ LOG(LS_VERBOSE) << __FUNCTION__;
- HRESULT hr = S_OK;
- IMMDevice *pDevice = NULL;
+ HRESULT hr = S_OK;
+ IMMDevice* pDevice = NULL;
- assert(dir == eRender || dir == eCapture);
- assert(role == eConsole || role == eCommunications);
- assert(_ptrEnumerator != NULL);
+ assert(dir == eRender || dir == eCapture);
+ assert(role == eConsole || role == eCommunications);
+ assert(_ptrEnumerator != NULL);
- hr = _ptrEnumerator->GetDefaultAudioEndpoint(
- dir,
- role,
- &pDevice);
+ hr = _ptrEnumerator->GetDefaultAudioEndpoint(dir, role, &pDevice);
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- SAFE_RELEASE(pDevice);
- return -1;
- }
-
- int32_t res = _GetDeviceName(pDevice, szBuffer, bufferLen);
+ if (FAILED(hr)) {
+ _TraceCOMError(hr);
SAFE_RELEASE(pDevice);
- return res;
+ return -1;
+ }
+
+ int32_t res = _GetDeviceName(pDevice, szBuffer, bufferLen);
+ SAFE_RELEASE(pDevice);
+ return res;
}
// ----------------------------------------------------------------------------
@@ -4280,34 +3807,32 @@
// in _RefreshDeviceList().
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::_GetListDeviceID(EDataFlow dir, int index, LPWSTR szBuffer, int bufferLen)
-{
- LOG(LS_VERBOSE) << __FUNCTION__;
+int32_t AudioDeviceWindowsCore::_GetListDeviceID(EDataFlow dir,
+ int index,
+ LPWSTR szBuffer,
+ int bufferLen) {
+ LOG(LS_VERBOSE) << __FUNCTION__;
- HRESULT hr = S_OK;
- IMMDevice *pDevice = NULL;
+ HRESULT hr = S_OK;
+ IMMDevice* pDevice = NULL;
- assert(dir == eRender || dir == eCapture);
+ assert(dir == eRender || dir == eCapture);
- if (eRender == dir && NULL != _ptrRenderCollection)
- {
- hr = _ptrRenderCollection->Item(index, &pDevice);
- }
- else if (NULL != _ptrCaptureCollection)
- {
- hr = _ptrCaptureCollection->Item(index, &pDevice);
- }
+ if (eRender == dir && NULL != _ptrRenderCollection) {
+ hr = _ptrRenderCollection->Item(index, &pDevice);
+ } else if (NULL != _ptrCaptureCollection) {
+ hr = _ptrCaptureCollection->Item(index, &pDevice);
+ }
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- SAFE_RELEASE(pDevice);
- return -1;
- }
-
- int32_t res = _GetDeviceID(pDevice, szBuffer, bufferLen);
+ if (FAILED(hr)) {
+ _TraceCOMError(hr);
SAFE_RELEASE(pDevice);
- return res;
+ return -1;
+ }
+
+ int32_t res = _GetDeviceID(pDevice, szBuffer, bufferLen);
+ SAFE_RELEASE(pDevice);
+ return res;
}
// ----------------------------------------------------------------------------
@@ -4319,114 +3844,99 @@
// Uses: _ptrEnumerator
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::_GetDefaultDeviceID(EDataFlow dir, ERole role, LPWSTR szBuffer, int bufferLen)
-{
- LOG(LS_VERBOSE) << __FUNCTION__;
+int32_t AudioDeviceWindowsCore::_GetDefaultDeviceID(EDataFlow dir,
+ ERole role,
+ LPWSTR szBuffer,
+ int bufferLen) {
+ LOG(LS_VERBOSE) << __FUNCTION__;
- HRESULT hr = S_OK;
- IMMDevice *pDevice = NULL;
+ HRESULT hr = S_OK;
+ IMMDevice* pDevice = NULL;
- assert(dir == eRender || dir == eCapture);
- assert(role == eConsole || role == eCommunications);
- assert(_ptrEnumerator != NULL);
+ assert(dir == eRender || dir == eCapture);
+ assert(role == eConsole || role == eCommunications);
+ assert(_ptrEnumerator != NULL);
- hr = _ptrEnumerator->GetDefaultAudioEndpoint(
- dir,
- role,
- &pDevice);
+ hr = _ptrEnumerator->GetDefaultAudioEndpoint(dir, role, &pDevice);
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- SAFE_RELEASE(pDevice);
- return -1;
- }
-
- int32_t res = _GetDeviceID(pDevice, szBuffer, bufferLen);
+ if (FAILED(hr)) {
+ _TraceCOMError(hr);
SAFE_RELEASE(pDevice);
- return res;
+ return -1;
+ }
+
+ int32_t res = _GetDeviceID(pDevice, szBuffer, bufferLen);
+ SAFE_RELEASE(pDevice);
+ return res;
}
int32_t AudioDeviceWindowsCore::_GetDefaultDeviceIndex(EDataFlow dir,
ERole role,
- int* index)
-{
- LOG(LS_VERBOSE) << __FUNCTION__;
+ int* index) {
+ LOG(LS_VERBOSE) << __FUNCTION__;
- HRESULT hr = S_OK;
- WCHAR szDefaultDeviceID[MAX_PATH] = {0};
- WCHAR szDeviceID[MAX_PATH] = {0};
+ HRESULT hr = S_OK;
+ WCHAR szDefaultDeviceID[MAX_PATH] = {0};
+ WCHAR szDeviceID[MAX_PATH] = {0};
- const size_t kDeviceIDLength = sizeof(szDeviceID)/sizeof(szDeviceID[0]);
- assert(kDeviceIDLength ==
- sizeof(szDefaultDeviceID) / sizeof(szDefaultDeviceID[0]));
+ const size_t kDeviceIDLength = sizeof(szDeviceID) / sizeof(szDeviceID[0]);
+ assert(kDeviceIDLength ==
+ sizeof(szDefaultDeviceID) / sizeof(szDefaultDeviceID[0]));
- if (_GetDefaultDeviceID(dir,
- role,
- szDefaultDeviceID,
- kDeviceIDLength) == -1)
+ if (_GetDefaultDeviceID(dir, role, szDefaultDeviceID, kDeviceIDLength) ==
+ -1) {
+ return -1;
+ }
+
+ IMMDeviceCollection* collection = _ptrCaptureCollection;
+ if (dir == eRender) {
+ collection = _ptrRenderCollection;
+ }
+
+ if (!collection) {
+ LOG(LS_ERROR) << "Device collection not valid";
+ return -1;
+ }
+
+ UINT count = 0;
+ hr = collection->GetCount(&count);
+ if (FAILED(hr)) {
+ _TraceCOMError(hr);
+ return -1;
+ }
+
+ *index = -1;
+ for (UINT i = 0; i < count; i++) {
+ memset(szDeviceID, 0, sizeof(szDeviceID));
+ rtc::scoped_refptr<IMMDevice> device;
{
- return -1;
- }
-
- IMMDeviceCollection* collection = _ptrCaptureCollection;
- if (dir == eRender)
- {
- collection = _ptrRenderCollection;
- }
-
- if (!collection)
- {
- LOG(LS_ERROR) << "Device collection not valid";
- return -1;
- }
-
- UINT count = 0;
- hr = collection->GetCount(&count);
- if (FAILED(hr))
- {
+ IMMDevice* ptrDevice = NULL;
+ hr = collection->Item(i, &ptrDevice);
+ if (FAILED(hr) || ptrDevice == NULL) {
_TraceCOMError(hr);
return -1;
+ }
+ device = ptrDevice;
+ SAFE_RELEASE(ptrDevice);
}
- *index = -1;
- for (UINT i = 0; i < count; i++)
- {
- memset(szDeviceID, 0, sizeof(szDeviceID));
- rtc::scoped_refptr<IMMDevice> device;
- {
- IMMDevice* ptrDevice = NULL;
- hr = collection->Item(i, &ptrDevice);
- if (FAILED(hr) || ptrDevice == NULL)
- {
- _TraceCOMError(hr);
- return -1;
- }
- device = ptrDevice;
- SAFE_RELEASE(ptrDevice);
- }
-
- if (_GetDeviceID(device, szDeviceID, kDeviceIDLength) == -1)
- {
- return -1;
- }
-
- if (wcsncmp(szDefaultDeviceID, szDeviceID, kDeviceIDLength) == 0)
- {
- // Found a match.
- *index = i;
- break;
- }
-
+ if (_GetDeviceID(device, szDeviceID, kDeviceIDLength) == -1) {
+ return -1;
}
- if (*index == -1)
- {
- LOG(LS_ERROR) << "Unable to find collection index for default device";
- return -1;
+ if (wcsncmp(szDefaultDeviceID, szDeviceID, kDeviceIDLength) == 0) {
+ // Found a match.
+ *index = i;
+ break;
}
+ }
- return 0;
+ if (*index == -1) {
+ LOG(LS_ERROR) << "Unable to find collection index for default device";
+ return -1;
+ }
+
+ return 0;
}
// ----------------------------------------------------------------------------
@@ -4435,401 +3945,363 @@
int32_t AudioDeviceWindowsCore::_GetDeviceName(IMMDevice* pDevice,
LPWSTR pszBuffer,
- int bufferLen)
-{
- LOG(LS_VERBOSE) << __FUNCTION__;
+ int bufferLen) {
+ LOG(LS_VERBOSE) << __FUNCTION__;
- static const WCHAR szDefault[] = L"<Device not available>";
+ static const WCHAR szDefault[] = L"<Device not available>";
- HRESULT hr = E_FAIL;
- IPropertyStore *pProps = NULL;
- PROPVARIANT varName;
+ HRESULT hr = E_FAIL;
+ IPropertyStore* pProps = NULL;
+ PROPVARIANT varName;
- assert(pszBuffer != NULL);
- assert(bufferLen > 0);
+ assert(pszBuffer != NULL);
+ assert(bufferLen > 0);
- if (pDevice != NULL)
- {
- hr = pDevice->OpenPropertyStore(STGM_READ, &pProps);
- if (FAILED(hr))
- {
- LOG(LS_ERROR) << "IMMDevice::OpenPropertyStore failed, hr = 0x"
- << std::hex << hr << std::dec;
- }
+ if (pDevice != NULL) {
+ hr = pDevice->OpenPropertyStore(STGM_READ, &pProps);
+ if (FAILED(hr)) {
+ LOG(LS_ERROR) << "IMMDevice::OpenPropertyStore failed, hr = 0x"
+ << std::hex << hr << std::dec;
}
+ }
- // Initialize container for property value.
- PropVariantInit(&varName);
+ // Initialize container for property value.
+ PropVariantInit(&varName);
- if (SUCCEEDED(hr))
- {
- // Get the endpoint device's friendly-name property.
- hr = pProps->GetValue(PKEY_Device_FriendlyName, &varName);
- if (FAILED(hr))
- {
- LOG(LS_ERROR) << "IPropertyStore::GetValue failed, hr = 0x"
- << std::hex << hr << std::dec;
- }
+ if (SUCCEEDED(hr)) {
+ // Get the endpoint device's friendly-name property.
+ hr = pProps->GetValue(PKEY_Device_FriendlyName, &varName);
+ if (FAILED(hr)) {
+ LOG(LS_ERROR) << "IPropertyStore::GetValue failed, hr = 0x" << std::hex
+ << hr << std::dec;
}
+ }
- if ((SUCCEEDED(hr)) && (VT_EMPTY == varName.vt))
- {
- hr = E_FAIL;
- LOG(LS_ERROR) << "IPropertyStore::GetValue returned no value,"
- << " hr = 0x" << std::hex << hr << std::dec;
- }
+ if ((SUCCEEDED(hr)) && (VT_EMPTY == varName.vt)) {
+ hr = E_FAIL;
+ LOG(LS_ERROR) << "IPropertyStore::GetValue returned no value,"
+ << " hr = 0x" << std::hex << hr << std::dec;
+ }
- if ((SUCCEEDED(hr)) && (VT_LPWSTR != varName.vt))
- {
- // The returned value is not a wide null terminated string.
- hr = E_UNEXPECTED;
- LOG(LS_ERROR) << "IPropertyStore::GetValue returned unexpected"
- << " type, hr = 0x" << std::hex << hr << std::dec;
- }
+ if ((SUCCEEDED(hr)) && (VT_LPWSTR != varName.vt)) {
+ // The returned value is not a wide null terminated string.
+ hr = E_UNEXPECTED;
+ LOG(LS_ERROR) << "IPropertyStore::GetValue returned unexpected"
+ << " type, hr = 0x" << std::hex << hr << std::dec;
+ }
- if (SUCCEEDED(hr) && (varName.pwszVal != NULL))
- {
- // Copy the valid device name to the provided ouput buffer.
- wcsncpy_s(pszBuffer, bufferLen, varName.pwszVal, _TRUNCATE);
- }
- else
- {
- // Failed to find the device name.
- wcsncpy_s(pszBuffer, bufferLen, szDefault, _TRUNCATE);
- }
+ if (SUCCEEDED(hr) && (varName.pwszVal != NULL)) {
+ // Copy the valid device name to the provided ouput buffer.
+ wcsncpy_s(pszBuffer, bufferLen, varName.pwszVal, _TRUNCATE);
+ } else {
+ // Failed to find the device name.
+ wcsncpy_s(pszBuffer, bufferLen, szDefault, _TRUNCATE);
+ }
- PropVariantClear(&varName);
- SAFE_RELEASE(pProps);
+ PropVariantClear(&varName);
+ SAFE_RELEASE(pProps);
- return 0;
+ return 0;
}
// ----------------------------------------------------------------------------
// _GetDeviceID
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::_GetDeviceID(IMMDevice* pDevice, LPWSTR pszBuffer, int bufferLen)
-{
- LOG(LS_VERBOSE) << __FUNCTION__;
+int32_t AudioDeviceWindowsCore::_GetDeviceID(IMMDevice* pDevice,
+ LPWSTR pszBuffer,
+ int bufferLen) {
+ LOG(LS_VERBOSE) << __FUNCTION__;
- static const WCHAR szDefault[] = L"<Device not available>";
+ static const WCHAR szDefault[] = L"<Device not available>";
- HRESULT hr = E_FAIL;
- LPWSTR pwszID = NULL;
+ HRESULT hr = E_FAIL;
+ LPWSTR pwszID = NULL;
- assert(pszBuffer != NULL);
- assert(bufferLen > 0);
+ assert(pszBuffer != NULL);
+ assert(bufferLen > 0);
- if (pDevice != NULL)
- {
- hr = pDevice->GetId(&pwszID);
- }
+ if (pDevice != NULL) {
+ hr = pDevice->GetId(&pwszID);
+ }
- if (hr == S_OK)
- {
- // Found the device ID.
- wcsncpy_s(pszBuffer, bufferLen, pwszID, _TRUNCATE);
- }
- else
- {
- // Failed to find the device ID.
- wcsncpy_s(pszBuffer, bufferLen, szDefault, _TRUNCATE);
- }
+ if (hr == S_OK) {
+ // Found the device ID.
+ wcsncpy_s(pszBuffer, bufferLen, pwszID, _TRUNCATE);
+ } else {
+ // Failed to find the device ID.
+ wcsncpy_s(pszBuffer, bufferLen, szDefault, _TRUNCATE);
+ }
- CoTaskMemFree(pwszID);
- return 0;
+ CoTaskMemFree(pwszID);
+ return 0;
}
// ----------------------------------------------------------------------------
// _GetDefaultDevice
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::_GetDefaultDevice(EDataFlow dir, ERole role, IMMDevice** ppDevice)
-{
- LOG(LS_VERBOSE) << __FUNCTION__;
+int32_t AudioDeviceWindowsCore::_GetDefaultDevice(EDataFlow dir,
+ ERole role,
+ IMMDevice** ppDevice) {
+ LOG(LS_VERBOSE) << __FUNCTION__;
- HRESULT hr(S_OK);
+ HRESULT hr(S_OK);
- assert(_ptrEnumerator != NULL);
+ assert(_ptrEnumerator != NULL);
- hr = _ptrEnumerator->GetDefaultAudioEndpoint(
- dir,
- role,
- ppDevice);
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- return -1;
- }
+ hr = _ptrEnumerator->GetDefaultAudioEndpoint(dir, role, ppDevice);
+ if (FAILED(hr)) {
+ _TraceCOMError(hr);
+ return -1;
+ }
- return 0;
+ return 0;
}
// ----------------------------------------------------------------------------
// _GetListDevice
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::_GetListDevice(EDataFlow dir, int index, IMMDevice** ppDevice)
-{
- HRESULT hr(S_OK);
+int32_t AudioDeviceWindowsCore::_GetListDevice(EDataFlow dir,
+ int index,
+ IMMDevice** ppDevice) {
+ HRESULT hr(S_OK);
- assert(_ptrEnumerator != NULL);
+ assert(_ptrEnumerator != NULL);
- IMMDeviceCollection *pCollection = NULL;
+ IMMDeviceCollection* pCollection = NULL;
- hr = _ptrEnumerator->EnumAudioEndpoints(
- dir,
- DEVICE_STATE_ACTIVE, // only active endpoints are OK
- &pCollection);
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- SAFE_RELEASE(pCollection);
- return -1;
- }
+ hr = _ptrEnumerator->EnumAudioEndpoints(
+ dir,
+ DEVICE_STATE_ACTIVE, // only active endpoints are OK
+ &pCollection);
+ if (FAILED(hr)) {
+ _TraceCOMError(hr);
+ SAFE_RELEASE(pCollection);
+ return -1;
+ }
- hr = pCollection->Item(
- index,
- ppDevice);
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- SAFE_RELEASE(pCollection);
- return -1;
- }
+ hr = pCollection->Item(index, ppDevice);
+ if (FAILED(hr)) {
+ _TraceCOMError(hr);
+ SAFE_RELEASE(pCollection);
+ return -1;
+ }
- return 0;
+ return 0;
}
// ----------------------------------------------------------------------------
// _EnumerateEndpointDevicesAll
// ----------------------------------------------------------------------------
-int32_t AudioDeviceWindowsCore::_EnumerateEndpointDevicesAll(EDataFlow dataFlow) const
-{
- LOG(LS_VERBOSE) << __FUNCTION__;
+int32_t AudioDeviceWindowsCore::_EnumerateEndpointDevicesAll(
+ EDataFlow dataFlow) const {
+ LOG(LS_VERBOSE) << __FUNCTION__;
- assert(_ptrEnumerator != NULL);
+ assert(_ptrEnumerator != NULL);
- HRESULT hr = S_OK;
- IMMDeviceCollection *pCollection = NULL;
- IMMDevice *pEndpoint = NULL;
- IPropertyStore *pProps = NULL;
- IAudioEndpointVolume* pEndpointVolume = NULL;
- LPWSTR pwszID = NULL;
+ HRESULT hr = S_OK;
+ IMMDeviceCollection* pCollection = NULL;
+ IMMDevice* pEndpoint = NULL;
+ IPropertyStore* pProps = NULL;
+ IAudioEndpointVolume* pEndpointVolume = NULL;
+ LPWSTR pwszID = NULL;
- // Generate a collection of audio endpoint devices in the system.
- // Get states for *all* endpoint devices.
- // Output: IMMDeviceCollection interface.
- hr = _ptrEnumerator->EnumAudioEndpoints(
- dataFlow, // data-flow direction (input parameter)
- DEVICE_STATE_ACTIVE | DEVICE_STATE_DISABLED | DEVICE_STATE_UNPLUGGED,
- &pCollection); // release interface when done
+ // Generate a collection of audio endpoint devices in the system.
+ // Get states for *all* endpoint devices.
+ // Output: IMMDeviceCollection interface.
+ hr = _ptrEnumerator->EnumAudioEndpoints(
+ dataFlow, // data-flow direction (input parameter)
+ DEVICE_STATE_ACTIVE | DEVICE_STATE_DISABLED | DEVICE_STATE_UNPLUGGED,
+ &pCollection); // release interface when done
- EXIT_ON_ERROR(hr);
+ EXIT_ON_ERROR(hr);
- // use the IMMDeviceCollection interface...
+ // use the IMMDeviceCollection interface...
- UINT count = 0;
+ UINT count = 0;
- // Retrieve a count of the devices in the device collection.
- hr = pCollection->GetCount(&count);
- EXIT_ON_ERROR(hr);
- if (dataFlow == eRender)
- LOG(LS_VERBOSE) << "#rendering endpoint devices (counting all): "
- << count;
- else if (dataFlow == eCapture)
- LOG(LS_VERBOSE) << "#capturing endpoint devices (counting all): "
- << count;
+ // Retrieve a count of the devices in the device collection.
+ hr = pCollection->GetCount(&count);
+ EXIT_ON_ERROR(hr);
+ if (dataFlow == eRender)
+ LOG(LS_VERBOSE) << "#rendering endpoint devices (counting all): " << count;
+ else if (dataFlow == eCapture)
+ LOG(LS_VERBOSE) << "#capturing endpoint devices (counting all): " << count;
- if (count == 0)
- {
- return 0;
- }
-
- // Each loop prints the name of an endpoint device.
- for (ULONG i = 0; i < count; i++)
- {
- LOG(LS_VERBOSE) << "Endpoint " << i << ":";
-
- // Get pointer to endpoint number i.
- // Output: IMMDevice interface.
- hr = pCollection->Item(
- i,
- &pEndpoint);
- CONTINUE_ON_ERROR(hr);
-
- // use the IMMDevice interface of the specified endpoint device...
-
- // Get the endpoint ID string (uniquely identifies the device among all audio endpoint devices)
- hr = pEndpoint->GetId(&pwszID);
- CONTINUE_ON_ERROR(hr);
- LOG(LS_VERBOSE) << "ID string : " << pwszID;
-
- // Retrieve an interface to the device's property store.
- // Output: IPropertyStore interface.
- hr = pEndpoint->OpenPropertyStore(
- STGM_READ,
- &pProps);
- CONTINUE_ON_ERROR(hr);
-
- // use the IPropertyStore interface...
-
- PROPVARIANT varName;
- // Initialize container for property value.
- PropVariantInit(&varName);
-
- // Get the endpoint's friendly-name property.
- // Example: "Speakers (Realtek High Definition Audio)"
- hr = pProps->GetValue(
- PKEY_Device_FriendlyName,
- &varName);
- CONTINUE_ON_ERROR(hr);
- LOG(LS_VERBOSE) << "friendly name: \"" << varName.pwszVal << "\"";
-
- // Get the endpoint's current device state
- DWORD dwState;
- hr = pEndpoint->GetState(&dwState);
- CONTINUE_ON_ERROR(hr);
- if (dwState & DEVICE_STATE_ACTIVE)
- LOG(LS_VERBOSE) << "state (0x" << std::hex << dwState << std::dec
- << ") : *ACTIVE*";
- if (dwState & DEVICE_STATE_DISABLED)
- LOG(LS_VERBOSE) << "state (0x" << std::hex << dwState << std::dec
- << ") : DISABLED";
- if (dwState & DEVICE_STATE_NOTPRESENT)
- LOG(LS_VERBOSE) << "state (0x" << std::hex << dwState << std::dec
- << ") : NOTPRESENT";
- if (dwState & DEVICE_STATE_UNPLUGGED)
- LOG(LS_VERBOSE) << "state (0x" << std::hex << dwState << std::dec
- << ") : UNPLUGGED";
-
- // Check the hardware volume capabilities.
- DWORD dwHwSupportMask = 0;
- hr = pEndpoint->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL,
- NULL, (void**)&pEndpointVolume);
- CONTINUE_ON_ERROR(hr);
- hr = pEndpointVolume->QueryHardwareSupport(&dwHwSupportMask);
- CONTINUE_ON_ERROR(hr);
- if (dwHwSupportMask & ENDPOINT_HARDWARE_SUPPORT_VOLUME)
- // The audio endpoint device supports a hardware volume control
- LOG(LS_VERBOSE) << "hwmask (0x" << std::hex << dwHwSupportMask
- << std::dec << ") : HARDWARE_SUPPORT_VOLUME";
- if (dwHwSupportMask & ENDPOINT_HARDWARE_SUPPORT_MUTE)
- // The audio endpoint device supports a hardware mute control
- LOG(LS_VERBOSE) << "hwmask (0x" << std::hex << dwHwSupportMask
- << std::dec << ") : HARDWARE_SUPPORT_MUTE";
- if (dwHwSupportMask & ENDPOINT_HARDWARE_SUPPORT_METER)
- // The audio endpoint device supports a hardware peak meter
- LOG(LS_VERBOSE) << "hwmask (0x" << std::hex << dwHwSupportMask
- << std::dec << ") : HARDWARE_SUPPORT_METER";
-
- // Check the channel count (#channels in the audio stream that enters or leaves the audio endpoint device)
- UINT nChannelCount(0);
- hr = pEndpointVolume->GetChannelCount(
- &nChannelCount);
- CONTINUE_ON_ERROR(hr);
- LOG(LS_VERBOSE) << "#channels : " << nChannelCount;
-
- if (dwHwSupportMask & ENDPOINT_HARDWARE_SUPPORT_VOLUME)
- {
- // Get the volume range.
- float fLevelMinDB(0.0);
- float fLevelMaxDB(0.0);
- float fVolumeIncrementDB(0.0);
- hr = pEndpointVolume->GetVolumeRange(
- &fLevelMinDB,
- &fLevelMaxDB,
- &fVolumeIncrementDB);
- CONTINUE_ON_ERROR(hr);
- LOG(LS_VERBOSE) << "volume range : " << fLevelMinDB << " (min), "
- << fLevelMaxDB << " (max), " << fVolumeIncrementDB
- << " (inc) [dB]";
-
- // The volume range from vmin = fLevelMinDB to vmax = fLevelMaxDB is divided
- // into n uniform intervals of size vinc = fVolumeIncrementDB, where
- // n = (vmax ?vmin) / vinc.
- // The values vmin, vmax, and vinc are measured in decibels. The client can set
- // the volume level to one of n + 1 discrete values in the range from vmin to vmax.
- int n = (int)((fLevelMaxDB-fLevelMinDB)/fVolumeIncrementDB);
- LOG(LS_VERBOSE) << "#intervals : " << n;
-
- // Get information about the current step in the volume range.
- // This method represents the volume level of the audio stream that enters or leaves
- // the audio endpoint device as an index or "step" in a range of discrete volume levels.
- // Output value nStepCount is the number of steps in the range. Output value nStep
- // is the step index of the current volume level. If the number of steps is n = nStepCount,
- // then step index nStep can assume values from 0 (minimum volume) to n ?1 (maximum volume).
- UINT nStep(0);
- UINT nStepCount(0);
- hr = pEndpointVolume->GetVolumeStepInfo(
- &nStep,
- &nStepCount);
- CONTINUE_ON_ERROR(hr);
- LOG(LS_VERBOSE) << "volume steps : " << nStep << " (nStep), "
- << nStepCount << " (nStepCount)";
- }
-Next:
- if (FAILED(hr)) {
- LOG(LS_VERBOSE) << "Error when logging device information";
- }
- CoTaskMemFree(pwszID);
- pwszID = NULL;
- PropVariantClear(&varName);
- SAFE_RELEASE(pProps);
- SAFE_RELEASE(pEndpoint);
- SAFE_RELEASE(pEndpointVolume);
- }
- SAFE_RELEASE(pCollection);
+ if (count == 0) {
return 0;
+ }
-Exit:
- _TraceCOMError(hr);
+ // Each loop prints the name of an endpoint device.
+ for (ULONG i = 0; i < count; i++) {
+ LOG(LS_VERBOSE) << "Endpoint " << i << ":";
+
+ // Get pointer to endpoint number i.
+ // Output: IMMDevice interface.
+ hr = pCollection->Item(i, &pEndpoint);
+ CONTINUE_ON_ERROR(hr);
+
+ // use the IMMDevice interface of the specified endpoint device...
+
+ // Get the endpoint ID string (uniquely identifies the device among all
+ // audio endpoint devices)
+ hr = pEndpoint->GetId(&pwszID);
+ CONTINUE_ON_ERROR(hr);
+ LOG(LS_VERBOSE) << "ID string : " << pwszID;
+
+ // Retrieve an interface to the device's property store.
+ // Output: IPropertyStore interface.
+ hr = pEndpoint->OpenPropertyStore(STGM_READ, &pProps);
+ CONTINUE_ON_ERROR(hr);
+
+ // use the IPropertyStore interface...
+
+ PROPVARIANT varName;
+ // Initialize container for property value.
+ PropVariantInit(&varName);
+
+ // Get the endpoint's friendly-name property.
+ // Example: "Speakers (Realtek High Definition Audio)"
+ hr = pProps->GetValue(PKEY_Device_FriendlyName, &varName);
+ CONTINUE_ON_ERROR(hr);
+ LOG(LS_VERBOSE) << "friendly name: \"" << varName.pwszVal << "\"";
+
+ // Get the endpoint's current device state
+ DWORD dwState;
+ hr = pEndpoint->GetState(&dwState);
+ CONTINUE_ON_ERROR(hr);
+ if (dwState & DEVICE_STATE_ACTIVE)
+ LOG(LS_VERBOSE) << "state (0x" << std::hex << dwState << std::dec
+ << ") : *ACTIVE*";
+ if (dwState & DEVICE_STATE_DISABLED)
+ LOG(LS_VERBOSE) << "state (0x" << std::hex << dwState << std::dec
+ << ") : DISABLED";
+ if (dwState & DEVICE_STATE_NOTPRESENT)
+ LOG(LS_VERBOSE) << "state (0x" << std::hex << dwState << std::dec
+ << ") : NOTPRESENT";
+ if (dwState & DEVICE_STATE_UNPLUGGED)
+ LOG(LS_VERBOSE) << "state (0x" << std::hex << dwState << std::dec
+ << ") : UNPLUGGED";
+
+ // Check the hardware volume capabilities.
+ DWORD dwHwSupportMask = 0;
+ hr = pEndpoint->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL, NULL,
+ (void**)&pEndpointVolume);
+ CONTINUE_ON_ERROR(hr);
+ hr = pEndpointVolume->QueryHardwareSupport(&dwHwSupportMask);
+ CONTINUE_ON_ERROR(hr);
+ if (dwHwSupportMask & ENDPOINT_HARDWARE_SUPPORT_VOLUME)
+ // The audio endpoint device supports a hardware volume control
+ LOG(LS_VERBOSE) << "hwmask (0x" << std::hex << dwHwSupportMask << std::dec
+ << ") : HARDWARE_SUPPORT_VOLUME";
+ if (dwHwSupportMask & ENDPOINT_HARDWARE_SUPPORT_MUTE)
+ // The audio endpoint device supports a hardware mute control
+ LOG(LS_VERBOSE) << "hwmask (0x" << std::hex << dwHwSupportMask << std::dec
+ << ") : HARDWARE_SUPPORT_MUTE";
+ if (dwHwSupportMask & ENDPOINT_HARDWARE_SUPPORT_METER)
+ // The audio endpoint device supports a hardware peak meter
+ LOG(LS_VERBOSE) << "hwmask (0x" << std::hex << dwHwSupportMask << std::dec
+ << ") : HARDWARE_SUPPORT_METER";
+
+ // Check the channel count (#channels in the audio stream that enters or
+ // leaves the audio endpoint device)
+ UINT nChannelCount(0);
+ hr = pEndpointVolume->GetChannelCount(&nChannelCount);
+ CONTINUE_ON_ERROR(hr);
+ LOG(LS_VERBOSE) << "#channels : " << nChannelCount;
+
+ if (dwHwSupportMask & ENDPOINT_HARDWARE_SUPPORT_VOLUME) {
+ // Get the volume range.
+ float fLevelMinDB(0.0);
+ float fLevelMaxDB(0.0);
+ float fVolumeIncrementDB(0.0);
+ hr = pEndpointVolume->GetVolumeRange(&fLevelMinDB, &fLevelMaxDB,
+ &fVolumeIncrementDB);
+ CONTINUE_ON_ERROR(hr);
+ LOG(LS_VERBOSE) << "volume range : " << fLevelMinDB << " (min), "
+ << fLevelMaxDB << " (max), " << fVolumeIncrementDB
+ << " (inc) [dB]";
+
+ // The volume range from vmin = fLevelMinDB to vmax = fLevelMaxDB is
+ // divided into n uniform intervals of size vinc = fVolumeIncrementDB,
+ // where n = (vmax ?vmin) / vinc. The values vmin, vmax, and vinc are
+ // measured in decibels. The client can set the volume level to one of n +
+ // 1 discrete values in the range from vmin to vmax.
+ int n = (int)((fLevelMaxDB - fLevelMinDB) / fVolumeIncrementDB);
+ LOG(LS_VERBOSE) << "#intervals : " << n;
+
+ // Get information about the current step in the volume range.
+ // This method represents the volume level of the audio stream that enters
+ // or leaves the audio endpoint device as an index or "step" in a range of
+ // discrete volume levels. Output value nStepCount is the number of steps
+ // in the range. Output value nStep is the step index of the current
+ // volume level. If the number of steps is n = nStepCount, then step index
+ // nStep can assume values from 0 (minimum volume) to n ?1 (maximum
+ // volume).
+ UINT nStep(0);
+ UINT nStepCount(0);
+ hr = pEndpointVolume->GetVolumeStepInfo(&nStep, &nStepCount);
+ CONTINUE_ON_ERROR(hr);
+ LOG(LS_VERBOSE) << "volume steps : " << nStep << " (nStep), "
+ << nStepCount << " (nStepCount)";
+ }
+ Next:
+ if (FAILED(hr)) {
+ LOG(LS_VERBOSE) << "Error when logging device information";
+ }
CoTaskMemFree(pwszID);
pwszID = NULL;
- SAFE_RELEASE(pCollection);
+ PropVariantClear(&varName);
+ SAFE_RELEASE(pProps);
SAFE_RELEASE(pEndpoint);
SAFE_RELEASE(pEndpointVolume);
- SAFE_RELEASE(pProps);
- return -1;
+ }
+ SAFE_RELEASE(pCollection);
+ return 0;
+
+Exit:
+ _TraceCOMError(hr);
+ CoTaskMemFree(pwszID);
+ pwszID = NULL;
+ SAFE_RELEASE(pCollection);
+ SAFE_RELEASE(pEndpoint);
+ SAFE_RELEASE(pEndpointVolume);
+ SAFE_RELEASE(pProps);
+ return -1;
}
// ----------------------------------------------------------------------------
// _TraceCOMError
// ----------------------------------------------------------------------------
-void AudioDeviceWindowsCore::_TraceCOMError(HRESULT hr) const
-{
- TCHAR buf[MAXERRORLENGTH];
- TCHAR errorText[MAXERRORLENGTH];
+void AudioDeviceWindowsCore::_TraceCOMError(HRESULT hr) const {
+ TCHAR buf[MAXERRORLENGTH];
+ TCHAR errorText[MAXERRORLENGTH];
- const DWORD dwFlags = FORMAT_MESSAGE_FROM_SYSTEM |
- FORMAT_MESSAGE_IGNORE_INSERTS;
- const DWORD dwLangID = MAKELANGID(LANG_ENGLISH, SUBLANG_ENGLISH_US);
+ const DWORD dwFlags =
+ FORMAT_MESSAGE_FROM_SYSTEM | FORMAT_MESSAGE_IGNORE_INSERTS;
+ const DWORD dwLangID = MAKELANGID(LANG_ENGLISH, SUBLANG_ENGLISH_US);
- // Gets the system's human readable message string for this HRESULT.
- // All error message in English by default.
- DWORD messageLength = ::FormatMessageW(dwFlags,
- 0,
- hr,
- dwLangID,
- errorText,
- MAXERRORLENGTH,
- NULL);
+ // Gets the system's human readable message string for this HRESULT.
+ // All error message in English by default.
+ DWORD messageLength = ::FormatMessageW(dwFlags, 0, hr, dwLangID, errorText,
+ MAXERRORLENGTH, NULL);
- assert(messageLength <= MAXERRORLENGTH);
+ assert(messageLength <= MAXERRORLENGTH);
- // Trims tailing white space (FormatMessage() leaves a trailing cr-lf.).
- for (; messageLength && ::isspace(errorText[messageLength - 1]);
- --messageLength)
- {
- errorText[messageLength - 1] = '\0';
- }
+ // Trims tailing white space (FormatMessage() leaves a trailing cr-lf.).
+ for (; messageLength && ::isspace(errorText[messageLength - 1]);
+ --messageLength) {
+ errorText[messageLength - 1] = '\0';
+ }
- LOG(LS_ERROR) << "Core Audio method failed (hr=" << hr << ")";
- StringCchPrintf(buf, MAXERRORLENGTH, TEXT("Error details: "));
- StringCchCat(buf, MAXERRORLENGTH, errorText);
- LOG(LS_ERROR) << WideToUTF8(buf);
+ LOG(LS_ERROR) << "Core Audio method failed (hr=" << hr << ")";
+ StringCchPrintf(buf, MAXERRORLENGTH, TEXT("Error details: "));
+ StringCchCat(buf, MAXERRORLENGTH, errorText);
+ LOG(LS_ERROR) << WideToUTF8(buf);
}
// ----------------------------------------------------------------------------
@@ -4838,29 +4310,27 @@
char* AudioDeviceWindowsCore::WideToUTF8(const TCHAR* src) const {
#ifdef UNICODE
- const size_t kStrLen = sizeof(_str);
- memset(_str, 0, kStrLen);
- // Get required size (in bytes) to be able to complete the conversion.
- unsigned int required_size = (unsigned int)WideCharToMultiByte(CP_UTF8, 0, src, -1, _str, 0, 0, 0);
- if (required_size <= kStrLen)
- {
- // Process the entire input string, including the terminating null char.
- if (WideCharToMultiByte(CP_UTF8, 0, src, -1, _str, kStrLen, 0, 0) == 0)
- memset(_str, 0, kStrLen);
- }
- return _str;
+ const size_t kStrLen = sizeof(_str);
+ memset(_str, 0, kStrLen);
+ // Get required size (in bytes) to be able to complete the conversion.
+ unsigned int required_size =
+ (unsigned int)WideCharToMultiByte(CP_UTF8, 0, src, -1, _str, 0, 0, 0);
+ if (required_size <= kStrLen) {
+ // Process the entire input string, including the terminating null char.
+ if (WideCharToMultiByte(CP_UTF8, 0, src, -1, _str, kStrLen, 0, 0) == 0)
+ memset(_str, 0, kStrLen);
+ }
+ return _str;
#else
- return const_cast<char*>(src);
+ return const_cast<char*>(src);
#endif
}
-
-bool AudioDeviceWindowsCore::KeyPressed() const{
-
+bool AudioDeviceWindowsCore::KeyPressed() const {
int key_down = 0;
for (int key = VK_SPACE; key < VK_NUMLOCK; key++) {
short res = GetAsyncKeyState(key);
- key_down |= res & 0x1; // Get the LSB
+ key_down |= res & 0x1; // Get the LSB
}
return (key_down > 0);
}