audio/media: convert to using the audio HAL and new audio defs
Change-Id: Ibc637918637329e4f2b62f4ac7781102fbc269f5
Signed-off-by: Dima Zavin <dima@android.com>
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 04cfa08..301f467 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -33,6 +33,7 @@
#include <utils/threads.h>
#include <utils/Atomic.h>
+#include <cutils/bitops.h>
#include <cutils/properties.h>
#include <media/AudioTrack.h>
@@ -41,15 +42,13 @@
#include <private/media/AudioTrackShared.h>
#include <private/media/AudioEffectShared.h>
-#include <hardware_legacy/AudioHardwareInterface.h>
+
+#include <hardware/audio.h>
+#include <hardware/audio_hal.h>
#include "AudioMixer.h"
#include "AudioFlinger.h"
-#ifdef WITH_A2DP
-#include "A2dpAudioInterface.h"
-#endif
-
#include <media/EffectsFactoryApi.h>
#include <media/EffectVisualizerApi.h>
@@ -141,22 +140,37 @@
AudioFlinger::AudioFlinger()
: BnAudioFlinger(),
- mAudioHardware(0), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1)
+ mAudioHardwareDev(0), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1)
{
+ const hw_module_t *module;
+ int rc;
+ char mod_name[PATH_MAX];
+
Mutex::Autolock _l(mLock);
mHardwareStatus = AUDIO_HW_IDLE;
- mAudioHardware = AudioHardwareInterface::create();
+ /* get the audio hw module and create an audio_hw device */
+ snprintf(mod_name, PATH_MAX, "%s.%s", AUDIO_HARDWARE_MODULE_ID, "primary");
+ rc = hw_get_module(mod_name, &module);
+ if (rc)
+ return;
+
+ rc = audio_hw_device_open(module, &mAudioHardwareDev);
+ LOGE_IF(rc, "couldn't open audio hw device (%s)", strerror(-rc));
+ if (rc)
+ return;
mHardwareStatus = AUDIO_HW_INIT;
- if (mAudioHardware->initCheck() == NO_ERROR) {
+
+ rc = mAudioHardwareDev->init_check(mAudioHardwareDev);
+ if (rc == 0) {
AutoMutex lock(mHardwareLock);
- mMode = AudioSystem::MODE_NORMAL;
+ mMode = AUDIO_MODE_NORMAL;
mHardwareStatus = AUDIO_HW_SET_MODE;
- mAudioHardware->setMode(mMode);
+ mAudioHardwareDev->set_mode(mAudioHardwareDev, mMode);
mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
- mAudioHardware->setMasterVolume(1.0f);
+ mAudioHardwareDev->set_master_volume(mAudioHardwareDev, 1.0f);
mHardwareStatus = AUDIO_HW_IDLE;
} else {
LOGE("Couldn't even initialize the stubbed audio hardware!");
@@ -173,8 +187,8 @@
// closeOutput() will remove first entry from mPlaybackThreads
closeOutput(mPlaybackThreads.keyAt(0));
}
- if (mAudioHardware) {
- delete mAudioHardware;
+ if (mAudioHardwareDev) {
+ audio_hw_device_close(mAudioHardwareDev);
}
}
@@ -277,8 +291,8 @@
mRecordThreads.valueAt(i)->dump(fd, args);
}
- if (mAudioHardware) {
- mAudioHardware->dumpState(fd, args);
+ if (mAudioHardwareDev) {
+ mAudioHardwareDev->dump(mAudioHardwareDev, fd);
}
if (locked) mLock.unlock();
}
@@ -309,7 +323,7 @@
status_t lStatus;
int lSessionId;
- if (streamType >= AudioSystem::NUM_STREAM_TYPES) {
+ if (streamType >= AUDIO_STREAM_CNT) {
LOGE("invalid stream type");
lStatus = BAD_VALUE;
goto Exit;
@@ -335,7 +349,7 @@
}
LOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId);
- if (sessionId != NULL && *sessionId != AudioSystem::SESSION_OUTPUT_MIX) {
+ if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
if (mPlaybackThreads.keyAt(i) != output) {
@@ -454,7 +468,7 @@
{ // scope for the lock
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
- if (mAudioHardware->setMasterVolume(value) == NO_ERROR) {
+ if (mAudioHardwareDev->set_master_volume(mAudioHardwareDev, value) == NO_ERROR) {
value = 1.0f;
}
mHardwareStatus = AUDIO_HW_IDLE;
@@ -476,7 +490,7 @@
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
- if ((mode < 0) || (mode >= AudioSystem::NUM_MODES)) {
+ if ((mode < 0) || (mode >= AUDIO_MODE_CNT)) {
LOGW("Illegal value: setMode(%d)", mode);
return BAD_VALUE;
}
@@ -484,7 +498,7 @@
{ // scope for the lock
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_SET_MODE;
- ret = mAudioHardware->setMode(mode);
+ ret = mAudioHardwareDev->set_mode(mAudioHardwareDev, mode);
mHardwareStatus = AUDIO_HW_IDLE;
}
@@ -507,16 +521,16 @@
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
- status_t ret = mAudioHardware->setMicMute(state);
+ status_t ret = mAudioHardwareDev->set_mic_mute(mAudioHardwareDev, state);
mHardwareStatus = AUDIO_HW_IDLE;
return ret;
}
bool AudioFlinger::getMicMute() const
{
- bool state = AudioSystem::MODE_INVALID;
+ bool state = AUDIO_MODE_INVALID;
mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
- mAudioHardware->getMicMute(&state);
+ mAudioHardwareDev->get_mic_mute(mAudioHardwareDev, &state);
mHardwareStatus = AUDIO_HW_IDLE;
return state;
}
@@ -553,7 +567,7 @@
return PERMISSION_DENIED;
}
- if (stream < 0 || uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) {
+ if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT) {
return BAD_VALUE;
}
@@ -586,8 +600,8 @@
return PERMISSION_DENIED;
}
- if (stream < 0 || uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES ||
- uint32_t(stream) == AudioSystem::ENFORCED_AUDIBLE) {
+ if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT ||
+ uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) {
return BAD_VALUE;
}
@@ -601,7 +615,7 @@
float AudioFlinger::streamVolume(int stream, int output) const
{
- if (stream < 0 || uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) {
+ if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT) {
return 0.0f;
}
@@ -622,7 +636,7 @@
bool AudioFlinger::streamMute(int stream) const
{
- if (stream < 0 || stream >= (int)AudioSystem::NUM_STREAM_TYPES) {
+ if (stream < 0 || stream >= (int)AUDIO_STREAM_CNT) {
return true;
}
@@ -644,7 +658,7 @@
if (ioHandle == 0) {
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_SET_PARAMETER;
- result = mAudioHardware->setParameters(keyValuePairs);
+ result = mAudioHardwareDev->set_parameters(mAudioHardwareDev, keyValuePairs.string());
mHardwareStatus = AUDIO_HW_IDLE;
return result;
}
@@ -672,7 +686,13 @@
// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid());
if (ioHandle == 0) {
- return mAudioHardware->getParameters(keys);
+ char *s;
+ String8 out_s8;
+
+ s = mAudioHardwareDev->get_parameters(mAudioHardwareDev, keys.string());
+ out_s8 = String8(s);
+ free(s);
+ return out_s8;
}
Mutex::Autolock _l(mLock);
@@ -690,7 +710,7 @@
size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
{
- return mAudioHardware->getInputBufferSize(sampleRate, format, channelCount);
+ return mAudioHardwareDev->get_input_buffer_size(mAudioHardwareDev, sampleRate, format, channelCount);
}
unsigned int AudioFlinger::getInputFramesLost(int ioHandle)
@@ -717,7 +737,7 @@
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_SET_VOICE_VOLUME;
- status_t ret = mAudioHardware->setVoiceVolume(value);
+ status_t ret = mAudioHardwareDev->set_voice_volume(mAudioHardwareDev, value);
mHardwareStatus = AUDIO_HW_IDLE;
return ret;
@@ -957,7 +977,7 @@
// ----------------------------------------------------------------------------
-AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
+AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, struct audio_stream_out* output, int id, uint32_t device)
: ThreadBase(audioFlinger, id),
mMixBuffer(0), mSuspended(0), mBytesWritten(0), mOutput(output),
mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false),
@@ -968,7 +988,7 @@
mMasterVolume = mAudioFlinger->masterVolume();
mMasterMute = mAudioFlinger->masterMute();
- for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
+ for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream);
mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream);
}
@@ -1131,12 +1151,12 @@
// conflicts will happen when tracks are moved from one output to another by audio policy
// manager
uint32_t strategy =
- AudioSystem::getStrategyForStream((AudioSystem::stream_type)streamType);
+ AudioSystem::getStrategyForStream((audio_stream_type_t)streamType);
for (size_t i = 0; i < mTracks.size(); ++i) {
sp<Track> t = mTracks[i];
if (t != 0) {
if (sessionId == t->sessionId() &&
- strategy != AudioSystem::getStrategyForStream((AudioSystem::stream_type)t->type())) {
+ strategy != AudioSystem::getStrategyForStream((audio_stream_type_t)t->type())) {
lStatus = BAD_VALUE;
goto Exit;
}
@@ -1155,7 +1175,7 @@
if (chain != 0) {
LOGV("createTrack_l() setting main buffer %p", chain->inBuffer());
track->setMainBuffer(chain->inBuffer());
- chain->setStrategy(AudioSystem::getStrategyForStream((AudioSystem::stream_type)track->type()));
+ chain->setStrategy(AudioSystem::getStrategyForStream((audio_stream_type_t)track->type()));
}
}
lStatus = NO_ERROR;
@@ -1170,7 +1190,7 @@
uint32_t AudioFlinger::PlaybackThread::latency() const
{
if (mOutput) {
- return mOutput->latency();
+ return mOutput->get_latency(mOutput);
}
else {
return 0;
@@ -1264,7 +1284,13 @@
String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys)
{
- return mOutput->getParameters(keys);
+ String8 out_s8;
+ char *s;
+
+ s = mOutput->common.get_parameters(&mOutput->common, keys.string());
+ out_s8 = String8(s);
+ free(s);
+ return out_s8;
}
// destroyTrack_l() must be called with AudioFlinger::mLock held
@@ -1296,12 +1322,12 @@
void AudioFlinger::PlaybackThread::readOutputParameters()
{
- mSampleRate = mOutput->sampleRate();
- mChannels = mOutput->channels();
- mChannelCount = (uint16_t)AudioSystem::popCount(mChannels);
- mFormat = mOutput->format();
- mFrameSize = (uint16_t)mOutput->frameSize();
- mFrameCount = mOutput->bufferSize() / mFrameSize;
+ mSampleRate = mOutput->common.get_sample_rate(&mOutput->common);
+ mChannels = mOutput->common.get_channels(&mOutput->common);
+ mChannelCount = (uint16_t)popcount(mChannels);
+ mFormat = mOutput->common.get_format(&mOutput->common);
+ mFrameSize = (uint16_t)audio_stream_frame_size(&mOutput->common);
+ mFrameCount = mOutput->common.get_buffer_size(&mOutput->common) / mFrameSize;
// FIXME - Current mixer implementation only supports stereo output: Always
// Allocate a stereo buffer even if HW output is mono.
@@ -1329,9 +1355,9 @@
if (mOutput == 0) {
return INVALID_OPERATION;
}
- *halFrames = mBytesWritten/mOutput->frameSize();
+ *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->common);
- return mOutput->getRenderPosition(dspFrames);
+ return mOutput->get_render_position(mOutput, dspFrames);
}
uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId)
@@ -1356,19 +1382,19 @@
uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId)
{
- // session AudioSystem::SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that
+ // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that
// it is moved to correct output by audio policy manager when A2DP is connected or disconnected
- if (sessionId == AudioSystem::SESSION_OUTPUT_MIX) {
- return AudioSystem::getStrategyForStream(AudioSystem::MUSIC);
+ if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
+ return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
}
for (size_t i = 0; i < mTracks.size(); i++) {
sp<Track> track = mTracks[i];
if (sessionId == track->sessionId() &&
!(track->mCblk->flags & CBLK_INVALID_MSK)) {
- return AudioSystem::getStrategyForStream((AudioSystem::stream_type) track->type());
+ return AudioSystem::getStrategyForStream((audio_stream_type_t) track->type());
}
}
- return AudioSystem::getStrategyForStream(AudioSystem::MUSIC);
+ return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
}
sp<AudioFlinger::EffectChain> AudioFlinger::PlaybackThread::getEffectChain(int sessionId)
@@ -1402,7 +1428,7 @@
// ----------------------------------------------------------------------------
-AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
+AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, struct audio_stream_out* output, int id, uint32_t device)
: PlaybackThread(audioFlinger, output, id, device),
mAudioMixer(0)
{
@@ -1461,7 +1487,7 @@
mSuspended) {
if (!mStandby) {
LOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended);
- mOutput->standby();
+ mOutput->common.standby(&mOutput->common);
mStandby = true;
mBytesWritten = 0;
}
@@ -1538,7 +1564,7 @@
mInWrite = true;
mBytesWritten += mixBufferSize;
- int bytesWritten = (int)mOutput->write(mMixBuffer, mixBufferSize);
+ int bytesWritten = (int)mOutput->write(mOutput, mMixBuffer, mixBufferSize);
if (bytesWritten < 0) mBytesWritten -= mixBufferSize;
mNumWrites++;
mInWrite = false;
@@ -1573,7 +1599,7 @@
}
if (!mStandby) {
- mOutput->standby();
+ mOutput->common.standby(&mOutput->common);
}
LOGV("MixerThread %p exiting", this);
@@ -1597,7 +1623,7 @@
masterVolume = 0;
}
// Delegate master volume control to effect in output mix effect chain if needed
- sp<EffectChain> chain = getEffectChain_l(AudioSystem::SESSION_OUTPUT_MIX);
+ sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
if (chain != 0) {
uint32_t v = (uint32_t)(masterVolume * (1 << 24));
chain->setVolume_l(&v, &v);
@@ -1823,14 +1849,14 @@
reconfig = true;
}
if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
- if (value != AudioSystem::PCM_16_BIT) {
+ if (value != AUDIO_FORMAT_PCM_16_BIT) {
status = BAD_VALUE;
} else {
reconfig = true;
}
}
if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
- if (value != AudioSystem::CHANNEL_OUT_STEREO) {
+ if (value != AUDIO_CHANNEL_OUT_STEREO) {
status = BAD_VALUE;
} else {
reconfig = true;
@@ -1852,12 +1878,12 @@
if (mDevice != value) {
uint32_t params = 0;
// check whether speaker is on
- if (value & AudioSystem::DEVICE_OUT_SPEAKER) {
+ if (value & AUDIO_DEVICE_OUT_SPEAKER) {
params |= IMediaPlayerService::kBatteryDataSpeakerOn;
}
int deviceWithoutSpeaker
- = AudioSystem::DEVICE_OUT_ALL & ~AudioSystem::DEVICE_OUT_SPEAKER;
+ = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER;
// check if any other device (except speaker) is on
if (value & deviceWithoutSpeaker ) {
params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn;
@@ -1877,12 +1903,14 @@
}
if (status == NO_ERROR) {
- status = mOutput->setParameters(keyValuePair);
+ status = mOutput->common.set_parameters(&mOutput->common,
+ keyValuePair.string());
if (!mStandby && status == INVALID_OPERATION) {
- mOutput->standby();
+ mOutput->common.standby(&mOutput->common);
mStandby = true;
mBytesWritten = 0;
- status = mOutput->setParameters(keyValuePair);
+ status = mOutput->common.set_parameters(&mOutput->common,
+ keyValuePair.string());
}
if (status == NO_ERROR && reconfig) {
delete mAudioMixer;
@@ -1926,7 +1954,7 @@
uint32_t AudioFlinger::MixerThread::activeSleepTimeUs()
{
- return (uint32_t)(mOutput->latency() * 1000) / 2;
+ return (uint32_t)(mOutput->get_latency(mOutput) * 1000) / 2;
}
uint32_t AudioFlinger::MixerThread::idleSleepTimeUs()
@@ -1940,7 +1968,7 @@
}
// ----------------------------------------------------------------------------
-AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
+AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, struct audio_stream_out* output, int id, uint32_t device)
: PlaybackThread(audioFlinger, output, id, device)
{
mType = PlaybackThread::DIRECT;
@@ -1976,12 +2004,12 @@
void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp)
{
// Do not apply volume on compressed audio
- if (!AudioSystem::isLinearPCM(mFormat)) {
+ if (!audio_is_linear_pcm(mFormat)) {
return;
}
// convert to signed 16 bit before volume calculation
- if (mFormat == AudioSystem::PCM_8_BIT) {
+ if (mFormat == AUDIO_FORMAT_PCM_8_BIT) {
size_t count = mFrameCount * mChannelCount;
uint8_t *src = (uint8_t *)mMixBuffer + count-1;
int16_t *dst = mMixBuffer + count-1;
@@ -2034,7 +2062,7 @@
}
// convert back to unsigned 8 bit after volume calculation
- if (mFormat == AudioSystem::PCM_8_BIT) {
+ if (mFormat == AUDIO_FORMAT_PCM_8_BIT) {
size_t count = mFrameCount * mChannelCount;
int16_t *src = mMixBuffer;
uint8_t *dst = (uint8_t *)mMixBuffer;
@@ -2090,7 +2118,7 @@
// wait until we have something to do...
if (!mStandby) {
LOGV("Audio hardware entering standby, mixer %p\n", this);
- mOutput->standby();
+ mOutput->common.standby(&mOutput->common);
mStandby = true;
mBytesWritten = 0;
}
@@ -2175,7 +2203,7 @@
// If audio HAL implements volume control,
// force software volume to nominal value
- if (mOutput->setVolume(left, right) == NO_ERROR) {
+ if (mOutput->set_volume(mOutput, left, right) == NO_ERROR) {
left = 1.0f;
right = 1.0f;
}
@@ -2276,7 +2304,7 @@
} else {
sleepTime = idleSleepTime;
}
- } else if (mBytesWritten != 0 && AudioSystem::isLinearPCM(mFormat)) {
+ } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) {
memset (mMixBuffer, 0, mFrameCount * mFrameSize);
sleepTime = 0;
}
@@ -2298,7 +2326,7 @@
mLastWriteTime = systemTime();
mInWrite = true;
mBytesWritten += mixBufferSize;
- int bytesWritten = (int)mOutput->write(mMixBuffer, mixBufferSize);
+ int bytesWritten = (int)mOutput->write(mOutput, mMixBuffer, mixBufferSize);
if (bytesWritten < 0) mBytesWritten -= mixBufferSize;
mNumWrites++;
mInWrite = false;
@@ -2320,7 +2348,7 @@
}
if (!mStandby) {
- mOutput->standby();
+ mOutput->common.standby(&mOutput->common);
}
LOGV("DirectOutputThread %p exiting", this);
@@ -2360,12 +2388,14 @@
}
}
if (status == NO_ERROR) {
- status = mOutput->setParameters(keyValuePair);
+ status = mOutput->common.set_parameters(&mOutput->common,
+ keyValuePair.string());
if (!mStandby && status == INVALID_OPERATION) {
- mOutput->standby();
+ mOutput->common.standby(&mOutput->common);
mStandby = true;
mBytesWritten = 0;
- status = mOutput->setParameters(keyValuePair);
+ status = mOutput->common.set_parameters(&mOutput->common,
+ keyValuePair.string());
}
if (status == NO_ERROR && reconfig) {
readOutputParameters();
@@ -2385,8 +2415,8 @@
uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs()
{
uint32_t time;
- if (AudioSystem::isLinearPCM(mFormat)) {
- time = (uint32_t)(mOutput->latency() * 1000) / 2;
+ if (audio_is_linear_pcm(mFormat)) {
+ time = (uint32_t)(mOutput->get_latency(mOutput) * 1000) / 2;
} else {
time = 10000;
}
@@ -2396,7 +2426,7 @@
uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs()
{
uint32_t time;
- if (AudioSystem::isLinearPCM(mFormat)) {
+ if (audio_is_linear_pcm(mFormat)) {
time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
} else {
time = 10000;
@@ -2407,7 +2437,7 @@
uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs()
{
uint32_t time;
- if (AudioSystem::isLinearPCM(mFormat)) {
+ if (audio_is_linear_pcm(mFormat)) {
time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
} else {
time = 10000;
@@ -2588,7 +2618,7 @@
mChannelCount,
frameCount);
if (outputTrack->cblk() != NULL) {
- thread->setStreamVolume(AudioSystem::NUM_STREAM_TYPES, 1.0f);
+ thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f);
mOutputTracks.add(outputTrack);
LOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread);
updateWaitTime();
@@ -2833,7 +2863,7 @@
mStreamType = streamType;
// NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of
// 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack
- mCblk->frameSize = AudioSystem::isLinearPCM(format) ? channelCount * sizeof(int16_t) : sizeof(int8_t);
+ mCblk->frameSize = audio_is_linear_pcm(format) ? channelCount * sizeof(int16_t) : sizeof(int8_t);
}
}
@@ -2864,7 +2894,7 @@
if (!isOutputTrack()) {
if (mState == ACTIVE || mState == RESUMING) {
AudioSystem::stopOutput(thread->id(),
- (AudioSystem::stream_type)mStreamType,
+ (audio_stream_type_t)mStreamType,
mSessionId);
// to track the speaker usage
@@ -2976,7 +3006,7 @@
if (!isOutputTrack() && state != ACTIVE && state != RESUMING) {
thread->mLock.unlock();
status = AudioSystem::startOutput(thread->id(),
- (AudioSystem::stream_type)mStreamType,
+ (audio_stream_type_t)mStreamType,
mSessionId);
thread->mLock.lock();
@@ -3016,7 +3046,7 @@
if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) {
thread->mLock.unlock();
AudioSystem::stopOutput(thread->id(),
- (AudioSystem::stream_type)mStreamType,
+ (audio_stream_type_t)mStreamType,
mSessionId);
thread->mLock.lock();
@@ -3038,7 +3068,7 @@
if (!isOutputTrack()) {
thread->mLock.unlock();
AudioSystem::stopOutput(thread->id(),
- (AudioSystem::stream_type)mStreamType,
+ (audio_stream_type_t)mStreamType,
mSessionId);
thread->mLock.lock();
@@ -3132,9 +3162,9 @@
{
if (mCblk != NULL) {
LOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer);
- if (format == AudioSystem::PCM_16_BIT) {
+ if (format == AUDIO_FORMAT_PCM_16_BIT) {
mCblk->frameSize = channelCount * sizeof(int16_t);
- } else if (format == AudioSystem::PCM_8_BIT) {
+ } else if (format == AUDIO_FORMAT_PCM_8_BIT) {
mCblk->frameSize = channelCount * sizeof(int8_t);
} else {
mCblk->frameSize = sizeof(int8_t);
@@ -3237,7 +3267,7 @@
int format,
int channelCount,
int frameCount)
- : Track(thread, NULL, AudioSystem::NUM_STREAM_TYPES, sampleRate, format, channelCount, frameCount, NULL, 0),
+ : Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelCount, frameCount, NULL, 0),
mActive(false), mSourceThread(sourceThread)
{
@@ -3609,7 +3639,7 @@
}
// If no audio session id is provided, create one here
- if (sessionId != NULL && *sessionId != AudioSystem::SESSION_OUTPUT_MIX) {
+ if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
lSessionId = *sessionId;
} else {
lSessionId = nextUniqueId_l();
@@ -3675,11 +3705,11 @@
// ----------------------------------------------------------------------------
-AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, AudioStreamIn *input, uint32_t sampleRate, uint32_t channels, int id) :
+AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, struct audio_stream_in *input, uint32_t sampleRate, uint32_t channels, int id) :
ThreadBase(audioFlinger, id),
mInput(input), mResampler(0), mRsmpOutBuffer(0), mRsmpInBuffer(0)
{
- mReqChannelCount = AudioSystem::popCount(channels);
+ mReqChannelCount = popcount(channels);
mReqSampleRate = sampleRate;
readInputParameters();
}
@@ -3721,7 +3751,7 @@
checkForNewParameters_l();
if (mActiveTrack == 0 && mConfigEvents.isEmpty()) {
if (!mStandby) {
- mInput->standby();
+ mInput->common.standby(&mInput->common);
mStandby = true;
}
@@ -3736,7 +3766,7 @@
if (mActiveTrack != 0) {
if (mActiveTrack->mState == TrackBase::PAUSING) {
if (!mStandby) {
- mInput->standby();
+ mInput->common.standby(&mInput->common);
mStandby = true;
}
mActiveTrack.clear();
@@ -3781,7 +3811,7 @@
mRsmpInIndex += framesIn;
framesOut -= framesIn;
if ((int)mChannelCount == mReqChannelCount ||
- mFormat != AudioSystem::PCM_16_BIT) {
+ mFormat != AUDIO_FORMAT_PCM_16_BIT) {
memcpy(dst, src, framesIn * mFrameSize);
} else {
int16_t *src16 = (int16_t *)src;
@@ -3801,11 +3831,11 @@
}
if (framesOut && mFrameCount == mRsmpInIndex) {
if (framesOut == mFrameCount &&
- ((int)mChannelCount == mReqChannelCount || mFormat != AudioSystem::PCM_16_BIT)) {
- mBytesRead = mInput->read(buffer.raw, mInputBytes);
+ ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) {
+ mBytesRead = mInput->read(mInput, buffer.raw, mInputBytes);
framesOut = 0;
} else {
- mBytesRead = mInput->read(mRsmpInBuffer, mInputBytes);
+ mBytesRead = mInput->read(mInput, mRsmpInBuffer, mInputBytes);
mRsmpInIndex = 0;
}
if (mBytesRead < 0) {
@@ -3813,7 +3843,7 @@
if (mActiveTrack->mState == TrackBase::ACTIVE) {
// Force input into standby so that it tries to
// recover at next read attempt
- mInput->standby();
+ mInput->common.standby(&mInput->common);
usleep(5000);
}
mRsmpInIndex = mFrameCount;
@@ -3868,7 +3898,7 @@
}
if (!mStandby) {
- mInput->standby();
+ mInput->common.standby(&mInput->common);
}
mActiveTrack.clear();
@@ -4000,13 +4030,13 @@
int channelCount;
if (framesReady == 0) {
- mBytesRead = mInput->read(mRsmpInBuffer, mInputBytes);
+ mBytesRead = mInput->read(mInput, mRsmpInBuffer, mInputBytes);
if (mBytesRead < 0) {
LOGE("RecordThread::getNextBuffer() Error reading audio input");
if (mActiveTrack->mState == TrackBase::ACTIVE) {
// Force input into standby so that it tries to
// recover at next read attempt
- mInput->standby();
+ mInput->common.standby(&mInput->common);
usleep(5000);
}
buffer->raw = 0;
@@ -4059,7 +4089,7 @@
reconfig = true;
}
if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
- reqChannelCount = AudioSystem::popCount(value);
+ reqChannelCount = popcount(value);
reconfig = true;
}
if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
@@ -4073,16 +4103,18 @@
}
}
if (status == NO_ERROR) {
- status = mInput->setParameters(keyValuePair);
+ status = mInput->common.set_parameters(&mInput->common, keyValuePair.string());
if (status == INVALID_OPERATION) {
- mInput->standby();
- status = mInput->setParameters(keyValuePair);
+ mInput->common.standby(&mInput->common);
+ status = mInput->common.set_parameters(&mInput->common, keyValuePair.string());
}
if (reconfig) {
if (status == BAD_VALUE &&
- reqFormat == mInput->format() && reqFormat == AudioSystem::PCM_16_BIT &&
- ((int)mInput->sampleRate() <= 2 * reqSamplingRate) &&
- (AudioSystem::popCount(mInput->channels()) < 3) && (reqChannelCount < 3)) {
+ reqFormat == mInput->common.get_format(&mInput->common) &&
+ reqFormat == AUDIO_FORMAT_PCM_16_BIT &&
+ ((int)mInput->common.get_sample_rate(&mInput->common) <= (2 * reqSamplingRate)) &&
+ (popcount(mInput->common.get_channels(&mInput->common)) < 3) &&
+ (reqChannelCount < 3)) {
status = NO_ERROR;
}
if (status == NO_ERROR) {
@@ -4103,7 +4135,13 @@
String8 AudioFlinger::RecordThread::getParameters(const String8& keys)
{
- return mInput->getParameters(keys);
+ char *s;
+ String8 out_s8;
+
+ s = mInput->common.get_parameters(&mInput->common, keys.string());
+ out_s8 = String8(s);
+ free(s);
+ return out_s8;
}
void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) {
@@ -4135,12 +4173,12 @@
if (mResampler) delete mResampler;
mResampler = 0;
- mSampleRate = mInput->sampleRate();
- mChannels = mInput->channels();
- mChannelCount = (uint16_t)AudioSystem::popCount(mChannels);
- mFormat = mInput->format();
- mFrameSize = (uint16_t)mInput->frameSize();
- mInputBytes = mInput->bufferSize();
+ mSampleRate = mInput->common.get_sample_rate(&mInput->common);
+ mChannels = mInput->common.get_channels(&mInput->common);
+ mChannelCount = (uint16_t)popcount(mChannels);
+ mFormat = mInput->common.get_format(&mInput->common);
+ mFrameSize = (uint16_t)audio_stream_frame_size(&mInput->common);
+ mInputBytes = mInput->common.get_buffer_size(&mInput->common);
mFrameCount = mInputBytes / mFrameSize;
mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount];
@@ -4170,7 +4208,7 @@
unsigned int AudioFlinger::RecordThread::getInputFramesLost()
{
- return mInput->getInputFramesLost();
+ return mInput->get_input_frames_lost(mInput);
}
// ----------------------------------------------------------------------------
@@ -4189,6 +4227,7 @@
uint32_t format = pFormat ? *pFormat : 0;
uint32_t channels = pChannels ? *pChannels : 0;
uint32_t latency = pLatencyMs ? *pLatencyMs : 0;
+ struct audio_stream_out *output;
LOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x",
pDevices ? *pDevices : 0,
@@ -4202,11 +4241,11 @@
}
Mutex::Autolock _l(mLock);
- AudioStreamOut *output = mAudioHardware->openOutputStream(*pDevices,
- (int *)&format,
- &channels,
- &samplingRate,
- &status);
+ status = mAudioHardwareDev->open_output_stream(mAudioHardwareDev, *pDevices,
+ (int *)&format,
+ &channels,
+ &samplingRate,
+ &output);
LOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d",
output,
samplingRate,
@@ -4217,9 +4256,9 @@
mHardwareStatus = AUDIO_HW_IDLE;
if (output != 0) {
int id = nextUniqueId_l();
- if ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) ||
- (format != AudioSystem::PCM_16_BIT) ||
- (channels != AudioSystem::CHANNEL_OUT_STEREO)) {
+ if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) ||
+ (format != AUDIO_FORMAT_PCM_16_BIT) ||
+ (channels != AUDIO_CHANNEL_OUT_STEREO)) {
thread = new DirectOutputThread(this, output, id, *pDevices);
LOGV("openOutput() created direct output: ID %d thread %p", id, thread);
} else {
@@ -4290,7 +4329,7 @@
thread->exit();
if (thread->type() != PlaybackThread::DUPLICATING) {
- mAudioHardware->closeOutputStream(thread->getOutput());
+ mAudioHardwareDev->close_output_stream(mAudioHardwareDev, thread->getOutput());
}
return NO_ERROR;
}
@@ -4340,18 +4379,20 @@
uint32_t reqSamplingRate = samplingRate;
uint32_t reqFormat = format;
uint32_t reqChannels = channels;
+ struct audio_stream_in *input;
if (pDevices == NULL || *pDevices == 0) {
return 0;
}
Mutex::Autolock _l(mLock);
- AudioStreamIn *input = mAudioHardware->openInputStream(*pDevices,
- (int *)&format,
- &channels,
- &samplingRate,
- &status,
- (AudioSystem::audio_in_acoustics)acoustics);
+ status = mAudioHardwareDev->open_input_stream(mAudioHardwareDev,
+ *pDevices,
+ (int *)&format,
+ &channels,
+ &samplingRate,
+ (audio_in_acoustics_t)acoustics,
+ &input);
LOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d",
input,
samplingRate,
@@ -4364,16 +4405,17 @@
// try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo
// or stereo to mono conversions on 16 bit PCM inputs.
if (input == 0 && status == BAD_VALUE &&
- reqFormat == format && format == AudioSystem::PCM_16_BIT &&
+ reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT &&
(samplingRate <= 2 * reqSamplingRate) &&
- (AudioSystem::popCount(channels) < 3) && (AudioSystem::popCount(reqChannels) < 3)) {
+ (popcount(channels) < 3) && (popcount(reqChannels) < 3)) {
LOGV("openInput() reopening with proposed sampling rate and channels");
- input = mAudioHardware->openInputStream(*pDevices,
- (int *)&format,
- &channels,
- &samplingRate,
- &status,
- (AudioSystem::audio_in_acoustics)acoustics);
+ status = mAudioHardwareDev->open_input_stream(mAudioHardwareDev,
+ *pDevices,
+ (int *)&format,
+ &channels,
+ &samplingRate,
+ (audio_in_acoustics_t)acoustics,
+ &input);
}
if (input != 0) {
@@ -4386,7 +4428,7 @@
if (pFormat) *pFormat = format;
if (pChannels) *pChannels = reqChannels;
- input->standby();
+ input->common.standby(&input->common);
// notify client processes of the new input creation
thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED);
@@ -4415,7 +4457,7 @@
}
thread->exit();
- mAudioHardware->closeInputStream(thread->getInput());
+ mAudioHardwareDev->close_input_stream(mAudioHardwareDev, thread->getInput());
return NO_ERROR;
}
@@ -4569,14 +4611,14 @@
}
// check audio settings permission for global effects
- if (sessionId == AudioSystem::SESSION_OUTPUT_MIX && !settingsAllowed()) {
+ if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) {
lStatus = PERMISSION_DENIED;
goto Exit;
}
- // Session AudioSystem::SESSION_OUTPUT_STAGE is reserved for output stage effects
+ // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects
// that can only be created by audio policy manager (running in same process)
- if (sessionId == AudioSystem::SESSION_OUTPUT_STAGE && getpid() != pid) {
+ if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid() != pid) {
lStatus = PERMISSION_DENIED;
goto Exit;
}
@@ -4590,12 +4632,12 @@
}
if (output == 0) {
- if (sessionId == AudioSystem::SESSION_OUTPUT_STAGE) {
+ if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
// output must be specified by AudioPolicyManager when using session
- // AudioSystem::SESSION_OUTPUT_STAGE
+ // AUDIO_SESSION_OUTPUT_STAGE
lStatus = BAD_VALUE;
goto Exit;
- } else if (sessionId == AudioSystem::SESSION_OUTPUT_MIX) {
+ } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
// if the output returned by getOutputForEffect() is removed before we lock the
// mutex below, the call to checkPlaybackThread_l(output) below will detect it
// and we will exit safely
@@ -4643,7 +4685,7 @@
// an auxiliary version of this effect type is available
found = true;
memcpy(&d, &desc, sizeof(effect_descriptor_t));
- if (sessionId != AudioSystem::SESSION_OUTPUT_MIX ||
+ if (sessionId != AUDIO_SESSION_OUTPUT_MIX ||
(desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
break;
}
@@ -4656,14 +4698,14 @@
}
// For same effect type, chose auxiliary version over insert version if
// connect to output mix (Compliance to OpenSL ES)
- if (sessionId == AudioSystem::SESSION_OUTPUT_MIX &&
+ if (sessionId == AUDIO_SESSION_OUTPUT_MIX &&
(d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) {
memcpy(&desc, &d, sizeof(effect_descriptor_t));
}
}
// Do not allow auxiliary effects on a session different from 0 (output mix)
- if (sessionId != AudioSystem::SESSION_OUTPUT_MIX &&
+ if (sessionId != AUDIO_SESSION_OUTPUT_MIX &&
(desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
lStatus = INVALID_OPERATION;
goto Exit;
@@ -4836,7 +4878,7 @@
// Do not allow auxiliary effect on session other than 0
if ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY &&
- sessionId != AudioSystem::SESSION_OUTPUT_MIX) {
+ sessionId != AUDIO_SESSION_OUTPUT_MIX) {
LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d",
desc->name, sessionId);
lStatus = BAD_VALUE;
@@ -4845,7 +4887,7 @@
// Do not allow effects with session ID 0 on direct output or duplicating threads
// TODO: add rule for hw accelerated effects on direct outputs with non PCM format
- if (sessionId == AudioSystem::SESSION_OUTPUT_MIX && mType != MIXER) {
+ if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) {
LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d",
desc->name, sessionId);
lStatus = BAD_VALUE;
@@ -5032,13 +5074,13 @@
chain->setInBuffer(buffer, ownsBuffer);
chain->setOutBuffer(mMixBuffer);
- // Effect chain for session AudioSystem::SESSION_OUTPUT_STAGE is inserted at end of effect
+ // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect
// chains list in order to be processed last as it contains output stage effects
- // Effect chain for session AudioSystem::SESSION_OUTPUT_MIX is inserted before
- // session AudioSystem::SESSION_OUTPUT_STAGE to be processed
+ // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before
+ // session AUDIO_SESSION_OUTPUT_STAGE to be processed
// after track specific effects and before output stage
- // It is therefore mandatory that AudioSystem::SESSION_OUTPUT_MIX == 0 and
- // that AudioSystem::SESSION_OUTPUT_STAGE < AudioSystem::SESSION_OUTPUT_MIX
+ // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and
+ // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX
// Effect chain for other sessions are inserted at beginning of effect
// chains list to be processed before output mix effects. Relative order between other
// sessions is not important
@@ -5118,8 +5160,8 @@
if (EffectId == 0) {
track->setAuxBuffer(0, NULL);
} else {
- // Auxiliary effects are always in audio session AudioSystem::SESSION_OUTPUT_MIX
- sp<EffectModule> effect = getEffect_l(AudioSystem::SESSION_OUTPUT_MIX, EffectId);
+ // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX
+ sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId);
if (effect != 0) {
if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer());
@@ -5403,7 +5445,7 @@
mConfig.outputCfg.bufferProvider.releaseBuffer = NULL;
mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ;
// Insert effect:
- // - in session AudioSystem::SESSION_OUTPUT_MIX or AudioSystem::SESSION_OUTPUT_STAGE,
+ // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE,
// always overwrites output buffer: input buffer == output buffer
// - in other sessions:
// last effect in the chain accumulates in output buffer: input buffer != output buffer
@@ -5684,17 +5726,17 @@
// update this table when AudioSystem::audio_devices or audio_device_e (in EffectApi.h) are modified
const uint32_t AudioFlinger::EffectModule::sDeviceConvTable[] = {
- DEVICE_EARPIECE, // AudioSystem::DEVICE_OUT_EARPIECE
- DEVICE_SPEAKER, // AudioSystem::DEVICE_OUT_SPEAKER
- DEVICE_WIRED_HEADSET, // case AudioSystem::DEVICE_OUT_WIRED_HEADSET
- DEVICE_WIRED_HEADPHONE, // AudioSystem::DEVICE_OUT_WIRED_HEADPHONE
- DEVICE_BLUETOOTH_SCO, // AudioSystem::DEVICE_OUT_BLUETOOTH_SCO
- DEVICE_BLUETOOTH_SCO_HEADSET, // AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET
- DEVICE_BLUETOOTH_SCO_CARKIT, // AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT
- DEVICE_BLUETOOTH_A2DP, // AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP
- DEVICE_BLUETOOTH_A2DP_HEADPHONES, // AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES
- DEVICE_BLUETOOTH_A2DP_SPEAKER, // AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER
- DEVICE_AUX_DIGITAL // AudioSystem::DEVICE_OUT_AUX_DIGITAL
+ DEVICE_EARPIECE, // AUDIO_DEVICE_OUT_EARPIECE
+ DEVICE_SPEAKER, // AUDIO_DEVICE_OUT_SPEAKER
+ DEVICE_WIRED_HEADSET, // case AUDIO_DEVICE_OUT_WIRED_HEADSET
+ DEVICE_WIRED_HEADPHONE, // AUDIO_DEVICE_OUT_WIRED_HEADPHONE
+ DEVICE_BLUETOOTH_SCO, // AUDIO_DEVICE_OUT_BLUETOOTH_SCO
+ DEVICE_BLUETOOTH_SCO_HEADSET, // AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET
+ DEVICE_BLUETOOTH_SCO_CARKIT, // AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT
+ DEVICE_BLUETOOTH_A2DP, // AUDIO_DEVICE_OUT_BLUETOOTH_A2DP
+ DEVICE_BLUETOOTH_A2DP_HEADPHONES, // AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES
+ DEVICE_BLUETOOTH_A2DP_SPEAKER, // AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER
+ DEVICE_AUX_DIGITAL // AUDIO_DEVICE_OUT_AUX_DIGITAL
};
uint32_t AudioFlinger::EffectModule::deviceAudioSystemToEffectApi(uint32_t device)
@@ -5714,10 +5756,10 @@
// update this table when AudioSystem::audio_mode or audio_mode_e (in EffectApi.h) are modified
const uint32_t AudioFlinger::EffectModule::sModeConvTable[] = {
- AUDIO_MODE_NORMAL, // AudioSystem::MODE_NORMAL
- AUDIO_MODE_RINGTONE, // AudioSystem::MODE_RINGTONE
- AUDIO_MODE_IN_CALL, // AudioSystem::MODE_IN_CALL
- AUDIO_MODE_IN_CALL // AudioSystem::MODE_IN_COMMUNICATION, same conversion as for MODE_IN_CALL
+ AUDIO_EFFECT_MODE_NORMAL, // AUDIO_MODE_NORMAL
+ AUDIO_EFFECT_MODE_RINGTONE, // AUDIO_MODE_RINGTONE
+ AUDIO_EFFECT_MODE_IN_CALL, // AUDIO_MODE_IN_CALL
+ AUDIO_EFFECT_MODE_IN_CALL // AUDIO_MODE_IN_COMMUNICATION, same conversion as for AUDIO_MODE_IN_CALL
};
int AudioFlinger::EffectModule::modeAudioSystemToEffectApi(uint32_t mode)
@@ -6027,7 +6069,7 @@
mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX),
mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX)
{
- mStrategy = AudioSystem::getStrategyForStream(AudioSystem::MUSIC);
+ mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
}
AudioFlinger::EffectChain::~EffectChain()
@@ -6078,8 +6120,8 @@
return;
}
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
- bool isGlobalSession = (mSessionId == AudioSystem::SESSION_OUTPUT_MIX) ||
- (mSessionId == AudioSystem::SESSION_OUTPUT_STAGE);
+ bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) ||
+ (mSessionId == AUDIO_SESSION_OUTPUT_STAGE);
bool tracksOnSession = false;
if (!isGlobalSession) {
tracksOnSession =