Update the debug recordings to use protobufs.
Also modify the unittest proto based to correspond with the changes. process_test is a bit of a hack job, but it works fine and isn't too unreadable. We should refactor it properly later.
Review URL: http://webrtc-codereview.appspot.com/98007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@296 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/src/modules/audio_processing/main/source/audio_processing_impl.cc b/src/modules/audio_processing/main/source/audio_processing_impl.cc
index 6440e36..660ac22 100644
--- a/src/modules/audio_processing/main/source/audio_processing_impl.cc
+++ b/src/modules/audio_processing/main/source/audio_processing_impl.cc
@@ -10,36 +10,24 @@
#include "audio_processing_impl.h"
-#include <cassert>
-
-#include "module_common_types.h"
-
-#include "critical_section_wrapper.h"
-#include "file_wrapper.h"
+#include <assert.h>
#include "audio_buffer.h"
+#include "critical_section_wrapper.h"
#include "echo_cancellation_impl.h"
#include "echo_control_mobile_impl.h"
+#include "file_wrapper.h"
#include "high_pass_filter_impl.h"
#include "gain_control_impl.h"
#include "level_estimator_impl.h"
+#include "module_common_types.h"
#include "noise_suppression_impl.h"
#include "processing_component.h"
#include "splitting_filter.h"
#include "voice_detection_impl.h"
+#include "webrtc/audio_processing/debug.pb.h"
namespace webrtc {
-namespace {
-
-enum Events {
- kInitializeEvent,
- kRenderEvent,
- kCaptureEvent
-};
-
-const char kMagicNumber[] = "#!vqetrace1.2";
-} // namespace
-
AudioProcessing* AudioProcessing::Create(int id) {
/*WEBRTC_TRACE(webrtc::kTraceModuleCall,
webrtc::kTraceAudioProcessing,
@@ -69,6 +57,7 @@
noise_suppression_(NULL),
voice_detection_(NULL),
debug_file_(FileWrapper::Create()),
+ event_msg_(new audioproc::Event()),
crit_(CriticalSectionWrapper::CreateCriticalSection()),
render_audio_(NULL),
capture_audio_(NULL),
@@ -77,9 +66,9 @@
samples_per_channel_(sample_rate_hz_ / 100),
stream_delay_ms_(0),
was_stream_delay_set_(false),
- num_render_input_channels_(1),
- num_capture_input_channels_(1),
- num_capture_output_channels_(1) {
+ num_reverse_channels_(1),
+ num_input_channels_(1),
+ num_output_channels_(1) {
echo_cancellation_ = new EchoCancellationImpl(this);
component_list_.push_back(echo_cancellation_);
@@ -117,15 +106,18 @@
delete debug_file_;
debug_file_ = NULL;
+ delete event_msg_;
+ event_msg_ = NULL;
+
delete crit_;
crit_ = NULL;
- if (render_audio_ != NULL) {
+ if (render_audio_) {
delete render_audio_;
render_audio_ = NULL;
}
- if (capture_audio_ != NULL) {
+ if (capture_audio_) {
delete capture_audio_;
capture_audio_ = NULL;
}
@@ -155,9 +147,9 @@
capture_audio_ = NULL;
}
- render_audio_ = new AudioBuffer(num_render_input_channels_,
+ render_audio_ = new AudioBuffer(num_reverse_channels_,
samples_per_channel_);
- capture_audio_ = new AudioBuffer(num_capture_input_channels_,
+ capture_audio_ = new AudioBuffer(num_input_channels_,
samples_per_channel_);
was_stream_delay_set_ = false;
@@ -171,6 +163,13 @@
}
}
+ if (debug_file_->Open()) {
+ int err = WriteInitMessage();
+ if (err != kNoError) {
+ return err;
+ }
+ }
+
return kNoError;
}
@@ -205,13 +204,13 @@
return kBadParameterError;
}
- num_render_input_channels_ = channels;
+ num_reverse_channels_ = channels;
return InitializeLocked();
}
int AudioProcessingImpl::num_reverse_channels() const {
- return num_render_input_channels_;
+ return num_reverse_channels_;
}
int AudioProcessingImpl::set_num_channels(
@@ -231,18 +230,18 @@
return kBadParameterError;
}
- num_capture_input_channels_ = input_channels;
- num_capture_output_channels_ = output_channels;
+ num_input_channels_ = input_channels;
+ num_output_channels_ = output_channels;
return InitializeLocked();
}
int AudioProcessingImpl::num_input_channels() const {
- return num_capture_input_channels_;
+ return num_input_channels_;
}
int AudioProcessingImpl::num_output_channels() const {
- return num_capture_output_channels_;
+ return num_output_channels_;
}
int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
@@ -258,7 +257,7 @@
return kBadSampleRateError;
}
- if (frame->_audioChannel != num_capture_input_channels_) {
+ if (frame->_audioChannel != num_input_channels_) {
return kBadNumberChannelsError;
}
@@ -267,44 +266,28 @@
}
if (debug_file_->Open()) {
- WebRtc_UWord8 event = kCaptureEvent;
- if (!debug_file_->Write(&event, sizeof(event))) {
- return kFileError;
- }
-
- if (!debug_file_->Write(&frame->_frequencyInHz,
- sizeof(frame->_frequencyInHz))) {
- return kFileError;
- }
-
- if (!debug_file_->Write(&frame->_audioChannel,
- sizeof(frame->_audioChannel))) {
- return kFileError;
- }
-
- if (!debug_file_->Write(&frame->_payloadDataLengthInSamples,
- sizeof(frame->_payloadDataLengthInSamples))) {
- return kFileError;
- }
-
- if (!debug_file_->Write(frame->_payloadData,
- sizeof(WebRtc_Word16) * frame->_payloadDataLengthInSamples *
- frame->_audioChannel)) {
- return kFileError;
- }
+ event_msg_->set_type(audioproc::Event::STREAM);
+ audioproc::Stream* msg = event_msg_->mutable_stream();
+ const size_t data_size = sizeof(WebRtc_Word16) *
+ frame->_payloadDataLengthInSamples *
+ frame->_audioChannel;
+ msg->set_input_data(frame->_payloadData, data_size);
+ msg->set_delay(stream_delay_ms_);
+ msg->set_drift(echo_cancellation_->stream_drift_samples());
+ msg->set_level(gain_control_->stream_analog_level());
}
capture_audio_->DeinterleaveFrom(frame);
// TODO(ajm): experiment with mixing and AEC placement.
- if (num_capture_output_channels_ < num_capture_input_channels_) {
- capture_audio_->Mix(num_capture_output_channels_);
+ if (num_output_channels_ < num_input_channels_) {
+ capture_audio_->Mix(num_output_channels_);
- frame->_audioChannel = num_capture_output_channels_;
+ frame->_audioChannel = num_output_channels_;
}
if (sample_rate_hz_ == kSampleRate32kHz) {
- for (int i = 0; i < num_capture_input_channels_; i++) {
+ for (int i = 0; i < num_input_channels_; i++) {
// Split into a low and high band.
SplittingFilterAnalysis(capture_audio_->data(i),
capture_audio_->low_pass_split_data(i),
@@ -360,7 +343,7 @@
//}
if (sample_rate_hz_ == kSampleRate32kHz) {
- for (int i = 0; i < num_capture_output_channels_; i++) {
+ for (int i = 0; i < num_output_channels_; i++) {
// Recombine low and high bands.
SplittingFilterSynthesis(capture_audio_->low_pass_split_data(i),
capture_audio_->high_pass_split_data(i),
@@ -372,6 +355,18 @@
capture_audio_->InterleaveTo(frame);
+ if (debug_file_->Open()) {
+ audioproc::Stream* msg = event_msg_->mutable_stream();
+ const size_t data_size = sizeof(WebRtc_Word16) *
+ frame->_payloadDataLengthInSamples *
+ frame->_audioChannel;
+ msg->set_output_data(frame->_payloadData, data_size);
+ err = WriteMessageToDebugFile();
+ if (err != kNoError) {
+ return err;
+ }
+ }
+
return kNoError;
}
@@ -388,7 +383,7 @@
return kBadSampleRateError;
}
- if (frame->_audioChannel != num_render_input_channels_) {
+ if (frame->_audioChannel != num_reverse_channels_) {
return kBadNumberChannelsError;
}
@@ -397,30 +392,15 @@
}
if (debug_file_->Open()) {
- WebRtc_UWord8 event = kRenderEvent;
- if (!debug_file_->Write(&event, sizeof(event))) {
- return kFileError;
- }
-
- if (!debug_file_->Write(&frame->_frequencyInHz,
- sizeof(frame->_frequencyInHz))) {
- return kFileError;
- }
-
- if (!debug_file_->Write(&frame->_audioChannel,
- sizeof(frame->_audioChannel))) {
- return kFileError;
- }
-
- if (!debug_file_->Write(&frame->_payloadDataLengthInSamples,
- sizeof(frame->_payloadDataLengthInSamples))) {
- return kFileError;
- }
-
- if (!debug_file_->Write(frame->_payloadData,
- sizeof(WebRtc_Word16) * frame->_payloadDataLengthInSamples *
- frame->_audioChannel)) {
- return kFileError;
+ event_msg_->set_type(audioproc::Event::REVERSE_STREAM);
+ audioproc::ReverseStream* msg = event_msg_->mutable_reverse_stream();
+ const size_t data_size = sizeof(WebRtc_Word16) *
+ frame->_payloadDataLengthInSamples *
+ frame->_audioChannel;
+ msg->set_data(frame->_payloadData, data_size);
+ err = WriteMessageToDebugFile();
+ if (err != kNoError) {
+ return err;
}
}
@@ -428,7 +408,7 @@
// TODO(ajm): turn the splitting filter into a component?
if (sample_rate_hz_ == kSampleRate32kHz) {
- for (int i = 0; i < num_render_input_channels_; i++) {
+ for (int i = 0; i < num_reverse_channels_; i++) {
// Split into low and high band.
SplittingFilterAnalysis(render_audio_->data(i),
render_audio_->low_pass_split_data(i),
@@ -508,20 +488,9 @@
return kFileError;
}
- if (debug_file_->WriteText("%s\n", kMagicNumber) == -1) {
- debug_file_->CloseFile();
- return kFileError;
- }
-
- // TODO(ajm): should we do this? If so, we need the number of channels etc.
- // Record the default sample rate.
- WebRtc_UWord8 event = kInitializeEvent;
- if (!debug_file_->Write(&event, sizeof(event))) {
- return kFileError;
- }
-
- if (!debug_file_->Write(&sample_rate_hz_, sizeof(sample_rate_hz_))) {
- return kFileError;
+ int err = WriteInitMessage();
+ if (err != kNoError) {
+ return err;
}
return kNoError;
@@ -578,7 +547,7 @@
}
memset(&version[position], 0, bytes_remaining);
- WebRtc_Word8 my_version[] = "AudioProcessing 1.0.0";
+ char my_version[] = "AudioProcessing 1.0.0";
// Includes null termination.
WebRtc_UWord32 length = static_cast<WebRtc_UWord32>(strlen(my_version));
if (bytes_remaining < length) {
@@ -633,4 +602,48 @@
return kNoError;
}
+
+int AudioProcessingImpl::WriteMessageToDebugFile() {
+ int32_t size = event_msg_->ByteSize();
+ if (size <= 0) {
+ return kUnspecifiedError;
+ }
+#if defined(WEBRTC_BIG_ENDIAN)
+ // TODO(ajm): Use little-endian "on the wire". For the moment, we can be
+ // pretty safe in assuming little-endian.
+#endif
+
+ if (!event_msg_->SerializeToString(&event_str_)) {
+ return kUnspecifiedError;
+ }
+
+ // Write message preceded by its size.
+ if (!debug_file_->Write(&size, sizeof(int32_t))) {
+ return kFileError;
+ }
+ if (!debug_file_->Write(event_str_.data(), event_str_.length())) {
+ return kFileError;
+ }
+
+ event_msg_->Clear();
+
+ return 0;
+}
+
+int AudioProcessingImpl::WriteInitMessage() {
+ event_msg_->set_type(audioproc::Event::INIT);
+ audioproc::Init* msg = event_msg_->mutable_init();
+ msg->set_sample_rate(sample_rate_hz_);
+ msg->set_device_sample_rate(echo_cancellation_->device_sample_rate_hz());
+ msg->set_num_input_channels(num_input_channels_);
+ msg->set_num_output_channels(num_output_channels_);
+ msg->set_num_reverse_channels(num_reverse_channels_);
+
+ int err = WriteMessageToDebugFile();
+ if (err != kNoError) {
+ return err;
+ }
+
+ return kNoError;
+}
} // namespace webrtc