AecDump interface

webrtc/modules/audio_processing/audio_processing_impl.cc

Index: webrtc/modules/audio_processing/audio_processing_impl.cc
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
index 816210f34f8738eb7a042adee69f87f5ed1a99d8..45b0cc24066aa1b7c38d2e6e08ef93965c75bdbd 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -152,6 +152,23 @@ class HighPassFilterImpl : public HighPassFilter {
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(HighPassFilterImpl);
 };
 
+webrtc::InternalAPMStreamsConfig ToStreamsConfig(
+    const ProcessingConfig& api_format) {
+  webrtc::InternalAPMStreamsConfig result;
+  result.input_sample_rate = api_format.input_stream().sample_rate_hz();
+  result.input_num_channels = api_format.input_stream().num_channels();
+  result.output_num_channels = api_format.output_stream().num_channels();
+  result.render_input_num_channels =
+      api_format.reverse_input_stream().num_channels();
+  result.render_input_sample_rate =
+      api_format.reverse_input_stream().sample_rate_hz();
+  result.output_sample_rate = api_format.output_stream().sample_rate_hz();
+  result.render_output_sample_rate =
+      api_format.reverse_output_stream().sample_rate_hz();
+  result.render_output_num_channels =
+      api_format.reverse_output_stream().num_channels();
+  return result;
+}
 }  // namespace
 
 // Throughout webrtc, it's assumed that success is represented by zero.
@@ -526,7 +543,9 @@ int AudioProcessingImpl::InitializeLocked() {
     }
   }
 #endif
-
+  if (aec_dump_) {
+    aec_dump_->WriteInitMessage(ToStreamsConfig(formats_.api_format));
+  }
   return kNoError;
 }
 
@@ -824,6 +843,11 @@ int AudioProcessingImpl::ProcessStream(const float* const* src,
   }
 #endif
 
+  std::unique_ptr<AecDump::CaptureStreamInfo> stream_info;
+  if (aec_dump_) {
+    stream_info = RecordUnprocessedCaptureStream(src);
+  }
+
   capture_.capture_audio->CopyFrom(src, formats_.api_format.input_stream());
   RETURN_ON_ERR(ProcessCaptureStreamLocked());
   capture_.capture_audio->CopyTo(formats_.api_format.output_stream(), dest);
@@ -841,7 +865,9 @@ int AudioProcessingImpl::ProcessStream(const float* const* src,
                                           &crit_debug_, &debug_dump_.capture));
   }
 #endif
-
+  if (aec_dump_) {
+    RecordProcessedCaptureStream(dest, std::move(stream_info));
+  }
   return kNoError;
 }
 
@@ -1078,6 +1104,11 @@ int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
     return kBadDataLengthError;
   }
 
+  std::unique_ptr<AecDump::CaptureStreamInfo> stream_info;
+  if (aec_dump_) {
+    stream_info = RecordUnprocessedCaptureStream(*frame);
+  }
+
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   if (debug_dump_.debug_file->is_open()) {
     RETURN_ON_ERR(WriteConfigMessage(false));
@@ -1095,6 +1126,9 @@ int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
   capture_.capture_audio->InterleaveTo(
       frame, submodule_states_.CaptureMultiBandProcessingActive());
 
+  if (aec_dump_) {
+    RecordProcessedCaptureStream(*frame, std::move(stream_info));
+  }
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   if (debug_dump_.debug_file->is_open()) {
     audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream();
@@ -1376,7 +1410,14 @@ int AudioProcessingImpl::AnalyzeReverseStreamLocked(
                                           &crit_debug_, &debug_dump_.render));
   }
 #endif
-
+  if (aec_dump_) {
+    const size_t channel_size =
+        formats_.api_format.reverse_input_stream().num_frames();
+    const size_t num_channels =
+        formats_.api_format.reverse_input_stream().num_channels();
+    aec_dump_->WriteRenderStreamMessage(
+        FloatAudioFrame(src, num_channels, channel_size));
+  }
   render_.render_audio->CopyFrom(src,
                                  formats_.api_format.reverse_input_stream());
   return ProcessRenderStreamLocked();
@@ -1429,6 +1470,10 @@ int AudioProcessingImpl::ProcessReverseStream(AudioFrame* frame) {
                                           &crit_debug_, &debug_dump_.render));
   }
 #endif
+  if (aec_dump_) {
+    aec_dump_->WriteRenderStreamMessage(*frame);
+  }
+
   render_.render_audio->DeinterleaveFrom(frame);
   RETURN_ON_ERR(ProcessRenderStreamLocked());
   render_.render_audio->InterleaveTo(
@@ -1512,6 +1557,17 @@ int AudioProcessingImpl::delay_offset_ms() const {
   return capture_.delay_offset_ms;
 }
 
+void AudioProcessingImpl::StartDebugRecording(
+    std::unique_ptr<AecDump> aec_dump) {
+  rtc::CritScope cs_render(&crit_render_);
+  rtc::CritScope cs_capture(&crit_capture_);
+  RTC_DCHECK(aec_dump);
+  aec_dump_ = std::move(aec_dump);
+
+  aec_dump_->WriteConfig(CollectApmConfig(), true);
+  aec_dump_->WriteInitMessage(ToStreamsConfig(formats_.api_format));
+}
+
 int AudioProcessingImpl::StartDebugRecording(
     const char filename[AudioProcessing::kMaxFilenameSize],
     int64_t max_log_size_bytes) {
@@ -1586,6 +1642,7 @@ int AudioProcessingImpl::StopDebugRecording() {
   // Run in a single-threaded manner.
   rtc::CritScope cs_render(&crit_render_);
   rtc::CritScope cs_capture(&crit_capture_);
+  aec_dump_.reset();
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   // We just return if recording hasn't started.
@@ -1837,6 +1894,124 @@ void AudioProcessingImpl::UpdateHistogramsOnCallEnd() {
   capture_.last_aec_system_delay_ms = 0;
 }
 
+InternalAPMConfig AudioProcessingImpl::CollectApmConfig() const {
+  std::string experiments_description =
+      public_submodules_->echo_cancellation->GetExperimentsDescription();
+  // TODO(peah): Add semicolon-separated concatenations of experiment
+  // descriptions for other submodules.
+  if (capture_nonlocked_.level_controller_enabled) {
+    experiments_description += "LevelController;";
+  }
+  if (constants_.agc_clipped_level_min != kClippedLevelMin) {
+    experiments_description += "AgcClippingLevelExperiment;";
+  }
+  if (capture_nonlocked_.echo_canceller3_enabled) {
+    experiments_description += "EchoCanceller3;";
+  }
+
+  InternalAPMConfig apm_config;
+
+  apm_config.aec_enabled = public_submodules_->echo_cancellation->is_enabled();
+  apm_config.aec_delay_agnostic_enabled =
+      public_submodules_->echo_cancellation->is_delay_agnostic_enabled();
+  apm_config.aec_drift_compensation_enabled =
+      public_submodules_->echo_cancellation->is_drift_compensation_enabled();
+  apm_config.aec_extended_filter_enabled =
+      public_submodules_->echo_cancellation->is_extended_filter_enabled();
+  apm_config.aec_suppression_level = static_cast<int>(
+      public_submodules_->echo_cancellation->suppression_level());
+
+  apm_config.aecm_enabled =
+      public_submodules_->echo_control_mobile->is_enabled();
+  apm_config.aecm_comfort_noise_enabled =
+      public_submodules_->echo_control_mobile->is_comfort_noise_enabled();
+  apm_config.aecm_routing_mode =
+      static_cast<int>(public_submodules_->echo_control_mobile->routing_mode());
+
+  apm_config.agc_enabled = public_submodules_->gain_control->is_enabled();
+  apm_config.agc_mode =
+      static_cast<int>(public_submodules_->gain_control->mode());
+  apm_config.agc_limiter_enabled =
+      public_submodules_->gain_control->is_limiter_enabled();
+  apm_config.noise_robust_agc_enabled = constants_.use_experimental_agc;
+
+  apm_config.hpf_enabled = config_.high_pass_filter.enabled;
+
+  apm_config.ns_enabled = public_submodules_->noise_suppression->is_enabled();
+  apm_config.ns_level =
+      static_cast<int>(public_submodules_->noise_suppression->level());
+
+  apm_config.transient_suppression_enabled =
+      capture_.transient_suppressor_enabled;
+  apm_config.intelligibility_enhancer_enabled =
+      capture_nonlocked_.intelligibility_enabled;
+  apm_config.experiments_description = experiments_description;
+  return apm_config;
+}
+
+std::unique_ptr<AecDump::CaptureStreamInfo>
+AudioProcessingImpl::RecordUnprocessedCaptureStream(
+    const float* const* src) const {
+  RTC_DCHECK(aec_dump_);
+  aec_dump_->WriteConfig(CollectApmConfig(), false);
+  auto stream_info = aec_dump_->GetCaptureStreamInfo();
+  RTC_DCHECK(stream_info);
+
+  const size_t channel_size = formats_.api_format.input_stream().num_frames();
+  const size_t num_channels = formats_.api_format.input_stream().num_channels();
+  stream_info->AddInput(FloatAudioFrame(src, num_channels, channel_size));
+  PopulateStreamInfoWithConfig(stream_info.get());
+  return stream_info;
+}
+
+std::unique_ptr<AecDump::CaptureStreamInfo>
+AudioProcessingImpl::RecordUnprocessedCaptureStream(
+    const AudioFrame& capture_frame) const {
+  RTC_DCHECK(aec_dump_);
+  auto stream_info = aec_dump_->GetCaptureStreamInfo();
+  RTC_DCHECK(stream_info);
+
+  stream_info->AddInput(capture_frame);
+  PopulateStreamInfoWithConfig(stream_info.get());
+  aec_dump_->WriteConfig(CollectApmConfig(), false);
+  return stream_info;
+}
+
+void AudioProcessingImpl::RecordProcessedCaptureStream(
+    const float* const* processed_capture_stream,
+    std::unique_ptr<AecDump::CaptureStreamInfo> stream_info) const {
+  RTC_DCHECK(stream_info);
+  RTC_DCHECK(aec_dump_);
+
+  const size_t channel_size = formats_.api_format.output_stream().num_frames();
+  const size_t num_channels =
+      formats_.api_format.output_stream().num_channels();
+  stream_info->AddOutput(
+      FloatAudioFrame(processed_capture_stream, num_channels, channel_size));
+  aec_dump_->WriteCaptureStreamMessage(std::move(stream_info));
+}
+
+void AudioProcessingImpl::RecordProcessedCaptureStream(
+    const AudioFrame& processed_capture_frame,
+    std::unique_ptr<AecDump::CaptureStreamInfo> stream_info) const {
+  RTC_DCHECK(stream_info);
+  RTC_DCHECK(aec_dump_);
+
+  stream_info->AddOutput(processed_capture_frame);
+  aec_dump_->WriteCaptureStreamMessage(std::move(stream_info));
+}
+
+void AudioProcessingImpl::PopulateStreamInfoWithConfig(

[peah-webrtc, 2017/05/04 06:19:08]

This naming is a big ambiguous as we have WriteConfig that stores the
apm_config, and the config referred to in this call is something different.
Furthermore, I don't think it is a configuration that is stored, rather than a
state.

Could this be renamed to make it more clear? 
What about 
PopulateStreamInfoWithState
PopulateStreamInfoMetadata
PopulateStreamInfo

[aleloi, 2017/05/04 08:56:40]

I like PopulateStreamInfoWithState the most. Done.

+    AecDump::CaptureStreamInfo* stream_info) const {
+  RTC_DCHECK(stream_info);
+
+  stream_info->set_delay(capture_nonlocked_.stream_delay_ms);

[peah-webrtc, 2017/05/04 06:19:08]

Would it be possible to instead bundle these 4 setters into a single set_ call
in stream_info that takes 4 arguments? That would be cheaper, would avoid the
need for this method and would make the code in audio_processing_impl.cc
shorter.

[aleloi, 2017/05/04 08:56:40]

I'd rather not. The setter names serve as documentation for what is logged. The
style guide has an example in the 'Function Argument Comments' section
https://google.github.io/styleguide/cppguide.html#Implementation_Comments that
argues against several parameters.

[peah-webrtc, 2017/05/04 10:57:51]

I definitely like the documentation part. But this comes at an additional cost
of up to 180 cycles as these are virtual methods. Please see the further
comments in aec_dump.h about this.

+  stream_info->set_drift(
+      public_submodules_->echo_cancellation->stream_drift_samples());
+  stream_info->set_level(gain_control()->stream_analog_level());
+  stream_info->set_keypress(capture_.key_pressed);
+}
+
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
 int AudioProcessingImpl::WriteMessageToDebugFile(
     FileWrapper* debug_file,