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 88229b4829aea7f0a4a9e68893c96f643e7944d1..fccfd4c8e6ef66c039a5e428ffd999f4dfe9ad2f 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -305,6 +305,7 @@ AudioProcessingImpl::AudioProcessingImpl(const webrtc::Config& config)
 AudioProcessingImpl::AudioProcessingImpl(const webrtc::Config& config,
                                          NonlinearBeamformer* beamformer)
     : high_pass_filter_impl_(new HighPassFilterImpl(this)),
+      aec_dump_(AecDump::CreateNullDump()),
       public_submodules_(new ApmPublicSubmodules()),
       private_submodules_(new ApmPrivateSubmodules(beamformer)),
       constants_(config.Get<ExperimentalAgc>().startup_min_volume,
@@ -525,7 +526,7 @@ int AudioProcessingImpl::InitializeLocked() {
     }
   }
 #endif
-
+  aec_dump_->WriteInitMessage(formats_.api_format);
   return kNoError;
 }
 
@@ -816,7 +817,30 @@ int AudioProcessingImpl::ProcessStream(const float* const* src,
   }
 #endif
 
+  std::unique_ptr<AecDump::CaptureStreamInfo> stream_info =

[peah-webrtc, 2017/04/07 12:57:15]

I definitely like the aspect of the NullImplementation that it does not require
conditionals. The extra cost is, however, quite significant.

Between lines 820 and 870, for the case where no aecdumping is active
this variant actually adds: 
-9 virtual function calls
-6 ordinary function calls
-2 copies of all the data in the capture buffer.
Only the function calls are about 700 cycles. Some of these can maybe be inlined
by the compiler but as there still are the extra copies of data I don't think
this overhead warrants the not having to use conditionals.

[aleloi, 2017/04/12 11:05:29]

I agree. Now everything is wrapped with if (aec_dump_)

+      aec_dump_->GetCaptureStreamInfo();
+  const size_t channel_size =
+      sizeof(float) * formats_.api_format.input_stream().num_frames();
+
+  {
+    std::vector<rtc::ArrayView<const float>> src_view;
+    for (size_t i = 0; i < formats_.api_format.input_stream().num_channels();
+         ++i) {
+      src_view.emplace_back(src[i], channel_size);
+    }
+    stream_info->AddInput(src_view);
+  }
   capture_.capture_audio->CopyFrom(src, formats_.api_format.input_stream());
+
+  RTC_DCHECK(!(public_submodules_->echo_cancellation->is_enabled() &&
+               public_submodules_->echo_control_mobile->is_enabled()));
+
+  stream_info->set_delay(capture_nonlocked_.stream_delay_ms);
+  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);
+
   RETURN_ON_ERR(ProcessCaptureStreamLocked());
   capture_.capture_audio->CopyTo(formats_.api_format.output_stream(), dest);
 
@@ -833,6 +857,17 @@ int AudioProcessingImpl::ProcessStream(const float* const* src,
                                           &crit_debug_, &debug_dump_.capture));
   }
 #endif
+  {
+    const size_t channel_size =
+        sizeof(float) * formats_.api_format.output_stream().num_frames();
+    std::vector<rtc::ArrayView<const float>> dest_view;
+    for (size_t i = 0; i < formats_.api_format.output_stream().num_channels();
+         ++i) {
+      dest_view.emplace_back(dest[i], channel_size);
+    }
+    stream_info->AddOutput(dest_view);
+  }
+  aec_dump_->WriteCaptureStreamMessage(std::move(stream_info));
 
   return kNoError;
 }
@@ -1070,6 +1105,9 @@ int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
     return kBadDataLengthError;
   }
 
+  std::unique_ptr<AecDump::CaptureStreamInfo> stream_info =
+      aec_dump_->GetCaptureStreamInfo();

[peah-webrtc, 2017/04/07 12:57:15]

Same as above.

[aleloi, 2017/04/12 11:05:29]

Acknowledged.

+  stream_info->AddInput(*frame);
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   if (debug_dump_.debug_file->is_open()) {
     RETURN_ON_ERR(WriteConfigMessage(false));
@@ -1083,10 +1121,22 @@ int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
 #endif
 
   capture_.capture_audio->DeinterleaveFrom(frame);
+
+  RTC_DCHECK(!(public_submodules_->echo_cancellation->is_enabled() &&
+               public_submodules_->echo_control_mobile->is_enabled()));
+
+  stream_info->set_delay(capture_nonlocked_.stream_delay_ms);
+  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);
+
   RETURN_ON_ERR(ProcessCaptureStreamLocked());
   capture_.capture_audio->InterleaveTo(
       frame, submodule_states_.CaptureMultiBandProcessingActive());
 
+  stream_info->AddOutput(*frame);
+  aec_dump_->WriteCaptureStreamMessage(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();
@@ -1363,6 +1413,15 @@ int AudioProcessingImpl::AnalyzeReverseStreamLocked(
                                           &crit_debug_, &debug_dump_.render));
   }
 #endif
+  std::vector<rtc::ArrayView<const float>> src_view;
+  const size_t channel_size =

[peah-webrtc, 2017/04/07 12:57:15]

Same as above.

[aleloi, 2017/04/12 11:05:29]

Acknowledged.

+      sizeof(float) * formats_.api_format.reverse_input_stream().num_frames();
+
+  for (size_t i = 0;
+       i < formats_.api_format.reverse_input_stream().num_channels(); ++i) {
+    src_view.emplace_back(src[i], channel_size);
+  }
+  aec_dump_->WriteRenderStreamMessage(src_view);
 
   render_.render_audio->CopyFrom(src,
                                  formats_.api_format.reverse_input_stream());
@@ -1416,6 +1475,8 @@ int AudioProcessingImpl::ProcessReverseStream(AudioFrame* frame) {
                                           &crit_debug_, &debug_dump_.render));
   }
 #endif
+  aec_dump_->WriteRenderStreamMessage(*frame);
+
   render_.render_audio->DeinterleaveFrom(frame);
   RETURN_ON_ERR(ProcessRenderStreamLocked());
   render_.render_audio->InterleaveTo(
@@ -1499,6 +1560,20 @@ 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_);
+  aec_dump_ = std::move(aec_dump);
+
+#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
+  const int error = WriteConfigMessage(true);
+  RTC_DCHECK(error);
+#endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
+
+  aec_dump_->WriteInitMessage(formats_.api_format);
+}
+
 int AudioProcessingImpl::StartDebugRecording(
     const char filename[AudioProcessing::kMaxFilenameSize],
     int64_t max_log_size_bytes) {
@@ -1511,6 +1586,7 @@ int AudioProcessingImpl::StartDebugRecording(
     return kNullPointerError;
   }
 
+  aec_dump_->WriteInitMessage(formats_.api_format);

[peah-webrtc, 2017/04/07 12:57:15]

Is this needed? Nothing is anyway written until the new StartDebugRecording is
called. And at that point, the init message is written as well.

[aleloi, 2017/04/12 11:05:29]

Good point. Removed.

 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   debug_dump_.num_bytes_left_for_log_ = max_log_size_bytes;
   // Stop any ongoing recording.
@@ -1538,6 +1614,7 @@ int AudioProcessingImpl::StartDebugRecording(FILE* handle,
     return kNullPointerError;
   }
 
+  aec_dump_->WriteInitMessage(formats_.api_format);

[peah-webrtc, 2017/04/07 12:57:15]

Same as above.

[aleloi, 2017/04/12 11:05:29]

Acknowledged.

 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   debug_dump_.num_bytes_left_for_log_ = max_log_size_bytes;
 
@@ -1573,6 +1650,7 @@ int AudioProcessingImpl::StopDebugRecording() {
   // Run in a single-threaded manner.
   rtc::CritScope cs_render(&crit_render_);
   rtc::CritScope cs_capture(&crit_capture_);
+  aec_dump_ = AecDump::CreateNullDump();
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   // We just return if recording hasn't started.
@@ -1937,6 +2015,43 @@ int AudioProcessingImpl::WriteConfigMessage(bool forced) {
   config.set_intelligibility_enhancer_enabled(
       capture_nonlocked_.intelligibility_enabled);
 
+  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;
+
   std::string experiments_description =
       public_submodules_->echo_cancellation->GetExperimentsDescription();
   // TODO(peah): Add semicolon-separated concatenations of experiment
@@ -1952,6 +2067,9 @@ int AudioProcessingImpl::WriteConfigMessage(bool forced) {
   }
   config.set_experiments_description(experiments_description);
 
+  apm_config.experiments_description = experiments_description;
+  aec_dump_->WriteConfig(apm_config, forced);
+
   std::string serialized_config = config.SerializeAsString();
   if (!forced &&
       debug_dump_.capture.last_serialized_config == serialized_config) {