Implementation of new AecDump interface.

webrtc/modules/audio_processing/audio_processing_impl.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 825 matching lines @@
     debug_dump_.capture.event_msg->set_type(audioproc::Event::STREAM);
     audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream();
     const size_t channel_size =
         sizeof(float) * formats_.api_format.input_stream().num_frames();
     for (size_t i = 0; i < formats_.api_format.input_stream().num_channels();
          ++i)
       msg->add_input_channel(src[i], channel_size);
   }
 #endif
 
-  std::unique_ptr<AecDump::CaptureStreamInfo> stream_info;
+  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);
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   if (debug_dump_.debug_file->is_open()) {
     audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream();
     const size_t channel_size =
         sizeof(float) * formats_.api_format.output_stream().num_frames();
     for (size_t i = 0; i < formats_.api_format.output_stream().num_channels();
          ++i)
       msg->add_output_channel(dest[i], channel_size);
     RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
                                           &debug_dump_.num_bytes_left_for_log_,
                                           &crit_debug_, &debug_dump_.capture));
   }
 #endif
   if (aec_dump_) {
-    RecordProcessedCaptureStream(dest, std::move(stream_info));
+    RecordProcessedCaptureStream(dest, stream_info);
   }
   return kNoError;
 }
 
 void AudioProcessingImpl::QueueRenderAudio(AudioBuffer* audio) {
   EchoCancellationImpl::PackRenderAudioBuffer(audio, num_output_channels(),
                                               num_reverse_channels(),
                                               &aec_render_queue_buffer_);
 
   RTC_DCHECK_GE(160, audio->num_frames_per_band());
@@ skipping 217 matching lines @@
     rtc::CritScope cs_render(&crit_render_);
     RETURN_ON_ERR(
         MaybeInitializeCapture(processing_config, reinitialization_required));
   }
   rtc::CritScope cs_capture(&crit_capture_);
   if (frame->samples_per_channel_ !=
       formats_.api_format.input_stream().num_frames()) {
     return kBadDataLengthError;
   }
 
-  std::unique_ptr<AecDump::CaptureStreamInfo> stream_info;
+  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));
 
     debug_dump_.capture.event_msg->set_type(audioproc::Event::STREAM);
     audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream();
     const size_t data_size =
         sizeof(int16_t) * frame->samples_per_channel_ * frame->num_channels_;
     msg->set_input_data(frame->data_, data_size);
   }
 #endif
 
   capture_.capture_audio->DeinterleaveFrom(frame);
   RETURN_ON_ERR(ProcessCaptureStreamLocked());
   capture_.capture_audio->InterleaveTo(
       frame, submodule_states_.CaptureMultiBandProcessingActive());
 
   if (aec_dump_) {
-    RecordProcessedCaptureStream(*frame, std::move(stream_info));
+    RecordProcessedCaptureStream(*frame, stream_info);
   }
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   if (debug_dump_.debug_file->is_open()) {
     audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream();
     const size_t data_size =
         sizeof(int16_t) * frame->samples_per_channel_ * frame->num_channels_;
     msg->set_output_data(frame->data_, data_size);
     RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
                                           &debug_dump_.num_bytes_left_for_log_,
                                           &crit_debug_, &debug_dump_.capture));
@@ skipping 806 matching lines @@
       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(
+AecDump::CaptureStreamInfo* AudioProcessingImpl::RecordUnprocessedCaptureStream(
     const float* const* src) const {
   RTC_DCHECK(aec_dump_);
   aec_dump_->WriteConfig(CollectApmConfig(), false);
-  auto stream_info = aec_dump_->CreateCaptureStreamInfo();
+  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));
-  PopulateStreamInfoWithState(stream_info.get());
+  PopulateStreamInfoWithState(stream_info);
   return stream_info;
 }
 
-std::unique_ptr<AecDump::CaptureStreamInfo>
-AudioProcessingImpl::RecordUnprocessedCaptureStream(
+AecDump::CaptureStreamInfo* AudioProcessingImpl::RecordUnprocessedCaptureStream(
     const AudioFrame& capture_frame) const {
   RTC_DCHECK(aec_dump_);
-  auto stream_info = aec_dump_->CreateCaptureStreamInfo();
+  auto* stream_info = aec_dump_->GetCaptureStreamInfo();
   RTC_DCHECK(stream_info);
 
   stream_info->AddInput(capture_frame);
-  PopulateStreamInfoWithState(stream_info.get());
+  PopulateStreamInfoWithState(stream_info);
   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 {
+    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));
+  aec_dump_->WriteCaptureStreamMessage();
 }
 
 void AudioProcessingImpl::RecordProcessedCaptureStream(
     const AudioFrame& processed_capture_frame,
-    std::unique_ptr<AecDump::CaptureStreamInfo> stream_info) const {
+    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));
+  aec_dump_->WriteCaptureStreamMessage();
 }
 
 void AudioProcessingImpl::PopulateStreamInfoWithState(
     AecDump::CaptureStreamInfo* stream_info) const {
   RTC_DCHECK(stream_info);
 
   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());
@@ skipping 166 matching lines @@
       previous_agc_level(0),
       echo_path_gain_change(false) {}
 
 AudioProcessingImpl::ApmCaptureState::~ApmCaptureState() = default;
 
 AudioProcessingImpl::ApmRenderState::ApmRenderState() = default;
 
 AudioProcessingImpl::ApmRenderState::~ApmRenderState() = default;
 
 }  // namespace webrtc