commit 09ceed2165137c4bea4e02e8d3db31970d0bf273
author Olga Sharonova <olka@webrtc.org> Wed Sep 30 16:27:39 2020
committer Commit Bot <commit-bot@chromium.org> Fri Oct 02 12:33:34 2020
tree c3bb9fdd930fe9977bd2f3c34726e33cee9ee4c3
parent eb79dd9ffdc41e4ca86803bfc1317e0961a8a8a6

Async audio processing API

API to injecting a heavy audio processing operation into WebRTC audio capture pipeline

Bug: webrtc:12003
Change-Id: I9f6f58f468bd84efd0a9d53d703db6229a03959e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165788
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32291}

audio/audio_transport_impl.h

diff --git a/audio/audio_transport_impl.h b/audio/audio_transport_impl.h
index 5b885bd..f3ca2fa 100644
--- a/audio/audio_transport_impl.h
+++ b/audio/audio_transport_impl.h
@@ -11,11 +11,13 @@
 #ifndef AUDIO_AUDIO_TRANSPORT_IMPL_H_
 #define AUDIO_AUDIO_TRANSPORT_IMPL_H_
 
+#include <memory>
 #include <vector>
 
 #include "api/audio/audio_mixer.h"
 #include "api/scoped_refptr.h"
 #include "common_audio/resampler/include/push_resampler.h"
+#include "modules/async_audio_processing/async_audio_processing.h"
 #include "modules/audio_device/include/audio_device.h"
 #include "modules/audio_processing/include/audio_processing.h"
 #include "modules/audio_processing/typing_detection.h"
@@ -28,7 +30,10 @@
 
 class AudioTransportImpl : public AudioTransport {
  public:
-  AudioTransportImpl(AudioMixer* mixer, AudioProcessing* audio_processing);
+  AudioTransportImpl(
+      AudioMixer* mixer,
+      AudioProcessing* audio_processing,
+      AsyncAudioProcessing::Factory* async_audio_processing_factory);
 
   AudioTransportImpl() = delete;
   AudioTransportImpl(const AudioTransportImpl&) = delete;
@@ -71,10 +76,16 @@
   bool typing_noise_detected() const;
 
  private:
+  void SendProcessedData(std::unique_ptr<AudioFrame> audio_frame);
+
   // Shared.
   AudioProcessing* audio_processing_ = nullptr;
 
   // Capture side.
+
+  // Thread-safe.
+  const std::unique_ptr<AsyncAudioProcessing> async_audio_processing_;
+
   mutable Mutex capture_lock_;
   std::vector<AudioSender*> audio_senders_ RTC_GUARDED_BY(capture_lock_);
   int send_sample_rate_hz_ RTC_GUARDED_BY(capture_lock_) = 8000;
@@ -85,6 +96,7 @@
   TypingDetection typing_detection_;
 
   // Render side.
+
   rtc::scoped_refptr<AudioMixer> mixer_;
   AudioFrame mixed_frame_;
   // Converts mixed audio to the audio device output rate.