Made AudioReceiveStream a mixer participant.

Methods to facilitate this are added to ChannelProxy and voe::Channel.

BUG=webrtc:6346

Review-Url: https://codereview.webrtc.org/2378143004
Cr-Commit-Position: refs/heads/master@{#14707}

webrtc/audio/BUILD.gn

@@ -27,6 +27,7 @@ rtc_static_library("audio") {
 
   deps = [
     "..:webrtc_common",
+    "../api:audio_mixer_api",
     "../api:call_api",
     "../system_wrappers",
     "../voice_engine",

webrtc/audio/audio_receive_stream.cc

@@ -272,6 +272,15 @@ bool AudioReceiveStream::DeliverRtp(const uint8_t* packet,
   return channel_proxy_->ReceivedRTPPacket(packet, length, packet_time);
 }
 
+AudioMixer::Source::AudioFrameWithInfo
+AudioReceiveStream::GetAudioFrameWithInfo(int sample_rate_hz) {
+  return channel_proxy_->GetAudioFrameWithInfo(sample_rate_hz);
+}
+
+int AudioReceiveStream::Ssrc() {
+  return config_.rtp.local_ssrc;
+}
+
 VoiceEngine* AudioReceiveStream::voice_engine() const {
   internal::AudioState* audio_state =
       static_cast<internal::AudioState*>(audio_state_.get());

webrtc/audio/audio_receive_stream.h

@@ -13,6 +13,7 @@
 
 #include <memory>
 
+#include "webrtc/api/audio/audio_mixer.h"
 #include "webrtc/api/call/audio_receive_stream.h"
 #include "webrtc/api/call/audio_state.h"
 #include "webrtc/base/constructormagic.h"
@@ -30,7 +31,8 @@ class ChannelProxy;
 
 namespace internal {
 
-class AudioReceiveStream final : public webrtc::AudioReceiveStream {
+class AudioReceiveStream final : public webrtc::AudioReceiveStream,
+                                 public AudioMixer::Source {
  public:
   AudioReceiveStream(CongestionController* congestion_controller,
                      const webrtc::AudioReceiveStream::Config& config,
@@ -52,6 +54,10 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream {
                   const PacketTime& packet_time);
   const webrtc::AudioReceiveStream::Config& config() const;
 
+  // AudioMixer::Source
+  AudioFrameWithInfo GetAudioFrameWithInfo(int sample_rate_hz) override;
+  int Ssrc() override;
+
  private:
   VoiceEngine* voice_engine() const;
 

webrtc/audio/webrtc_audio.gypi

@@ -8,6 +8,7 @@
 {
   'variables': {
     'webrtc_audio_dependencies': [
+      '<(webrtc_root)/api/api.gyp:audio_mixer_api',
       '<(webrtc_root)/api/api.gyp:call_api',
       '<(webrtc_root)/common.gyp:webrtc_common',
       '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',

webrtc/voice_engine/BUILD.gn

@@ -87,6 +87,7 @@ rtc_static_library("voice_engine") {
   deps = [
     ":level_indicator",
     "..:webrtc_common",
+    "../api:audio_mixer_api",
     "../api:call_api",
     "../base:rtc_base_approved",
     "../common_audio",

webrtc/voice_engine/channel.cc

@@ -712,6 +712,28 @@ MixerParticipant::AudioFrameInfo Channel::GetAudioFrameWithMuted(
                : MixerParticipant::AudioFrameInfo::kNormal;
 }
 
+AudioMixer::Source::AudioFrameWithInfo Channel::GetAudioFrameWithInfo(
+    int sample_rate_hz) {
+  mix_audio_frame_.sample_rate_hz_ = sample_rate_hz;
+
+  const auto frame_info = GetAudioFrameWithMuted(-1, &mix_audio_frame_);
+
+  using FrameInfo = AudioMixer::Source::AudioFrameInfo;
+  FrameInfo new_audio_frame_info = FrameInfo::kError;
+  switch (frame_info) {
+    case MixerParticipant::AudioFrameInfo::kNormal:
+      new_audio_frame_info = FrameInfo::kNormal;
+      break;
+    case MixerParticipant::AudioFrameInfo::kMuted:
+      new_audio_frame_info = FrameInfo::kMuted;
+      break;
+    case MixerParticipant::AudioFrameInfo::kError:
+      new_audio_frame_info = FrameInfo::kError;
+      break;
+  }
+  return {&mix_audio_frame_, new_audio_frame_info};
+}
+
 int32_t Channel::NeededFrequency(int32_t id) const {
   WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::NeededFrequency(id=%d)", id);

webrtc/voice_engine/channel.h

@@ -13,6 +13,7 @@
 
 #include <memory>
 
+#include "webrtc/api/audio/audio_mixer.h"
 #include "webrtc/api/call/audio_sink.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/optional.h"
@@ -377,6 +378,10 @@ class Channel
       AudioFrame* audioFrame) override;
   int32_t NeededFrequency(int32_t id) const override;
 
+  // From AudioMixer::Source.
+  AudioMixer::Source::AudioFrameWithInfo GetAudioFrameWithInfo(
+      int sample_rate_hz);
+
   // From FileCallback
   void PlayNotification(int32_t id, uint32_t durationMs) override;
   void RecordNotification(int32_t id, uint32_t durationMs) override;
@@ -470,6 +475,7 @@ class Channel
   AudioLevel _outputAudioLevel;
   bool _externalTransport;
   AudioFrame _audioFrame;
+  AudioFrame mix_audio_frame_;
   // Downsamples to the codec rate if necessary.
   PushResampler<int16_t> input_resampler_;
   std::unique_ptr<FilePlayer> input_file_player_;

webrtc/voice_engine/channel_proxy.cc

@@ -214,6 +214,12 @@ void ChannelProxy::SetRtcEventLog(RtcEventLog* event_log) {
   channel()->SetRtcEventLog(event_log);
 }
 
+AudioMixer::Source::AudioFrameWithInfo ChannelProxy::GetAudioFrameWithInfo(
+    int sample_rate_hz) {
+  RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
+  return channel()->GetAudioFrameWithInfo(sample_rate_hz);
+}
+
 Channel* ChannelProxy::channel() const {
   RTC_DCHECK(channel_owner_.channel());
   return channel_owner_.channel();

webrtc/voice_engine/channel_proxy.h

@@ -11,7 +11,9 @@
 #ifndef WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_
 #define WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_
 
+#include "webrtc/api/audio/audio_mixer.h"
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/base/race_checker.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/voice_engine/channel_manager.h"
 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
@@ -91,10 +93,14 @@ class ChannelProxy {
 
   virtual void SetRtcEventLog(RtcEventLog* event_log);
 
+  virtual AudioMixer::Source::AudioFrameWithInfo GetAudioFrameWithInfo(
+      int sample_rate_hz);
+
  private:
   Channel* channel() const;
 
   rtc::ThreadChecker thread_checker_;
+  rtc::RaceChecker race_checker_;
   ChannelOwner channel_owner_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(ChannelProxy);

webrtc/voice_engine/voice_engine.gyp

@@ -15,6 +15,7 @@
       'target_name': 'voice_engine',
       'type': 'static_library',
       'dependencies': [
+        '<(webrtc_root)/api/api.gyp:audio_mixer_api',
         '<(webrtc_root)/api/api.gyp:call_api',
         '<(webrtc_root)/base/base.gyp:rtc_base_approved',
         '<(webrtc_root)/common.gyp:webrtc_common',