Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly i…

webrtc/audio/audio_receive_stream.cc

Index: webrtc/audio/audio_receive_stream.cc
diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc
index 34197c3ff7fd011b02864e91dc9c6e29d425753f..4621f7ed9ebcfa087055f1774f671293fa5feb64 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -12,6 +12,7 @@
 
 #include <string>
 
+#include "webrtc/audio/audio_state.h"
 #include "webrtc/audio/conversion.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
@@ -62,17 +63,16 @@ namespace internal {
 AudioReceiveStream::AudioReceiveStream(
       RemoteBitrateEstimator* remote_bitrate_estimator,
       const webrtc::AudioReceiveStream::Config& config,
-      VoiceEngine* voice_engine)
+      AudioState* audio_state)
     : remote_bitrate_estimator_(remote_bitrate_estimator),
       config_(config),
-      voice_engine_(voice_engine),
-      voe_base_(voice_engine),
+      audio_state_(audio_state),
       rtp_header_parser_(RtpHeaderParser::Create()) {
   LOG(LS_INFO) << "AudioReceiveStream: " << config_.ToString();
-  RTC_DCHECK(config.voe_channel_id != -1);
-  RTC_DCHECK(remote_bitrate_estimator_ != nullptr);
-  RTC_DCHECK(voice_engine_ != nullptr);
-  RTC_DCHECK(rtp_header_parser_ != nullptr);
+  RTC_DCHECK_NE(config_.voe_channel_id, -1);
+  RTC_DCHECK(remote_bitrate_estimator_);
+  RTC_DCHECK(audio_state_);
+  RTC_DCHECK(rtp_header_parser_);
   for (const auto& ext : config.rtp.extensions) {
     // One-byte-extension local identifiers are in the range 1-14 inclusive.
     RTC_DCHECK_GE(ext.id, 1);
@@ -101,11 +101,12 @@ webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   webrtc::AudioReceiveStream::Stats stats;
   stats.remote_ssrc = config_.rtp.remote_ssrc;
-  ScopedVoEInterface<VoECodec> codec(voice_engine_);
-  ScopedVoEInterface<VoENetEqStats> neteq(voice_engine_);
-  ScopedVoEInterface<VoERTP_RTCP> rtp(voice_engine_);
-  ScopedVoEInterface<VoEVideoSync> sync(voice_engine_);
-  ScopedVoEInterface<VoEVolumeControl> volume(voice_engine_);
+  VoiceEngine* voice_engine = audio_state_->voice_engine();
+  ScopedVoEInterface<VoECodec> codec(voice_engine);
+  ScopedVoEInterface<VoENetEqStats> neteq(voice_engine);
+  ScopedVoEInterface<VoERTP_RTCP> rtp(voice_engine);
+  ScopedVoEInterface<VoEVideoSync> sync(voice_engine);
+  ScopedVoEInterface<VoEVolumeControl> volume(voice_engine);
   unsigned int ssrc = 0;
   webrtc::CallStatistics call_stats = {0};
   webrtc::CodecInst codec_inst = {0};