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

webrtc/audio/audio_receive_stream.cc

 /*
  *  Copyright (c) 2015 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.
  */
 
 #include "webrtc/audio/audio_receive_stream.h"
 
 #include <string>
 
+#include "webrtc/audio/audio_state.h"
 #include "webrtc/audio/conversion.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/include/voe_codec.h"
 #include "webrtc/voice_engine/include/voe_neteq_stats.h"
 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
 #include "webrtc/voice_engine/include/voe_video_sync.h"
@@ skipping 28 matching lines @@
     ss << ", sync_group: " << sync_group;
   }
   ss << ", combined_audio_video_bwe: "
      << (combined_audio_video_bwe ? "true" : "false");
   ss << '}';
   return ss.str();
 }
 
 namespace internal {
 AudioReceiveStream::AudioReceiveStream(
-      RemoteBitrateEstimator* remote_bitrate_estimator,
-      const webrtc::AudioReceiveStream::Config& config,
-      VoiceEngine* voice_engine)
+    RemoteBitrateEstimator* remote_bitrate_estimator,
+    const webrtc::AudioReceiveStream::Config& config,
+    const rtc::scoped_refptr<webrtc::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_.get());
+  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);
     RTC_DCHECK_LE(ext.id, 14);
     if (ext.name == RtpExtension::kAudioLevel) {
       RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension(
           kRtpExtensionAudioLevel, ext.id));
     } else if (ext.name == RtpExtension::kAbsSendTime) {
       RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension(
           kRtpExtensionAbsoluteSendTime, ext.id));
     } else if (ext.name == RtpExtension::kTransportSequenceNumber) {
       RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension(
           kRtpExtensionTransportSequenceNumber, ext.id));
     } else {
       RTC_NOTREACHED() << "Unsupported RTP extension.";
     }
   }
 }
 
 AudioReceiveStream::~AudioReceiveStream() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString();
 }
 
 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_);
+  internal::AudioState* audio_state =
+      static_cast<internal::AudioState*>(audio_state_.get());
+  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};
   // Only collect stats if we have seen some traffic with the SSRC.
   if (rtp->GetRemoteSSRC(config_.voe_channel_id, ssrc) == -1 ||
       rtp->GetRTCPStatistics(config_.voe_channel_id, call_stats) == -1 ||
       codec->GetRecCodec(config_.voe_channel_id, codec_inst) == -1) {
     return stats;
   }
 
   stats.bytes_rcvd = call_stats.bytesReceived;
   stats.packets_rcvd = call_stats.packetsReceived;
   stats.packets_lost = call_stats.cumulativeLost;
   stats.fraction_lost = Q8ToFloat(call_stats.fractionLost);
   if (codec_inst.pltype != -1) {
     stats.codec_name = codec_inst.plname;
   }
   stats.ext_seqnum = call_stats.extendedMax;
   if (codec_inst.plfreq / 1000 > 0) {
     stats.jitter_ms = call_stats.jitterSamples / (codec_inst.plfreq / 1000);
   }
   {
     int jitter_buffer_delay_ms = 0;
     int playout_buffer_delay_ms = 0;
     sync->GetDelayEstimate(config_.voe_channel_id, &jitter_buffer_delay_ms,
                            &playout_buffer_delay_ms);
-    stats.delay_estimate_ms =
-        jitter_buffer_delay_ms + playout_buffer_delay_ms;
+    stats.delay_estimate_ms = jitter_buffer_delay_ms + playout_buffer_delay_ms;
   }
   {
     unsigned int level = 0;
-    if (volume->GetSpeechOutputLevelFullRange(config_.voe_channel_id, level)
-        != -1) {
+    if (volume->GetSpeechOutputLevelFullRange(config_.voe_channel_id, level) !=
+        -1) {
       stats.audio_level = static_cast<int32_t>(level);
     }
   }
 
   webrtc::NetworkStatistics ns = {0};
   if (neteq->GetNetworkStatistics(config_.voe_channel_id, ns) != -1) {
     // Get jitter buffer and total delay (alg + jitter + playout) stats.
     stats.jitter_buffer_ms = ns.currentBufferSize;
     stats.jitter_buffer_preferred_ms = ns.preferredBufferSize;
     stats.expand_rate = Q14ToFloat(ns.currentExpandRate);
     stats.speech_expand_rate = Q14ToFloat(ns.currentSpeechExpandRate);
     stats.secondary_decoded_rate = Q14ToFloat(ns.currentSecondaryDecodedRate);
     stats.accelerate_rate = Q14ToFloat(ns.currentAccelerateRate);
     stats.preemptive_expand_rate = Q14ToFloat(ns.currentPreemptiveRate);
   }
 
   webrtc::AudioDecodingCallStats ds;
   if (neteq->GetDecodingCallStatistics(config_.voe_channel_id, &ds) != -1) {
-    stats.decoding_calls_to_silence_generator =
-        ds.calls_to_silence_generator;
+    stats.decoding_calls_to_silence_generator = ds.calls_to_silence_generator;
     stats.decoding_calls_to_neteq = ds.calls_to_neteq;
     stats.decoding_normal = ds.decoded_normal;
     stats.decoding_plc = ds.decoded_plc;
     stats.decoding_cng = ds.decoded_cng;
     stats.decoding_plc_cng = ds.decoded_plc_cng;
   }
 
   stats.capture_start_ntp_time_ms = call_stats.capture_start_ntp_time_ms_;
 
   return stats;
@@ skipping 45 matching lines @@
     if (packet_time.timestamp >= 0)
       arrival_time_ms = (packet_time.timestamp + 500) / 1000;
     size_t payload_size = length - header.headerLength;
     remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_size,
                                               header, false);
   }
   return true;
 }
 }  // namespace internal
 }  // namespace webrtc