Propagate muted info from VoE Channel to AudioConferenceMixer

webrtc/voice_engine/channel.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 458 matching lines @@
                  "IncomingPacket invalid RTP header");
     return false;
   }
   header.payload_type_frequency =
       rtp_payload_registry_->GetPayloadTypeFrequency(header.payloadType);
   if (header.payload_type_frequency < 0)
     return false;
   return ReceivePacket(rtp_packet, rtp_packet_length, header, false);
 }
 
-int32_t Channel::GetAudioFrame(int32_t id, AudioFrame* audioFrame) {
+MixerParticipant::AudioFrameInfo Channel::GetAudioFrameWithMuted(
+    int32_t id,
+    AudioFrame* audioFrame) {
   if (event_log_) {
     unsigned int ssrc;
     RTC_CHECK_EQ(GetLocalSSRC(ssrc), 0);
     event_log_->LogAudioPlayout(ssrc);
   }
   // Get 10ms raw PCM data from the ACM (mixer limits output frequency)
   bool muted;
   if (audio_coding_->PlayoutData10Ms(audioFrame->sample_rate_hz_, audioFrame,
                                      &muted) == -1) {
     WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, _channelId),
                  "Channel::GetAudioFrame() PlayoutData10Ms() failed!");
     // In all likelihood, the audio in this frame is garbage. We return an
     // error so that the audio mixer module doesn't add it to the mix. As
     // a result, it won't be played out and the actions skipped here are
     // irrelevant.
-    return -1;
+    return MixerParticipant::AudioFrameInfo::kError;
   }
   RTC_DCHECK(!muted);
 
   if (_RxVadDetection) {
     UpdateRxVadDetection(*audioFrame);
   }
 
   // Convert module ID to internal VoE channel ID
   audioFrame->id_ = VoEChannelId(audioFrame->id_);
   // Store speech type for dead-or-alive detection
@@ skipping 108 matching lines @@
       // |ntp_time_ms_| won't be valid until at least 2 RTCP SRs are received.
       if (audioFrame->ntp_time_ms_ > 0) {
         // Compute |capture_start_ntp_time_ms_| so that
         // |capture_start_ntp_time_ms_| + |elapsed_time_ms_| == |ntp_time_ms_|
         capture_start_ntp_time_ms_ =
             audioFrame->ntp_time_ms_ - audioFrame->elapsed_time_ms_;
       }
     }
   }
 
-  return 0;
+  return muted ? MixerParticipant::AudioFrameInfo::kMuted
+               : MixerParticipant::AudioFrameInfo::kNormal;
 }
 
 int32_t Channel::NeededFrequency(int32_t id) const {
   WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::NeededFrequency(id=%d)", id);
 
   int highestNeeded = 0;
 
   // Determine highest needed receive frequency
   int32_t receiveFrequency = audio_coding_->ReceiveFrequency();
@@ skipping 2913 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc