Add mute state field to AudioFrame.

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 642 matching lines @@
   ChannelState::State state = channel_state_.Get();
 
   {
     // Pass the audio buffers to an optional sink callback, before applying
     // scaling/panning, as that applies to the mix operation.
     // External recipients of the audio (e.g. via AudioTrack), will do their
     // own mixing/dynamic processing.
     rtc::CritScope cs(&_callbackCritSect);
     if (audio_sink_) {
       AudioSinkInterface::Data data(
-          &audioFrame->data_[0], audioFrame->samples_per_channel_,
+          audioFrame->data(), audioFrame->samples_per_channel_,
           audioFrame->sample_rate_hz_, audioFrame->num_channels_,
           audioFrame->timestamp_);
       audio_sink_->OnData(data);
     }
   }
 
   float output_gain = 1.0f;
   {
     rtc::CritScope cs(&volume_settings_critsect_);
     output_gain = _outputGain;
@@ skipping 2105 matching lines @@
   if (channel_state_.Get().input_file_playing) {
     MixOrReplaceAudioWithFile(audio_input);
   }
 
   bool is_muted = InputMute();
   AudioFrameOperations::Mute(audio_input, previous_frame_muted_, is_muted);
 
   if (_includeAudioLevelIndication) {
     size_t length =
         audio_input->samples_per_channel_ * audio_input->num_channels_;
-    RTC_CHECK_LE(length, sizeof(audio_input->data_));
+    RTC_CHECK_LE(length, AudioFrame::kMaxDataSizeBytes);
     if (is_muted && previous_frame_muted_) {
       rms_level_.AnalyzeMuted(length);
     } else {
       rms_level_.Analyze(
-          rtc::ArrayView<const int16_t>(audio_input->data_, length));
+          rtc::ArrayView<const int16_t>(audio_input->data(), length));
     }
   }
   previous_frame_muted_ = is_muted;
 
   // Add 10ms of raw (PCM) audio data to the encoder @ 32kHz.
 
   // The ACM resamples internally.
   audio_input->timestamp_ = _timeStamp;
   // This call will trigger AudioPacketizationCallback::SendData if encoding
   // is done and payload is ready for packetization and transmission.
@@ skipping 139 matching lines @@
                    "Channel::MixOrReplaceAudioWithFile() file is ended");
       return 0;
     }
   }
 
   RTC_DCHECK_EQ(audio_input->samples_per_channel_, fileSamples);
 
   if (_mixFileWithMicrophone) {
     // Currently file stream is always mono.
     // TODO(xians): Change the code when FilePlayer supports real stereo.
-    MixWithSat(audio_input->data_, audio_input->num_channels_, fileBuffer.get(),
-               1, fileSamples);
+    MixWithSat(audio_input->mutable_data(), audio_input->num_channels_,
+               fileBuffer.get(), 1, fileSamples);
   } else {
     // Replace ACM audio with file.
     // Currently file stream is always mono.
     // TODO(xians): Change the code when FilePlayer supports real stereo.
     audio_input->UpdateFrame(
         _channelId, 0xFFFFFFFF, fileBuffer.get(), fileSamples, mixingFrequency,
         AudioFrame::kNormalSpeech, AudioFrame::kVadUnknown, 1);
   }
   return 0;
 }
@@ skipping 18 matching lines @@
             fileBuffer.get(), &fileSamples, mixingFrequency) == -1) {
       WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
                    "Channel::MixAudioWithFile() file mixing failed");
       return -1;
     }
   }
 
   if (audioFrame.samples_per_channel_ == fileSamples) {
     // Currently file stream is always mono.
     // TODO(xians): Change the code when FilePlayer supports real stereo.
-    MixWithSat(audioFrame.data_, audioFrame.num_channels_, fileBuffer.get(), 1,
-               fileSamples);
+    MixWithSat(audioFrame.mutable_data(), audioFrame.num_channels_,
+               fileBuffer.get(), 1, fileSamples);
   } else {
     WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
                  "Channel::MixAudioWithFile() samples_per_channel_(%" PRIuS
                  ") != "
                  "fileSamples(%" PRIuS ")",
                  audioFrame.samples_per_channel_, fileSamples);
     return -1;
   }
 
   return 0;
@@ skipping 134 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