Add mute state field to AudioFrame.

webrtc/audio/audio_transport_proxy.cc

Index: webrtc/audio/audio_transport_proxy.cc
diff --git a/webrtc/audio/audio_transport_proxy.cc b/webrtc/audio/audio_transport_proxy.cc
index 4d2f9e30e1c217bae2381b3f75ee27b8aeb5fe85..d6ce9397c71fdd2d597f36754aef0deb45f6b3fa 100644
--- a/webrtc/audio/audio_transport_proxy.cc
+++ b/webrtc/audio/audio_transport_proxy.cc
@@ -25,9 +25,11 @@ int Resample(const AudioFrame& frame,
   resampler->InitializeIfNeeded(frame.sample_rate_hz_, destination_sample_rate,
                                 number_of_channels);
 
+  // TODO(yujo): make resampler take an AudioFrame, and add special case
+  // handling of muted frames.
   return resampler->Resample(
-      frame.data_, frame.samples_per_channel_ * number_of_channels, destination,
-      number_of_channels * target_number_of_samples_per_channel);
+      frame.data(), frame.samples_per_channel_ * number_of_channels,
+      destination, number_of_channels * target_number_of_samples_per_channel);
 }
 }  // namespace
 
@@ -77,7 +79,7 @@ int32_t AudioTransportProxy::NeedMorePlayData(const size_t nSamples,
   // 100 = 1 second / data duration (10 ms).
   RTC_DCHECK_EQ(nSamples * 100, samplesPerSec);
   RTC_DCHECK_LE(nBytesPerSample * nSamples * nChannels,
-                sizeof(AudioFrame::data_));
+                AudioFrame::kMaxDataSizeBytes);
 
   mixer_->Mix(nChannels, &mixed_frame_);
   *elapsed_time_ms = mixed_frame_.elapsed_time_ms_;
@@ -120,7 +122,7 @@ void AudioTransportProxy::PullRenderData(int bits_per_sample,
 
   // 8 = bits per byte.
   RTC_DCHECK_LE(bits_per_sample / 8 * number_of_frames * number_of_channels,
-                sizeof(AudioFrame::data_));
+                AudioFrame::kMaxDataSizeBytes);
   mixer_->Mix(number_of_channels, &mixed_frame_);
   *elapsed_time_ms = mixed_frame_.elapsed_time_ms_;
   *ntp_time_ms = mixed_frame_.ntp_time_ms_;