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.

[the sun, 2017/03/23 19:34:38]

Note: That would mean the resampler needs to maintain the muted state also, and
possibly run for a while on zeroed input, to allow IR and filter state to
converge...

[yujo, 2017/03/24 07:30:14]

I'm not clear on how the resampler works, but maybe it's best to let it work
through the muted frames normally, then.

I think the important thing is actually to have it take an audio frame and write
an audio frame, and to make sure that muted input always gives muted output.
Does that sound right?

   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_;