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..8ae7005b0a2a6ba76bd4941d3c6081d07c0c7ecd 100644
--- a/webrtc/audio/audio_transport_proxy.cc
+++ b/webrtc/audio/audio_transport_proxy.cc
@@ -25,9 +25,13 @@ int Resample(const AudioFrame& frame,
   resampler->InitializeIfNeeded(frame.sample_rate_hz_, destination_sample_rate,
                                 number_of_channels);
 
+  // TODO(yujo): for muted input frames, don't resample. Either 1) allow

[hlundin-webrtc, 2017/03/16 14:47:48]

I would suggest that this function simply writes the correct number of zeros to
destination if frame is muted.

EDIT: This might not work, since you would then not be priming the resampler
filters with the correct data (i.e., it will be missing out on the zero
samples). I will have to think about his.

[yujo, 2017/03/16 23:37:21]

I think the correct answer is to push AudioFrame deeper, what do you think?
(Also sorry, the TODO made no sense. I didn't see that destination_sample_rate
was available as a local.)

[hlundin-webrtc, 2017/03/17 14:29:38]

I think the TODO now is good enough for this CL. I agree that we should push
AudioFrame deeper later on.

+  // resampler to return output length without doing the resample, so we know
+  // how much to zero here; or 2) make resampler accept a hint that the input is
+  // zeroed.
   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 +81,7 @@ 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 +124,7 @@ 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_;