Replace AudioConferenceMixer with AudioMixer.

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 ed72200379d1d07fab29e04482b761ea732cb6e2..b372ead4ffe704c6966a70c09011b5e3042d47c7 100644
--- a/webrtc/audio/audio_transport_proxy.cc
+++ b/webrtc/audio/audio_transport_proxy.cc
@@ -12,12 +12,32 @@
 
 namespace webrtc {
 
+namespace {
+// Resample audio in |frame| to given sample rate preserving the
+// channel count and place the result in |destination|.
+int Resample(const AudioFrame& frame,
+             const int destination_sample_rate,
+             PushResampler<int16_t>* resampler,
+             int16_t* destination) {
+  const int number_of_channels = frame.num_channels_;
+  const int target_number_of_samples_per_channel =
+      destination_sample_rate / 100;
+  resampler->InitializeIfNeeded(frame.sample_rate_hz_, destination_sample_rate,
+                                number_of_channels);
+
+  return resampler->Resample(
+      frame.data_, frame.samples_per_channel_ * number_of_channels, destination,
+      number_of_channels * target_number_of_samples_per_channel);
+}
+}  // namespace
+
 AudioTransportProxy::AudioTransportProxy(AudioTransport* voe_audio_transport,
                                          AudioProcessing* apm,
                                          AudioMixer* mixer)
-    : voe_audio_transport_(voe_audio_transport) {
+    : voe_audio_transport_(voe_audio_transport), apm_(apm), mixer_(mixer) {
   RTC_DCHECK(voe_audio_transport);
   RTC_DCHECK(apm);
+  RTC_DCHECK(mixer);
 }
 
 AudioTransportProxy::~AudioTransportProxy() {}
@@ -47,20 +67,30 @@ int32_t AudioTransportProxy::NeedMorePlayData(const size_t nSamples,
                                               size_t& nSamplesOut,
                                               int64_t* elapsed_time_ms,
                                               int64_t* ntp_time_ms) {
-  RTC_DCHECK_EQ(sizeof(int16_t) * nChannels, nBytesPerSample);
+  // 2 = bytes in sample.
+  RTC_DCHECK_EQ(2 * nChannels, nBytesPerSample);

[the sun, 2016/11/16 20:22:58]

Here I think sizeof(int16_t) made the code more readable.

   RTC_DCHECK_GE(nChannels, 1u);
   RTC_DCHECK_LE(nChannels, 2u);
   RTC_DCHECK_GE(
       samplesPerSec,
       static_cast<uint32_t>(AudioProcessing::NativeRate::kSampleRate8kHz));
+
+  // 100 = 1 second / data duration.
   RTC_DCHECK_EQ(nSamples * 100, samplesPerSec);
   RTC_DCHECK_LE(nBytesPerSample * nSamples * nChannels,
                 sizeof(AudioFrame::data_));
 
-  // Pass call through to original audio transport instance.
-  return voe_audio_transport_->NeedMorePlayData(
-      nSamples, nBytesPerSample, nChannels, samplesPerSec, audioSamples,
-      nSamplesOut, elapsed_time_ms, ntp_time_ms);
+  mixer_->Mix(nChannels, &mixed_frame_);
+  *elapsed_time_ms = mixed_frame_.elapsed_time_ms_;
+  *ntp_time_ms = mixed_frame_.ntp_time_ms_;
+
+  const auto error = apm_->ProcessReverseStream(&mixed_frame_);
+  RTC_DCHECK_EQ(error, AudioProcessing::kNoError);
+
+  nSamplesOut = Resample(mixed_frame_, samplesPerSec, &resampler_,
+                         static_cast<int16_t*>(audioSamples));
+  RTC_DCHECK_EQ(nSamplesOut, nChannels * nSamples);
+  return 0;
 }
 
 void AudioTransportProxy::PushCaptureData(int voe_channel,
@@ -81,17 +111,26 @@ void AudioTransportProxy::PullRenderData(int bits_per_sample,
                                          void* audio_data,
                                          int64_t* elapsed_time_ms,
                                          int64_t* ntp_time_ms) {
-  RTC_DCHECK_EQ(static_cast<size_t>(bits_per_sample), 8 * sizeof(int16_t));
+  // 8 * 2 = bits per byte * bytes in sample.

[the sun, 2016/11/16 20:22:57]

Sorry for not being clear. I thought you could just DCHECK against "16" instead
of calculating the number of bits in an int16_t ... I don't think a comment
would be needed then.

[aleloi, 2016/11/17 18:12:26]

Ah, of course...

+  RTC_DCHECK_EQ(static_cast<size_t>(bits_per_sample), 8 * 2);
   RTC_DCHECK_GE(number_of_channels, 1u);
   RTC_DCHECK_LE(number_of_channels, 2u);
   RTC_DCHECK_GE(static_cast<int>(sample_rate),
                 AudioProcessing::NativeRate::kSampleRate8kHz);
+
+  // 100 = 1 second / data duration.

[the sun, 2016/11/16 20:22:57]

1 s / 10 ms ?

[aleloi, 2016/11/17 18:12:26]

Acknowledged.

   RTC_DCHECK_EQ(static_cast<int>(number_of_frames * 100), sample_rate);
+
+  // 8 = bits per byte.
   RTC_DCHECK_LE(bits_per_sample / 8 * number_of_frames * number_of_channels,
                 sizeof(AudioFrame::data_));
-  voe_audio_transport_->PullRenderData(
-      bits_per_sample, sample_rate, number_of_channels, number_of_frames,
-      audio_data, elapsed_time_ms, ntp_time_ms);
+  mixer_->Mix(number_of_channels, &mixed_frame_);
+  *elapsed_time_ms = mixed_frame_.elapsed_time_ms_;
+  *ntp_time_ms = mixed_frame_.ntp_time_ms_;
+
+  const auto output_samples = Resample(mixed_frame_, sample_rate, &resampler_,
+                                       static_cast<int16_t*>(audio_data));
+  RTC_DCHECK_EQ(output_samples, number_of_channels * number_of_frames);
 }
 
 }  // namespace webrtc