Fix the maximum native sample rate in AudioProcessing

webrtc/voice_engine/transmit_mixer.cc

Index: webrtc/voice_engine/transmit_mixer.cc
diff --git a/webrtc/voice_engine/transmit_mixer.cc b/webrtc/voice_engine/transmit_mixer.cc
index a02f298509fc521845f8ae602c0886dedaefb620..30b7ee4eea5a9dde6a94de1182c6bfc8f3d8100d 100644
--- a/webrtc/voice_engine/transmit_mixer.cc
+++ b/webrtc/voice_engine/transmit_mixer.cc
@@ -1136,11 +1136,7 @@ void TransmitMixer::GenerateAudioFrame(const int16_t* audio,
   int codec_rate;
   int num_codec_channels;
   GetSendCodecInfo(&codec_rate, &num_codec_channels);
-  // TODO(ajm): This currently restricts the sample rate to 32 kHz.
-  // See: https://code.google.com/p/webrtc/issues/detail?id=3146
-  // When 48 kHz is supported natively by AudioProcessing, this will have
-  // to be changed to handle 44.1 kHz.
-  int max_sample_rate_hz = kAudioProcMaxNativeSampleRateHz;
+  int max_sample_rate_hz = AudioProcessing::kMaxNativeSampleRateHz;
   if (audioproc_->echo_control_mobile()->is_enabled()) {
     // AECM only supports 8 and 16 kHz.
     max_sample_rate_hz = 16000;
@@ -1148,19 +1144,18 @@ void TransmitMixer::GenerateAudioFrame(const int16_t* audio,
   codec_rate = std::min(codec_rate, max_sample_rate_hz);

[Andrew MacDonald, 2015/09/23 17:31:47]

Sorry, just noticed this. Can you remove this check and integrate below? You can
remove max_sample_rate_hz then as it will already be handled by the loop:

const int min_processing_rate = std::min(sample_rate_hz, codec_rate);
for (...) {
  ...
}
if (audioproc_->echo_control_mobile()->is_enabled()) {
  // AECM only supports 8 and 16 kHz.
  _audioFrame.sample_rate_hz = std::min(_audioFrame.sample_rate_hz, 16000);
}

where we might want to advertise the 16000 as a const in the APM interface, up
to you.

[aluebs-webrtc, 2015/09/23 19:48:54]

Great point. Done.

   stereo_codec_ = num_codec_channels == 2;
 
-  if (!mono_buffer_.get()) {
-    // Temporary space for DownConvertToCodecFormat.
-    mono_buffer_.reset(new int16_t[kMaxMonoDataSizeSamples]);
+  // We want to process at the lowest rate possible without losing information.
+  // Choose the lowest native rate at least equal to the input and codec rates.
+  const int min_processing_rate = std::min(sample_rate_hz, codec_rate);
+  for (size_t i = 0; i < AudioProcessing::kNumNativeSampleRates; ++i) {
+    _audioFrame.sample_rate_hz_ = AudioProcessing::kNativeSampleRatesHz[i];
+    if (_audioFrame.sample_rate_hz_ >= min_processing_rate) {
+      break;
+    }
   }
-  DownConvertToCodecFormat(audio,
-                           samples_per_channel,
-                           num_channels,
-                           sample_rate_hz,
-                           num_codec_channels,
-                           codec_rate,
-                           mono_buffer_.get(),
-                           &resampler_,
-                           &_audioFrame);
+  _audioFrame.num_channels_ = std::min(num_channels, num_codec_channels);
+  RemixAndResample(audio, samples_per_channel, num_channels, sample_rate_hz,
+                   &resampler_, &_audioFrame);
 }
 
 int32_t TransmitMixer::RecordAudioToFile(