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..586ab8a875f21643b90c48b6c194e83a0b78892a 100644
--- a/webrtc/voice_engine/transmit_mixer.cc
+++ b/webrtc/voice_engine/transmit_mixer.cc
@@ -1136,31 +1136,25 @@ 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;
-  if (audioproc_->echo_control_mobile()->is_enabled()) {
-    // AECM only supports 8 and 16 kHz.
-    max_sample_rate_hz = 16000;
-  }
-  codec_rate = std::min(codec_rate, max_sample_rate_hz);
   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;
+    }
+  }
+  if (audioproc_->echo_control_mobile()->is_enabled()) {
+    // AECM only supports 8 and 16 kHz.
+    _audioFrame.sample_rate_hz_ = std::min(
+        _audioFrame.sample_rate_hz_, AudioProcessing::kMaxAECMSampleRateHz);
   }
-  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(