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..8b0a479b6fa1b6431b50143c83fc1719fea79cb2 100644 |
--- a/webrtc/voice_engine/transmit_mixer.cc |
+++ b/webrtc/voice_engine/transmit_mixer.cc |
@@ -1136,10 +1136,6 @@ 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. |