| Index: webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
|
| diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
|
| index 7f63d880de94c001d821e0ef7e4d4be7105ede7f..49d0704e07f559dcbcbe6c858e3715c95578418a 100644
|
| --- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
|
| +++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
|
| @@ -659,17 +659,6 @@ int AudioCodingModuleImpl::SetMaximumPlayoutDelay(int time_ms) {
|
| return receiver_.SetMaximumDelay(time_ms);
|
| }
|
|
|
| -// Set playout mode for: voice, fax, streaming or off.
|
| -int AudioCodingModuleImpl::SetPlayoutMode(AudioPlayoutMode mode) {
|
| - receiver_.SetPlayoutMode(mode);
|
| - return 0; // TODO(turajs): return value is for backward compatibility.
|
| -}
|
| -
|
| -// Get playout mode voice, fax, streaming or off.
|
| -AudioPlayoutMode AudioCodingModuleImpl::PlayoutMode() const {
|
| - return receiver_.PlayoutMode();
|
| -}
|
| -
|
| // Get 10 milliseconds of raw audio data to play out.
|
| // Automatic resample to the requested frequency.
|
| int AudioCodingModuleImpl::PlayoutData10Ms(int desired_freq_hz,
|
|
|