| Index: webrtc/modules/audio_coding/include/audio_coding_module.h
|
| diff --git a/webrtc/modules/audio_coding/include/audio_coding_module.h b/webrtc/modules/audio_coding/include/audio_coding_module.h
|
| index fc8ae1ed513cb5098c2406f09a4d63a2e1f9720a..615e8d949658e51b1b5ed08f418391ddf1a3ae13 100644
|
| --- a/webrtc/modules/audio_coding/include/audio_coding_module.h
|
| +++ b/webrtc/modules/audio_coding/include/audio_coding_module.h
|
| @@ -252,6 +252,9 @@ class AudioCodingModule {
|
| ///////////////////////////////////////////////////////////////////////////
|
| // Sets the bitrate to the specified value in bits/sec. If the value is not
|
| // supported by the codec, it will choose another appropriate value.
|
| + //
|
| + // This is only used in audio_coding_module_unittest_oldapi.cc.
|
| + // TODO(minyue): Remove it when possible.
|
| virtual void SetBitRate(int bitrate_bps) = 0;
|
|
|
| // int32_t RegisterTransportCallback()
|
| @@ -371,6 +374,8 @@ class AudioCodingModule {
|
| // -1 if failed to set packet loss rate,
|
| // 0 if succeeded.
|
| //
|
| + // This is only used in audio_coding_module_unittest_oldapi.cc.
|
| + // TODO(minyue): Remove it when possible.
|
| virtual int SetPacketLossRate(int packet_loss_rate) = 0;
|
|
|
| ///////////////////////////////////////////////////////////////////////////
|
|
|