| 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 844bd57cd19a81d8f516cc361e519f26adf6531b..f5af65a13d5d12b84ecf86c992175d821eb402cc 100644
|
| --- a/webrtc/modules/audio_coding/include/audio_coding_module.h
|
| +++ b/webrtc/modules/audio_coding/include/audio_coding_module.h
|
| @@ -11,6 +11,7 @@
|
| #ifndef WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_
|
| #define WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_
|
|
|
| +#include <string>
|
| #include <vector>
|
|
|
| #include "webrtc/base/optional.h"
|
| @@ -471,10 +472,14 @@ class AudioCodingModule {
|
| //
|
| virtual int RegisterReceiveCodec(const CodecInst& receive_codec) = 0;
|
|
|
| + // Registers an external decoder. The name is only used to provide information
|
| + // back to the caller about the decoder. Hence, the name is arbitrary, and may
|
| + // be empty.
|
| virtual int RegisterExternalReceiveCodec(int rtp_payload_type,
|
| AudioDecoder* external_decoder,
|
| int sample_rate_hz,
|
| - int num_channels) = 0;
|
| + int num_channels,
|
| + const std::string& name) = 0;
|
|
|
| ///////////////////////////////////////////////////////////////////////////
|
| // int32_t UnregisterReceiveCodec()
|
|
|