| Index: webrtc/modules/audio_coding/main/interface/audio_coding_module.h
|
| diff --git a/webrtc/modules/audio_coding/main/interface/audio_coding_module.h b/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
|
| index b7d9a91b2f8dcaa3c7355a0d65b3c0653532f728..29a1e854de3d76da736d10f2f80cca91db482c70 100644
|
| --- a/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
|
| +++ b/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
|
| @@ -26,9 +26,10 @@ namespace webrtc {
|
| // forward declarations
|
| struct CodecInst;
|
| struct WebRtcRTPHeader;
|
| +class AudioEncoderMutable;
|
| class AudioFrame;
|
| +class RtcEventLog;
|
| class RTPFragmentationHeader;
|
| -class AudioEncoderMutable;
|
|
|
| #define WEBRTC_10MS_PCM_AUDIO 960 // 16 bits super wideband 48 kHz
|
|
|
| @@ -978,6 +979,16 @@ class AudioCodingModule {
|
| // Disable NACK.
|
| virtual void DisableNack() = 0;
|
|
|
| + ///////////////////////////////////////////////////////////////////////////
|
| + // void SetEventLog(RtcEventLog* event_log)
|
| + //
|
| + // Set an RtcEventLog object to enable logging of debug events inside the
|
| + // audio coding module.
|
| + //
|
| + // Input:
|
| + // -event_log : pointer to logging object.
|
| + virtual void SetEventLog(RtcEventLog* event_log) = 0;
|
| +
|
| //
|
| // Get a list of packets to be retransmitted. |round_trip_time_ms| is an
|
| // estimate of the round-trip-time (in milliseconds). Missing packets which
|
|
|