| Index: webrtc/modules/audio_coding/acm2/acm_receiver.h
|
| diff --git a/webrtc/modules/audio_coding/acm2/acm_receiver.h b/webrtc/modules/audio_coding/acm2/acm_receiver.h
|
| index f5ceb61fbdccdb41d27d804045160cd8787bad1a..476f29dd7187885f53d005f87975378d5849f07b 100644
|
| --- a/webrtc/modules/audio_coding/acm2/acm_receiver.h
|
| +++ b/webrtc/modules/audio_coding/acm2/acm_receiver.h
|
| @@ -211,11 +211,6 @@ class AcmReceiver {
|
| int RemoveAllCodecs();
|
|
|
| //
|
| - // Set ID.
|
| - //
|
| - void set_id(int id); // TODO(turajs): can be inline.
|
| -
|
| - //
|
| // Gets the RTP timestamp of the last sample delivered by GetAudio().
|
| // Returns true if the RTP timestamp is valid, otherwise false.
|
| //
|
| @@ -282,7 +277,6 @@ class AcmReceiver {
|
| uint32_t NowInTimestamp(int decoder_sampling_rate) const;
|
|
|
| rtc::CriticalSection crit_sect_;
|
| - int id_; // TODO(henrik.lundin) Make const.
|
| const Decoder* last_audio_decoder_ GUARDED_BY(crit_sect_);
|
| AudioFrame::VADActivity previous_audio_activity_ GUARDED_BY(crit_sect_);
|
| ACMResampler resampler_ GUARDED_BY(crit_sect_);
|
|
|