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_); |