diff --git a/webrtc/modules/audio_coding/acm2/acm_receiver.cc b/webrtc/modules/audio_coding/acm2/acm_receiver.cc index 53ec8841e5..39d8a0d175 100644 --- a/webrtc/modules/audio_coding/acm2/acm_receiver.cc +++ b/webrtc/modules/audio_coding/acm2/acm_receiver.cc @@ -118,8 +118,7 @@ bool IsCng(int codec_id) { } // namespace AcmReceiver::AcmReceiver(const AudioCodingModule::Config& config) - : id_(config.id), - last_audio_decoder_(nullptr), + : last_audio_decoder_(nullptr), previous_audio_activity_(AudioFrame::kVadPassive), audio_buffer_(new int16_t[AudioFrame::kMaxDataSizeSamples]), last_audio_buffer_(new int16_t[AudioFrame::kMaxDataSizeSamples]), @@ -419,11 +418,6 @@ int AcmReceiver::RemoveCodec(uint8_t payload_type) { return 0; } -void AcmReceiver::set_id(int id) { - rtc::CritScope lock(&crit_sect_); - id_ = id; -} - bool AcmReceiver::GetPlayoutTimestamp(uint32_t* timestamp) { return neteq_->GetPlayoutTimestamp(timestamp); } diff --git a/webrtc/modules/audio_coding/acm2/acm_receiver.h b/webrtc/modules/audio_coding/acm2/acm_receiver.h index f5ceb61fbd..476f29dd71 100644 --- a/webrtc/modules/audio_coding/acm2/acm_receiver.h +++ b/webrtc/modules/audio_coding/acm2/acm_receiver.h @@ -210,11 +210,6 @@ 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_); diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc index 3262d48b6e..ff779a28b1 100644 --- a/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc +++ b/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc @@ -544,7 +544,6 @@ int AudioCodingModuleImpl::InitializeReceiverSafe() { if (receiver_.RemoveAllCodecs() < 0) return -1; } - receiver_.set_id(id_); receiver_.ResetInitialDelay(); receiver_.SetMinimumDelay(0); receiver_.SetMaximumDelay(0);