Remove various IDs

voice_engine/channel.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 629 matching lines @@
     return AudioMixer::Source::AudioFrameInfo::kError;
   }
 
   if (muted) {
     // TODO(henrik.lundin): We should be able to do better than this. But we
     // will have to go through all the cases below where the audio samples may
     // be used, and handle the muted case in some way.
     AudioFrameOperations::Mute(audio_frame);
   }
 
-  // Convert module ID to internal VoE channel ID
-  audio_frame->id_ = VoEChannelId(audio_frame->id_);
   // Store speech type for dead-or-alive detection
   _outputSpeechType = audio_frame->speech_type_;
 
   {
     // Pass the audio buffers to an optional sink callback, before applying
     // scaling/panning, as that applies to the mix operation.
     // External recipients of the audio (e.g. via AudioTrack), will do their
     // own mixing/dynamic processing.
     rtc::CritScope cs(&_callbackCritSect);
     if (audio_sink_) {
@@ skipping 128 matching lines @@
       seq_num_allocator_proxy_(new TransportSequenceNumberProxy()),
       rtp_packet_sender_proxy_(new RtpPacketSenderProxy()),
       retransmission_rate_limiter_(new RateLimiter(Clock::GetRealTimeClock(),
                                                    kMaxRetransmissionWindowMs)),
       decoder_factory_(config.acm_config.decoder_factory),
       use_twcc_plr_for_ana_(
           webrtc::field_trial::FindFullName("UseTwccPlrForAna") == "Enabled") {
   WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::Channel() - ctor");
   AudioCodingModule::Config acm_config(config.acm_config);
-  acm_config.id = VoEModuleId(instanceId, channelId);
   acm_config.neteq_config.enable_muted_state = true;
   audio_coding_.reset(AudioCodingModule::Create(acm_config));
 
   _outputAudioLevel.Clear();
 
   RtpRtcp::Configuration configuration;
   configuration.audio = true;
   configuration.outgoing_transport = this;
   configuration.overhead_observer = this;
   configuration.receive_statistics = rtp_receive_statistics_.get();
@@ skipping 825 matching lines @@
 void Channel::ProcessAndEncodeAudio(const AudioFrame& audio_input) {
   // Avoid posting any new tasks if sending was already stopped in StopSend().
   rtc::CritScope cs(&encoder_queue_lock_);
   if (!encoder_queue_is_active_) {
     return;
   }
   std::unique_ptr<AudioFrame> audio_frame(new AudioFrame());
   // TODO(henrika): try to avoid copying by moving ownership of audio frame
   // either into pool of frames or into the task itself.
   audio_frame->CopyFrom(audio_input);
-  audio_frame->id_ = ChannelId();
   // Profile time between when the audio frame is added to the task queue and
   // when the task is actually executed.
   audio_frame->UpdateProfileTimeStamp();
   encoder_queue_->PostTask(std::unique_ptr<rtc::QueuedTask>(
       new ProcessAndEncodeAudioTask(std::move(audio_frame), this)));
 }
 
 void Channel::ProcessAndEncodeAudio(const int16_t* audio_data,
                                     int sample_rate,
                                     size_t number_of_frames,
                                     size_t number_of_channels) {
   // Avoid posting as new task if sending was already stopped in StopSend().
   rtc::CritScope cs(&encoder_queue_lock_);
   if (!encoder_queue_is_active_) {
     return;
   }
   CodecInst codec;
   const int result = GetSendCodec(codec);
   std::unique_ptr<AudioFrame> audio_frame(new AudioFrame());
-  audio_frame->id_ = ChannelId();
   // TODO(ossu): Investigate how this could happen. b/62909493
   if (result == 0) {
     audio_frame->sample_rate_hz_ = std::min(codec.plfreq, sample_rate);
     audio_frame->num_channels_ = std::min(number_of_channels, codec.channels);
   } else {
     audio_frame->sample_rate_hz_ = sample_rate;
     audio_frame->num_channels_ = number_of_channels;
     LOG(LS_WARNING) << "Unable to get send codec for channel " << ChannelId();
     RTC_NOTREACHED();
   }
   RemixAndResample(audio_data, number_of_frames, number_of_channels,
                    sample_rate, &input_resampler_, audio_frame.get());
   encoder_queue_->PostTask(std::unique_ptr<rtc::QueuedTask>(
       new ProcessAndEncodeAudioTask(std::move(audio_frame), this)));
 }
 
 void Channel::ProcessAndEncodeAudioOnTaskQueue(AudioFrame* audio_input) {
   RTC_DCHECK_RUN_ON(encoder_queue_);
   RTC_DCHECK_GT(audio_input->samples_per_channel_, 0);
   RTC_DCHECK_LE(audio_input->num_channels_, 2);
-  RTC_DCHECK_EQ(audio_input->id_, ChannelId());
 
   // Measure time between when the audio frame is added to the task queue and
   // when the task is actually executed. Goal is to keep track of unwanted
   // extra latency added by the task queue.
   RTC_HISTOGRAM_COUNTS_10000("WebRTC.Audio.EncodingTaskQueueLatencyMs",
                              audio_input->ElapsedProfileTimeMs());
 
   bool is_muted = InputMute();
   AudioFrameOperations::Mute(audio_input, previous_frame_muted_, is_muted);
 
@@ skipping 268 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc