Remove the ID from AcmReceiver

webrtc/modules/audio_coding/acm2/acm_receiver.cc

 /*
  *  Copyright (c) 2013 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 100 matching lines @@
   auto i = RentACodec::CodecIdFromIndex(codec_id);
   return (i && (*i == RentACodec::CodecId::kCNNB ||
                 *i == RentACodec::CodecId::kCNWB ||
                 *i == RentACodec::CodecId::kCNSWB ||
                 *i == RentACodec::CodecId::kCNFB));
 }
 
 }  // 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]),
       neteq_(NetEq::Create(config.neteq_config)),
       vad_enabled_(config.neteq_config.enable_post_decode_vad),
       clock_(config.clock),
       resampled_last_output_frame_(true) {
   assert(clock_);
   memset(audio_buffer_.get(), 0, AudioFrame::kMaxDataSizeSamples);
   memset(last_audio_buffer_.get(), 0, AudioFrame::kMaxDataSizeSamples);
@@ skipping 279 matching lines @@
     return -1;
   }
   if (last_audio_decoder_ == &it->second) {
     last_audio_decoder_ = nullptr;
     last_packet_sample_rate_hz_ = rtc::Optional<int>();
   }
   decoders_.erase(it);
   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);
 }
 
 int AcmReceiver::LastAudioCodec(CodecInst* codec) const {
   rtc::CritScope lock(&crit_sect_);
   if (!last_audio_decoder_) {
     return -1;
   }
   *codec = *RentACodec::CodecInstById(
@@ skipping 93 matching lines @@
 
 void AcmReceiver::GetDecodingCallStatistics(
     AudioDecodingCallStats* stats) const {
   rtc::CritScope lock(&crit_sect_);
   *stats = call_stats_.GetDecodingStatistics();
 }
 
 }  // namespace acm2
 
 }  // namespace webrtc