Remove various IDs

modules/audio_coding/acm2/audio_coding_module.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 251 matching lines @@
   int PreprocessToAddData(const AudioFrame& in_frame,
                           const AudioFrame** ptr_out)
       RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
 
   // Change required states after starting to receive the codec corresponding
   // to |index|.
   int UpdateUponReceivingCodec(int index);
 
   rtc::CriticalSection acm_crit_sect_;
   rtc::Buffer encode_buffer_ RTC_GUARDED_BY(acm_crit_sect_);
-  int id_;  // TODO(henrik.lundin) Make const.
   uint32_t expected_codec_ts_ RTC_GUARDED_BY(acm_crit_sect_);
   uint32_t expected_in_ts_ RTC_GUARDED_BY(acm_crit_sect_);
   acm2::ACMResampler resampler_ RTC_GUARDED_BY(acm_crit_sect_);
   acm2::AcmReceiver receiver_;  // AcmReceiver has it's own internal lock.
   ChangeLogger bitrate_logger_ RTC_GUARDED_BY(acm_crit_sect_);
 
   std::unique_ptr<EncoderFactory> encoder_factory_
       RTC_GUARDED_BY(acm_crit_sect_);
 
   // Current encoder stack, either obtained from
@@ skipping 166 matching lines @@
 void AudioCodingModuleImpl::ChangeLogger::MaybeLog(int value) {
   if (value != last_value_ || first_time_) {
     first_time_ = false;
     last_value_ = value;
     RTC_HISTOGRAM_COUNTS_SPARSE_100(histogram_name_, value);
   }
 }
 
 AudioCodingModuleImpl::AudioCodingModuleImpl(
     const AudioCodingModule::Config& config)
-    : id_(config.id),
-      expected_codec_ts_(0xD87F3F9F),
+    : expected_codec_ts_(0xD87F3F9F),
       expected_in_ts_(0xD87F3F9F),
       receiver_(config),
       bitrate_logger_("WebRTC.Audio.TargetBitrateInKbps"),
       encoder_factory_(new EncoderFactory),
       encoder_stack_(nullptr),
       previous_pltype_(255),
       receiver_initialized_(false),
       first_10ms_data_(false),
       first_frame_(true),
       packetization_callback_(NULL),
@@ skipping 642 matching lines @@
 // Get 10 milliseconds of raw audio data to play out.
 // Automatic resample to the requested frequency.
 int AudioCodingModuleImpl::PlayoutData10Ms(int desired_freq_hz,
                                            AudioFrame* audio_frame,
                                            bool* muted) {
   // GetAudio always returns 10 ms, at the requested sample rate.
   if (receiver_.GetAudio(desired_freq_hz, audio_frame, muted) != 0) {
     LOG(LS_ERROR) << "PlayoutData failed, RecOut Failed";
     return -1;
   }
-  audio_frame->id_ = id_;
   return 0;
 }
 
 int AudioCodingModuleImpl::PlayoutData10Ms(int desired_freq_hz,
                                            AudioFrame* audio_frame) {
   bool muted;
   int ret = PlayoutData10Ms(desired_freq_hz, audio_frame, &muted);
   RTC_DCHECK(!muted);
   return ret;
 }
@@ skipping 239 matching lines @@
 
 // Checks the validity of the parameters of the given codec
 bool AudioCodingModule::IsCodecValid(const CodecInst& codec) {
   bool valid = acm2::RentACodec::IsCodecValid(codec);
   if (!valid)
     LOG(LS_ERROR) << "Invalid codec setting";
   return valid;
 }
 
 }  // namespace webrtc