From 0023fdffd0156d7cce71a0817e964689d6af762b Mon Sep 17 00:00:00 2001
From: "henrik.lundin" <henrik.lundin@webrtc.org>
Date: Thu, 3 Mar 2016 23:05:39 -0800
Subject: [PATCH] Remove the ID from AcmReceiver

This is an artifact from the past, and is no longer used.

R=ivoc@webrtc.org

Review URL: https://codereview.webrtc.org/1764623002

Cr-Commit-Position: refs/heads/master@{#11866}
---
 webrtc/modules/audio_coding/acm2/acm_receiver.cc          | 8 +-------
 webrtc/modules/audio_coding/acm2/acm_receiver.h           | 6 ------
 .../modules/audio_coding/acm2/audio_coding_module_impl.cc | 1 -
 3 files changed, 1 insertion(+), 14 deletions(-)

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);