Remove various IDs

modules/audio_coding/acm2/audio_coding_module_unittest.cc

 /*
  *  Copyright (c) 2014 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 139 matching lines @@
   FrameType last_frame_type_ RTC_GUARDED_BY(crit_sect_);
   int last_payload_type_ RTC_GUARDED_BY(crit_sect_);
   uint32_t last_timestamp_ RTC_GUARDED_BY(crit_sect_);
   std::vector<uint8_t> last_payload_vec_ RTC_GUARDED_BY(crit_sect_);
   rtc::CriticalSection crit_sect_;
 };
 
 class AudioCodingModuleTestOldApi : public ::testing::Test {
  protected:
   AudioCodingModuleTestOldApi()
-      : id_(1),
-        rtp_utility_(new RtpUtility(kFrameSizeSamples, kPayloadType)),
+      : rtp_utility_(new RtpUtility(kFrameSizeSamples, kPayloadType)),
         clock_(Clock::GetRealTimeClock()) {}
 
   ~AudioCodingModuleTestOldApi() {}
 
   void TearDown() {}
 
   void SetUp() {
-    acm_.reset(AudioCodingModule::Create(id_, clock_));
+    acm_.reset(AudioCodingModule::Create(clock_));
 
     rtp_utility_->Populate(&rtp_header_);
 
     input_frame_.sample_rate_hz_ = kSampleRateHz;
     input_frame_.num_channels_ = 1;
     input_frame_.samples_per_channel_ = kSampleRateHz * 10 / 1000;  // 10 ms.
     static_assert(kSampleRateHz * 10 / 1000 <= AudioFrame::kMaxDataSizeSamples,
                   "audio frame too small");
     input_frame_.Mute();
 
@@ skipping 43 matching lines @@
     int last_length = packet_cb_.last_payload_len_bytes();
     EXPECT_TRUE(last_length == 2 * codec_.pacsize || last_length == 0)
         << "Last encoded packet was " << last_length << " bytes.";
   }
 
   virtual void InsertAudioAndVerifyEncoding() {
     InsertAudio();
     VerifyEncoding();
   }
 
-  const int id_;
   std::unique_ptr<RtpUtility> rtp_utility_;
   std::unique_ptr<AudioCodingModule> acm_;
   PacketizationCallbackStubOldApi packet_cb_;
   WebRtcRTPHeader rtp_header_;
   AudioFrame input_frame_;
 
   // These two have to be kept in sync for now. In the future, we'll be able to
   // eliminate the CodecInst and keep only the SdpAudioFormat.
   rtc::Optional<SdpAudioFormat> audio_format_;
   CodecInst codec_;
@@ skipping 63 matching lines @@
   EXPECT_EQ(0, stats.decoded_muted_output);
   // TODO(henrik.lundin) Add a test with muted state enabled.
 }
 
 TEST_F(AudioCodingModuleTestOldApi, VerifyOutputFrame) {
   AudioFrame audio_frame;
   const int kSampleRateHz = 32000;
   bool muted;
   EXPECT_EQ(0, acm_->PlayoutData10Ms(kSampleRateHz, &audio_frame, &muted));
   ASSERT_FALSE(muted);
-  EXPECT_EQ(id_, audio_frame.id_);
   EXPECT_EQ(0u, audio_frame.timestamp_);
   EXPECT_GT(audio_frame.num_channels_, 0u);
   EXPECT_EQ(static_cast<size_t>(kSampleRateHz / 100),
             audio_frame.samples_per_channel_);
   EXPECT_EQ(kSampleRateHz, audio_frame.sample_rate_hz_);
 }
 
 // The below test is temporarily disabled on Windows due to problems
 // with clang debug builds.
 // TODO(tommi): Re-enable when we've figured out what the problem is.
@@ skipping 1628 matching lines @@
   Run(16000, 8000, 1000);
 }
 
 TEST_F(AcmSwitchingOutputFrequencyOldApi, Toggle8KhzTo16Khz) {
   Run(8000, 16000, 1000);
 }
 
 #endif
 
 }  // namespace webrtc