Remove various IDs:

- AudioFrame
- AudioCodingModule

BUG=webrtc:4690
TBR=kwiberg@webrtc.org

Review-Url: https://codereview.webrtc.org/3019543002
Cr-Commit-Position: refs/heads/master@{#20005}
diff --git a/modules/audio_coding/acm2/acm_receive_test.cc b/modules/audio_coding/acm2/acm_receive_test.cc
index 6cfe464..082506a 100644
--- a/modules/audio_coding/acm2/acm_receive_test.cc
+++ b/modules/audio_coding/acm2/acm_receive_test.cc
@@ -110,7 +110,6 @@
     Clock* clock,
     rtc::scoped_refptr<AudioDecoderFactory> decoder_factory) {
   AudioCodingModule::Config config;
-  config.id = 0;
   config.clock = clock;
   config.decoder_factory = std::move(decoder_factory);
   return config;
diff --git a/modules/audio_coding/acm2/acm_send_test.cc b/modules/audio_coding/acm2/acm_send_test.cc
index d5f196b..307c906 100644
--- a/modules/audio_coding/acm2/acm_send_test.cc
+++ b/modules/audio_coding/acm2/acm_send_test.cc
@@ -28,7 +28,7 @@
                                      int source_rate_hz,
                                      int test_duration_ms)
     : clock_(0),
-      acm_(webrtc::AudioCodingModule::Create(0, &clock_)),
+      acm_(webrtc::AudioCodingModule::Create(&clock_)),
       audio_source_(audio_source),
       source_rate_hz_(source_rate_hz),
       input_block_size_samples_(
diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc
index 5997d12..c48fbef 100644
--- a/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/modules/audio_coding/acm2/audio_coding_module.cc
@@ -269,7 +269,6 @@
 
   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_);
@@ -456,8 +455,7 @@
 
 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"),
@@ -1120,7 +1118,6 @@
     LOG(LS_ERROR) << "PlayoutData failed, RecOut Failed";
     return -1;
   }
-  audio_frame->id_ = id_;
   return 0;
 }
 
@@ -1286,7 +1283,7 @@
 }  // namespace
 
 AudioCodingModule::Config::Config()
-    : id(0), neteq_config(), clock(Clock::GetRealTimeClock()) {
+    : neteq_config(), clock(Clock::GetRealTimeClock()) {
   // Post-decode VAD is disabled by default in NetEq, however, Audio
   // Conference Mixer relies on VAD decisions and fails without them.
   neteq_config.enable_post_decode_vad = true;
@@ -1296,17 +1293,15 @@
 AudioCodingModule::Config::~Config() = default;
 
 // Create module
-AudioCodingModule* AudioCodingModule::Create(int id) {
+AudioCodingModule* AudioCodingModule::Create() {
   Config config;
-  config.id = id;
   config.clock = Clock::GetRealTimeClock();
   config.decoder_factory = CreateBuiltinAudioDecoderFactory();
   return Create(config);
 }
 
-AudioCodingModule* AudioCodingModule::Create(int id, Clock* clock) {
+AudioCodingModule* AudioCodingModule::Create(Clock* clock) {
   Config config;
-  config.id = id;
   config.clock = clock;
   config.decoder_factory = CreateBuiltinAudioDecoderFactory();
   return Create(config);
diff --git a/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
index 80fc4d8..a010619 100644
--- a/modules/audio_coding/acm2/audio_coding_module_unittest.cc
+++ b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
@@ -157,8 +157,7 @@
 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() {}
@@ -166,7 +165,7 @@
   void TearDown() {}
 
   void SetUp() {
-    acm_.reset(AudioCodingModule::Create(id_, clock_));
+    acm_.reset(AudioCodingModule::Create(clock_));
 
     rtp_utility_->Populate(&rtp_header_);
 
@@ -230,7 +229,6 @@
     VerifyEncoding();
   }
 
-  const int id_;
   std::unique_ptr<RtpUtility> rtp_utility_;
   std::unique_ptr<AudioCodingModule> acm_;
   PacketizationCallbackStubOldApi packet_cb_;
@@ -314,7 +312,6 @@
   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),
diff --git a/modules/audio_coding/include/audio_coding_module.h b/modules/audio_coding/include/audio_coding_module.h
index 944ad60..63af3ab 100644
--- a/modules/audio_coding/include/audio_coding_module.h
+++ b/modules/audio_coding/include/audio_coding_module.h
@@ -70,7 +70,6 @@
     Config(const Config&);
     ~Config();
 
-    int id;
     NetEq::Config neteq_config;
     Clock* clock;
     rtc::scoped_refptr<AudioDecoderFactory> decoder_factory;
@@ -83,8 +82,8 @@
   // injected into ACM. ACM will take the ownership of the object clock and
   // delete it when destroyed.
   //
-  static AudioCodingModule* Create(int id);
-  static AudioCodingModule* Create(int id, Clock* clock);
+  static AudioCodingModule* Create();
+  static AudioCodingModule* Create(Clock* clock);
   static AudioCodingModule* Create(const Config& config);
   virtual ~AudioCodingModule() = default;
 
diff --git a/modules/audio_coding/test/APITest.cc b/modules/audio_coding/test/APITest.cc
index 5418342..b29e84e 100644
--- a/modules/audio_coding/test/APITest.cc
+++ b/modules/audio_coding/test/APITest.cc
@@ -48,8 +48,8 @@
 }
 
 APITest::APITest()
-    : _acmA(AudioCodingModule::Create(1)),
-      _acmB(AudioCodingModule::Create(2)),
+    : _acmA(AudioCodingModule::Create()),
+      _acmB(AudioCodingModule::Create()),
       _channel_A2B(NULL),
       _channel_B2A(NULL),
       _writeToFile(true),
diff --git a/modules/audio_coding/test/EncodeDecodeTest.cc b/modules/audio_coding/test/EncodeDecodeTest.cc
index 8257ed9..2b6b4ac 100644
--- a/modules/audio_coding/test/EncodeDecodeTest.cc
+++ b/modules/audio_coding/test/EncodeDecodeTest.cc
@@ -281,7 +281,7 @@
   codePars[1] = 0;
   codePars[2] = 0;
 
-  std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create(0));
+  std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create());
   struct CodecInst sendCodecTmp;
   numCodecs = acm->NumberOfCodecs();
 
@@ -337,7 +337,7 @@
                                            int codeId,
                                            int* codePars,
                                            int testMode) {
-  std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create(1));
+  std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create());
   RTPFile rtpFile;
   std::string fileName = webrtc::test::TempFilename(webrtc::test::OutputPath(),
                                                     "encode_decode_rtp");
diff --git a/modules/audio_coding/test/PacketLossTest.cc b/modules/audio_coding/test/PacketLossTest.cc
index c80615a..a6c56fa7 100644
--- a/modules/audio_coding/test/PacketLossTest.cc
+++ b/modules/audio_coding/test/PacketLossTest.cc
@@ -127,7 +127,7 @@
 #ifndef WEBRTC_CODEC_OPUS
   return;
 #else
-  std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create(0));
+  std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create());
 
   int codec_id = acm->Codec("opus", 48000, channels_);
 
diff --git a/modules/audio_coding/test/TestAllCodecs.cc b/modules/audio_coding/test/TestAllCodecs.cc
index 74319c2..ff28a28 100644
--- a/modules/audio_coding/test/TestAllCodecs.cc
+++ b/modules/audio_coding/test/TestAllCodecs.cc
@@ -104,8 +104,8 @@
 }
 
 TestAllCodecs::TestAllCodecs(int test_mode)
-    : acm_a_(AudioCodingModule::Create(0)),
-      acm_b_(AudioCodingModule::Create(1)),
+    : acm_a_(AudioCodingModule::Create()),
+      acm_b_(AudioCodingModule::Create()),
       channel_a_to_b_(NULL),
       test_count_(0),
       packet_size_samples_(0),
diff --git a/modules/audio_coding/test/TestRedFec.cc b/modules/audio_coding/test/TestRedFec.cc
index 3e88290..58561c6 100644
--- a/modules/audio_coding/test/TestRedFec.cc
+++ b/modules/audio_coding/test/TestRedFec.cc
@@ -48,8 +48,8 @@
 }
 
 TestRedFec::TestRedFec()
-    : _acmA(AudioCodingModule::Create(0)),
-      _acmB(AudioCodingModule::Create(1)),
+    : _acmA(AudioCodingModule::Create()),
+      _acmB(AudioCodingModule::Create()),
       _channelA2B(NULL),
       _testCntr(0) {
 }
diff --git a/modules/audio_coding/test/TestStereo.cc b/modules/audio_coding/test/TestStereo.cc
index d598191..eca81f8 100644
--- a/modules/audio_coding/test/TestStereo.cc
+++ b/modules/audio_coding/test/TestStereo.cc
@@ -108,8 +108,8 @@
 }
 
 TestStereo::TestStereo(int test_mode)
-    : acm_a_(AudioCodingModule::Create(0)),
-      acm_b_(AudioCodingModule::Create(1)),
+    : acm_a_(AudioCodingModule::Create()),
+      acm_b_(AudioCodingModule::Create()),
       channel_a2b_(NULL),
       test_cntr_(0),
       pack_size_samp_(0),
diff --git a/modules/audio_coding/test/TestVADDTX.cc b/modules/audio_coding/test/TestVADDTX.cc
index 1aa00b5..628582d 100644
--- a/modules/audio_coding/test/TestVADDTX.cc
+++ b/modules/audio_coding/test/TestVADDTX.cc
@@ -62,8 +62,8 @@
 }
 
 TestVadDtx::TestVadDtx()
-    : acm_send_(AudioCodingModule::Create(0)),
-      acm_receive_(AudioCodingModule::Create(1)),
+    : acm_send_(AudioCodingModule::Create()),
+      acm_receive_(AudioCodingModule::Create()),
       channel_(new Channel),
       monitor_(new ActivityMonitor) {
   EXPECT_EQ(0, acm_send_->RegisterTransportCallback(channel_.get()));
diff --git a/modules/audio_coding/test/TwoWayCommunication.cc b/modules/audio_coding/test/TwoWayCommunication.cc
index addb717..8049436 100644
--- a/modules/audio_coding/test/TwoWayCommunication.cc
+++ b/modules/audio_coding/test/TwoWayCommunication.cc
@@ -34,16 +34,14 @@
 #define MAX_FILE_NAME_LENGTH_BYTE 500
 
 TwoWayCommunication::TwoWayCommunication(int testMode)
-    : _acmA(AudioCodingModule::Create(1)),
-      _acmRefA(AudioCodingModule::Create(3)),
+    : _acmA(AudioCodingModule::Create()),
+      _acmRefA(AudioCodingModule::Create()),
       _testMode(testMode) {
   AudioCodingModule::Config config;
   // The clicks will be more obvious in FAX mode. TODO(henrik.lundin) Really?
   config.neteq_config.playout_mode = kPlayoutFax;
-  config.id = 2;
   config.decoder_factory = CreateBuiltinAudioDecoderFactory();
   _acmB.reset(AudioCodingModule::Create(config));
-  config.id = 4;
   _acmRefB.reset(AudioCodingModule::Create(config));
 }
 
@@ -62,7 +60,7 @@
 
 void TwoWayCommunication::ChooseCodec(uint8_t* codecID_A,
                                       uint8_t* codecID_B) {
-  std::unique_ptr<AudioCodingModule> tmpACM(AudioCodingModule::Create(0));
+  std::unique_ptr<AudioCodingModule> tmpACM(AudioCodingModule::Create());
   uint8_t noCodec = tmpACM->NumberOfCodecs();
   CodecInst codecInst;
   printf("List of Supported Codecs\n");
diff --git a/modules/audio_coding/test/delay_test.cc b/modules/audio_coding/test/delay_test.cc
index 3f78ea6..407f709 100644
--- a/modules/audio_coding/test/delay_test.cc
+++ b/modules/audio_coding/test/delay_test.cc
@@ -64,8 +64,8 @@
 class DelayTest {
  public:
   DelayTest()
-      : acm_a_(AudioCodingModule::Create(0)),
-        acm_b_(AudioCodingModule::Create(1)),
+      : acm_a_(AudioCodingModule::Create()),
+        acm_b_(AudioCodingModule::Create()),
         channel_a2b_(new Channel),
         test_cntr_(0),
         encoding_sample_rate_hz_(8000) {}
diff --git a/modules/audio_coding/test/iSACTest.cc b/modules/audio_coding/test/iSACTest.cc
index a14f795..a44259f 100644
--- a/modules/audio_coding/test/iSACTest.cc
+++ b/modules/audio_coding/test/iSACTest.cc
@@ -67,8 +67,8 @@
 }
 
 ISACTest::ISACTest(int testMode)
-    : _acmA(AudioCodingModule::Create(1)),
-      _acmB(AudioCodingModule::Create(2)),
+    : _acmA(AudioCodingModule::Create()),
+      _acmB(AudioCodingModule::Create()),
       _testMode(testMode) {}
 
 ISACTest::~ISACTest() {}
diff --git a/modules/audio_coding/test/insert_packet_with_timing.cc b/modules/audio_coding/test/insert_packet_with_timing.cc
index 500375c..2c0e54b 100644
--- a/modules/audio_coding/test/insert_packet_with_timing.cc
+++ b/modules/audio_coding/test/insert_packet_with_timing.cc
@@ -61,8 +61,8 @@
   InsertPacketWithTiming()
       : sender_clock_(new SimulatedClock(0)),
         receiver_clock_(new SimulatedClock(0)),
-        send_acm_(AudioCodingModule::Create(0, sender_clock_)),
-        receive_acm_(AudioCodingModule::Create(0, receiver_clock_)),
+        send_acm_(AudioCodingModule::Create(sender_clock_)),
+        receive_acm_(AudioCodingModule::Create(receiver_clock_)),
         channel_(new Channel),
         seq_num_fid_(fopen(FLAG_seq_num, "rt")),
         send_ts_fid_(fopen(FLAG_send_ts, "rt")),
diff --git a/modules/audio_coding/test/opus_test.cc b/modules/audio_coding/test/opus_test.cc
index 7b54668..b7acc0f 100644
--- a/modules/audio_coding/test/opus_test.cc
+++ b/modules/audio_coding/test/opus_test.cc
@@ -27,7 +27,7 @@
 namespace webrtc {
 
 OpusTest::OpusTest()
-    : acm_receiver_(AudioCodingModule::Create(0)),
+    : acm_receiver_(AudioCodingModule::Create()),
       channel_a2b_(NULL),
       counter_(0),
       payload_type_(255),
diff --git a/modules/audio_coding/test/target_delay_unittest.cc b/modules/audio_coding/test/target_delay_unittest.cc
index 2a75706..03135da 100644
--- a/modules/audio_coding/test/target_delay_unittest.cc
+++ b/modules/audio_coding/test/target_delay_unittest.cc
@@ -22,7 +22,7 @@
 
 class TargetDelayTest : public ::testing::Test {
  protected:
-  TargetDelayTest() : acm_(AudioCodingModule::Create(0)) {}
+  TargetDelayTest() : acm_(AudioCodingModule::Create()) {}
 
   ~TargetDelayTest() {}