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}
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@ -48,8 +48,8 @@ void APITest::Wait(uint32_t waitLengthMs) {
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}
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APITest::APITest()
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: _acmA(AudioCodingModule::Create(1)),
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_acmB(AudioCodingModule::Create(2)),
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: _acmA(AudioCodingModule::Create()),
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_acmB(AudioCodingModule::Create()),
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_channel_A2B(NULL),
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_channel_B2A(NULL),
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_writeToFile(true),
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@ -281,7 +281,7 @@ void EncodeDecodeTest::Perform() {
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codePars[1] = 0;
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codePars[2] = 0;
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std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create(0));
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std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create());
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struct CodecInst sendCodecTmp;
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numCodecs = acm->NumberOfCodecs();
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@ -337,7 +337,7 @@ std::string EncodeDecodeTest::EncodeToFile(int fileType,
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int codeId,
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int* codePars,
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int testMode) {
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std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create(1));
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std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create());
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RTPFile rtpFile;
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std::string fileName = webrtc::test::TempFilename(webrtc::test::OutputPath(),
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"encode_decode_rtp");
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@ -127,7 +127,7 @@ void PacketLossTest::Perform() {
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#ifndef WEBRTC_CODEC_OPUS
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return;
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#else
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std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create(0));
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std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create());
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int codec_id = acm->Codec("opus", 48000, channels_);
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@ -104,8 +104,8 @@ void TestPack::reset_payload_size() {
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}
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TestAllCodecs::TestAllCodecs(int test_mode)
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: acm_a_(AudioCodingModule::Create(0)),
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acm_b_(AudioCodingModule::Create(1)),
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: acm_a_(AudioCodingModule::Create()),
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acm_b_(AudioCodingModule::Create()),
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channel_a_to_b_(NULL),
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test_count_(0),
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packet_size_samples_(0),
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@ -48,8 +48,8 @@ namespace {
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}
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TestRedFec::TestRedFec()
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: _acmA(AudioCodingModule::Create(0)),
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_acmB(AudioCodingModule::Create(1)),
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: _acmA(AudioCodingModule::Create()),
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_acmB(AudioCodingModule::Create()),
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_channelA2B(NULL),
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_testCntr(0) {
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}
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@ -108,8 +108,8 @@ void TestPackStereo::set_lost_packet(bool lost) {
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}
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TestStereo::TestStereo(int test_mode)
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: acm_a_(AudioCodingModule::Create(0)),
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acm_b_(AudioCodingModule::Create(1)),
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: acm_a_(AudioCodingModule::Create()),
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acm_b_(AudioCodingModule::Create()),
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channel_a2b_(NULL),
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test_cntr_(0),
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pack_size_samp_(0),
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@ -62,8 +62,8 @@ void ActivityMonitor::GetStatistics(uint32_t* counter) {
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}
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TestVadDtx::TestVadDtx()
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: acm_send_(AudioCodingModule::Create(0)),
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acm_receive_(AudioCodingModule::Create(1)),
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: acm_send_(AudioCodingModule::Create()),
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acm_receive_(AudioCodingModule::Create()),
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channel_(new Channel),
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monitor_(new ActivityMonitor) {
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EXPECT_EQ(0, acm_send_->RegisterTransportCallback(channel_.get()));
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@ -34,16 +34,14 @@ namespace webrtc {
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#define MAX_FILE_NAME_LENGTH_BYTE 500
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TwoWayCommunication::TwoWayCommunication(int testMode)
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: _acmA(AudioCodingModule::Create(1)),
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_acmRefA(AudioCodingModule::Create(3)),
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: _acmA(AudioCodingModule::Create()),
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_acmRefA(AudioCodingModule::Create()),
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_testMode(testMode) {
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AudioCodingModule::Config config;
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// The clicks will be more obvious in FAX mode. TODO(henrik.lundin) Really?
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config.neteq_config.playout_mode = kPlayoutFax;
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config.id = 2;
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config.decoder_factory = CreateBuiltinAudioDecoderFactory();
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_acmB.reset(AudioCodingModule::Create(config));
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config.id = 4;
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_acmRefB.reset(AudioCodingModule::Create(config));
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}
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@ -62,7 +60,7 @@ TwoWayCommunication::~TwoWayCommunication() {
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void TwoWayCommunication::ChooseCodec(uint8_t* codecID_A,
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uint8_t* codecID_B) {
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std::unique_ptr<AudioCodingModule> tmpACM(AudioCodingModule::Create(0));
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std::unique_ptr<AudioCodingModule> tmpACM(AudioCodingModule::Create());
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uint8_t noCodec = tmpACM->NumberOfCodecs();
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CodecInst codecInst;
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printf("List of Supported Codecs\n");
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@ -64,8 +64,8 @@ struct TestSettings {
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class DelayTest {
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public:
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DelayTest()
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: acm_a_(AudioCodingModule::Create(0)),
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acm_b_(AudioCodingModule::Create(1)),
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: acm_a_(AudioCodingModule::Create()),
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acm_b_(AudioCodingModule::Create()),
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channel_a2b_(new Channel),
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test_cntr_(0),
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encoding_sample_rate_hz_(8000) {}
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@ -67,8 +67,8 @@ int16_t SetISAConfig(ACMTestISACConfig& isacConfig, AudioCodingModule* acm,
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}
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ISACTest::ISACTest(int testMode)
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: _acmA(AudioCodingModule::Create(1)),
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_acmB(AudioCodingModule::Create(2)),
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: _acmA(AudioCodingModule::Create()),
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_acmB(AudioCodingModule::Create()),
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_testMode(testMode) {}
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ISACTest::~ISACTest() {}
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@ -61,8 +61,8 @@ class InsertPacketWithTiming {
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InsertPacketWithTiming()
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: sender_clock_(new SimulatedClock(0)),
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receiver_clock_(new SimulatedClock(0)),
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send_acm_(AudioCodingModule::Create(0, sender_clock_)),
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receive_acm_(AudioCodingModule::Create(0, receiver_clock_)),
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send_acm_(AudioCodingModule::Create(sender_clock_)),
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receive_acm_(AudioCodingModule::Create(receiver_clock_)),
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channel_(new Channel),
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seq_num_fid_(fopen(FLAG_seq_num, "rt")),
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send_ts_fid_(fopen(FLAG_send_ts, "rt")),
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@ -27,7 +27,7 @@
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namespace webrtc {
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OpusTest::OpusTest()
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: acm_receiver_(AudioCodingModule::Create(0)),
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: acm_receiver_(AudioCodingModule::Create()),
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channel_a2b_(NULL),
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counter_(0),
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payload_type_(255),
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@ -22,7 +22,7 @@ namespace webrtc {
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class TargetDelayTest : public ::testing::Test {
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protected:
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TargetDelayTest() : acm_(AudioCodingModule::Create(0)) {}
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TargetDelayTest() : acm_(AudioCodingModule::Create()) {}
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~TargetDelayTest() {}
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