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| NetEqDecodingTest () |
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virtual void | SetUp () |
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virtual void | TearDown () |
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void | SelectDecoders (NetEqDecoder *used_codec) |
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void | OpenInputFile (const std::string &rtp_file) |
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void | Process () |
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void | DecodeAndCompare (const std::string &rtp_file, const std::string &output_checksum, const std::string &network_stats_checksum, const std::string &rtcp_stats_checksum, bool gen_ref) |
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void | WrapTest (uint16_t start_seq_no, uint32_t start_timestamp, const std::set< uint16_t > &drop_seq_numbers, bool expect_seq_no_wrap, bool expect_timestamp_wrap) |
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void | LongCngWithClockDrift (double drift_factor, double network_freeze_ms, bool pull_audio_during_freeze, int delay_tolerance_ms, int max_time_to_speech_ms) |
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void | DuplicateCng () |
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rtc::Optional< uint32_t > | PlayoutTimestamp () |
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| Test () |
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◆ NetEqDecodingTest()
webrtc::NetEqDecodingTest::NetEqDecodingTest |
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◆ DecodeAndCompare()
◆ DuplicateCng()
void webrtc::NetEqDecodingTest::DuplicateCng |
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◆ LongCngWithClockDrift()
void webrtc::NetEqDecodingTest::LongCngWithClockDrift |
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double |
drift_factor, |
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double |
network_freeze_ms, |
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bool |
pull_audio_during_freeze, |
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int |
delay_tolerance_ms, |
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int |
max_time_to_speech_ms |
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◆ OpenInputFile()
void webrtc::NetEqDecodingTest::OpenInputFile |
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const std::string & |
rtp_file | ) |
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◆ PlayoutTimestamp()
◆ PopulateCng()
◆ PopulateRtpInfo()
void webrtc::NetEqDecodingTest::PopulateRtpInfo |
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int |
frame_index, |
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int |
timestamp, |
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WebRtcRTPHeader * |
rtp_info |
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) |
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staticprotected |
◆ Process()
void webrtc::NetEqDecodingTest::Process |
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◆ SelectDecoders()
◆ SetUp()
void webrtc::NetEqDecodingTest::SetUp |
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◆ TearDown()
void webrtc::NetEqDecodingTest::TearDown |
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◆ WrapTest()
void webrtc::NetEqDecodingTest::WrapTest |
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uint16_t |
start_seq_no, |
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uint32_t |
start_timestamp, |
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const std::set< uint16_t > & |
drop_seq_numbers, |
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bool |
expect_seq_no_wrap, |
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bool |
expect_timestamp_wrap |
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◆ algorithmic_delay_ms_
int webrtc::NetEqDecodingTest::algorithmic_delay_ms_ |
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◆ config_
◆ kBlockSize16kHz
◆ kBlockSize32kHz
◆ kBlockSize48kHz
◆ kBlockSize8kHz
◆ kInitSampleRateHz
const int webrtc::NetEqDecodingTest::kInitSampleRateHz = 8000 |
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◆ kTimeStepMs
const int webrtc::NetEqDecodingTest::kTimeStepMs = 10 |
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◆ neteq_
NetEq* webrtc::NetEqDecodingTest::neteq_ |
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◆ out_frame_
◆ output_sample_rate_
int webrtc::NetEqDecodingTest::output_sample_rate_ |
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◆ packet_
std::unique_ptr<test::Packet> webrtc::NetEqDecodingTest::packet_ |
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◆ rtp_source_
◆ sim_clock_
unsigned int webrtc::NetEqDecodingTest::sim_clock_ |
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The documentation for this class was generated from the following file:
- Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_unittest.cc