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| NetEqDecodingTest () |
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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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static void | PopulateRtpInfo (int frame_index, int timestamp, WebRtcRTPHeader *rtp_info) |
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static void | PopulateCng (int frame_index, int timestamp, WebRtcRTPHeader *rtp_info, uint8_t *payload, size_t *payload_len) |
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static const int | kTimeStepMs = 10 |
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static const size_t | kBlockSize8kHz = kTimeStepMs * 8 |
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static const size_t | kBlockSize16kHz = kTimeStepMs * 16 |
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static const size_t | kBlockSize32kHz = kTimeStepMs * 32 |
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static const size_t | kBlockSize48kHz = kTimeStepMs * 48 |
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static const int | kInitSampleRateHz = 8000 |
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◆ NetEqDecodingTestTwoInstances()
webrtc::NetEqDecodingTestTwoInstances::NetEqDecodingTestTwoInstances |
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◆ CreateSecondInstance()
void webrtc::NetEqDecodingTestTwoInstances::CreateSecondInstance |
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◆ SetUp()
void webrtc::NetEqDecodingTestTwoInstances::SetUp |
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inlineoverridevirtual |
◆ config2_
◆ neteq2_
std::unique_ptr<NetEq> webrtc::NetEqDecodingTestTwoInstances::neteq2_ |
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protected |
The documentation for this class was generated from the following file:
- Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_unittest.cc