#include <opus_test.h>
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| OpusTest () |
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void | TestDtxEffect (bool dtx, int block_length_ms) |
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void | PrepareSpeechData (size_t channel, int block_length_ms, int loop_length_ms) |
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int | EncodeDecode (WebRtcOpusEncInst *encoder, rtc::ArrayView< const int16_t > input_audio, WebRtcOpusDecInst *decoder, int16_t *output_audio, int16_t *audio_type) |
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void | SetMaxPlaybackRate (WebRtcOpusEncInst *encoder, opus_int32 expect, int32_t set) |
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void | CheckAudioBounded (const int16_t *audio, size_t samples, size_t channels, uint16_t bound) const |
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◆ OpusTest() [1/3]
webrtc::OpusTest::OpusTest |
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◆ ~OpusTest() [1/2]
webrtc::OpusTest::~OpusTest |
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◆ OpusTest() [2/3]
webrtc::OpusTest::OpusTest |
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protected |
◆ OpusTest() [3/3]
webrtc::OpusTest::OpusTest |
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◆ ~OpusTest() [2/2]
webrtc::OpusTest::~OpusTest |
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◆ CheckAudioBounded()
◆ EncodeDecode()
◆ Perform() [1/2]
void webrtc::OpusTest::Perform |
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virtual |
◆ Perform() [2/2]
void webrtc::OpusTest::Perform |
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virtual |
◆ PrepareSpeechData()
void webrtc::OpusTest::PrepareSpeechData |
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size_t |
channel, |
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int |
block_length_ms, |
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int |
loop_length_ms |
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protected |
◆ SetMaxPlaybackRate()
◆ TestDtxEffect()
void webrtc::OpusTest::TestDtxEffect |
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bool |
dtx, |
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int |
block_length_ms |
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protected |
◆ application_
int webrtc::OpusTest::application_ |
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protected |
◆ bitstream_
◆ channels_
size_t webrtc::OpusTest::channels_ |
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protected |
◆ encoded_bytes_
size_t webrtc::OpusTest::encoded_bytes_ |
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protected |
◆ opus_decoder_
◆ opus_encoder_
◆ speech_data_
The documentation for this class was generated from the following files:
- DerivedData/WebKit/Build/Products/Debug/usr/local/include/webrtc/modules/audio_coding/test/opus_test.h
- Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
- Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/opus_test.cc