| /external/webrtc/webrtc/modules/audio_coding/codecs/ilbc/ |
| D | audio_encoder_ilbc.cc | 27 config.frame_size_ms = codec_inst.pacsize / 8; in CreateConfig()
38 return (frame_size_ms == 20 || frame_size_ms == 30 || frame_size_ms == 40 || in IsOk()
39 frame_size_ms == 60) && in IsOk()
40 static_cast<size_t>(kSampleRateHz / 100 * (frame_size_ms / 10)) <= in IsOk()
47 static_cast<size_t>(config.frame_size_ms / 10)), in AudioEncoderIlbc()
136 const int encoder_frame_size_ms = config_.frame_size_ms > 30 in Reset()
137 ? config_.frame_size_ms / 2 in Reset()
138 : config_.frame_size_ms; in Reset()
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| D | audio_encoder_ilbc.h | 28 int frame_size_ms = 30; // Valid values are 20, 30, 40, and 60 ms. member
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| /external/webrtc/webrtc/modules/audio_coding/codecs/isac/ |
| D | unittest.cc | 107 int frame_size_ms) { in TestGetSetBandwidthInfo() argument
117 ASSERT_EQ(0, T::ControlBwe(encdec, bit_rate, frame_size_ms, false)); in TestGetSetBandwidthInfo()
119 ASSERT_EQ(0, T::Control(encdec, bit_rate, frame_size_ms)); in TestGetSetBandwidthInfo()
127 ASSERT_EQ(0, T::ControlBwe(enc, bit_rate, frame_size_ms, false)); in TestGetSetBandwidthInfo()
129 ASSERT_EQ(0, T::Control(enc, bit_rate, frame_size_ms)); in TestGetSetBandwidthInfo()
158 EXPECT_EQ(frame_size_ms, duration1_ms); in TestGetSetBandwidthInfo()
199 int frame_size_ms; member
205 << itp.frame_size_ms << '}'; in operator <<()
230 p.sample_rate_hz, p.frame_size_ms); in TEST_P()
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| D | audio_encoder_isac_t_impl.h | 28 config.frame_size_ms = in CreateIsacConfig()
50 return (frame_size_ms == 30 || frame_size_ms == 60) && in IsOk()
58 (frame_size_ms == 30 && in IsOk()
168 RTC_CHECK_EQ(0, T::ControlBwe(isac_state_, bit_rate, config.frame_size_ms, in RecreateEncoderInstance()
171 RTC_CHECK_EQ(0, T::Control(isac_state_, bit_rate, config.frame_size_ms)); in RecreateEncoderInstance()
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| D | audio_encoder_isac_t.h | 37 int frame_size_ms = 30; member
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| /external/webrtc/webrtc/modules/audio_coding/codecs/g711/ |
| D | audio_encoder_pcm.cc | 26 config.frame_size_ms = codec_inst.pacsize / 8; in CreateConfig()
35 return (frame_size_ms % 10 == 0) && (num_channels >= 1); in IsOk()
43 static_cast<size_t>(config.frame_size_ms / 10)), in AudioEncoderPcm()
45 config.num_channels * config.frame_size_ms * sample_rate_hz / 1000), in AudioEncoderPcm()
48 RTC_CHECK_EQ(config.frame_size_ms % 10, 0) in AudioEncoderPcm()
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| D | audio_encoder_pcm.h | 27 int frame_size_ms; member
33 : frame_size_ms(20), num_channels(1), payload_type(pt) {} in Config()
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| /external/webrtc/webrtc/modules/audio_coding/codecs/g722/ |
| D | audio_encoder_g722.cc | 27 config.frame_size_ms = codec_inst.pacsize / 16; in CreateConfig()
35 return (frame_size_ms > 0) && (frame_size_ms % 10 == 0) && in IsOk()
43 static_cast<size_t>(config.frame_size_ms / 10)), in AudioEncoderG722()
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| D | audio_encoder_g722.h | 29 int frame_size_ms = 20; member
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| /external/webrtc/webrtc/modules/audio_coding/neteq/ |
| D | audio_decoder_unittest.cc | 294 config.frame_size_ms = static_cast<int>(frame_size_ / 8); in AudioDecoderPcmUTest()
307 config.frame_size_ms = static_cast<int>(frame_size_ / 8); in AudioDecoderPcmATest()
323 config.frame_size_ms = in AudioDecoderPcm16BTest()
339 config.frame_size_ms = 30; in AudioDecoderIlbcTest()
374 config.frame_size_ms = in AudioDecoderIsacFloatTest()
391 config.frame_size_ms = in AudioDecoderIsacSwbTest()
408 config.frame_size_ms = in AudioDecoderIsacFixTest()
424 config.frame_size_ms = 10; in AudioDecoderG722Test()
441 config.frame_size_ms = 10; in AudioDecoderG722StereoTest()
456 config.frame_size_ms = static_cast<int>(frame_size_) / 48; in AudioDecoderOpusTest()
[all …]
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| /external/webrtc/webrtc/modules/audio_coding/codecs/opus/ |
| D | audio_encoder_opus.cc | 28 config.frame_size_ms = rtc::CheckedDivExact(codec_inst.pacsize, 48); in CreateConfig()
80 if (frame_size_ms <= 0 || frame_size_ms % 10 != 0) in IsOk()
216 return static_cast<size_t>(rtc::CheckedDivExact(config_.frame_size_ms, 10)); in Num10msFramesPerPacket()
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| D | audio_encoder_opus.h | 33 int frame_size_ms = 20; member
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| /external/webrtc/webrtc/modules/audio_coding/neteq/test/ |
| D | neteq_pcmu_quality_test.cc | 37 DEFINE_int32(frame_size_ms, 20, "Codec frame size (milliseconds).");
55 config.frame_size_ms = FLAGS_frame_size_ms; in SetUp()
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| D | neteq_ilbc_quality_test.cc | 37 DEFINE_int32(frame_size_ms, 20, "Codec frame size (milliseconds).");
55 config.frame_size_ms = FLAGS_frame_size_ms; in SetUp()
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| /external/webrtc/webrtc/modules/audio_coding/codecs/isac/fix/source/ |
| D | isac_fix_type.h | 30 int frame_size_ms, in ControlBwe() argument
32 return WebRtcIsacfix_ControlBwe(inst, rate_bps, frame_size_ms, in ControlBwe()
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| /external/webrtc/webrtc/modules/audio_coding/codecs/isac/main/source/ |
| D | isac_float_type.h | 28 int frame_size_ms, in ControlBwe()
30 return WebRtcIsac_ControlBwe(inst, rate_bps, frame_size_ms, in ControlBwe()
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| /external/webrtc/webrtc/modules/audio_coding/codecs/pcm16b/ |
| D | audio_encoder_pcm16b.cc | 34 config.frame_size_ms = rtc::CheckedDivExact( in CreateConfig()
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| /external/webrtc/webrtc/modules/audio_processing/ |
| D | voice_detection_impl.h | 40 int frame_size_ms() const override;
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| D | voice_detection_impl.cc | 150 int VoiceDetectionImpl::frame_size_ms() const { in frame_size_ms() function in webrtc::VoiceDetectionImpl
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| /external/webrtc/webrtc/modules/audio_processing/include/ |
| D | mock_audio_processing.h | 161 MOCK_CONST_METHOD0(frame_size_ms,
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| D | audio_processing.h | 957 virtual int frame_size_ms() const = 0;
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