| /external/webrtc/webrtc/modules/audio_coding/codecs/ilbc/ |
| audio_encoder_ilbc.cc | 27 config.frame_size_ms = codec_inst.pacsize / 8;
38 return (frame_size_ms == 20 || frame_size_ms == 30 || frame_size_ms == 40 ||
39 frame_size_ms == 60) &&
40 static_cast<size_t>(kSampleRateHz / 100 * (frame_size_ms / 10)) <=
47 static_cast<size_t>(config.frame_size_ms / 10)),
136 const int encoder_frame_size_ms = config_.frame_size_ms > 30
137 ? config_.frame_size_ms / 2
138 : config_.frame_size_ms;
[all...] |
| audio_encoder_ilbc.h | 28 int frame_size_ms = 30; // Valid values are 20, 30, 40, and 60 ms. member in struct:webrtc::final::Config
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/g711/ |
| audio_encoder_pcm.cc | 26 config.frame_size_ms = codec_inst.pacsize / 8;
35 return (frame_size_ms % 10 == 0) && (num_channels >= 1);
43 static_cast<size_t>(config.frame_size_ms / 10)),
45 config.num_channels * config.frame_size_ms * sample_rate_hz / 1000),
48 RTC_CHECK_EQ(config.frame_size_ms % 10, 0)
|
| audio_encoder_pcm.h | 27 int frame_size_ms; member in struct:webrtc::AudioEncoderPcm::Config
33 : frame_size_ms(20), num_channels(1), payload_type(pt) {}
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/isac/ |
| audio_encoder_isac_t_impl.h | 28 config.frame_size_ms =
50 return (frame_size_ms == 30 || frame_size_ms == 60) &&
58 (frame_size_ms == 30 &&
168 RTC_CHECK_EQ(0, T::ControlBwe(isac_state_, bit_rate, config.frame_size_ms,
171 RTC_CHECK_EQ(0, T::Control(isac_state_, bit_rate, config.frame_size_ms));
|
| unittest.cc | 107 int frame_size_ms) {
117 ASSERT_EQ(0, T::ControlBwe(encdec, bit_rate, frame_size_ms, false));
119 ASSERT_EQ(0, T::Control(encdec, bit_rate, frame_size_ms));
127 ASSERT_EQ(0, T::ControlBwe(enc, bit_rate, frame_size_ms, false));
129 ASSERT_EQ(0, T::Control(enc, bit_rate, frame_size_ms));
158 EXPECT_EQ(frame_size_ms, duration1_ms);
199 int frame_size_ms; member in struct:webrtc::__anon44084::IsacTestParam
205 << itp.frame_size_ms << '}';
230 p.sample_rate_hz, p.frame_size_ms);
|
| audio_encoder_isac_t.h | 37 int frame_size_ms = 30; member in struct:webrtc::final::Config
|
| /external/webrtc/webrtc/modules/audio_coding/neteq/test/ |
| 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;
|
| 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;
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/g722/ |
| audio_encoder_g722.cc | 27 config.frame_size_ms = codec_inst.pacsize / 16;
35 return (frame_size_ms > 0) && (frame_size_ms % 10 == 0) &&
43 static_cast<size_t>(config.frame_size_ms / 10)),
|
| audio_encoder_g722.h | 29 int frame_size_ms = 20; member in struct:webrtc::final::Config
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/pcm16b/ |
| audio_encoder_pcm16b.cc | 34 config.frame_size_ms = rtc::CheckedDivExact(
|
| /external/webrtc/webrtc/modules/audio_processing/ |
| voice_detection_impl.h | 40 int frame_size_ms() const override;
|
| voice_detection_impl.cc | 150 int VoiceDetectionImpl::frame_size_ms() const { function in class:webrtc::VoiceDetectionImpl
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/isac/fix/source/ |
| isac_fix_type.h | 30 int frame_size_ms,
32 return WebRtcIsacfix_ControlBwe(inst, rate_bps, frame_size_ms,
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/isac/main/source/ |
| isac_float_type.h | 28 int frame_size_ms,
30 return WebRtcIsac_ControlBwe(inst, rate_bps, frame_size_ms,
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/opus/ |
| audio_encoder_opus.cc | 28 config.frame_size_ms = rtc::CheckedDivExact(codec_inst.pacsize, 48);
80 if (frame_size_ms <= 0 || frame_size_ms % 10 != 0)
216 return static_cast<size_t>(rtc::CheckedDivExact(config_.frame_size_ms, 10));
|
| audio_encoder_opus.h | 33 int frame_size_ms = 20; member in struct:webrtc::final::Config
|
| /external/webrtc/webrtc/modules/audio_coding/neteq/ |
| audio_decoder_unittest.cc | 294 config.frame_size_ms = static_cast<int>(frame_size_ / 8);
307 config.frame_size_ms = static_cast<int>(frame_size_ / 8);
323 config.frame_size_ms =
339 config.frame_size_ms = 30;
374 config.frame_size_ms =
391 config.frame_size_ms =
408 config.frame_size_ms =
424 config.frame_size_ms = 10;
441 config.frame_size_ms = 10;
456 config.frame_size_ms = static_cast<int>(frame_size_) / 48
[all...] |
| /external/webrtc/webrtc/modules/audio_processing/include/ |
| mock_audio_processing.h | 161 MOCK_CONST_METHOD0(frame_size_ms,
|
| audio_processing.h | [all...] |
