| /external/webrtc/webrtc/modules/audio_coding/codecs/ilbc/ |
| H A D | 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
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| /external/webrtc/webrtc/modules/audio_coding/codecs/g722/ |
| H A D | audio_encoder_g722.h | 29 int frame_size_ms = 20; member in struct:webrtc::final::Config
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| /external/webrtc/webrtc/modules/audio_coding/codecs/g711/ |
| H A D | 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) {}
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| /external/webrtc/webrtc/modules/audio_coding/codecs/isac/ |
| H A D | audio_encoder_isac_t.h | 37 int frame_size_ms = 30; member in struct:webrtc::final::Config
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| H A D | 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::__anon30765::IsacTestParam
205 << itp.frame_size_ms << '}';
230 p.sample_rate_hz, p.frame_size_ms);
104 TestGetSetBandwidthInfo(const int16_t* speech_data, int rate_bits_per_second, int sample_rate_hz, int frame_size_ms) argument
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| /external/webrtc/webrtc/modules/audio_coding/codecs/opus/ |
| H A D | audio_encoder_opus.h | 33 int frame_size_ms = 20; member in struct:webrtc::final::Config
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| /external/webrtc/webrtc/modules/audio_processing/ |
| H A D | voice_detection_impl.cc | 150 int VoiceDetectionImpl::frame_size_ms() const { function in class:webrtc::VoiceDetectionImpl
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| /external/webrtc/webrtc/modules/audio_coding/codecs/isac/fix/source/ |
| H A D | isac_fix_type.h | 30 int frame_size_ms,
32 return WebRtcIsacfix_ControlBwe(inst, rate_bps, frame_size_ms,
28 ControlBwe(instance_type* inst, int32_t rate_bps, int frame_size_ms, int16_t enforce_frame_size) argument
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| /external/webrtc/webrtc/modules/audio_coding/codecs/isac/main/source/ |
| H A D | isac_float_type.h | 28 int frame_size_ms,
30 return WebRtcIsac_ControlBwe(inst, rate_bps, frame_size_ms,
26 ControlBwe(instance_type* inst, int32_t rate_bps, int frame_size_ms, int16_t enforce_frame_size) argument
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