/external/webrtc/modules/audio_coding/codecs/ |
D | builtin_audio_encoder_factory_unittest.cc | 51 EXPECT_EQ(encoder->RtpTimestampRateHz(), spec.format.clockrate_hz); in TEST_P() 81 blocks * encoder->RtpTimestampRateHz() / 100, audio, &out); in TEST_P() 91 blocks * encoder->RtpTimestampRateHz() / 100; in TEST_P() 95 blocks * encoder->RtpTimestampRateHz() / 100, audio, &out); in TEST_P()
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/external/webrtc/audio/voip/ |
D | audio_egress.cc | 51 encoder->RtpTimestampRateHz()); in SetEncoder() 53 encoder->RtpTimestampRateHz(), in SetEncoder()
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/external/webrtc/modules/audio_coding/codecs/cng/ |
D | audio_encoder_cng.cc | 41 int RtpTimestampRateHz() const override; 104 int AudioEncoderCng::RtpTimestampRateHz() const { in RtpTimestampRateHz() function in webrtc::__anonc964e42d0111::AudioEncoderCng 105 return speech_encoder_->RtpTimestampRateHz(); in RtpTimestampRateHz()
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/external/webrtc/modules/audio_coding/codecs/red/ |
D | audio_encoder_copy_red.cc | 44 int AudioEncoderCopyRed::RtpTimestampRateHz() const { in RtpTimestampRateHz() function in webrtc::AudioEncoderCopyRed 45 return speech_encoder_->RtpTimestampRateHz(); in RtpTimestampRateHz()
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D | audio_encoder_copy_red.h | 49 int RtpTimestampRateHz() const override;
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/external/webrtc/modules/audio_coding/codecs/g722/ |
D | audio_encoder_g722.h | 34 int RtpTimestampRateHz() const override;
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D | audio_encoder_g722.cc | 57 int AudioEncoderG722Impl::RtpTimestampRateHz() const { in RtpTimestampRateHz() function in webrtc::AudioEncoderG722Impl
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/external/webrtc/modules/audio_coding/neteq/tools/ |
D | encode_neteq_input.cc | 82 num_samples * encoder_->RtpTimestampRateHz() / in CreatePacket()
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D | rtp_encode.cc | 317 const int timestamp_rate_hz = codec->RtpTimestampRateHz(); in RunRtpEncode()
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/external/webrtc/test/ |
D | mock_audio_encoder.h | 28 MOCK_METHOD(int, RtpTimestampRateHz, (), (const, override));
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/external/webrtc/api/audio_codecs/ |
D | audio_encoder.cc | 31 int AudioEncoder::RtpTimestampRateHz() const { in RtpTimestampRateHz() function in webrtc::AudioEncoder
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D | audio_encoder.h | 126 virtual int RtpTimestampRateHz() const;
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/external/webrtc/modules/audio_coding/codecs/opus/ |
D | audio_encoder_opus.h | 66 int RtpTimestampRateHz() const override;
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D | audio_encoder_opus.cc | 397 int AudioEncoderOpusImpl::RtpTimestampRateHz() const { in RtpTimestampRateHz() function in webrtc::AudioEncoderOpusImpl
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/external/webrtc/audio/ |
D | channel_send.cc | 577 encoder->RtpTimestampRateHz()); in SetEncoder() 579 encoder->RtpTimestampRateHz(), in SetEncoder()
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D | audio_send_stream_unittest.cc | 109 ON_CALL(*encoder.get(), RtpTimestampRateHz()) in SetupAudioEncoderMock()
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/external/webrtc/modules/audio_coding/acm2/ |
D | audio_coding_module_unittest.cc | 1124 external_speech_encoder->RtpTimestampRateHz() / 100); in RegisterExternalSendCodec() 1764 EXPECT_CALL(*mock_encoder, RtpTimestampRateHz()) in TEST_F() 1766 .WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::RtpTimestampRateHz)); in TEST_F()
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D | audio_coding_module.cc | 253 encoder_stack_->RtpTimestampRateHz(), in Encode()
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