Searched refs:rtp_number (Results 1 – 10 of 10) sorted by relevance
/external/webrtc/webrtc/modules/audio_coding/codecs/isac/fix/test/ |
D | test_iSACfixfloat.c | 57 uint16_t rtp_number; member 79 BN_data->rtp_number++; in get_arrival_time() 279 BN_data.rtp_number = 0; in main() 464 stream_len, BN_data.rtp_number, in main() 484 stream_len, BN_data.rtp_number, in main() 487 stream_len, BN_data.rtp_number, in main() 525 stream_len, BN_data.rtp_number, in main() 529 stream_len, BN_data.rtp_number, in main() 549 ISACFIX_main_inst, streamdata, stream_len, BN_data.rtp_number, in main()
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D | kenny.cc | 49 uint16_t rtp_number; member 72 BN_data->rtp_number++; in get_arrival_time() 92 BN_data->rtp_number++; in get_arrival_time2() 487 BN_data.rtp_number = 0; in main() 707 BN_data.rtp_number, in main()
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/external/webrtc/webrtc/modules/audio_coding/codecs/isac/main/source/ |
D | bandwidth_estimator.c | 141 const uint16_t rtp_number, in WebRtcIsac_UpdateBandwidthEstimator() argument 193 bwest_str->prev_rec_rtp_number = rtp_number; in WebRtcIsac_UpdateBandwidthEstimator() 320 if ( rtp_number == bwest_str->prev_rec_rtp_number + 1 ) in WebRtcIsac_UpdateBandwidthEstimator() 464 bwest_str->prev_rec_rtp_number = rtp_number; in WebRtcIsac_UpdateBandwidthEstimator()
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D | bandwidth_estimator.h | 94 const uint16_t rtp_number,
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/external/webrtc/webrtc/modules/audio_coding/codecs/isac/fix/source/ |
D | bandwidth_estimator.h | 61 const uint16_t rtp_number,
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/external/webrtc/webrtc/modules/audio_coding/codecs/isac/main/util/ |
D | utility.h | 97 unsigned int rtp_number; member
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D | utility.c | 177 BN_data->rtp_number++; in get_arrival_time()
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/external/webrtc/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ |
D | ReleaseTest-API.cc | 468 BN_data.rtp_number = 0; in main() 754 BN_data.rtp_number--; in main() 767 stream_len, BN_data.rtp_number, BN_data.sample_count, in main()
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/external/webrtc/webrtc/modules/audio_coding/codecs/isac/main/test/ |
D | simpleKenny.c | 115 packetData.rtp_number = 0; in main() 361 ISAC_main_inst, payload, stream_len, packetData.rtp_number, in main()
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/external/webrtc/webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/ |
D | SwitchingSampRate.cc | 413 packetData[senderIdx]->rtp_number, in main()
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