/external/webrtc/src/modules/audio_coding/codecs/isac/fix/test/ |
Isac_test.cc | 30 WebRtc_UWord16 rtp_number; member in struct:__anon37132 51 BN_data->rtp_number++; 161 BN_data.rtp_number = 0; 203 BN_data.rtp_number,
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test_iSACfixfloat.c | 63 WebRtc_UWord16 rtp_number; member in struct:__anon37134 85 BN_data->rtp_number++; 313 BN_data.rtp_number = 0; 496 BN_data.rtp_number, 522 BN_data.rtp_number, 527 BN_data.rtp_number, 573 BN_data.rtp_number, 579 BN_data.rtp_number, 604 BN_data.rtp_number,
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kenny.c | 48 WebRtc_UWord16 rtp_number; member in struct:__anon37133 70 BN_data->rtp_number++; 90 BN_data->rtp_number++; 492 BN_data.rtp_number = 0; 712 BN_data.rtp_number,
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/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/fix/test/ |
test_iSACfixfloat.c | 63 uint16_t rtp_number; member in struct:__anon19678 85 BN_data->rtp_number++; 313 BN_data.rtp_number = 0; 496 BN_data.rtp_number, 522 BN_data.rtp_number, 527 BN_data.rtp_number, 573 BN_data.rtp_number, 579 BN_data.rtp_number, 604 BN_data.rtp_number,
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kenny.cc | 49 uint16_t rtp_number; member in struct:__anon19677 71 BN_data->rtp_number++; 91 BN_data->rtp_number++; 483 BN_data.rtp_number = 0; 703 BN_data.rtp_number,
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/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/ |
bandwidth_estimator.h | 61 const uint16_t rtp_number,
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/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/ |
bandwidth_estimator.h | 85 /* rtp_number - value from RTP packet, from NetEq */ 94 const uint16_t rtp_number,
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bandwidth_estimator.c | 127 /* rtp_number - value from RTP packet, from NetEq */ 136 const uint16_t rtp_number, 187 bwest_str->prev_rec_rtp_number = rtp_number; 314 if ( rtp_number == bwest_str->prev_rec_rtp_number + 1 ) 458 bwest_str->prev_rec_rtp_number = rtp_number;
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/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/main/util/ |
utility.h | 97 unsigned int rtp_number; member in struct:__anon19701
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utility.c | 177 BN_data->rtp_number++;
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/external/webrtc/src/modules/audio_coding/codecs/isac/fix/source/ |
bandwidth_estimator.h | 61 const WebRtc_UWord16 rtp_number,
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/external/webrtc/src/modules/audio_coding/codecs/isac/main/source/ |
bandwidth_estimator.h | 85 /* rtp_number - value from RTP packet, from NetEq */ 94 const WebRtc_UWord16 rtp_number,
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bandwidth_estimator.c | 127 /* rtp_number - value from RTP packet, from NetEq */ 136 const WebRtc_UWord16 rtp_number, 187 bwest_str->prev_rec_rtp_number = rtp_number; 314 if ( rtp_number == bwest_str->prev_rec_rtp_number + 1 ) 458 bwest_str->prev_rec_rtp_number = rtp_number;
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/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ |
ReleaseTest-API.cc | 506 BN_data.rtp_number = 0; 835 BN_data.rtp_number--; 852 stream_len, BN_data.rtp_number, BN_data.sample_count, [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/main/test/ |
simpleKenny.c | 116 packetData.rtp_number = 0; 401 payload, stream_len, packetData.rtp_number,
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/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/ |
SwitchingSampRate.cc | 412 bitStream, streamLen, packetData[senderIdx]->rtp_number,
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