| /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,
|
| 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,
|
| 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,
|
| /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,
|
| 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,
|
| /external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/ |
| bandwidth_estimator.h | 61 const uint16_t rtp_number,
|
| /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,
|
| 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;
|
| /external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/main/util/ |
| utility.h | 97 unsigned int rtp_number; member in struct:__anon19701
|
| utility.c | 177 BN_data->rtp_number++;
|
| /external/webrtc/src/modules/audio_coding/codecs/isac/fix/source/ |
| bandwidth_estimator.h | 61 const WebRtc_UWord16 rtp_number,
|
| /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,
|
| 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;
|
| /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,
|
| /external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/ |
| SwitchingSampRate.cc | 412 bitStream, streamLen, packetData[senderIdx]->rtp_number,
|
