| /external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
| packet_unittest.cc | 119 uint32_t timestamp_offset,
128 rtp_data[1] = timestamp_offset >> 6;
129 rtp_data[2] = (timestamp_offset & 0x3F) << 2;
154 uint32_t timestamp_offset = 100 * i; local
158 payload_type, timestamp_offset, block_length, last_block, payload_ptr);
|
| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| fec_receiver_impl.cc | 112 uint16_t timestamp_offset = local
114 timestamp_offset +=
116 timestamp_offset = timestamp_offset >> 2;
117 if (timestamp_offset != 0) {
|
| /external/webrtc/talk/media/base/ |
| rtpdataengine.h | 69 RtpClock(int clockrate, uint16_t first_seq_num, uint32_t timestamp_offset)
72 timestamp_offset_(timestamp_offset) {}
|
| /external/webrtc/webrtc/modules/remote_bitrate_estimator/ |
| remote_bitrate_estimator_unittest_helper.h | 69 uint32_t timestamp_offset, int64_t rtcp_receive_time);
|
| remote_bitrate_estimator_unittest_helper.cc | 34 uint32_t timestamp_offset,
42 rtp_timestamp_offset_(timestamp_offset),
|
| /external/webrtc/webrtc/modules/audio_coding/neteq/ |
| payload_splitter.cc | 66 int timestamp_offset = (payload_ptr[1] << 6) + local
69 timestamp_offset;
|
| nack_unittest.cc | 223 uint32_t timestamp_offset = local
230 timestamp_offset + kLostPackets[n] * kTimestampIncrement;
|
| payload_splitter_unittest.cc | 73 // |num_payloads|). Each redundant payload is |timestamp_offset| samples
77 int timestamp_offset,
99 int this_offset = (num_payloads - i - 1) * timestamp_offset;
[all...] |
