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Issue 1360913004: Fix BWE bug where audio has timestamps in us. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: . Created 5 years, 2 months ago
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1 /* 1 /*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
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95 if (!rtp_header_parser_->Parse(packet, length, &header)) { 95 if (!rtp_header_parser_->Parse(packet, length, &header)) {
96 return false; 96 return false;
97 } 97 }
98 98
99 // Only forward if the parsed header has absolute sender time. RTP timestamps 99 // Only forward if the parsed header has absolute sender time. RTP timestamps
100 // may have different rates for audio and video and shouldn't be mixed. 100 // may have different rates for audio and video and shouldn't be mixed.
101 if (config_.combined_audio_video_bwe && 101 if (config_.combined_audio_video_bwe &&
102 header.extension.hasAbsoluteSendTime) { 102 header.extension.hasAbsoluteSendTime) {
103 int64_t arrival_time_ms = TickTime::MillisecondTimestamp(); 103 int64_t arrival_time_ms = TickTime::MillisecondTimestamp();
104 if (packet_time.timestamp >= 0) 104 if (packet_time.timestamp >= 0)
105 arrival_time_ms = packet_time.timestamp; 105 arrival_time_ms = (packet_time.timestamp + 500) / 1000;
106 size_t payload_size = length - header.headerLength; 106 size_t payload_size = length - header.headerLength;
107 remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_size, 107 remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_size,
108 header, false); 108 header, false);
109 } 109 }
110 return true; 110 return true;
111 } 111 }
112 } // namespace internal 112 } // namespace internal
113 } // namespace webrtc 113 } // namespace webrtc
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