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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 |
11 #include "webrtc/audio/audio_receive_stream.h" | 11 #include "webrtc/audio/audio_receive_stream.h" |
12 | 12 |
13 #include <string> | 13 #include <string> |
14 | 14 |
15 #include "webrtc/audio/conversion.h" | 15 #include "webrtc/audio/conversion.h" |
16 #include "webrtc/base/checks.h" | 16 #include "webrtc/base/checks.h" |
17 #include "webrtc/base/logging.h" | 17 #include "webrtc/base/logging.h" |
18 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimat
or.h" | 18 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimat
or.h" |
19 #include "webrtc/system_wrappers/interface/tick_util.h" | 19 #include "webrtc/system_wrappers/include/tick_util.h" |
20 #include "webrtc/voice_engine/include/voe_base.h" | 20 #include "webrtc/voice_engine/include/voe_base.h" |
21 #include "webrtc/voice_engine/include/voe_codec.h" | 21 #include "webrtc/voice_engine/include/voe_codec.h" |
22 #include "webrtc/voice_engine/include/voe_neteq_stats.h" | 22 #include "webrtc/voice_engine/include/voe_neteq_stats.h" |
23 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h" | 23 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h" |
24 #include "webrtc/voice_engine/include/voe_video_sync.h" | 24 #include "webrtc/voice_engine/include/voe_video_sync.h" |
25 #include "webrtc/voice_engine/include/voe_volume_control.h" | 25 #include "webrtc/voice_engine/include/voe_volume_control.h" |
26 | 26 |
27 namespace webrtc { | 27 namespace webrtc { |
28 std::string AudioReceiveStream::Config::Rtp::ToString() const { | 28 std::string AudioReceiveStream::Config::Rtp::ToString() const { |
29 std::stringstream ss; | 29 std::stringstream ss; |
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217 if (packet_time.timestamp >= 0) | 217 if (packet_time.timestamp >= 0) |
218 arrival_time_ms = (packet_time.timestamp + 500) / 1000; | 218 arrival_time_ms = (packet_time.timestamp + 500) / 1000; |
219 size_t payload_size = length - header.headerLength; | 219 size_t payload_size = length - header.headerLength; |
220 remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_size, | 220 remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_size, |
221 header, false); | 221 header, false); |
222 } | 222 } |
223 return true; | 223 return true; |
224 } | 224 } |
225 } // namespace internal | 225 } // namespace internal |
226 } // namespace webrtc | 226 } // namespace webrtc |
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