OLD | NEW |
1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 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/modules/rtp_rtcp/source/rtp_sender_audio.h" | 11 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h" |
12 | 12 |
13 #include <string.h> | 13 #include <string.h> |
14 | 14 |
15 #include <memory> | 15 #include <memory> |
16 #include <utility> | 16 #include <utility> |
17 | 17 |
18 #include "webrtc/base/logging.h" | |
19 #include "webrtc/base/timeutils.h" | |
20 #include "webrtc/base/trace_event.h" | |
21 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" | 18 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
22 #include "webrtc/modules/rtp_rtcp/source/byte_io.h" | 19 #include "webrtc/modules/rtp_rtcp/source/byte_io.h" |
23 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h" | 20 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h" |
24 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h" | 21 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h" |
| 22 #include "webrtc/rtc_base/logging.h" |
| 23 #include "webrtc/rtc_base/timeutils.h" |
| 24 #include "webrtc/rtc_base/trace_event.h" |
25 | 25 |
26 namespace webrtc { | 26 namespace webrtc { |
27 | 27 |
28 RTPSenderAudio::RTPSenderAudio(Clock* clock, RTPSender* rtp_sender) | 28 RTPSenderAudio::RTPSenderAudio(Clock* clock, RTPSender* rtp_sender) |
29 : clock_(clock), | 29 : clock_(clock), |
30 rtp_sender_(rtp_sender) {} | 30 rtp_sender_(rtp_sender) {} |
31 | 31 |
32 RTPSenderAudio::~RTPSenderAudio() {} | 32 RTPSenderAudio::~RTPSenderAudio() {} |
33 | 33 |
34 int32_t RTPSenderAudio::RegisterAudioPayload( | 34 int32_t RTPSenderAudio::RegisterAudioPayload( |
(...skipping 305 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
340 TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "Audio::SendTelephoneEvent", | 340 TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "Audio::SendTelephoneEvent", |
341 "timestamp", packet->Timestamp(), "seqnum", packet->SequenceNumber()); | 341 "timestamp", packet->Timestamp(), "seqnum", packet->SequenceNumber()); |
342 result = rtp_sender_->SendToNetwork(std::move(packet), kAllowRetransmission, | 342 result = rtp_sender_->SendToNetwork(std::move(packet), kAllowRetransmission, |
343 RtpPacketSender::kHighPriority); | 343 RtpPacketSender::kHighPriority); |
344 send_count--; | 344 send_count--; |
345 } while (send_count > 0 && result); | 345 } while (send_count > 0 && result); |
346 | 346 |
347 return result; | 347 return result; |
348 } | 348 } |
349 } // namespace webrtc | 349 } // namespace webrtc |
OLD | NEW |