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1 /* | 1 /* |
2 * Copyright 2004 The WebRTC project authors. All Rights Reserved. | 2 * Copyright 2004 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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36 std::unique_ptr<webrtc::AudioSinkInterface>* sink) { | 36 std::unique_ptr<webrtc::AudioSinkInterface>* sink) { |
37 channel->SetRawAudioSink(ssrc, std::move(*sink)); | 37 channel->SetRawAudioSink(ssrc, std::move(*sink)); |
38 return true; | 38 return true; |
39 } | 39 } |
40 | 40 |
41 struct SendPacketMessageData : public rtc::MessageData { | 41 struct SendPacketMessageData : public rtc::MessageData { |
42 rtc::CopyOnWriteBuffer packet; | 42 rtc::CopyOnWriteBuffer packet; |
43 rtc::PacketOptions options; | 43 rtc::PacketOptions options; |
44 }; | 44 }; |
45 | 45 |
| 46 #if defined(ENABLE_EXTERNAL_AUTH) |
| 47 // Returns the named header extension if found among all extensions, |
| 48 // nullptr otherwise. |
| 49 const webrtc::RtpExtension* FindHeaderExtension( |
| 50 const std::vector<webrtc::RtpExtension>& extensions, |
| 51 const std::string& uri) { |
| 52 for (const auto& extension : extensions) { |
| 53 if (extension.uri == uri) |
| 54 return &extension; |
| 55 } |
| 56 return nullptr; |
| 57 } |
| 58 #endif |
| 59 |
46 } // namespace | 60 } // namespace |
47 | 61 |
48 enum { | 62 enum { |
49 MSG_EARLYMEDIATIMEOUT = 1, | 63 MSG_EARLYMEDIATIMEOUT = 1, |
50 MSG_SEND_RTP_PACKET, | 64 MSG_SEND_RTP_PACKET, |
51 MSG_SEND_RTCP_PACKET, | 65 MSG_SEND_RTCP_PACKET, |
52 MSG_CHANNEL_ERROR, | 66 MSG_CHANNEL_ERROR, |
53 MSG_READYTOSENDDATA, | 67 MSG_READYTOSENDDATA, |
54 MSG_DATARECEIVED, | 68 MSG_DATARECEIVED, |
55 MSG_FIRSTPACKETRECEIVED, | 69 MSG_FIRSTPACKETRECEIVED, |
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1383 SafeSetError(desc.str(), error_desc); | 1397 SafeSetError(desc.str(), error_desc); |
1384 ret = false; | 1398 ret = false; |
1385 } | 1399 } |
1386 } | 1400 } |
1387 } | 1401 } |
1388 remote_streams_ = streams; | 1402 remote_streams_ = streams; |
1389 return ret; | 1403 return ret; |
1390 } | 1404 } |
1391 | 1405 |
1392 void BaseChannel::MaybeCacheRtpAbsSendTimeHeaderExtension_w( | 1406 void BaseChannel::MaybeCacheRtpAbsSendTimeHeaderExtension_w( |
1393 const std::vector<RtpHeaderExtension>& extensions) { | 1407 const std::vector<webrtc::RtpExtension>& extensions) { |
1394 // Absolute Send Time extension id is used only with external auth, | 1408 // Absolute Send Time extension id is used only with external auth, |
1395 // so do not bother searching for it and making asyncronious call to set | 1409 // so do not bother searching for it and making asyncronious call to set |
1396 // something that is not used. | 1410 // something that is not used. |
1397 #if defined(ENABLE_EXTERNAL_AUTH) | 1411 #if defined(ENABLE_EXTERNAL_AUTH) |
1398 const RtpHeaderExtension* send_time_extension = | 1412 const webrtc::RtpExtension* send_time_extension = |
1399 FindHeaderExtension(extensions, kRtpAbsoluteSenderTimeHeaderExtension); | 1413 FindHeaderExtension(extensions, webrtc::RtpExtension::kAbsSendTimeUri); |
1400 int rtp_abs_sendtime_extn_id = | 1414 int rtp_abs_sendtime_extn_id = |
1401 send_time_extension ? send_time_extension->id : -1; | 1415 send_time_extension ? send_time_extension->id : -1; |
1402 invoker_.AsyncInvoke<void>( | 1416 invoker_.AsyncInvoke<void>( |
1403 network_thread_, Bind(&BaseChannel::CacheRtpAbsSendTimeHeaderExtension_n, | 1417 network_thread_, Bind(&BaseChannel::CacheRtpAbsSendTimeHeaderExtension_n, |
1404 this, rtp_abs_sendtime_extn_id)); | 1418 this, rtp_abs_sendtime_extn_id)); |
1405 #endif | 1419 #endif |
1406 } | 1420 } |
1407 | 1421 |
1408 void BaseChannel::CacheRtpAbsSendTimeHeaderExtension_n( | 1422 void BaseChannel::CacheRtpAbsSendTimeHeaderExtension_n( |
1409 int rtp_abs_sendtime_extn_id) { | 1423 int rtp_abs_sendtime_extn_id) { |
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2398 return data_channel_type_ == DCT_RTP && BaseChannel::ShouldSetupDtlsSrtp_n(); | 2412 return data_channel_type_ == DCT_RTP && BaseChannel::ShouldSetupDtlsSrtp_n(); |
2399 } | 2413 } |
2400 | 2414 |
2401 void DataChannel::OnStreamClosedRemotely(uint32_t sid) { | 2415 void DataChannel::OnStreamClosedRemotely(uint32_t sid) { |
2402 rtc::TypedMessageData<uint32_t>* message = | 2416 rtc::TypedMessageData<uint32_t>* message = |
2403 new rtc::TypedMessageData<uint32_t>(sid); | 2417 new rtc::TypedMessageData<uint32_t>(sid); |
2404 signaling_thread()->Post(this, MSG_STREAMCLOSEDREMOTELY, message); | 2418 signaling_thread()->Post(this, MSG_STREAMCLOSEDREMOTELY, message); |
2405 } | 2419 } |
2406 | 2420 |
2407 } // namespace cricket | 2421 } // namespace cricket |
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