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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 |
| 11 #include <utility> | 11 #include <utility> |
| 12 | 12 |
| 13 #include "webrtc/pc/channel.h" | 13 #include "webrtc/pc/channel.h" |
| 14 | 14 |
| 15 #include "webrtc/api/call/audio_sink.h" | 15 #include "webrtc/api/call/audio_sink.h" |
| 16 #include "webrtc/base/bind.h" | 16 #include "webrtc/base/bind.h" |
| 17 #include "webrtc/base/byteorder.h" | 17 #include "webrtc/base/byteorder.h" |
| 18 #include "webrtc/base/checks.h" | 18 #include "webrtc/base/checks.h" |
| 19 #include "webrtc/base/copyonwritebuffer.h" | 19 #include "webrtc/base/copyonwritebuffer.h" |
| 20 #include "webrtc/base/dscp.h" | 20 #include "webrtc/base/dscp.h" |
| 21 #include "webrtc/base/logging.h" | 21 #include "webrtc/base/logging.h" |
| 22 #include "webrtc/base/networkroute.h" | 22 #include "webrtc/base/networkroute.h" |
| 23 #include "webrtc/base/trace_event.h" | 23 #include "webrtc/base/trace_event.h" |
| 24 #include "webrtc/media/base/mediaconstants.h" | 24 #include "webrtc/media/base/mediaconstants.h" |
| 25 #include "webrtc/media/base/rtputils.h" | 25 #include "webrtc/media/base/rtputils.h" |
| 26 #include "webrtc/media/engine/webrtcvoiceengine.h" |
| 26 #include "webrtc/p2p/base/packettransportinternal.h" | 27 #include "webrtc/p2p/base/packettransportinternal.h" |
| 27 #include "webrtc/pc/channelmanager.h" | 28 #include "webrtc/pc/channelmanager.h" |
| 28 | 29 |
| 29 namespace cricket { | 30 namespace cricket { |
| 30 using rtc::Bind; | 31 using rtc::Bind; |
| 31 | 32 |
| 32 namespace { | 33 namespace { |
| 33 // See comment below for why we need to use a pointer to a unique_ptr. | 34 // See comment below for why we need to use a pointer to a unique_ptr. |
| 34 bool SetRawAudioSink_w(VoiceMediaChannel* channel, | 35 bool SetRawAudioSink_w(VoiceMediaChannel* channel, |
| 35 uint32_t ssrc, | 36 uint32_t ssrc, |
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| 1656 bool VoiceChannel::SetRtpReceiveParameters_w(uint32_t ssrc, | 1657 bool VoiceChannel::SetRtpReceiveParameters_w(uint32_t ssrc, |
| 1657 webrtc::RtpParameters parameters) { | 1658 webrtc::RtpParameters parameters) { |
| 1658 return media_channel()->SetRtpReceiveParameters(ssrc, parameters); | 1659 return media_channel()->SetRtpReceiveParameters(ssrc, parameters); |
| 1659 } | 1660 } |
| 1660 | 1661 |
| 1661 bool VoiceChannel::GetStats(VoiceMediaInfo* stats) { | 1662 bool VoiceChannel::GetStats(VoiceMediaInfo* stats) { |
| 1662 return InvokeOnWorker(RTC_FROM_HERE, Bind(&VoiceMediaChannel::GetStats, | 1663 return InvokeOnWorker(RTC_FROM_HERE, Bind(&VoiceMediaChannel::GetStats, |
| 1663 media_channel(), stats)); | 1664 media_channel(), stats)); |
| 1664 } | 1665 } |
| 1665 | 1666 |
| 1667 std::vector<webrtc::RtpSource> VoiceChannel::GetSources(uint32_t ssrc) const { |
| 1668 return worker_thread()->Invoke<std::vector<webrtc::RtpSource>>( |
| 1669 RTC_FROM_HERE, |
| 1670 Bind(&WebRtcVoiceMediaChannel::GetSources, |
| 1671 static_cast<WebRtcVoiceMediaChannel*>(media_channel()), ssrc)); |
| 1672 } |
| 1673 |
| 1666 void VoiceChannel::StartMediaMonitor(int cms) { | 1674 void VoiceChannel::StartMediaMonitor(int cms) { |
| 1667 media_monitor_.reset(new VoiceMediaMonitor(media_channel(), worker_thread(), | 1675 media_monitor_.reset(new VoiceMediaMonitor(media_channel(), worker_thread(), |
| 1668 rtc::Thread::Current())); | 1676 rtc::Thread::Current())); |
| 1669 media_monitor_->SignalUpdate.connect( | 1677 media_monitor_->SignalUpdate.connect( |
| 1670 this, &VoiceChannel::OnMediaMonitorUpdate); | 1678 this, &VoiceChannel::OnMediaMonitorUpdate); |
| 1671 media_monitor_->Start(cms); | 1679 media_monitor_->Start(cms); |
| 1672 } | 1680 } |
| 1673 | 1681 |
| 1674 void VoiceChannel::StopMediaMonitor() { | 1682 void VoiceChannel::StopMediaMonitor() { |
| 1675 if (media_monitor_) { | 1683 if (media_monitor_) { |
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| 2465 signaling_thread()->Post(RTC_FROM_HERE, this, MSG_READYTOSENDDATA, | 2473 signaling_thread()->Post(RTC_FROM_HERE, this, MSG_READYTOSENDDATA, |
| 2466 new DataChannelReadyToSendMessageData(writable)); | 2474 new DataChannelReadyToSendMessageData(writable)); |
| 2467 } | 2475 } |
| 2468 | 2476 |
| 2469 void RtpDataChannel::GetSrtpCryptoSuites_n( | 2477 void RtpDataChannel::GetSrtpCryptoSuites_n( |
| 2470 std::vector<int>* crypto_suites) const { | 2478 std::vector<int>* crypto_suites) const { |
| 2471 GetSupportedDataCryptoSuites(crypto_options(), crypto_suites); | 2479 GetSupportedDataCryptoSuites(crypto_options(), crypto_suites); |
| 2472 } | 2480 } |
| 2473 | 2481 |
| 2474 } // namespace cricket | 2482 } // namespace cricket |
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