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Issue 1917193008: Adding getParameters/setParameters APIs to RtpReceiver. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Adding nil check and removing unneeded methods. Created 4 years, 7 months ago
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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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1396 1396
1397 void VoiceChannel::SetRawAudioSink( 1397 void VoiceChannel::SetRawAudioSink(
1398 uint32_t ssrc, 1398 uint32_t ssrc,
1399 std::unique_ptr<webrtc::AudioSinkInterface> sink) { 1399 std::unique_ptr<webrtc::AudioSinkInterface> sink) {
1400 // We need to work around Bind's lack of support for unique_ptr and ownership 1400 // We need to work around Bind's lack of support for unique_ptr and ownership
1401 // passing. So we invoke to our own little routine that gets a pointer to 1401 // passing. So we invoke to our own little routine that gets a pointer to
1402 // our local variable. This is OK since we're synchronously invoking. 1402 // our local variable. This is OK since we're synchronously invoking.
1403 InvokeOnWorker(Bind(&SetRawAudioSink_w, media_channel(), ssrc, &sink)); 1403 InvokeOnWorker(Bind(&SetRawAudioSink_w, media_channel(), ssrc, &sink));
1404 } 1404 }
1405 1405
1406 webrtc::RtpParameters VoiceChannel::GetRtpParameters(uint32_t ssrc) const { 1406 webrtc::RtpParameters VoiceChannel::GetRtpSendParameters(uint32_t ssrc) const {
1407 return worker_thread()->Invoke<webrtc::RtpParameters>( 1407 return worker_thread()->Invoke<webrtc::RtpParameters>(
1408 Bind(&VoiceChannel::GetRtpParameters_w, this, ssrc)); 1408 Bind(&VoiceChannel::GetRtpSendParameters_w, this, ssrc));
1409 } 1409 }
1410 1410
1411 webrtc::RtpParameters VoiceChannel::GetRtpParameters_w(uint32_t ssrc) const { 1411 webrtc::RtpParameters VoiceChannel::GetRtpSendParameters_w(
1412 return media_channel()->GetRtpParameters(ssrc); 1412 uint32_t ssrc) const {
1413 return media_channel()->GetRtpSendParameters(ssrc);
1413 } 1414 }
1414 1415
1415 bool VoiceChannel::SetRtpParameters(uint32_t ssrc, 1416 bool VoiceChannel::SetRtpSendParameters(
1416 const webrtc::RtpParameters& parameters) { 1417 uint32_t ssrc,
1418 const webrtc::RtpParameters& parameters) {
1417 return InvokeOnWorker( 1419 return InvokeOnWorker(
1418 Bind(&VoiceChannel::SetRtpParameters_w, this, ssrc, parameters)); 1420 Bind(&VoiceChannel::SetRtpSendParameters_w, this, ssrc, parameters));
1419 } 1421 }
1420 1422
1421 bool VoiceChannel::SetRtpParameters_w(uint32_t ssrc, 1423 bool VoiceChannel::SetRtpSendParameters_w(uint32_t ssrc,
1422 webrtc::RtpParameters parameters) { 1424 webrtc::RtpParameters parameters) {
1423 return media_channel()->SetRtpParameters(ssrc, parameters); 1425 return media_channel()->SetRtpSendParameters(ssrc, parameters);
1426 }
1427
1428 webrtc::RtpParameters VoiceChannel::GetRtpReceiveParameters(
1429 uint32_t ssrc) const {
1430 return worker_thread()->Invoke<webrtc::RtpParameters>(
1431 Bind(&VoiceChannel::GetRtpReceiveParameters_w, this, ssrc));
1432 }
1433
1434 webrtc::RtpParameters VoiceChannel::GetRtpReceiveParameters_w(
1435 uint32_t ssrc) const {
1436 return media_channel()->GetRtpReceiveParameters(ssrc);
1437 }
1438
1439 bool VoiceChannel::SetRtpReceiveParameters(
1440 uint32_t ssrc,
1441 const webrtc::RtpParameters& parameters) {
1442 return InvokeOnWorker(
1443 Bind(&VoiceChannel::SetRtpReceiveParameters_w, this, ssrc, parameters));
1444 }
1445
1446 bool VoiceChannel::SetRtpReceiveParameters_w(uint32_t ssrc,
1447 webrtc::RtpParameters parameters) {
1448 return media_channel()->SetRtpReceiveParameters(ssrc, parameters);
1424 } 1449 }
1425 1450
1426 bool VoiceChannel::GetStats(VoiceMediaInfo* stats) { 1451 bool VoiceChannel::GetStats(VoiceMediaInfo* stats) {
1427 return InvokeOnWorker(Bind(&VoiceMediaChannel::GetStats, 1452 return InvokeOnWorker(Bind(&VoiceMediaChannel::GetStats,
1428 media_channel(), stats)); 1453 media_channel(), stats));
1429 } 1454 }
1430 1455
1431 void VoiceChannel::StartMediaMonitor(int cms) { 1456 void VoiceChannel::StartMediaMonitor(int cms) {
1432 media_monitor_.reset(new VoiceMediaMonitor(media_channel(), worker_thread(), 1457 media_monitor_.reset(new VoiceMediaMonitor(media_channel(), worker_thread(),
1433 rtc::Thread::Current())); 1458 rtc::Thread::Current()));
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1697 Bind(&VideoMediaChannel::SetSource, media_channel(), ssrc, source)); 1722 Bind(&VideoMediaChannel::SetSource, media_channel(), ssrc, source));
1698 } 1723 }
1699 1724
1700 bool VideoChannel::SetVideoSend(uint32_t ssrc, 1725 bool VideoChannel::SetVideoSend(uint32_t ssrc,
1701 bool mute, 1726 bool mute,
1702 const VideoOptions* options) { 1727 const VideoOptions* options) {
1703 return InvokeOnWorker(Bind(&VideoMediaChannel::SetVideoSend, media_channel(), 1728 return InvokeOnWorker(Bind(&VideoMediaChannel::SetVideoSend, media_channel(),
1704 ssrc, mute, options)); 1729 ssrc, mute, options));
1705 } 1730 }
1706 1731
1707 webrtc::RtpParameters VideoChannel::GetRtpParameters(uint32_t ssrc) const { 1732 webrtc::RtpParameters VideoChannel::GetRtpSendParameters(uint32_t ssrc) const {
1708 return worker_thread()->Invoke<webrtc::RtpParameters>( 1733 return worker_thread()->Invoke<webrtc::RtpParameters>(
1709 Bind(&VideoChannel::GetRtpParameters_w, this, ssrc)); 1734 Bind(&VideoChannel::GetRtpSendParameters_w, this, ssrc));
1710 } 1735 }
1711 1736
1712 webrtc::RtpParameters VideoChannel::GetRtpParameters_w(uint32_t ssrc) const { 1737 webrtc::RtpParameters VideoChannel::GetRtpSendParameters_w(
1713 return media_channel()->GetRtpParameters(ssrc); 1738 uint32_t ssrc) const {
1739 return media_channel()->GetRtpSendParameters(ssrc);
1714 } 1740 }
1715 1741
1716 bool VideoChannel::SetRtpParameters(uint32_t ssrc, 1742 bool VideoChannel::SetRtpSendParameters(
1717 const webrtc::RtpParameters& parameters) { 1743 uint32_t ssrc,
1744 const webrtc::RtpParameters& parameters) {
1718 return InvokeOnWorker( 1745 return InvokeOnWorker(
1719 Bind(&VideoChannel::SetRtpParameters_w, this, ssrc, parameters)); 1746 Bind(&VideoChannel::SetRtpSendParameters_w, this, ssrc, parameters));
1720 } 1747 }
1721 1748
1722 bool VideoChannel::SetRtpParameters_w(uint32_t ssrc, 1749 bool VideoChannel::SetRtpSendParameters_w(uint32_t ssrc,
1723 webrtc::RtpParameters parameters) { 1750 webrtc::RtpParameters parameters) {
1724 return media_channel()->SetRtpParameters(ssrc, parameters); 1751 return media_channel()->SetRtpSendParameters(ssrc, parameters);
1725 } 1752 }
1753
1754 webrtc::RtpParameters VideoChannel::GetRtpReceiveParameters(
1755 uint32_t ssrc) const {
1756 return worker_thread()->Invoke<webrtc::RtpParameters>(
1757 Bind(&VideoChannel::GetRtpReceiveParameters_w, this, ssrc));
1758 }
1759
1760 webrtc::RtpParameters VideoChannel::GetRtpReceiveParameters_w(
1761 uint32_t ssrc) const {
1762 return media_channel()->GetRtpReceiveParameters(ssrc);
1763 }
1764
1765 bool VideoChannel::SetRtpReceiveParameters(
1766 uint32_t ssrc,
1767 const webrtc::RtpParameters& parameters) {
1768 return InvokeOnWorker(
1769 Bind(&VideoChannel::SetRtpReceiveParameters_w, this, ssrc, parameters));
1770 }
1771
1772 bool VideoChannel::SetRtpReceiveParameters_w(uint32_t ssrc,
1773 webrtc::RtpParameters parameters) {
1774 return media_channel()->SetRtpReceiveParameters(ssrc, parameters);
1775 }
1776
1726 void VideoChannel::ChangeState() { 1777 void VideoChannel::ChangeState() {
1727 // Send outgoing data if we're the active call, we have the remote content, 1778 // Send outgoing data if we're the active call, we have the remote content,
1728 // and we have had some form of connectivity. 1779 // and we have had some form of connectivity.
1729 bool send = IsReadyToSend(); 1780 bool send = IsReadyToSend();
1730 if (!media_channel()->SetSend(send)) { 1781 if (!media_channel()->SetSend(send)) {
1731 LOG(LS_ERROR) << "Failed to SetSend on video channel"; 1782 LOG(LS_ERROR) << "Failed to SetSend on video channel";
1732 // TODO(gangji): Report error back to server. 1783 // TODO(gangji): Report error back to server.
1733 } 1784 }
1734 1785
1735 LOG(LS_INFO) << "Changing video state, send=" << send; 1786 LOG(LS_INFO) << "Changing video state, send=" << send;
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2203 return (data_channel_type_ == DCT_RTP) && BaseChannel::ShouldSetupDtlsSrtp(); 2254 return (data_channel_type_ == DCT_RTP) && BaseChannel::ShouldSetupDtlsSrtp();
2204 } 2255 }
2205 2256
2206 void DataChannel::OnStreamClosedRemotely(uint32_t sid) { 2257 void DataChannel::OnStreamClosedRemotely(uint32_t sid) {
2207 rtc::TypedMessageData<uint32_t>* message = 2258 rtc::TypedMessageData<uint32_t>* message =
2208 new rtc::TypedMessageData<uint32_t>(sid); 2259 new rtc::TypedMessageData<uint32_t>(sid);
2209 signaling_thread()->Post(this, MSG_STREAMCLOSEDREMOTELY, message); 2260 signaling_thread()->Post(this, MSG_STREAMCLOSEDREMOTELY, message);
2210 } 2261 }
2211 2262
2212 } // namespace cricket 2263 } // namespace cricket
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