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Side by Side Diff: remoting/protocol/webrtc_transport.cc

Issue 2782523003: [Remoting Host] Supporting WebRTC VP9 streaming (Closed)
Patch Set: Fix Feedback Created 3 years, 8 months ago
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1 // Copyright 2015 The Chromium Authors. All rights reserved. 1 // Copyright 2015 The Chromium Authors. All rights reserved.
2 // Use of this source code is governed by a BSD-style license that can be 2 // Use of this source code is governed by a BSD-style license that can be
3 // found in the LICENSE file. 3 // found in the LICENSE file.
4 4
5 #include "remoting/protocol/webrtc_transport.h" 5 #include "remoting/protocol/webrtc_transport.h"
6 6
7 #include <string> 7 #include <string>
8 #include <utility> 8 #include <utility>
9 #include <vector> 9 #include <vector>
10 10
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64 // - Setting min bitrate here enables padding. 64 // - Setting min bitrate here enables padding.
65 // - The default max bitrate is 600 kbps. Setting it to 100 Mbps allows to 65 // - The default max bitrate is 600 kbps. Setting it to 100 Mbps allows to
66 // use higher bandwidth when it's available. 66 // use higher bandwidth when it's available.
67 // 67 //
68 // TODO(sergeyu): Padding needs to be enabled to workaround BW estimator not 68 // TODO(sergeyu): Padding needs to be enabled to workaround BW estimator not
69 // handling spiky traffic patterns well. This won't be necessary with a 69 // handling spiky traffic patterns well. This won't be necessary with a
70 // better bandwidth estimator. 70 // better bandwidth estimator.
71 // TODO(isheriff): The need for this should go away once we have a proper 71 // TODO(isheriff): The need for this should go away once we have a proper
72 // API to provide max bitrate for the case of handing over encoded 72 // API to provide max bitrate for the case of handing over encoded
73 // frames to webrtc. 73 // frames to webrtc.
74 if (sdp_message->has_video() && 74 if (sdp_message->has_video()) {
75 !sdp_message->AddCodecParameter( 75 bool param_added = sdp_message->AddCodecParameter(
76 "VP8", "x-google-min-bitrate=1000; x-google-max-bitrate=100000")) { 76 "VP8", "x-google-min-bitrate=1000; x-google-max-bitrate=100000");
77 if (incoming) { 77 param_added |= sdp_message->AddCodecParameter(
78 LOG(WARNING) << "VP8 not found in an incoming SDP."; 78 "VP9", "x-google-min-bitrate=1000; x-google-max-bitrate=100000");
79 } else { 79 if (!param_added) {
80 LOG(FATAL) << "VP8 not found in SDP generated by WebRTC."; 80 if (incoming) {
81 LOG(WARNING) << "Neither of VP8/VP9 is found in an incoming SDP.";
82 } else {
83 LOG(FATAL) << "Neither of VP8/VP9 is found in SDP generated by WebRTC.";
84 }
81 } 85 }
82 } 86 }
83 87
84 // Update SDP format to use 160kbps stereo for opus codec. 88 // Update SDP format to use 160kbps stereo for opus codec.
85 if (sdp_message->has_audio() && 89 if (sdp_message->has_audio() &&
86 !sdp_message->AddCodecParameter("opus", 90 !sdp_message->AddCodecParameter("opus",
87 "stereo=1; maxaveragebitrate=163840")) { 91 "stereo=1; maxaveragebitrate=163840")) {
88 if (incoming) { 92 if (incoming) {
89 LOG(WARNING) << "Opus not found in an incoming SDP."; 93 LOG(WARNING) << "Opus not found in an incoming SDP.";
90 } else { 94 } else {
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714 // the stack and so it must be destroyed later. 718 // the stack and so it must be destroyed later.
715 base::ThreadTaskRunnerHandle::Get()->DeleteSoon( 719 base::ThreadTaskRunnerHandle::Get()->DeleteSoon(
716 FROM_HERE, peer_connection_wrapper_.release()); 720 FROM_HERE, peer_connection_wrapper_.release());
717 721
718 if (error != OK) 722 if (error != OK)
719 event_handler_->OnWebrtcTransportError(error); 723 event_handler_->OnWebrtcTransportError(error);
720 } 724 }
721 725
722 } // namespace protocol 726 } // namespace protocol
723 } // namespace remoting 727 } // namespace remoting
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