| /* |
| * libjingle |
| * Copyright 2014 Google Inc. |
| * |
| * Redistribution and use in source and binary forms, with or without |
| * modification, are permitted provided that the following conditions are met: |
| * |
| * 1. Redistributions of source code must retain the above copyright notice, |
| * this list of conditions and the following disclaimer. |
| * 2. Redistributions in binary form must reproduce the above copyright notice, |
| * this list of conditions and the following disclaimer in the documentation |
| * and/or other materials provided with the distribution. |
| * 3. The name of the author may not be used to endorse or promote products |
| * derived from this software without specific prior written permission. |
| * |
| * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED |
| * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF |
| * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO |
| * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, |
| * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, |
| * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; |
| * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, |
| * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR |
| * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF |
| * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. |
| */ |
| |
| #ifndef TALK_MEDIA_WEBRTC_SIMULCAST_H_ |
| #define TALK_MEDIA_WEBRTC_SIMULCAST_H_ |
| |
| #include <vector> |
| |
| #include "webrtc/base/basictypes.h" |
| #include "webrtc/config.h" |
| |
| namespace webrtc { |
| struct VideoCodec; |
| } |
| |
| namespace cricket { |
| struct VideoOptions; |
| struct StreamParams; |
| |
| enum SimulcastBitrateMode { |
| SBM_NORMAL = 0, |
| SBM_HIGH, |
| SBM_VERY_HIGH, |
| SBM_COUNT |
| }; |
| |
| // Config for use with screen cast when temporal layers are enabled. |
| struct ScreenshareLayerConfig { |
| public: |
| ScreenshareLayerConfig(int tl0_bitrate, int tl1_bitrate); |
| |
| // Bitrates, for temporal layers 0 and 1. |
| int tl0_bitrate_kbps; |
| int tl1_bitrate_kbps; |
| |
| static ScreenshareLayerConfig GetDefault(); |
| |
| // Parse bitrate from group name on format "(tl0_bitrate)-(tl1_bitrate)", |
| // eg. "100-1000" for the default rates. |
| static bool FromFieldTrialGroup(const std::string& group, |
| ScreenshareLayerConfig* config); |
| }; |
| |
| // TODO(pthatcher): Write unit tests just for these functions, |
| // independent of WebrtcVideoEngine. |
| |
| // Get the simulcast bitrate mode to use based on |
| // options.video_highest_bitrate. |
| SimulcastBitrateMode GetSimulcastBitrateMode( |
| const VideoOptions& options); |
| |
| // Get the ssrcs of the SIM group from the stream params. |
| void GetSimulcastSsrcs(const StreamParams& sp, std::vector<uint32>* ssrcs); |
| |
| // Get simulcast settings. |
| std::vector<webrtc::VideoStream> GetSimulcastConfig( |
| size_t max_streams, |
| SimulcastBitrateMode bitrate_mode, |
| int width, |
| int height, |
| int max_bitrate_bps, |
| int max_qp, |
| int max_framerate); |
| |
| // Set the codec->simulcastStreams, codec->width, and codec->height |
| // based on the number of ssrcs to use and the bitrate mode to use. |
| bool ConfigureSimulcastCodec(int number_ssrcs, |
| SimulcastBitrateMode bitrate_mode, |
| webrtc::VideoCodec* codec); |
| |
| // Set the codec->simulcastStreams, codec->width, and codec->height |
| // based on the video options (to get the simulcast bitrate mode) and |
| // the stream params (to get the number of ssrcs). This is really a |
| // convenience function. |
| bool ConfigureSimulcastCodec(const StreamParams& sp, |
| const VideoOptions& options, |
| webrtc::VideoCodec* codec); |
| |
| // Set the numberOfTemporalLayers in each codec->simulcastStreams[i]. |
| // Apparently it is useful to do this at a different time than |
| // ConfigureSimulcastCodec. |
| // TODO(pthatcher): Figure out why and put this code into |
| // ConfigureSimulcastCodec. |
| void ConfigureSimulcastTemporalLayers( |
| int num_temporal_layers, webrtc::VideoCodec* codec); |
| |
| // Turn off all simulcasting for the given codec. |
| void DisableSimulcastCodec(webrtc::VideoCodec* codec); |
| |
| // Log useful info about each of the simulcast substreams of the |
| // codec. |
| void LogSimulcastSubstreams(const webrtc::VideoCodec& codec); |
| |
| // Configure the codec's bitrate and temporal layers so that it's good |
| // for a screencast in conference mode. Technically, this shouldn't |
| // go in simulcast.cc. But it's closely related. |
| void ConfigureConferenceModeScreencastCodec(webrtc::VideoCodec* codec); |
| |
| } // namespace cricket |
| |
| #endif // TALK_MEDIA_WEBRTC_SIMULCAST_H_ |