buildbot@webrtc.org | a853077 | 2014-12-10 09:01:18 +0000 | [diff] [blame] | 1 | /* |
| 2 | * libjingle |
| 3 | * Copyright 2014 Google Inc. |
| 4 | * |
| 5 | * Redistribution and use in source and binary forms, with or without |
| 6 | * modification, are permitted provided that the following conditions are met: |
| 7 | * |
| 8 | * 1. Redistributions of source code must retain the above copyright notice, |
| 9 | * this list of conditions and the following disclaimer. |
| 10 | * 2. Redistributions in binary form must reproduce the above copyright notice, |
| 11 | * this list of conditions and the following disclaimer in the documentation |
| 12 | * and/or other materials provided with the distribution. |
| 13 | * 3. The name of the author may not be used to endorse or promote products |
| 14 | * derived from this software without specific prior written permission. |
| 15 | * |
| 16 | * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED |
| 17 | * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF |
| 18 | * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO |
| 19 | * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, |
| 20 | * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, |
| 21 | * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; |
| 22 | * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, |
| 23 | * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR |
| 24 | * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF |
| 25 | * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. |
| 26 | */ |
jlmiller@webrtc.org | 5f93d0a | 2015-01-20 21:36:13 +0000 | [diff] [blame] | 27 | |
buildbot@webrtc.org | a853077 | 2014-12-10 09:01:18 +0000 | [diff] [blame] | 28 | #ifndef TALK_MEDIA_WEBRTC_SIMULCAST_H_ |
| 29 | #define TALK_MEDIA_WEBRTC_SIMULCAST_H_ |
| 30 | |
| 31 | #include <vector> |
| 32 | |
| 33 | #include "webrtc/base/basictypes.h" |
| 34 | #include "webrtc/config.h" |
| 35 | |
| 36 | namespace webrtc { |
| 37 | struct VideoCodec; |
| 38 | } |
| 39 | |
| 40 | namespace cricket { |
| 41 | struct VideoOptions; |
| 42 | struct StreamParams; |
| 43 | |
| 44 | enum SimulcastBitrateMode { |
| 45 | SBM_NORMAL = 0, |
| 46 | SBM_HIGH, |
| 47 | SBM_VERY_HIGH, |
| 48 | SBM_COUNT |
| 49 | }; |
| 50 | |
sprang@webrtc.org | 46d4d29 | 2014-12-23 15:19:35 +0000 | [diff] [blame] | 51 | // Config for use with screen cast when temporal layers are enabled. |
| 52 | struct ScreenshareLayerConfig { |
| 53 | public: |
| 54 | ScreenshareLayerConfig(int tl0_bitrate, int tl1_bitrate); |
| 55 | |
| 56 | // Bitrates, for temporal layers 0 and 1. |
| 57 | int tl0_bitrate_kbps; |
| 58 | int tl1_bitrate_kbps; |
| 59 | |
| 60 | static ScreenshareLayerConfig GetDefault(); |
| 61 | |
| 62 | // Parse bitrate from group name on format "(tl0_bitrate)-(tl1_bitrate)", |
| 63 | // eg. "100-1000" for the default rates. |
| 64 | static bool FromFieldTrialGroup(const std::string& group, |
| 65 | ScreenshareLayerConfig* config); |
| 66 | }; |
| 67 | |
buildbot@webrtc.org | a853077 | 2014-12-10 09:01:18 +0000 | [diff] [blame] | 68 | // TODO(pthatcher): Write unit tests just for these functions, |
| 69 | // independent of WebrtcVideoEngine. |
| 70 | |
| 71 | // Get the simulcast bitrate mode to use based on |
| 72 | // options.video_highest_bitrate. |
| 73 | SimulcastBitrateMode GetSimulcastBitrateMode( |
| 74 | const VideoOptions& options); |
| 75 | |
| 76 | // Get the ssrcs of the SIM group from the stream params. |
Peter Boström | 0c4e06b | 2015-10-07 12:23:21 +0200 | [diff] [blame] | 77 | void GetSimulcastSsrcs(const StreamParams& sp, std::vector<uint32_t>* ssrcs); |
buildbot@webrtc.org | a853077 | 2014-12-10 09:01:18 +0000 | [diff] [blame] | 78 | |
| 79 | // Get simulcast settings. |
| 80 | std::vector<webrtc::VideoStream> GetSimulcastConfig( |
| 81 | size_t max_streams, |
| 82 | SimulcastBitrateMode bitrate_mode, |
| 83 | int width, |
| 84 | int height, |
buildbot@webrtc.org | a853077 | 2014-12-10 09:01:18 +0000 | [diff] [blame] | 85 | int max_bitrate_bps, |
| 86 | int max_qp, |
| 87 | int max_framerate); |
| 88 | |
| 89 | // Set the codec->simulcastStreams, codec->width, and codec->height |
| 90 | // based on the number of ssrcs to use and the bitrate mode to use. |
| 91 | bool ConfigureSimulcastCodec(int number_ssrcs, |
| 92 | SimulcastBitrateMode bitrate_mode, |
| 93 | webrtc::VideoCodec* codec); |
| 94 | |
| 95 | // Set the codec->simulcastStreams, codec->width, and codec->height |
| 96 | // based on the video options (to get the simulcast bitrate mode) and |
| 97 | // the stream params (to get the number of ssrcs). This is really a |
| 98 | // convenience function. |
| 99 | bool ConfigureSimulcastCodec(const StreamParams& sp, |
| 100 | const VideoOptions& options, |
| 101 | webrtc::VideoCodec* codec); |
| 102 | |
| 103 | // Set the numberOfTemporalLayers in each codec->simulcastStreams[i]. |
| 104 | // Apparently it is useful to do this at a different time than |
| 105 | // ConfigureSimulcastCodec. |
| 106 | // TODO(pthatcher): Figure out why and put this code into |
| 107 | // ConfigureSimulcastCodec. |
| 108 | void ConfigureSimulcastTemporalLayers( |
| 109 | int num_temporal_layers, webrtc::VideoCodec* codec); |
| 110 | |
| 111 | // Turn off all simulcasting for the given codec. |
| 112 | void DisableSimulcastCodec(webrtc::VideoCodec* codec); |
| 113 | |
| 114 | // Log useful info about each of the simulcast substreams of the |
| 115 | // codec. |
| 116 | void LogSimulcastSubstreams(const webrtc::VideoCodec& codec); |
| 117 | |
| 118 | // Configure the codec's bitrate and temporal layers so that it's good |
| 119 | // for a screencast in conference mode. Technically, this shouldn't |
| 120 | // go in simulcast.cc. But it's closely related. |
| 121 | void ConfigureConferenceModeScreencastCodec(webrtc::VideoCodec* codec); |
| 122 | |
| 123 | } // namespace cricket |
| 124 | |
| 125 | #endif // TALK_MEDIA_WEBRTC_SIMULCAST_H_ |