| /* |
| * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include "webrtc/media/engine/webrtcmediaengine.h" |
| |
| #include <algorithm> |
| |
| #include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h" |
| #include "webrtc/api/audio_codecs/builtin_audio_encoder_factory.h" |
| #include "webrtc/media/engine/webrtcvoiceengine.h" |
| |
| #ifdef HAVE_WEBRTC_VIDEO |
| #include "webrtc/media/engine/webrtcvideoengine.h" |
| #else |
| #include "webrtc/media/engine/nullwebrtcvideoengine.h" |
| #endif |
| |
| namespace cricket { |
| |
| namespace { |
| |
| MediaEngineInterface* CreateWebRtcMediaEngine( |
| webrtc::AudioDeviceModule* adm, |
| const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& |
| audio_encoder_factory, |
| const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& |
| audio_decoder_factory, |
| WebRtcVideoEncoderFactory* video_encoder_factory, |
| WebRtcVideoDecoderFactory* video_decoder_factory, |
| rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer, |
| rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing) { |
| #ifdef HAVE_WEBRTC_VIDEO |
| typedef WebRtcVideoEngine VideoEngine; |
| std::tuple<WebRtcVideoEncoderFactory*, WebRtcVideoDecoderFactory*> video_args( |
| video_encoder_factory, video_decoder_factory); |
| #else |
| typedef NullWebRtcVideoEngine VideoEngine; |
| std::tuple<> video_args; |
| #endif |
| return new CompositeMediaEngine<WebRtcVoiceEngine, VideoEngine>( |
| std::forward_as_tuple(adm, audio_encoder_factory, audio_decoder_factory, |
| audio_mixer, audio_processing), |
| std::move(video_args)); |
| } |
| |
| } // namespace |
| |
| // TODO(ossu): Backwards-compatible interface. Will be deprecated once the |
| // audio decoder factory is fully plumbed and used throughout WebRTC. |
| // See: crbug.com/webrtc/6000 |
| MediaEngineInterface* WebRtcMediaEngineFactory::Create( |
| webrtc::AudioDeviceModule* adm, |
| WebRtcVideoEncoderFactory* video_encoder_factory, |
| WebRtcVideoDecoderFactory* video_decoder_factory) { |
| return CreateWebRtcMediaEngine( |
| adm, webrtc::CreateBuiltinAudioEncoderFactory(), |
| webrtc::CreateBuiltinAudioDecoderFactory(), video_encoder_factory, |
| video_decoder_factory, nullptr, webrtc::AudioProcessing::Create()); |
| } |
| |
| MediaEngineInterface* WebRtcMediaEngineFactory::Create( |
| webrtc::AudioDeviceModule* adm, |
| const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& |
| audio_decoder_factory, |
| WebRtcVideoEncoderFactory* video_encoder_factory, |
| WebRtcVideoDecoderFactory* video_decoder_factory) { |
| return CreateWebRtcMediaEngine( |
| adm, webrtc::CreateBuiltinAudioEncoderFactory(), audio_decoder_factory, |
| video_encoder_factory, video_decoder_factory, nullptr, |
| webrtc::AudioProcessing::Create()); |
| } |
| |
| // Used by PeerConnectionFactory to create a media engine passed into |
| // ChannelManager. |
| MediaEngineInterface* WebRtcMediaEngineFactory::Create( |
| webrtc::AudioDeviceModule* adm, |
| const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& |
| audio_decoder_factory, |
| WebRtcVideoEncoderFactory* video_encoder_factory, |
| WebRtcVideoDecoderFactory* video_decoder_factory, |
| rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer, |
| rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing) { |
| return CreateWebRtcMediaEngine( |
| adm, webrtc::CreateBuiltinAudioEncoderFactory(), audio_decoder_factory, |
| video_encoder_factory, video_decoder_factory, audio_mixer, |
| audio_processing); |
| } |
| |
| MediaEngineInterface* WebRtcMediaEngineFactory::Create( |
| webrtc::AudioDeviceModule* adm, |
| const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& |
| audio_encoder_factory, |
| const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& |
| audio_decoder_factory, |
| WebRtcVideoEncoderFactory* video_encoder_factory, |
| WebRtcVideoDecoderFactory* video_decoder_factory) { |
| return CreateWebRtcMediaEngine( |
| adm, audio_encoder_factory, audio_decoder_factory, video_encoder_factory, |
| video_decoder_factory, nullptr, webrtc::AudioProcessing::Create()); |
| } |
| |
| MediaEngineInterface* WebRtcMediaEngineFactory::Create( |
| webrtc::AudioDeviceModule* adm, |
| const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& |
| audio_encoder_factory, |
| const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& |
| audio_decoder_factory, |
| WebRtcVideoEncoderFactory* video_encoder_factory, |
| WebRtcVideoDecoderFactory* video_decoder_factory, |
| rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer, |
| rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing) { |
| return CreateWebRtcMediaEngine( |
| adm, audio_encoder_factory, audio_decoder_factory, video_encoder_factory, |
| video_decoder_factory, audio_mixer, audio_processing); |
| } |
| |
| namespace { |
| // Remove mutually exclusive extensions with lower priority. |
| void DiscardRedundantExtensions( |
| std::vector<webrtc::RtpExtension>* extensions, |
| rtc::ArrayView<const char* const> extensions_decreasing_prio) { |
| RTC_DCHECK(extensions); |
| bool found = false; |
| for (const char* uri : extensions_decreasing_prio) { |
| auto it = std::find_if( |
| extensions->begin(), extensions->end(), |
| [uri](const webrtc::RtpExtension& rhs) { return rhs.uri == uri; }); |
| if (it != extensions->end()) { |
| if (found) { |
| extensions->erase(it); |
| } |
| found = true; |
| } |
| } |
| } |
| } // namespace |
| |
| bool ValidateRtpExtensions( |
| const std::vector<webrtc::RtpExtension>& extensions) { |
| bool id_used[14] = {false}; |
| for (const auto& extension : extensions) { |
| if (extension.id <= 0 || extension.id >= 15) { |
| LOG(LS_ERROR) << "Bad RTP extension ID: " << extension.ToString(); |
| return false; |
| } |
| if (id_used[extension.id - 1]) { |
| LOG(LS_ERROR) << "Duplicate RTP extension ID: " << extension.ToString(); |
| return false; |
| } |
| id_used[extension.id - 1] = true; |
| } |
| return true; |
| } |
| |
| std::vector<webrtc::RtpExtension> FilterRtpExtensions( |
| const std::vector<webrtc::RtpExtension>& extensions, |
| bool (*supported)(const std::string&), |
| bool filter_redundant_extensions) { |
| RTC_DCHECK(ValidateRtpExtensions(extensions)); |
| RTC_DCHECK(supported); |
| std::vector<webrtc::RtpExtension> result; |
| |
| // Ignore any extensions that we don't recognize. |
| for (const auto& extension : extensions) { |
| if (supported(extension.uri)) { |
| result.push_back(extension); |
| } else { |
| LOG(LS_WARNING) << "Unsupported RTP extension: " << extension.ToString(); |
| } |
| } |
| |
| // Sort by name, ascending (prioritise encryption), so that we don't reset |
| // extensions if they were specified in a different order (also allows us |
| // to use std::unique below). |
| std::sort(result.begin(), result.end(), |
| [](const webrtc::RtpExtension& rhs, |
| const webrtc::RtpExtension& lhs) { |
| return rhs.encrypt == lhs.encrypt ? rhs.uri < lhs.uri |
| : rhs.encrypt > lhs.encrypt; |
| }); |
| |
| // Remove unnecessary extensions (used on send side). |
| if (filter_redundant_extensions) { |
| auto it = std::unique( |
| result.begin(), result.end(), |
| [](const webrtc::RtpExtension& rhs, const webrtc::RtpExtension& lhs) { |
| return rhs.uri == lhs.uri && rhs.encrypt == lhs.encrypt; |
| }); |
| result.erase(it, result.end()); |
| |
| // Keep just the highest priority extension of any in the following list. |
| static const char* const kBweExtensionPriorities[] = { |
| webrtc::RtpExtension::kTransportSequenceNumberUri, |
| webrtc::RtpExtension::kAbsSendTimeUri, |
| webrtc::RtpExtension::kTimestampOffsetUri}; |
| DiscardRedundantExtensions(&result, kBweExtensionPriorities); |
| } |
| |
| return result; |
| } |
| |
| webrtc::Call::Config::BitrateConfig GetBitrateConfigForCodec( |
| const Codec& codec) { |
| webrtc::Call::Config::BitrateConfig config; |
| int bitrate_kbps = 0; |
| if (codec.GetParam(kCodecParamMinBitrate, &bitrate_kbps) && |
| bitrate_kbps > 0) { |
| config.min_bitrate_bps = bitrate_kbps * 1000; |
| } else { |
| config.min_bitrate_bps = 0; |
| } |
| if (codec.GetParam(kCodecParamStartBitrate, &bitrate_kbps) && |
| bitrate_kbps > 0) { |
| config.start_bitrate_bps = bitrate_kbps * 1000; |
| } else { |
| // Do not reconfigure start bitrate unless it's specified and positive. |
| config.start_bitrate_bps = -1; |
| } |
| if (codec.GetParam(kCodecParamMaxBitrate, &bitrate_kbps) && |
| bitrate_kbps > 0) { |
| config.max_bitrate_bps = bitrate_kbps * 1000; |
| } else { |
| config.max_bitrate_bps = -1; |
| } |
| return config; |
| } |
| } // namespace cricket |