diff options
Diffstat (limited to 'third_party/libwebrtc/webrtc/media/engine/webrtcmediaengine.cc')
| -rw-r--r-- | third_party/libwebrtc/webrtc/media/engine/webrtcmediaengine.cc | 227 |
1 files changed, 227 insertions, 0 deletions
diff --git a/third_party/libwebrtc/webrtc/media/engine/webrtcmediaengine.cc b/third_party/libwebrtc/webrtc/media/engine/webrtcmediaengine.cc new file mode 100644 index 0000000000..8d8b768170 --- /dev/null +++ b/third_party/libwebrtc/webrtc/media/engine/webrtcmediaengine.cc @@ -0,0 +1,227 @@ +/* + * 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 "media/engine/webrtcmediaengine.h" + +#include <algorithm> +#include <memory> +#include <tuple> +#include <utility> + +#include "api/video_codecs/video_decoder_factory.h" +#include "api/video_codecs/video_encoder_factory.h" +#include "media/engine/webrtcvoiceengine.h" + +#ifdef HAVE_WEBRTC_VIDEO +#include "media/engine/webrtcvideoengine.h" +#else +#include "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<std::unique_ptr<WebRtcVideoEncoderFactory>, + std::unique_ptr<WebRtcVideoDecoderFactory>> + video_args( + (std::unique_ptr<WebRtcVideoEncoderFactory>(video_encoder_factory)), + (std::unique_ptr<WebRtcVideoDecoderFactory>(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 + +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); +} + +std::unique_ptr<MediaEngineInterface> WebRtcMediaEngineFactory::Create( + rtc::scoped_refptr<webrtc::AudioDeviceModule> adm, + rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory, + rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory, + std::unique_ptr<webrtc::VideoEncoderFactory> video_encoder_factory, + std::unique_ptr<webrtc::VideoDecoderFactory> 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<std::unique_ptr<webrtc::VideoEncoderFactory>, + std::unique_ptr<webrtc::VideoDecoderFactory>> + video_args(std::move(video_encoder_factory), + std::move(video_decoder_factory)); +#else + typedef NullWebRtcVideoEngine VideoEngine; + std::tuple<> video_args; +#endif + return std::unique_ptr<MediaEngineInterface>( + new CompositeMediaEngine<WebRtcVoiceEngine, VideoEngine>( + std::forward_as_tuple(adm, audio_encoder_factory, + audio_decoder_factory, audio_mixer, + audio_processing), + std::move(video_args))); +} + +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) { + RTC_LOG(LS_ERROR) << "Bad RTP extension ID: " << extension.ToString(); + return false; + } + if (id_used[extension.id - 1]) { + RTC_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 { + RTC_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 |