| /* |
| * Copyright (c) 2011 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. |
| */ |
| |
| #ifndef MEDIA_ENGINE_WEBRTC_MEDIA_ENGINE_H_ |
| #define MEDIA_ENGINE_WEBRTC_MEDIA_ENGINE_H_ |
| |
| #include <memory> |
| #include <string> |
| #include <vector> |
| |
| #include "api/audio/audio_frame_processor.h" |
| #include "api/audio/audio_mixer.h" |
| #include "api/audio_codecs/audio_decoder_factory.h" |
| #include "api/audio_codecs/audio_encoder_factory.h" |
| #include "api/rtp_parameters.h" |
| #include "api/task_queue/task_queue_factory.h" |
| #include "api/transport/bitrate_settings.h" |
| #include "api/transport/field_trial_based_config.h" |
| #include "api/transport/webrtc_key_value_config.h" |
| #include "api/video_codecs/video_decoder_factory.h" |
| #include "api/video_codecs/video_encoder_factory.h" |
| #include "media/base/codec.h" |
| #include "media/base/media_engine.h" |
| #include "modules/audio_device/include/audio_device.h" |
| #include "modules/audio_processing/include/audio_processing.h" |
| #include "rtc_base/system/rtc_export.h" |
| |
| namespace cricket { |
| |
| struct MediaEngineDependencies { |
| MediaEngineDependencies() = default; |
| MediaEngineDependencies(const MediaEngineDependencies&) = delete; |
| MediaEngineDependencies(MediaEngineDependencies&&) = default; |
| MediaEngineDependencies& operator=(const MediaEngineDependencies&) = delete; |
| MediaEngineDependencies& operator=(MediaEngineDependencies&&) = default; |
| ~MediaEngineDependencies() = default; |
| |
| webrtc::TaskQueueFactory* task_queue_factory = nullptr; |
| rtc::scoped_refptr<webrtc::AudioDeviceModule> adm; |
| rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory; |
| rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory; |
| rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer; |
| rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing; |
| webrtc::AudioFrameProcessor* audio_frame_processor = nullptr; |
| |
| std::unique_ptr<webrtc::VideoEncoderFactory> video_encoder_factory; |
| std::unique_ptr<webrtc::VideoDecoderFactory> video_decoder_factory; |
| |
| const webrtc::WebRtcKeyValueConfig* trials = nullptr; |
| }; |
| |
| // CreateMediaEngine may be called on any thread, though the engine is |
| // only expected to be used on one thread, internally called the "worker |
| // thread". This is the thread Init must be called on. |
| RTC_EXPORT std::unique_ptr<MediaEngineInterface> CreateMediaEngine( |
| MediaEngineDependencies dependencies); |
| |
| // Verify that extension IDs are within 1-byte extension range and are not |
| // overlapping. |
| bool ValidateRtpExtensions(const std::vector<webrtc::RtpExtension>& extensions); |
| |
| // Discard any extensions not validated by the 'supported' predicate. Duplicate |
| // extensions are removed if 'filter_redundant_extensions' is set, and also any |
| // mutually exclusive extensions (see implementation for details) are removed. |
| std::vector<webrtc::RtpExtension> FilterRtpExtensions( |
| const std::vector<webrtc::RtpExtension>& extensions, |
| bool (*supported)(absl::string_view), |
| bool filter_redundant_extensions, |
| const webrtc::WebRtcKeyValueConfig& trials); |
| |
| webrtc::BitrateConstraints GetBitrateConfigForCodec(const Codec& codec); |
| |
| } // namespace cricket |
| |
| #endif // MEDIA_ENGINE_WEBRTC_MEDIA_ENGINE_H_ |