/* * Copyright (c) 2004 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_BASE_MEDIA_ENGINE_H_ #define MEDIA_BASE_MEDIA_ENGINE_H_ #include #include #include #include "api/audio_codecs/audio_decoder_factory.h" #include "api/audio_codecs/audio_encoder_factory.h" #include "api/crypto/crypto_options.h" #include "api/rtp_parameters.h" #include "api/transport/webrtc_key_value_config.h" #include "api/video/video_bitrate_allocator_factory.h" #include "call/audio_state.h" #include "media/base/codec.h" #include "media/base/media_channel.h" #include "media/base/video_common.h" #include "rtc_base/system/file_wrapper.h" namespace webrtc { class AudioDeviceModule; class AudioMixer; class AudioProcessing; class Call; } // namespace webrtc namespace cricket { webrtc::RTCError CheckRtpParametersValues( const webrtc::RtpParameters& new_parameters); webrtc::RTCError CheckRtpParametersInvalidModificationAndValues( const webrtc::RtpParameters& old_parameters, const webrtc::RtpParameters& new_parameters); struct RtpCapabilities { RtpCapabilities(); ~RtpCapabilities(); std::vector header_extensions; }; class RtpHeaderExtensionQueryInterface { public: virtual ~RtpHeaderExtensionQueryInterface() = default; // Returns a vector of RtpHeaderExtensionCapability, whose direction is // kStopped if the extension is stopped (not used) by default. virtual std::vector GetRtpHeaderExtensions() const = 0; }; class VoiceEngineInterface : public RtpHeaderExtensionQueryInterface { public: VoiceEngineInterface() = default; virtual ~VoiceEngineInterface() = default; RTC_DISALLOW_COPY_AND_ASSIGN(VoiceEngineInterface); // Initialization // Starts the engine. virtual void Init() = 0; // TODO(solenberg): Remove once VoE API refactoring is done. virtual rtc::scoped_refptr GetAudioState() const = 0; // MediaChannel creation // Creates a voice media channel. Returns NULL on failure. virtual VoiceMediaChannel* CreateMediaChannel( webrtc::Call* call, const MediaConfig& config, const AudioOptions& options, const webrtc::CryptoOptions& crypto_options) = 0; virtual const std::vector& send_codecs() const = 0; virtual const std::vector& recv_codecs() const = 0; // Starts AEC dump using existing file, a maximum file size in bytes can be // specified. Logging is stopped just before the size limit is exceeded. // If max_size_bytes is set to a value <= 0, no limit will be used. virtual bool StartAecDump(webrtc::FileWrapper file, int64_t max_size_bytes) = 0; // Stops recording AEC dump. virtual void StopAecDump() = 0; }; class VideoEngineInterface : public RtpHeaderExtensionQueryInterface { public: VideoEngineInterface() = default; virtual ~VideoEngineInterface() = default; RTC_DISALLOW_COPY_AND_ASSIGN(VideoEngineInterface); // Creates a video media channel, paired with the specified voice channel. // Returns NULL on failure. virtual VideoMediaChannel* CreateMediaChannel( webrtc::Call* call, const MediaConfig& config, const VideoOptions& options, const webrtc::CryptoOptions& crypto_options, webrtc::VideoBitrateAllocatorFactory* video_bitrate_allocator_factory) = 0; virtual std::vector send_codecs() const = 0; virtual std::vector recv_codecs() const = 0; }; // MediaEngineInterface is an abstraction of a media engine which can be // subclassed to support different media componentry backends. // It supports voice and video operations in the same class to facilitate // proper synchronization between both media types. class MediaEngineInterface { public: virtual ~MediaEngineInterface() {} // Initialization. Needs to be called on the worker thread. virtual bool Init() = 0; virtual VoiceEngineInterface& voice() = 0; virtual VideoEngineInterface& video() = 0; virtual const VoiceEngineInterface& voice() const = 0; virtual const VideoEngineInterface& video() const = 0; }; // CompositeMediaEngine constructs a MediaEngine from separate // voice and video engine classes. // Optionally owns a WebRtcKeyValueConfig trials map. class CompositeMediaEngine : public MediaEngineInterface { public: CompositeMediaEngine(std::unique_ptr trials, std::unique_ptr audio_engine, std::unique_ptr video_engine); CompositeMediaEngine(std::unique_ptr audio_engine, std::unique_ptr video_engine); ~CompositeMediaEngine() override; // Always succeeds. bool Init() override; VoiceEngineInterface& voice() override; VideoEngineInterface& video() override; const VoiceEngineInterface& voice() const override; const VideoEngineInterface& video() const override; private: const std::unique_ptr trials_; const std::unique_ptr voice_engine_; const std::unique_ptr video_engine_; }; webrtc::RtpParameters CreateRtpParametersWithOneEncoding(); webrtc::RtpParameters CreateRtpParametersWithEncodings(StreamParams sp); // Returns a vector of RTP extensions as visible from RtpSender/Receiver // GetCapabilities(). The returned vector only shows what will definitely be // offered by default, i.e. the list of extensions returned from // GetRtpHeaderExtensions() that are not kStopped. std::vector GetDefaultEnabledRtpHeaderExtensions( const RtpHeaderExtensionQueryInterface& query_interface); } // namespace cricket #endif // MEDIA_BASE_MEDIA_ENGINE_H_