2020-08-14 16:58:22 +00:00
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/*
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* Copyright (c) 2004 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MEDIA_BASE_MEDIA_ENGINE_H_
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#define MEDIA_BASE_MEDIA_ENGINE_H_
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#include <memory>
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#include <string>
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#include <vector>
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#include "api/audio_codecs/audio_decoder_factory.h"
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#include "api/audio_codecs/audio_encoder_factory.h"
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#include "api/crypto/crypto_options.h"
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#include "api/rtp_parameters.h"
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2020-12-23 07:48:30 +00:00
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#include "api/transport/webrtc_key_value_config.h"
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2020-08-14 16:58:22 +00:00
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#include "api/video/video_bitrate_allocator_factory.h"
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#include "call/audio_state.h"
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#include "media/base/codec.h"
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#include "media/base/media_channel.h"
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#include "media/base/video_common.h"
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#include "rtc_base/system/file_wrapper.h"
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namespace webrtc {
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class AudioDeviceModule;
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class AudioMixer;
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class AudioProcessing;
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class Call;
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} // namespace webrtc
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namespace cricket {
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webrtc::RTCError CheckRtpParametersValues(
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const webrtc::RtpParameters& new_parameters);
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webrtc::RTCError CheckRtpParametersInvalidModificationAndValues(
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const webrtc::RtpParameters& old_parameters,
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const webrtc::RtpParameters& new_parameters);
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struct RtpCapabilities {
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RtpCapabilities();
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~RtpCapabilities();
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std::vector<webrtc::RtpExtension> header_extensions;
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};
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class RtpHeaderExtensionQueryInterface {
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public:
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virtual ~RtpHeaderExtensionQueryInterface() = default;
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// Returns a vector of RtpHeaderExtensionCapability, whose direction is
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// kStopped if the extension is stopped (not used) by default.
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virtual std::vector<webrtc::RtpHeaderExtensionCapability>
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GetRtpHeaderExtensions() const = 0;
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};
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class VoiceEngineInterface : public RtpHeaderExtensionQueryInterface {
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public:
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VoiceEngineInterface() = default;
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virtual ~VoiceEngineInterface() = default;
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RTC_DISALLOW_COPY_AND_ASSIGN(VoiceEngineInterface);
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// Initialization
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// Starts the engine.
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virtual void Init() = 0;
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// TODO(solenberg): Remove once VoE API refactoring is done.
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virtual rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const = 0;
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// MediaChannel creation
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// Creates a voice media channel. Returns NULL on failure.
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virtual VoiceMediaChannel* CreateMediaChannel(
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webrtc::Call* call,
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const MediaConfig& config,
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const AudioOptions& options,
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const webrtc::CryptoOptions& crypto_options) = 0;
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virtual const std::vector<AudioCodec>& send_codecs() const = 0;
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virtual const std::vector<AudioCodec>& recv_codecs() const = 0;
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// Starts AEC dump using existing file, a maximum file size in bytes can be
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// specified. Logging is stopped just before the size limit is exceeded.
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// If max_size_bytes is set to a value <= 0, no limit will be used.
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virtual bool StartAecDump(webrtc::FileWrapper file,
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int64_t max_size_bytes) = 0;
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// Stops recording AEC dump.
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virtual void StopAecDump() = 0;
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};
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class VideoEngineInterface : public RtpHeaderExtensionQueryInterface {
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public:
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VideoEngineInterface() = default;
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virtual ~VideoEngineInterface() = default;
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RTC_DISALLOW_COPY_AND_ASSIGN(VideoEngineInterface);
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// Creates a video media channel, paired with the specified voice channel.
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// Returns NULL on failure.
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virtual VideoMediaChannel* CreateMediaChannel(
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webrtc::Call* call,
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const MediaConfig& config,
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const VideoOptions& options,
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const webrtc::CryptoOptions& crypto_options,
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webrtc::VideoBitrateAllocatorFactory*
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video_bitrate_allocator_factory) = 0;
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virtual std::vector<VideoCodec> send_codecs() const = 0;
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virtual std::vector<VideoCodec> recv_codecs() const = 0;
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};
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// MediaEngineInterface is an abstraction of a media engine which can be
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// subclassed to support different media componentry backends.
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// It supports voice and video operations in the same class to facilitate
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// proper synchronization between both media types.
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class MediaEngineInterface {
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public:
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virtual ~MediaEngineInterface() {}
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2021-06-25 00:43:10 +00:00
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// Initialization. Needs to be called on the worker thread.
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virtual bool Init() = 0;
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2020-08-14 16:58:22 +00:00
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virtual VoiceEngineInterface& voice() = 0;
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virtual VideoEngineInterface& video() = 0;
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virtual const VoiceEngineInterface& voice() const = 0;
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virtual const VideoEngineInterface& video() const = 0;
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};
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// CompositeMediaEngine constructs a MediaEngine from separate
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// voice and video engine classes.
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// Optionally owns a WebRtcKeyValueConfig trials map.
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class CompositeMediaEngine : public MediaEngineInterface {
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public:
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CompositeMediaEngine(std::unique_ptr<webrtc::WebRtcKeyValueConfig> trials,
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std::unique_ptr<VoiceEngineInterface> audio_engine,
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std::unique_ptr<VideoEngineInterface> video_engine);
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CompositeMediaEngine(std::unique_ptr<VoiceEngineInterface> audio_engine,
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std::unique_ptr<VideoEngineInterface> video_engine);
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~CompositeMediaEngine() override;
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// Always succeeds.
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bool Init() override;
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VoiceEngineInterface& voice() override;
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VideoEngineInterface& video() override;
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const VoiceEngineInterface& voice() const override;
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const VideoEngineInterface& video() const override;
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private:
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const std::unique_ptr<webrtc::WebRtcKeyValueConfig> trials_;
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const std::unique_ptr<VoiceEngineInterface> voice_engine_;
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const std::unique_ptr<VideoEngineInterface> video_engine_;
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};
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webrtc::RtpParameters CreateRtpParametersWithOneEncoding();
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webrtc::RtpParameters CreateRtpParametersWithEncodings(StreamParams sp);
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// Returns a vector of RTP extensions as visible from RtpSender/Receiver
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// GetCapabilities(). The returned vector only shows what will definitely be
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// offered by default, i.e. the list of extensions returned from
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// GetRtpHeaderExtensions() that are not kStopped.
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std::vector<webrtc::RtpExtension> GetDefaultEnabledRtpHeaderExtensions(
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const RtpHeaderExtensionQueryInterface& query_interface);
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} // namespace cricket
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#endif // MEDIA_BASE_MEDIA_ENGINE_H_
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