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This patch makes VideoBitrateAllocatorFactory injectable by adding to PeerConnectionDependencies instead of allowing it to be overridden using MediaEngine (on PeerConnectionFactory). With this patch VideoBitrateAllocatorFactory is owned by the PeerConnection. WANT_LGTM (examples) : sakal@ WANT_LGTM (api/pc) : steveanton@ Bug: webrtc:10547 Change-Id: I768d400a621f2b7a98795eb7f410adb48651bfd6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132706 Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27654}
164 lines
5.2 KiB
C++
164 lines
5.2 KiB
C++
/*
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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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#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
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#include <CoreAudio/CoreAudio.h>
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#endif
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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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#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/platform_file.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 VoiceEngineInterface {
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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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virtual RtpCapabilities GetCapabilities() 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(rtc::PlatformFile file, 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 {
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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> codecs() const = 0;
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virtual RtpCapabilities GetCapabilities() 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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// Initialization
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// Starts the engine.
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virtual bool Init() = 0;
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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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class CompositeMediaEngine : public MediaEngineInterface {
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public:
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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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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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std::unique_ptr<VoiceEngineInterface> voice_engine_;
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std::unique_ptr<VideoEngineInterface> video_engine_;
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};
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enum DataChannelType {
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DCT_NONE = 0,
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DCT_RTP = 1,
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DCT_SCTP = 2,
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DCT_MEDIA_TRANSPORT = 3
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};
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class DataEngineInterface {
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public:
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virtual ~DataEngineInterface() {}
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virtual DataMediaChannel* CreateChannel(const MediaConfig& config) = 0;
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virtual const std::vector<DataCodec>& data_codecs() = 0;
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};
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webrtc::RtpParameters CreateRtpParametersWithOneEncoding();
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webrtc::RtpParameters CreateRtpParametersWithEncodings(StreamParams sp);
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} // namespace cricket
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#endif // MEDIA_BASE_MEDIA_ENGINE_H_
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