webrtc/media/base/media_engine.h
Jonas Oreland a3aa9bd75b Make VideoBitrateAllocatorFactory injectable.
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}
2019-04-17 06:17:34 +00:00

164 lines
5.2 KiB
C++

/*
* 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_
#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
#include <CoreAudio/CoreAudio.h>
#endif
#include <memory>
#include <string>
#include <vector>
#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/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/platform_file.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<webrtc::RtpExtension> header_extensions;
};
class VoiceEngineInterface {
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<webrtc::AudioState> 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<AudioCodec>& send_codecs() const = 0;
virtual const std::vector<AudioCodec>& recv_codecs() const = 0;
virtual RtpCapabilities GetCapabilities() 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(rtc::PlatformFile file, int64_t max_size_bytes) = 0;
// Stops recording AEC dump.
virtual void StopAecDump() = 0;
};
class VideoEngineInterface {
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<VideoCodec> codecs() const = 0;
virtual RtpCapabilities GetCapabilities() 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
// Starts the engine.
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.
class CompositeMediaEngine : public MediaEngineInterface {
public:
CompositeMediaEngine(std::unique_ptr<VoiceEngineInterface> audio_engine,
std::unique_ptr<VideoEngineInterface> video_engine);
~CompositeMediaEngine() override;
bool Init() override;
VoiceEngineInterface& voice() override;
VideoEngineInterface& video() override;
const VoiceEngineInterface& voice() const override;
const VideoEngineInterface& video() const override;
private:
std::unique_ptr<VoiceEngineInterface> voice_engine_;
std::unique_ptr<VideoEngineInterface> video_engine_;
};
enum DataChannelType {
DCT_NONE = 0,
DCT_RTP = 1,
DCT_SCTP = 2,
DCT_MEDIA_TRANSPORT = 3
};
class DataEngineInterface {
public:
virtual ~DataEngineInterface() {}
virtual DataMediaChannel* CreateChannel(const MediaConfig& config) = 0;
virtual const std::vector<DataCodec>& data_codecs() = 0;
};
webrtc::RtpParameters CreateRtpParametersWithOneEncoding();
webrtc::RtpParameters CreateRtpParametersWithEncodings(StreamParams sp);
} // namespace cricket
#endif // MEDIA_BASE_MEDIA_ENGINE_H_