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/*
* 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 <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/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<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(webrtc::FileWrapper 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,
// Data channel transport over media transport.
DCT_MEDIA_TRANSPORT = 3,
// Data channel transport over datagram transport (with no fallback). This is
// the same behavior as data channel transport over media transport, and is
// usable without DTLS.
DCT_DATA_CHANNEL_TRANSPORT = 4,
// Data channel transport over datagram transport (with SCTP negotiation
// semantics and a fallback to SCTP). Only usable with DTLS.
DCT_DATA_CHANNEL_TRANSPORT_SCTP = 5,
};
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_