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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/field_trials_view.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/media_channel_impl.h"
#include "media/base/media_config.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 {
// Checks that the scalability_mode value of each encoding is supported by at
// least one video codec of the list. If the list is empty, no check is done.
webrtc::RTCError CheckScalabilityModeValues(
const webrtc::RtpParameters& new_parameters,
rtc::ArrayView<cricket::Codec> codec_preferences,
absl::optional<cricket::Codec> send_codec);
// Checks the parameters have valid and supported values, and checks parameters
// with CheckScalabilityModeValues().
webrtc::RTCError CheckRtpParametersValues(
const webrtc::RtpParameters& new_parameters,
rtc::ArrayView<cricket::Codec> codec_preferences,
absl::optional<cricket::Codec> send_codec);
// Checks that the immutable values have not changed in new_parameters and
// checks all parameters with CheckRtpParametersValues().
webrtc::RTCError CheckRtpParametersInvalidModificationAndValues(
const webrtc::RtpParameters& old_parameters,
const webrtc::RtpParameters& new_parameters,
rtc::ArrayView<cricket::Codec> codec_preferences,
absl::optional<cricket::Codec> send_codec);
// Checks that the immutable values have not changed in new_parameters and
// checks parameters (except SVC) with CheckRtpParametersValues(). It should
// usually be paired with a call to CheckScalabilityModeValues().
webrtc::RTCError CheckRtpParametersInvalidModificationAndValues(
const webrtc::RtpParameters& old_parameters,
const webrtc::RtpParameters& new_parameters);
struct RtpCapabilities {
RtpCapabilities();
~RtpCapabilities();
std::vector<webrtc::RtpExtension> 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<webrtc::RtpHeaderExtensionCapability>
GetRtpHeaderExtensions() const = 0;
};
class VoiceEngineInterface : public RtpHeaderExtensionQueryInterface {
public:
VoiceEngineInterface() = default;
virtual ~VoiceEngineInterface() = default;
VoiceEngineInterface(const VoiceEngineInterface&) = delete;
VoiceEngineInterface& operator=(const VoiceEngineInterface&) = delete;
// 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;
virtual std::unique_ptr<VoiceMediaSendChannelInterface> CreateSendChannel(
webrtc::Call* call,
const MediaConfig& config,
const AudioOptions& options,
const webrtc::CryptoOptions& crypto_options,
webrtc::AudioCodecPairId codec_pair_id) {
// TODO(hta): Make pure virtual when all downstream has updated
RTC_CHECK_NOTREACHED();
return nullptr;
}
virtual std::unique_ptr<VoiceMediaReceiveChannelInterface>
CreateReceiveChannel(webrtc::Call* call,
const MediaConfig& config,
const AudioOptions& options,
const webrtc::CryptoOptions& crypto_options,
webrtc::AudioCodecPairId codec_pair_id) {
// TODO(hta): Make pure virtual when all downstream has updated
RTC_CHECK_NOTREACHED();
return nullptr;
}
virtual const std::vector<AudioCodec>& send_codecs() const = 0;
virtual const std::vector<AudioCodec>& 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;
virtual absl::optional<webrtc::AudioDeviceModule::Stats>
GetAudioDeviceStats() = 0;
};
class VideoEngineInterface : public RtpHeaderExtensionQueryInterface {
public:
VideoEngineInterface() = default;
virtual ~VideoEngineInterface() = default;
VideoEngineInterface(const VideoEngineInterface&) = delete;
VideoEngineInterface& operator=(const VideoEngineInterface&) = delete;
virtual std::unique_ptr<VideoMediaSendChannelInterface> CreateSendChannel(
webrtc::Call* call,
const MediaConfig& config,
const VideoOptions& options,
const webrtc::CryptoOptions& crypto_options,
webrtc::VideoBitrateAllocatorFactory* video_bitrate_allocator_factory) {
// Default implementation, delete when all is updated
RTC_CHECK_NOTREACHED();
return nullptr;
}
virtual std::unique_ptr<VideoMediaReceiveChannelInterface>
CreateReceiveChannel(webrtc::Call* call,
const MediaConfig& config,
const VideoOptions& options,
const webrtc::CryptoOptions& crypto_options) {
// Default implementation, delete when all is updated
RTC_CHECK_NOTREACHED();
return nullptr;
}
// Retrieve list of supported codecs.
virtual std::vector<VideoCodec> send_codecs() const = 0;
virtual std::vector<VideoCodec> recv_codecs() const = 0;
// As above, but if include_rtx is false, don't include RTX codecs.
// TODO(bugs.webrtc.org/13931): Remove default implementation once
// upstream subclasses have converted.
virtual std::vector<VideoCodec> send_codecs(bool include_rtx) const {
RTC_DCHECK(include_rtx);
return send_codecs();
}
virtual std::vector<VideoCodec> recv_codecs(bool include_rtx) const {
RTC_DCHECK(include_rtx);
return recv_codecs();
}
};
// 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 FieldTrialsView trials map.
class CompositeMediaEngine : public MediaEngineInterface {
public:
CompositeMediaEngine(std::unique_ptr<webrtc::FieldTrialsView> trials,
std::unique_ptr<VoiceEngineInterface> audio_engine,
std::unique_ptr<VideoEngineInterface> video_engine);
CompositeMediaEngine(std::unique_ptr<VoiceEngineInterface> audio_engine,
std::unique_ptr<VideoEngineInterface> 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<webrtc::FieldTrialsView> trials_;
const std::unique_ptr<VoiceEngineInterface> voice_engine_;
const std::unique_ptr<VideoEngineInterface> 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<webrtc::RtpExtension> GetDefaultEnabledRtpHeaderExtensions(
const RtpHeaderExtensionQueryInterface& query_interface);
} // namespace cricket
#endif // MEDIA_BASE_MEDIA_ENGINE_H_
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