chromium/third_party/webrtc/media/base/media_engine.h

/*
 *  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 AudioMixer;
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> send_codecs,
    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> send_codecs,
    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> send_codecs,
    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 { … };

class RtpHeaderExtensionQueryInterface { … };

class VoiceEngineInterface : public RtpHeaderExtensionQueryInterface { … };

class VideoEngineInterface : public RtpHeaderExtensionQueryInterface { … };

// 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 { … };

// CompositeMediaEngine constructs a MediaEngine from separate
// voice and video engine classes.
// Optionally owns a FieldTrialsView trials map.
class CompositeMediaEngine : public MediaEngineInterface { … };

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_