api/media_constraints_interface.h

/*
 *  Copyright 2013 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.
 */

// This file contains the interface for MediaConstraints, corresponding to
// the definition at
// http://www.w3.org/TR/mediacapture-streams/#mediastreamconstraints and also
// used in WebRTC: http://dev.w3.org/2011/webrtc/editor/webrtc.html#constraints.

// Implementation of the w3c constraints spec is the responsibility of the
// browser. Chrome no longer uses the constraints api declared here, and it will
// be removed from WebRTC.
// https://bugs.chromium.org/p/webrtc/issues/detail?id=9239

#ifndef API_MEDIA_CONSTRAINTS_INTERFACE_H_
#define API_MEDIA_CONSTRAINTS_INTERFACE_H_

#include <stddef.h>
#include <string>
#include <vector>

#include "api/audio_options.h"
#include "api/peer_connection_interface.h"

namespace webrtc {

// Interface used for passing arguments about media constraints
// to the MediaStream and PeerConnection implementation.
//
// Constraints may be either "mandatory", which means that unless satisfied,
// the method taking the constraints should fail, or "optional", which means
// they may not be satisfied..
class MediaConstraintsInterface {
 public:
  struct Constraint {
    Constraint() {}
    Constraint(const std::string& key, const std::string value)
        : key(key), value(value) {}
    std::string key;
    std::string value;
  };

  class Constraints : public std::vector<Constraint> {
   public:
    bool FindFirst(const std::string& key, std::string* value) const;
  };

  // Constraint keys used by a local audio source.

  // These keys are google specific.
  static const char kGoogEchoCancellation[];  // googEchoCancellation

  static const char kExtendedFilterEchoCancellation[];  // googEchoCancellation2
  static const char kDAEchoCancellation[];            // googDAEchoCancellation
  static const char kAutoGainControl[];               // googAutoGainControl
  static const char kExperimentalAutoGainControl[];   // googAutoGainControl2
  static const char kNoiseSuppression[];              // googNoiseSuppression
  static const char kExperimentalNoiseSuppression[];  // googNoiseSuppression2
  static const char kHighpassFilter[];                // googHighpassFilter
  static const char kTypingNoiseDetection[];  // googTypingNoiseDetection
  static const char kAudioMirroring[];        // googAudioMirroring
  static const char
      kAudioNetworkAdaptorConfig[];  // goodAudioNetworkAdaptorConfig

  // Constraint keys for CreateOffer / CreateAnswer
  // Specified by the W3C PeerConnection spec
  static const char kOfferToReceiveVideo[];     // OfferToReceiveVideo
  static const char kOfferToReceiveAudio[];     // OfferToReceiveAudio
  static const char kVoiceActivityDetection[];  // VoiceActivityDetection
  static const char kIceRestart[];              // IceRestart
  // These keys are google specific.
  static const char kUseRtpMux[];  // googUseRtpMUX

  // Constraints values.
  static const char kValueTrue[];   // true
  static const char kValueFalse[];  // false

  // PeerConnection constraint keys.
  // Temporary pseudo-constraints used to enable DTLS-SRTP
  static const char kEnableDtlsSrtp[];  // Enable DTLS-SRTP
  // Temporary pseudo-constraints used to enable DataChannels
  static const char kEnableRtpDataChannels[];  // Enable RTP DataChannels
  // Google-specific constraint keys.
  // Temporary pseudo-constraint for enabling DSCP through JS.
  static const char kEnableDscp[];  // googDscp
  // Constraint to enable IPv6 through JS.
  static const char kEnableIPv6[];  // googIPv6
  // Temporary constraint to enable suspend below min bitrate feature.
  static const char kEnableVideoSuspendBelowMinBitrate[];
  // googSuspendBelowMinBitrate
  // Constraint to enable combined audio+video bandwidth estimation.
  static const char kCombinedAudioVideoBwe[];  // googCombinedAudioVideoBwe
  static const char kScreencastMinBitrate[];   // googScreencastMinBitrate
  static const char kCpuOveruseDetection[];    // googCpuOveruseDetection

  // Specifies number of simulcast layers for all video tracks
  // with a Plan B offer/answer
  // (see RTCOfferAnswerOptions::num_simulcast_layers).
  static const char kNumSimulcastLayers[];

  virtual ~MediaConstraintsInterface() = default;

  virtual const Constraints& GetMandatory() const = 0;
  virtual const Constraints& GetOptional() const = 0;
};

bool FindConstraint(const MediaConstraintsInterface* constraints,
                    const std::string& key,
                    bool* value,
                    size_t* mandatory_constraints);

bool FindConstraint(const MediaConstraintsInterface* constraints,
                    const std::string& key,
                    int* value,
                    size_t* mandatory_constraints);

// Copy all relevant constraints into an RTCConfiguration object.
void CopyConstraintsIntoRtcConfiguration(
    const MediaConstraintsInterface* constraints,
    PeerConnectionInterface::RTCConfiguration* configuration);

// Copy all relevant constraints into an AudioOptions object.
void CopyConstraintsIntoAudioOptions(
    const MediaConstraintsInterface* constraints,
    cricket::AudioOptions* options);

bool CopyConstraintsIntoOfferAnswerOptions(
    const MediaConstraintsInterface* constraints,
    PeerConnectionInterface::RTCOfferAnswerOptions* offer_answer_options);

}  // namespace webrtc

#endif  // API_MEDIA_CONSTRAINTS_INTERFACE_H_