Repo created

TMessagesProj/jni/voip/webrtc/sdk/media_constraints.h

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+/*
+ *  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.
+ */
+
+// 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 SDK_MEDIA_CONSTRAINTS_H_
+#define SDK_MEDIA_CONSTRAINTS_H_
+
+#include <stddef.h>
+
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "api/audio_options.h"
+#include "api/peer_connection_interface.h"
+
+namespace webrtc {
+
+// Class representing constraints, as used by the android and objc apis.
+//
+// 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 MediaConstraints {
+ 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:
+    Constraints() = default;
+    Constraints(std::initializer_list<Constraint> l)
+        : std::vector<Constraint>(l) {}
+
+    bool FindFirst(const std::string& key, std::string* value) const;
+  };
+
+  MediaConstraints() = default;
+  MediaConstraints(Constraints mandatory, Constraints optional)
+      : mandatory_(std::move(mandatory)), optional_(std::move(optional)) {}
+
+  // Constraint keys used by a local audio source.
+
+  // These keys are google specific.
+  static const char kGoogEchoCancellation[];  // googEchoCancellation
+
+  static const char kAutoGainControl[];   // googAutoGainControl
+  static const char kNoiseSuppression[];  // googNoiseSuppression
+  static const char kHighpassFilter[];    // googHighpassFilter
+  static const char kAudioMirroring[];    // googAudioMirroring
+  static const char
+      kAudioNetworkAdaptorConfig[];  // googAudioNetworkAdaptorConfig
+  static const char kInitAudioRecordingOnSend[];  // InitAudioRecordingOnSend;
+
+  // 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.
+  // 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[];
+  static const char kScreencastMinBitrate[];   // googScreencastMinBitrate
+  static const char kCpuOveruseDetection[];    // googCpuOveruseDetection
+
+  // Constraint to enable negotiating raw RTP packetization using attribute
+  // "a=packetization:<payload_type> raw" in the SDP for all video payload.
+  static const char kRawPacketizationForVideoEnabled[];
+
+  // Specifies number of simulcast layers for all video tracks
+  // with a Plan B offer/answer
+  // (see RTCOfferAnswerOptions::num_simulcast_layers).
+  static const char kNumSimulcastLayers[];
+
+  ~MediaConstraints() = default;
+
+  const Constraints& GetMandatory() const { return mandatory_; }
+  const Constraints& GetOptional() const { return optional_; }
+
+ private:
+  const Constraints mandatory_ = {};
+  const Constraints optional_ = {};
+};
+
+// Copy all relevant constraints into an RTCConfiguration object.
+void CopyConstraintsIntoRtcConfiguration(
+    const MediaConstraints* constraints,
+    PeerConnectionInterface::RTCConfiguration* configuration);
+
+// Copy all relevant constraints into an AudioOptions object.
+void CopyConstraintsIntoAudioOptions(const MediaConstraints* constraints,
+                                     cricket::AudioOptions* options);
+
+bool CopyConstraintsIntoOfferAnswerOptions(
+    const MediaConstraints* constraints,
+    PeerConnectionInterface::RTCOfferAnswerOptions* offer_answer_options);
+
+}  // namespace webrtc
+
+#endif  // SDK_MEDIA_CONSTRAINTS_H_