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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-19 00:47:55 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-19 00:47:55 +0000
commit26a029d407be480d791972afb5975cf62c9360a6 (patch)
treef435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/sdk/media_constraints.h
parentInitial commit. (diff)
downloadfirefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz
firefox-26a029d407be480d791972afb5975cf62c9360a6.zip
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
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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_