android webrtc libjingle_peerconnection

/* * Copyright 2014 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. */ package org.appspot.apprtc; import android.content.Context; import android.os.Environment; import android.os.ParcelFileDescriptor; import android.util.Log; import java.io.File; import java.io.IOException; import java.nio.ByteBuffer; import java.util.ArrayList; import java.util.Arrays; import java.util.Collections; import java.util.EnumSet; import java.util.Iterator; import java.util.LinkedList; import java.util.List; import java.util.Timer; import java.util.TimerTask; import java.util.concurrent.Executors; import java.util.concurrent.ScheduledExecutorService; import java.util.regex.Matcher; import java.util.regex.Pattern; import org.appspot.apprtc.AppRTCClient.SignalingParameters; import org.webrtc.AudioSource; import org.webrtc.AudioTrack; import org.webrtc.CameraVideoCapturer; import org.webrtc.DataChannel; import org.webrtc.EglBase; import org.webrtc.IceCandidate; import org.webrtc.Logging; import org.webrtc.MediaConstraints; import org.webrtc.MediaStream; import org.webrtc.PeerConnection; import org.webrtc.PeerConnection.IceConnectionState; import org.webrtc.PeerConnectionFactory; import org.webrtc.RtpParameters; import org.webrtc.RtpReceiver; import org.webrtc.RtpSender; import org.webrtc.SdpObserver; import org.webrtc.SessionDescription; import org.webrtc.StatsObserver; import org.webrtc.StatsReport; import org.webrtc.VideoCapturer; import org.webrtc.VideoRenderer; import org.webrtc.VideoSource; import org.webrtc.VideoTrack; import org.webrtc.voiceengine.WebRtcAudioManager; import org.webrtc.voiceengine.WebRtcAudioRecord; import org.webrtc.voiceengine.WebRtcAudioRecord.WebRtcAudioRecordErrorCallback; import org.webrtc.voiceengine.WebRtcAudioUtils; /** * Peer connection client implementation. * * <p>All public methods are routed to local looper thread. * All PeerConnectionEvents callbacks are invoked from the same looper thread. * This class is a singleton. */ public class PeerConnectionClient { public static final String VIDEO_TRACK_ID = "ARDAMSv0"; public static final String AUDIO_TRACK_ID = "ARDAMSa0"; public static final String VIDEO_TRACK_TYPE = "video"; private static final String TAG = "PCRTCClient"; private static final String VIDEO_CODEC_VP8 = "VP8"; private static final String VIDEO_CODEC_VP9 = "VP9"; private static final String VIDEO_CODEC_H264 = "H264"; private static final String VIDEO_CODEC_H264_BASELINE = "H264 Baseline"; private static final String VIDEO_CODEC_H264_HIGH = "H264 High"; private static final String AUDIO_CODEC_OPUS = "opus"; private static final String AUDIO_CODEC_ISAC = "ISAC"; private static final String VIDEO_CODEC_PARAM_START_BITRATE = "x-google-start-bitrate"; private static final String VIDEO_FLEXFEC_FIELDTRIAL = "WebRTC-FlexFEC-03/Enabled/"; private static final String VIDEO_VP8_INTEL_HW_ENCODER_FIELDTRIAL = "WebRTC-IntelVP8/Enabled/"; private static final String VIDEO_H264_HIGH_PROFILE_FIELDTRIAL = "WebRTC-H264HighProfile/Enabled/"; private static final String AUDIO_CODEC_PARAM_BITRATE = "maxaveragebitrate"; private static final String AUDIO_ECHO_CANCELLATION_CONSTRAINT = "googEchoCancellation"; private static final String AUDIO_AUTO_GAIN_CONTROL_CONSTRAINT = "googAutoGainControl"; private static final String AUDIO_HIGH_PASS_FILTER_CONSTRAINT = "googHighpassFilter"; private static final String AUDIO_NOISE_SUPPRESSION_CONSTRAINT = "googNoiseSuppression"; private static final String AUDIO_LEVEL_CONTROL_CONSTRAINT = "levelControl"; private static final String DTLS_SRTP_KEY_AGREEMENT_CONSTRAINT = "DtlsSrtpKeyAgreement"; private static final int HD_VIDEO_WIDTH = 1280; private static final int HD_VIDEO_HEIGHT = 720; private static final int BPS_IN_KBPS = 1000; private static final PeerConnectionClient instance = new PeerConnectionClient(); private final PCObserver pcObserver = new PCObserver(); private final SDPObserver sdpObserver = new SDPObserver(); private final ScheduledExecutorService executor; private PeerConnectionFactory factory; private PeerConnection peerConnection; PeerConnectionFactory.Options options = null; private AudioSource audioSource; private VideoSource videoSource; private boolean videoCallEnabled; private boolean preferIsac; private String preferredVideoCodec; private boolean videoCapturerStopped; private boolean isError; private Timer statsTimer; private VideoRenderer.Callbacks localRender; private List<VideoRenderer.Callbacks> remoteRenders; private SignalingParameters signalingParameters; private MediaConstraints pcConstraints; private int videoWidth; private int videoHeight; private int videoFps; private MediaConstraints audioConstraints; private ParcelFileDescriptor aecDumpFileDescriptor; private MediaConstraints sdpMediaConstraints; private PeerConnectionParameters peerConnectionParameters; // Queued remote ICE candidates are consumed only after both local and // remote descriptions are set. Similarly local ICE candidates are sent to // remote peer after both local and remote description are set. private LinkedList<IceCandidate> queuedRemoteCandidates; private PeerConnectionEvents events; private boolean isInitiator; private SessionDescription localSdp; // either offer or answer SDP private MediaStream mediaStream; private VideoCapturer videoCapturer; // enableVideo is set to true if video should be rendered and sent. private boolean renderVideo; private VideoTrack localVideoTrack; private VideoTrack remoteVideoTrack; private RtpSender localVideoSender; // enableAudio is set to true if audio should be sent. private boolean enableAudio; private AudioTrack localAudioTrack; private DataChannel dataChannel; private boolean dataChannelEnabled; /** * Peer connection parameters. */ public static class DataChannelParameters { public final boolean ordered; public final int maxRetransmitTimeMs; public final int maxRetransmits; public final String protocol; public final boolean negotiated; public final int id; public DataChannelParameters(boolean ordered, int maxRetransmitTimeMs, int maxRetransmits, String protocol, boolean negotiated, int id) { this.ordered = ordered; this.maxRetransmitTimeMs = maxRetransmitTimeMs; this.maxRetransmits = maxRetransmits; this.protocol = protocol; this.negotiated = negotiated; this.id = id; } } /** * Peer connection parameters. */ public static class PeerConnectionParameters { public final boolean videoCallEnabled; public final boolean loopback; public final boolean tracing; public final int videoWidth; public final int videoHeight; public final int videoFps; public final int videoMaxBitrate; public final String videoCodec; public final boolean videoCodecHwAcceleration; public final boolean videoFlexfecEnabled; public final int audioStartBitrate; public final String audioCodec; public final boolean noAudioProcessing; public final boolean aecDump; public final boolean useOpenSLES; public final boolean disableBuiltInAEC; public final boolean disableBuiltInAGC; public final boolean disableBuiltInNS; public final boolean enableLevelControl; private final DataChannelParameters dataChannelParameters; public PeerConnectionParameters(boolean videoCallEnabled, boolean loopback, boolean tracing, int videoWidth, int videoHeight, int videoFps, int videoMaxBitrate, String videoCodec, boolean videoCodecHwAcceleration, boolea