webrtc-m89-android.7z

/* * 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.support.annotation.Nullable; import android.util.Log; import java.io.File; import java.io.IOException; import java.nio.ByteBuffer; import java.nio.charset.Charset; import java.text.DateFormat; import java.text.SimpleDateFormat; import java.util.ArrayList; import java.util.Arrays; import java.util.Collections; import java.util.Date; import java.util.Iterator; import java.util.List; import java.util.Locale; import java.util.Timer; import java.util.TimerTask; import java.util.concurrent.ExecutorService; import java.util.concurrent.Executors; 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.CandidatePairChangeEvent; import org.webrtc.DataChannel; import org.webrtc.DefaultVideoDecoderFactory; import org.webrtc.DefaultVideoEncoderFactory; import org.webrtc.EglBase; import org.webrtc.IceCandidate; import org.webrtc.Logging; import org.webrtc.MediaConstraints; import org.webrtc.MediaStream; import org.webrtc.MediaStreamTrack; import org.webrtc.PeerConnection; import org.webrtc.PeerConnection.IceConnectionState; import org.webrtc.PeerConnection.PeerConnectionState; import org.webrtc.PeerConnectionFactory; import org.webrtc.RtpParameters; import org.webrtc.RtpReceiver; import org.webrtc.RtpSender; import org.webrtc.RtpTransceiver; import org.webrtc.SdpObserver; import org.webrtc.SessionDescription; import org.webrtc.SoftwareVideoDecoderFactory; import org.webrtc.SoftwareVideoEncoderFactory; import org.webrtc.StatsObserver; import org.webrtc.StatsReport; import org.webrtc.SurfaceTextureHelper; import org.webrtc.VideoCapturer; import org.webrtc.VideoDecoderFactory; import org.webrtc.VideoEncoderFactory; import org.webrtc.VideoSink; import org.webrtc.VideoSource; import org.webrtc.VideoTrack; import org.webrtc.audio.AudioDeviceModule; import org.webrtc.audio.JavaAudioDeviceModule; import org.webrtc.audio.JavaAudioDeviceModule.AudioRecordErrorCallback; import org.webrtc.audio.JavaAudioDeviceModule.AudioRecordStateCallback; import org.webrtc.audio.JavaAudioDeviceModule.AudioTrackErrorCallback; import org.webrtc.audio.JavaAudioDeviceModule.AudioTrackStateCallback; /** * 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-Advertised/Enabled/WebRTC-FlexFEC-03/Enabled/"; private static final String VIDEO_VP8_INTEL_HW_ENCODER_FIELDTRIAL = "WebRTC-IntelVP8/Enabled/"; private static final String DISABLE_WEBRTC_AGC_FIELDTRIAL = "WebRTC-Audio-MinimizeResamplingOnMobile/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 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 String RTCEVENTLOG_OUTPUT_DIR_NAME = "rtc_event_log"; // Executor thread is started once in private ctor and is used for all // peer connection API calls to ensure new peer connection factory is // created on the same thread as previously destroyed factory. private static final ExecutorService executor = Executors.newSingleThreadExecutor(); private final PCObserver pcObserver = new PCObserver(); private final SDPObserver sdpObserver = new SDPObserver(); private final Timer statsTimer = new Timer(); private final EglBase rootEglBase; private final Context appContext; private final PeerConnectionParameters peerConnectionParameters; private final PeerConnectionEvents events; @Nullable private PeerConnectionFactory factory; @Nullable private PeerConnection peerConnection; @Nullable private AudioSource audioSource; @Nullable private SurfaceTextureHelper surfaceTextureHelper; @Nullable private VideoSource videoSource; private boolean preferIsac; private boolean videoCapturerStopped; private boolean isError; @Nullable private VideoSink localRender; @Nullable private List<VideoSink> remoteRenders; private SignalingParameters signalingParameters; private int videoWidth; private int videoHeight; private int videoFps; private MediaConstraints audioConstraints; private MediaConstraints sdpMediaConstraints; // 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. @Nullable private List<IceCandidate> queuedRemoteCandidates; private boolean isInitiator; @Nullable private SessionDescription localDescription; // either offer or answer description @Nullable private VideoCapturer videoCapturer; // enableVideo is set to true if video should be rendered and sent. private boolean renderVideo = true; @Nullable private VideoTrack localVideoTrack; @Nullable private VideoTrack remoteVideoTrack; @Nullable private RtpSender localVideoSender; // enableAudio is set to true if audio should be sent. private boolean enableAudio = true; @Nullable private AudioTrack localAudioTrack; @Nullable private DataChannel dataChannel; private final boolean dataChannelEnabled; // Enable RtcEventLog. @Nullable private RtcEventLog rtcEventLog; // Implements the WebRtcAudioRecordSamplesReadyCallback interface and writes // recorded audio samples to an output file. @Nullable private RecordedAudioToFileController saveRecordedAudioToFile; /** * 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;