| /* |
| * 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.appspot.apprtc.RecordedAudioToFileController; |
| import org.webrtc.AddIceObserver; |
| 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 VIDEO_CODEC_AV1 = "AV1"; |
| private static final String VIDEO_CODEC_AV1_SDP_CODEC_NAME = "AV1X"; |
| 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> remoteSinks; |
| 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; |
| 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 saveInputAudioToFile; |
| public final boolean useOpenSLES; |
| public final boolean disableBuiltInAEC; |
| public final boolean disableBuiltInAGC; |
| public final boolean disableBuiltInNS; |
| public final boolean disableWebRtcAGCAndHPF; |
| public final boolean enableRtcEventLog; |
| private final DataChannelParameters dataChannelParameters; |
| |
| public PeerConnectionParameters(boolean videoCallEnabled, boolean loopback, boolean tracing, |
| int videoWidth, int videoHeight, int videoFps, int videoMaxBitrate, String videoCodec, |
| boolean videoCodecHwAcceleration, boolean videoFlexfecEnabled, int audioStartBitrate, |
| String audioCodec, boolean noAudioProcessing, boolean aecDump, boolean saveInputAudioToFile, |
| boolean useOpenSLES, boolean disableBuiltInAEC, boolean disableBuiltInAGC, |
| boolean disableBuiltInNS, boolean disableWebRtcAGCAndHPF, boolean enableRtcEventLog, |
| DataChannelParameters dataChannelParameters) { |
| this.videoCallEnabled = videoCallEnabled; |
| this.loopback = loopback; |
| this.tracing = tracing; |
| this.videoWidth = videoWidth; |
| this.videoHeight = videoHeight; |
| this.videoFps = videoFps; |
| this.videoMaxBitrate = videoMaxBitrate; |
| this.videoCodec = videoCodec; |
| this.videoFlexfecEnabled = videoFlexfecEnabled; |
| this.videoCodecHwAcceleration = videoCodecHwAcceleration; |
| this.audioStartBitrate = audioStartBitrate; |
| this.audioCodec = audioCodec; |
| this.noAudioProcessing = noAudioProcessing; |
| this.aecDump = aecDump; |
| this.saveInputAudioToFile = saveInputAudioToFile; |
| this.useOpenSLES = useOpenSLES; |
| this.disableBuiltInAEC = disableBuiltInAEC; |
| this.disableBuiltInAGC = disableBuiltInAGC; |
| this.disableBuiltInNS = disableBuiltInNS; |
| this.disableWebRtcAGCAndHPF = disableWebRtcAGCAndHPF; |
| this.enableRtcEventLog = enableRtcEventLog; |
| this.dataChannelParameters = dataChannelParameters; |
| } |
| } |
| |
| /** |
| * Peer connection events. |
| */ |
| public interface PeerConnectionEvents { |
| /** |
| * Callback fired once local SDP is created and set. |
| */ |
| void onLocalDescription(final SessionDescription sdp); |
| |
| /** |
| * Callback fired once local Ice candidate is generated. |
| */ |
| void onIceCandidate(final IceCandidate candidate); |
| |
| /** |
| * Callback fired once local ICE candidates are removed. |
| */ |
| void onIceCandidatesRemoved(final IceCandidate[] candidates); |
| |
| /** |
| * Callback fired once connection is established (IceConnectionState is |
| * CONNECTED). |
| */ |
| void onIceConnected(); |
| |
| /** |
| * Callback fired once connection is disconnected (IceConnectionState is |
| * DISCONNECTED). |
| */ |
| void onIceDisconnected(); |
| |
| /** |
| * Callback fired once DTLS connection is established (PeerConnectionState |
| * is CONNECTED). |
| */ |
| void onConnected(); |
| |
| /** |
| * Callback fired once DTLS connection is disconnected (PeerConnectionState |
| * is DISCONNECTED). |
| */ |
| void onDisconnected(); |
| |
| /** |
| * Callback fired once peer connection is closed. |
| */ |
| void onPeerConnectionClosed(); |
| |
| /** |
| * Callback fired once peer connection statistics is ready. |
| */ |
| void onPeerConnectionStatsReady(final StatsReport[] reports); |
| |
| /** |
| * Callback fired once peer connection error happened. |
| */ |
| void onPeerConnectionError(final String description); |
| } |
| |
| /** |
| * Create a PeerConnectionClient with the specified parameters. PeerConnectionClient takes |
| * ownership of `eglBase`. |
| */ |
| public PeerConnectionClient(Context appContext, EglBase eglBase, |
| PeerConnectionParameters peerConnectionParameters, PeerConnectionEvents events) { |
| this.rootEglBase = eglBase; |
| this.appContext = appContext; |
| this.events = events; |
| this.peerConnectionParameters = peerConnectionParameters; |
| this.dataChannelEnabled = peerConnectionParameters.dataChannelParameters != null; |
| |
| Log.d(TAG, "Preferred video codec: " + getSdpVideoCodecName(peerConnectionParameters)); |
| |
| final String fieldTrials = getFieldTrials(peerConnectionParameters); |
| executor.execute(() -> { |
| Log.d(TAG, "Initialize WebRTC. Field trials: " + fieldTrials); |
| PeerConnectionFactory.initialize( |
| PeerConnectionFactory.InitializationOptions.builder(appContext) |
| .setFieldTrials(fieldTrials) |
| .setEnableInternalTracer(true) |
| .createInitializationOptions()); |
| }); |
| } |
| |
| /** |
| * This function should only be called once. |
| */ |
| public void createPeerConnectionFactory(PeerConnectionFactory.Options options) { |
| if (factory != null) { |
| throw new IllegalStateException("PeerConnectionFactory has already been constructed"); |
| } |
| executor.execute(() -> createPeerConnectionFactoryInternal(options)); |
| } |
| |
| public void createPeerConnection(final VideoSink localRender, final VideoSink remoteSink, |
| final VideoCapturer videoCapturer, final SignalingParameters signalingParameters) { |
| if (peerConnectionParameters.videoCallEnabled && videoCapturer == null) { |
| Log.w(TAG, "Video call enabled but no video capturer provided."); |
| } |
| createPeerConnection( |
| localRender, Collections.singletonList(remoteSink), videoCapturer, signalingParameters); |
| } |
| |
| public void createPeerConnection(final VideoSink localRender, final List<VideoSink> remoteSinks, |
| final VideoCapturer videoCapturer, final SignalingParameters signalingParameters) { |
| if (peerConnectionParameters == null) { |
| Log.e(TAG, "Creating peer connection without initializing factory."); |
| return; |
| } |
| this.localRender = localRender; |
| this.remoteSinks = remoteSinks; |
| this.videoCapturer = videoCapturer; |
| this.signalingParameters = signalingParameters; |
| executor.execute(() -> { |
| try { |
| createMediaConstraintsInternal(); |
| createPeerConnectionInternal(); |
| maybeCreateAndStartRtcEventLog(); |
| } catch (Exception e) { |
| reportError("Failed to create peer connection: " + e.getMessage()); |
| throw e; |
| } |
| }); |
| } |
| |
| public void close() { |
| executor.execute(this ::closeInternal); |
| } |
| |
| private boolean isVideoCallEnabled() { |
| return peerConnectionParameters.videoCallEnabled && videoCapturer != null; |
| } |
| |
| private void createPeerConnectionFactoryInternal(PeerConnectionFactory.Options options) { |
| isError = false; |
| |
| if (peerConnectionParameters.tracing) { |
| PeerConnectionFactory.startInternalTracingCapture( |
| Environment.getExternalStorageDirectory().getAbsolutePath() + File.separator |
| + "webrtc-trace.txt"); |
| } |
| |
| // Check if ISAC is used by default. |
| preferIsac = peerConnectionParameters.audioCodec != null |
| && peerConnectionParameters.audioCodec.equals(AUDIO_CODEC_ISAC); |
| |
| // It is possible to save a copy in raw PCM format on a file by checking |
| // the "Save input audio to file" checkbox in the Settings UI. A callback |
| // interface is set when this flag is enabled. As a result, a copy of recorded |
| // audio samples are provided to this client directly from the native audio |
| // layer in Java. |
| if (peerConnectionParameters.saveInputAudioToFile) { |
| if (!peerConnectionParameters.useOpenSLES) { |
| Log.d(TAG, "Enable recording of microphone input audio to file"); |
| saveRecordedAudioToFile = new RecordedAudioToFileController(executor); |
| } else { |
| // TODO(henrika): ensure that the UI reflects that if OpenSL ES is selected, |
| // then the "Save inut audio to file" option shall be grayed out. |
| Log.e(TAG, "Recording of input audio is not supported for OpenSL ES"); |
| } |
| } |
| |
| final AudioDeviceModule adm = createJavaAudioDevice(); |
| |
| // Create peer connection factory. |
| if (options != null) { |
| Log.d(TAG, "Factory networkIgnoreMask option: " + options.networkIgnoreMask); |
| } |
| final boolean enableH264HighProfile = |
| VIDEO_CODEC_H264_HIGH.equals(peerConnectionParameters.videoCodec); |
| final VideoEncoderFactory encoderFactory; |
| final VideoDecoderFactory decoderFactory; |
| |
| if (peerConnectionParameters.videoCodecHwAcceleration) { |
| encoderFactory = new DefaultVideoEncoderFactory( |
| rootEglBase.getEglBaseContext(), true /* enableIntelVp8Encoder */, enableH264HighProfile); |
| decoderFactory = new DefaultVideoDecoderFactory(rootEglBase.getEglBaseContext()); |
| } else { |
| encoderFactory = new SoftwareVideoEncoderFactory(); |
| decoderFactory = new SoftwareVideoDecoderFactory(); |
| } |
| |
| factory = PeerConnectionFactory.builder() |
| .setOptions(options) |
| .setAudioDeviceModule(adm) |
| .setVideoEncoderFactory(encoderFactory) |
| .setVideoDecoderFactory(decoderFactory) |
| .createPeerConnectionFactory(); |
| Log.d(TAG, "Peer connection factory created."); |
| adm.release(); |
| } |
| |
| AudioDeviceModule createJavaAudioDevice() { |
| // Enable/disable OpenSL ES playback. |
| if (!peerConnectionParameters.useOpenSLES) { |
| Log.w(TAG, "External OpenSLES ADM not implemented yet."); |
| // TODO(magjed): Add support for external OpenSLES ADM. |
| } |
| |
| // Set audio record error callbacks. |
| AudioRecordErrorCallback audioRecordErrorCallback = new AudioRecordErrorCallback() { |
| @Override |
| public void onWebRtcAudioRecordInitError(String errorMessage) { |
| Log.e(TAG, "onWebRtcAudioRecordInitError: " + errorMessage); |
| reportError(errorMessage); |
| } |
| |
| @Override |
| public void onWebRtcAudioRecordStartError( |
| JavaAudioDeviceModule.AudioRecordStartErrorCode errorCode, String errorMessage) { |
| Log.e(TAG, "onWebRtcAudioRecordStartError: " + errorCode + ". " + errorMessage); |
| reportError(errorMessage); |
| } |
| |
| @Override |
| public void onWebRtcAudioRecordError(String errorMessage) { |
| Log.e(TAG, "onWebRtcAudioRecordError: " + errorMessage); |
| reportError(errorMessage); |
| } |
| }; |
| |
| AudioTrackErrorCallback audioTrackErrorCallback = new AudioTrackErrorCallback() { |
| @Override |
| public void onWebRtcAudioTrackInitError(String errorMessage) { |
| Log.e(TAG, "onWebRtcAudioTrackInitError: " + errorMessage); |
| reportError(errorMessage); |
| } |
| |
| @Override |
| public void onWebRtcAudioTrackStartError( |
| JavaAudioDeviceModule.AudioTrackStartErrorCode errorCode, String errorMessage) { |
| Log.e(TAG, "onWebRtcAudioTrackStartError: " + errorCode + ". " + errorMessage); |
| reportError(errorMessage); |
| } |
| |
| @Override |
| public void onWebRtcAudioTrackError(String errorMessage) { |
| Log.e(TAG, "onWebRtcAudioTrackError: " + errorMessage); |
| reportError(errorMessage); |
| } |
| }; |
| |
| // Set audio record state callbacks. |
| AudioRecordStateCallback audioRecordStateCallback = new AudioRecordStateCallback() { |
| @Override |
| public void onWebRtcAudioRecordStart() { |
| Log.i(TAG, "Audio recording starts"); |
| } |
| |
| @Override |
| public void onWebRtcAudioRecordStop() { |
| Log.i(TAG, "Audio recording stops"); |
| } |
| }; |
| |
| // Set audio track state callbacks. |
| AudioTrackStateCallback audioTrackStateCallback = new AudioTrackStateCallback() { |
| @Override |
| public void onWebRtcAudioTrackStart() { |
| Log.i(TAG, "Audio playout starts"); |
| } |
| |
| @Override |
| public void onWebRtcAudioTrackStop() { |
| Log.i(TAG, "Audio playout stops"); |
| } |
| }; |
| |
| return JavaAudioDeviceModule.builder(appContext) |
| .setSamplesReadyCallback(saveRecordedAudioToFile) |
| .setUseHardwareAcousticEchoCanceler(!peerConnectionParameters.disableBuiltInAEC) |
| .setUseHardwareNoiseSuppressor(!peerConnectionParameters.disableBuiltInNS) |
| .setAudioRecordErrorCallback(audioRecordErrorCallback) |
| .setAudioTrackErrorCallback(audioTrackErrorCallback) |
| .setAudioRecordStateCallback(audioRecordStateCallback) |
| .setAudioTrackStateCallback(audioTrackStateCallback) |
| .createAudioDeviceModule(); |
| } |
| |
| private void createMediaConstraintsInternal() { |
| // Create video constraints if video call is enabled. |
| if (isVideoCallEnabled()) { |
| videoWidth = peerConnectionParameters.videoWidth; |
| videoHeight = peerConnectionParameters.videoHeight; |
| videoFps = peerConnectionParameters.videoFps; |
| |
| // If video resolution is not specified, default to HD. |
| if (videoWidth == 0 || videoHeight == 0) { |
| videoWidth = HD_VIDEO_WIDTH; |
| videoHeight = HD_VIDEO_HEIGHT; |
| } |
| |
| // If fps is not specified, default to 30. |
| if (videoFps == 0) { |
| videoFps = 30; |
| } |
| Logging.d(TAG, "Capturing format: " + videoWidth + "x" + videoHeight + "@" + videoFps); |
| } |
| |
| // Create audio constraints. |
| audioConstraints = new MediaConstraints(); |
| // added for audio performance measurements |
| if (peerConnectionParameters.noAudioProcessing) { |
| Log.d(TAG, "Disabling audio processing"); |
| audioConstraints.mandatory.add( |
| new MediaConstraints.KeyValuePair(AUDIO_ECHO_CANCELLATION_CONSTRAINT, "false")); |
| audioConstraints.mandatory.add( |
| new MediaConstraints.KeyValuePair(AUDIO_AUTO_GAIN_CONTROL_CONSTRAINT, "false")); |
| audioConstraints.mandatory.add( |
| new MediaConstraints.KeyValuePair(AUDIO_HIGH_PASS_FILTER_CONSTRAINT, "false")); |
| audioConstraints.mandatory.add( |
| new MediaConstraints.KeyValuePair(AUDIO_NOISE_SUPPRESSION_CONSTRAINT, "false")); |
| } |
| // Create SDP constraints. |
| sdpMediaConstraints = new MediaConstraints(); |
| sdpMediaConstraints.mandatory.add( |
| new MediaConstraints.KeyValuePair("OfferToReceiveAudio", "true")); |
| sdpMediaConstraints.mandatory.add(new MediaConstraints.KeyValuePair( |
| "OfferToReceiveVideo", Boolean.toString(isVideoCallEnabled()))); |
| } |
| |
| private void createPeerConnectionInternal() { |
| if (factory == null || isError) { |
| Log.e(TAG, "Peerconnection factory is not created"); |
| return; |
| } |
| Log.d(TAG, "Create peer connection."); |
| |
| queuedRemoteCandidates = new ArrayList<>(); |
| |
| PeerConnection.RTCConfiguration rtcConfig = |
| new PeerConnection.RTCConfiguration(signalingParameters.iceServers); |
| // TCP candidates are only useful when connecting to a server that supports |
| // ICE-TCP. |
| rtcConfig.tcpCandidatePolicy = PeerConnection.TcpCandidatePolicy.DISABLED; |
| rtcConfig.bundlePolicy = PeerConnection.BundlePolicy.MAXBUNDLE; |
| rtcConfig.rtcpMuxPolicy = PeerConnection.RtcpMuxPolicy.REQUIRE; |
| rtcConfig.continualGatheringPolicy = PeerConnection.ContinualGatheringPolicy.GATHER_CONTINUALLY; |
| // Use ECDSA encryption. |
| rtcConfig.keyType = PeerConnection.KeyType.ECDSA; |
| // Enable DTLS for normal calls and disable for loopback calls. |
| rtcConfig.enableDtlsSrtp = !peerConnectionParameters.loopback; |
| rtcConfig.sdpSemantics = PeerConnection.SdpSemantics.UNIFIED_PLAN; |
| |
| peerConnection = factory.createPeerConnection(rtcConfig, pcObserver); |
| |
| if (dataChannelEnabled) { |
| DataChannel.Init init = new DataChannel.Init(); |
| init.ordered = peerConnectionParameters.dataChannelParameters.ordered; |
| init.negotiated = peerConnectionParameters.dataChannelParameters.negotiated; |
| init.maxRetransmits = peerConnectionParameters.dataChannelParameters.maxRetransmits; |
| init.maxRetransmitTimeMs = peerConnectionParameters.dataChannelParameters.maxRetransmitTimeMs; |
| init.id = peerConnectionParameters.dataChannelParameters.id; |
| init.protocol = peerConnectionParameters.dataChannelParameters.protocol; |
| dataChannel = peerConnection.createDataChannel("ApprtcDemo data", init); |
| } |
| isInitiator = false; |
| |
| // Set INFO libjingle logging. |
| // NOTE: this _must_ happen while `factory` is alive! |
| Logging.enableLogToDebugOutput(Logging.Severity.LS_INFO); |
| |
| List<String> mediaStreamLabels = Collections.singletonList("ARDAMS"); |
| if (isVideoCallEnabled()) { |
| peerConnection.addTrack(createVideoTrack(videoCapturer), mediaStreamLabels); |
| // We can add the renderers right away because we don't need to wait for an |
| // answer to get the remote track. |
| remoteVideoTrack = getRemoteVideoTrack(); |
| remoteVideoTrack.setEnabled(renderVideo); |
| for (VideoSink remoteSink : remoteSinks) { |
| remoteVideoTrack.addSink(remoteSink); |
| } |
| } |
| peerConnection.addTrack(createAudioTrack(), mediaStreamLabels); |
| if (isVideoCallEnabled()) { |
| findVideoSender(); |
| } |
| |
| if (peerConnectionParameters.aecDump) { |
| try { |
| ParcelFileDescriptor aecDumpFileDescriptor = |
| ParcelFileDescriptor.open(new File(Environment.getExternalStorageDirectory().getPath() |
| + File.separator + "Download/audio.aecdump"), |
| ParcelFileDescriptor.MODE_READ_WRITE | ParcelFileDescriptor.MODE_CREATE |
| | ParcelFileDescriptor.MODE_TRUNCATE); |
| factory.startAecDump(aecDumpFileDescriptor.detachFd(), -1); |
| } catch (IOException e) { |
| Log.e(TAG, "Can not open aecdump file", e); |
| } |
| } |
| |
| if (saveRecordedAudioToFile != null) { |
| if (saveRecordedAudioToFile.start()) { |
| Log.d(TAG, "Recording input audio to file is activated"); |
| } |
| } |
| Log.d(TAG, "Peer connection created."); |
| } |
| |
| private File createRtcEventLogOutputFile() { |
| DateFormat dateFormat = new SimpleDateFormat("yyyyMMdd_hhmm_ss", Locale.getDefault()); |
| Date date = new Date(); |
| final String outputFileName = "event_log_" + dateFormat.format(date) + ".log"; |
| return new File( |
| appContext.getDir(RTCEVENTLOG_OUTPUT_DIR_NAME, Context.MODE_PRIVATE), outputFileName); |
| } |
| |
| private void maybeCreateAndStartRtcEventLog() { |
| if (appContext == null || peerConnection == null) { |
| return; |
| } |
| if (!peerConnectionParameters.enableRtcEventLog) { |
| Log.d(TAG, "RtcEventLog is disabled."); |
| return; |
| } |
| rtcEventLog = new RtcEventLog(peerConnection); |
| rtcEventLog.start(createRtcEventLogOutputFile()); |
| } |
| |
| private void closeInternal() { |
| if (factory != null && peerConnectionParameters.aecDump) { |
| factory.stopAecDump(); |
| } |
| Log.d(TAG, "Closing peer connection."); |
| statsTimer.cancel(); |
| if (dataChannel != null) { |
| dataChannel.dispose(); |
| dataChannel = null; |
| } |
| if (rtcEventLog != null) { |
| // RtcEventLog should stop before the peer connection is disposed. |
| rtcEventLog.stop(); |
| rtcEventLog = null; |
| } |
| if (peerConnection != null) { |
| peerConnection.dispose(); |
| peerConnection = null; |
| } |
| Log.d(TAG, "Closing audio source."); |
| if (audioSource != null) { |
| audioSource.dispose(); |
| audioSource = null; |
| } |
| Log.d(TAG, "Stopping capture."); |
| if (videoCapturer != null) { |
| try { |
| videoCapturer.stopCapture(); |
| } catch (InterruptedException e) { |
| throw new RuntimeException(e); |
| } |
| videoCapturerStopped = true; |
| videoCapturer.dispose(); |
| videoCapturer = null; |
| } |
| Log.d(TAG, "Closing video source."); |
| if (videoSource != null) { |
| videoSource.dispose(); |
| videoSource = null; |
| } |
| if (surfaceTextureHelper != null) { |
| surfaceTextureHelper.dispose(); |
| surfaceTextureHelper = null; |
| } |
| if (saveRecordedAudioToFile != null) { |
| Log.d(TAG, "Closing audio file for recorded input audio."); |
| saveRecordedAudioToFile.stop(); |
| saveRecordedAudioToFile = null; |
| } |
| localRender = null; |
| remoteSinks = null; |
| Log.d(TAG, "Closing peer connection factory."); |
| if (factory != null) { |
| factory.dispose(); |
| factory = null; |
| } |
| rootEglBase.release(); |
| Log.d(TAG, "Closing peer connection done."); |
| events.onPeerConnectionClosed(); |
| PeerConnectionFactory.stopInternalTracingCapture(); |
| PeerConnectionFactory.shutdownInternalTracer(); |
| } |
| |
| public boolean isHDVideo() { |
| return isVideoCallEnabled() && videoWidth * videoHeight >= 1280 * 720; |
| } |
| |
| @SuppressWarnings("deprecation") // TODO(sakal): getStats is deprecated. |
| private void getStats() { |
| if (peerConnection == null || isError) { |
| return; |
| } |
| boolean success = peerConnection.getStats(new StatsObserver() { |
| @Override |
| public void onComplete(final StatsReport[] reports) { |
| events.onPeerConnectionStatsReady(reports); |
| } |
| }, null); |
| if (!success) { |
| Log.e(TAG, "getStats() returns false!"); |
| } |
| } |
| |
| public void enableStatsEvents(boolean enable, int periodMs) { |
| if (enable) { |
| try { |
| statsTimer.schedule(new TimerTask() { |
| @Override |
| public void run() { |
| executor.execute(() -> getStats()); |
| } |
| }, 0, periodMs); |
| } catch (Exception e) { |
| Log.e(TAG, "Can not schedule statistics timer", e); |
| } |
| } else { |
| statsTimer.cancel(); |
| } |
| } |
| |
| public void setAudioEnabled(final boolean enable) { |
| executor.execute(() -> { |
| enableAudio = enable; |
| if (localAudioTrack != null) { |
| localAudioTrack.setEnabled(enableAudio); |
| } |
| }); |
| } |
| |
| public void setVideoEnabled(final boolean enable) { |
| executor.execute(() -> { |
| renderVideo = enable; |
| if (localVideoTrack != null) { |
| localVideoTrack.setEnabled(renderVideo); |
| } |
| if (remoteVideoTrack != null) { |
| remoteVideoTrack.setEnabled(renderVideo); |
| } |
| }); |
| } |
| |
| public void createOffer() { |
| executor.execute(() -> { |
| if (peerConnection != null && !isError) { |
| Log.d(TAG, "PC Create OFFER"); |
| isInitiator = true; |
| peerConnection.createOffer(sdpObserver, sdpMediaConstraints); |
| } |
| }); |
| } |
| |
| public void createAnswer() { |
| executor.execute(() -> { |
| if (peerConnection != null && !isError) { |
| Log.d(TAG, "PC create ANSWER"); |
| isInitiator = false; |
| peerConnection.createAnswer(sdpObserver, sdpMediaConstraints); |
| } |
| }); |
| } |
| |
| public void addRemoteIceCandidate(final IceCandidate candidate) { |
| executor.execute(() -> { |
| if (peerConnection != null && !isError) { |
| if (queuedRemoteCandidates != null) { |
| queuedRemoteCandidates.add(candidate); |
| } else { |
| peerConnection.addIceCandidate(candidate, new AddIceObserver() { |
| @Override |
| public void onAddSuccess() { |
| Log.d(TAG, "Candidate " + candidate + " successfully added."); |
| } |
| @Override |
| public void onAddFailure(String error) { |
| Log.d(TAG, "Candidate " + candidate + " addition failed: " + error); |
| } |
| }); |
| } |
| } |
| }); |
| } |
| |
| public void removeRemoteIceCandidates(final IceCandidate[] candidates) { |
| executor.execute(() -> { |
| if (peerConnection == null || isError) { |
| return; |
| } |
| // Drain the queued remote candidates if there is any so that |
| // they are processed in the proper order. |
| drainCandidates(); |
| peerConnection.removeIceCandidates(candidates); |
| }); |
| } |
| |
| public void setRemoteDescription(final SessionDescription desc) { |
| executor.execute(() -> { |
| if (peerConnection == null || isError) { |
| return; |
| } |
| String sdp = desc.description; |
| if (preferIsac) { |
| sdp = preferCodec(sdp, AUDIO_CODEC_ISAC, true); |
| } |
| if (isVideoCallEnabled()) { |
| sdp = preferCodec(sdp, getSdpVideoCodecName(peerConnectionParameters), false); |
| } |
| if (peerConnectionParameters.audioStartBitrate > 0) { |
| sdp = setStartBitrate( |
| AUDIO_CODEC_OPUS, false, sdp, peerConnectionParameters.audioStartBitrate); |
| } |
| Log.d(TAG, "Set remote SDP."); |
| SessionDescription sdpRemote = new SessionDescription(desc.type, sdp); |
| peerConnection.setRemoteDescription(sdpObserver, sdpRemote); |
| }); |
| } |
| |
| public void stopVideoSource() { |
| executor.execute(() -> { |
| if (videoCapturer != null && !videoCapturerStopped) { |
| Log.d(TAG, "Stop video source."); |
| try { |
| videoCapturer.stopCapture(); |
| } catch (InterruptedException e) { |
| } |
| videoCapturerStopped = true; |
| } |
| }); |
| } |
| |
| public void startVideoSource() { |
| executor.execute(() -> { |
| if (videoCapturer != null && videoCapturerStopped) { |
| Log.d(TAG, "Restart video source."); |
| videoCapturer.startCapture(videoWidth, videoHeight, videoFps); |
| videoCapturerStopped = false; |
| } |
| }); |
| } |
| |
| public void setVideoMaxBitrate(@Nullable final Integer maxBitrateKbps) { |
| executor.execute(() -> { |
| if (peerConnection == null || localVideoSender == null || isError) { |
| return; |
| } |
| Log.d(TAG, "Requested max video bitrate: " + maxBitrateKbps); |
| if (localVideoSender == null) { |
| Log.w(TAG, "Sender is not ready."); |
| return; |
| } |
| |
| RtpParameters parameters = localVideoSender.getParameters(); |
| if (parameters.encodings.size() == 0) { |
| Log.w(TAG, "RtpParameters are not ready."); |
| return; |
| } |
| |
| for (RtpParameters.Encoding encoding : parameters.encodings) { |
| // Null value means no limit. |
| encoding.maxBitrateBps = maxBitrateKbps == null ? null : maxBitrateKbps * BPS_IN_KBPS; |
| } |
| if (!localVideoSender.setParameters(parameters)) { |
| Log.e(TAG, "RtpSender.setParameters failed."); |
| } |
| Log.d(TAG, "Configured max video bitrate to: " + maxBitrateKbps); |
| }); |
| } |
| |
| private void reportError(final String errorMessage) { |
| Log.e(TAG, "Peerconnection error: " + errorMessage); |
| executor.execute(() -> { |
| if (!isError) { |
| events.onPeerConnectionError(errorMessage); |
| isError = true; |
| } |
| }); |
| } |
| |
| @Nullable |
| private AudioTrack createAudioTrack() { |
| audioSource = factory.createAudioSource(audioConstraints); |
| localAudioTrack = factory.createAudioTrack(AUDIO_TRACK_ID, audioSource); |
| localAudioTrack.setEnabled(enableAudio); |
| return localAudioTrack; |
| } |
| |
| @Nullable |
| private VideoTrack createVideoTrack(VideoCapturer capturer) { |
| surfaceTextureHelper = |
| SurfaceTextureHelper.create("CaptureThread", rootEglBase.getEglBaseContext()); |
| videoSource = factory.createVideoSource(capturer.isScreencast()); |
| capturer.initialize(surfaceTextureHelper, appContext, videoSource.getCapturerObserver()); |
| capturer.startCapture(videoWidth, videoHeight, videoFps); |
| |
| localVideoTrack = factory.createVideoTrack(VIDEO_TRACK_ID, videoSource); |
| localVideoTrack.setEnabled(renderVideo); |
| localVideoTrack.addSink(localRender); |
| return localVideoTrack; |
| } |
| |
| private void findVideoSender() { |
| for (RtpSender sender : peerConnection.getSenders()) { |
| if (sender.track() != null) { |
| String trackType = sender.track().kind(); |
| if (trackType.equals(VIDEO_TRACK_TYPE)) { |
| Log.d(TAG, "Found video sender."); |
| localVideoSender = sender; |
| } |
| } |
| } |
| } |
| |
| // Returns the remote VideoTrack, assuming there is only one. |
| private @Nullable VideoTrack getRemoteVideoTrack() { |
| for (RtpTransceiver transceiver : peerConnection.getTransceivers()) { |
| MediaStreamTrack track = transceiver.getReceiver().track(); |
| if (track instanceof VideoTrack) { |
| return (VideoTrack) track; |
| } |
| } |
| return null; |
| } |
| |
| private static String getSdpVideoCodecName(PeerConnectionParameters parameters) { |
| switch (parameters.videoCodec) { |
| case VIDEO_CODEC_VP8: |
| return VIDEO_CODEC_VP8; |
| case VIDEO_CODEC_VP9: |
| return VIDEO_CODEC_VP9; |
| case VIDEO_CODEC_AV1: |
| return VIDEO_CODEC_AV1_SDP_CODEC_NAME; |
| case VIDEO_CODEC_H264_HIGH: |
| case VIDEO_CODEC_H264_BASELINE: |
| return VIDEO_CODEC_H264; |
| default: |
| return VIDEO_CODEC_VP8; |
| } |
| } |
| |
| private static String getFieldTrials(PeerConnectionParameters peerConnectionParameters) { |
| String fieldTrials = ""; |
| if (peerConnectionParameters.videoFlexfecEnabled) { |
| fieldTrials += VIDEO_FLEXFEC_FIELDTRIAL; |
| Log.d(TAG, "Enable FlexFEC field trial."); |
| } |
| fieldTrials += VIDEO_VP8_INTEL_HW_ENCODER_FIELDTRIAL; |
| if (peerConnectionParameters.disableWebRtcAGCAndHPF) { |
| fieldTrials += DISABLE_WEBRTC_AGC_FIELDTRIAL; |
| Log.d(TAG, "Disable WebRTC AGC field trial."); |
| } |
| return fieldTrials; |
| } |
| |
| @SuppressWarnings("StringSplitter") |
| private static String setStartBitrate( |
| String codec, boolean isVideoCodec, String sdp, int bitrateKbps) { |
| String[] lines = sdp.split("\r\n"); |
| int rtpmapLineIndex = -1; |
| boolean sdpFormatUpdated = false; |
| String codecRtpMap = null; |
| // Search for codec rtpmap in format |
| // a=rtpmap:<payload type> <encoding name>/<clock rate> [/<encoding parameters>] |
| String regex = "^a=rtpmap:(\\d+) " + codec + "(/\\d+)+[\r]?$"; |
| Pattern codecPattern = Pattern.compile(regex); |
| for (int i = 0; i < lines.length; i++) { |
| Matcher codecMatcher = codecPattern.matcher(lines[i]); |
| if (codecMatcher.matches()) { |
| codecRtpMap = codecMatcher.group(1); |
| rtpmapLineIndex = i; |
| break; |
| } |
| } |
| if (codecRtpMap == null) { |
| Log.w(TAG, "No rtpmap for " + codec + " codec"); |
| return sdp; |
| } |
| Log.d(TAG, "Found " + codec + " rtpmap " + codecRtpMap + " at " + lines[rtpmapLineIndex]); |
| |
| // Check if a=fmtp string already exist in remote SDP for this codec and |
| // update it with new bitrate parameter. |
| regex = "^a=fmtp:" + codecRtpMap + " \\w+=\\d+.*[\r]?$"; |
| codecPattern = Pattern.compile(regex); |
| for (int i = 0; i < lines.length; i++) { |
| Matcher codecMatcher = codecPattern.matcher(lines[i]); |
| if (codecMatcher.matches()) { |
| Log.d(TAG, "Found " + codec + " " + lines[i]); |
| if (isVideoCodec) { |
| lines[i] += "; " + VIDEO_CODEC_PARAM_START_BITRATE + "=" + bitrateKbps; |
| } else { |
| lines[i] += "; " + AUDIO_CODEC_PARAM_BITRATE + "=" + (bitrateKbps * 1000); |
| } |
| Log.d(TAG, "Update remote SDP line: " + lines[i]); |
| sdpFormatUpdated = true; |
| break; |
| } |
| } |
| |
| StringBuilder newSdpDescription = new StringBuilder(); |
| for (int i = 0; i < lines.length; i++) { |
| newSdpDescription.append(lines[i]).append("\r\n"); |
| // Append new a=fmtp line if no such line exist for a codec. |
| if (!sdpFormatUpdated && i == rtpmapLineIndex) { |
| String bitrateSet; |
| if (isVideoCodec) { |
| bitrateSet = |
| "a=fmtp:" + codecRtpMap + " " + VIDEO_CODEC_PARAM_START_BITRATE + "=" + bitrateKbps; |
| } else { |
| bitrateSet = "a=fmtp:" + codecRtpMap + " " + AUDIO_CODEC_PARAM_BITRATE + "=" |
| + (bitrateKbps * 1000); |
| } |
| Log.d(TAG, "Add remote SDP line: " + bitrateSet); |
| newSdpDescription.append(bitrateSet).append("\r\n"); |
| } |
| } |
| return newSdpDescription.toString(); |
| } |
| |
| /** Returns the line number containing "m=audio|video", or -1 if no such line exists. */ |
| private static int findMediaDescriptionLine(boolean isAudio, String[] sdpLines) { |
| final String mediaDescription = isAudio ? "m=audio " : "m=video "; |
| for (int i = 0; i < sdpLines.length; ++i) { |
| if (sdpLines[i].startsWith(mediaDescription)) { |
| return i; |
| } |
| } |
| return -1; |
| } |
| |
| private static String joinString( |
| Iterable<? extends CharSequence> s, String delimiter, boolean delimiterAtEnd) { |
| Iterator<? extends CharSequence> iter = s.iterator(); |
| if (!iter.hasNext()) { |
| return ""; |
| } |
| StringBuilder buffer = new StringBuilder(iter.next()); |
| while (iter.hasNext()) { |
| buffer.append(delimiter).append(iter.next()); |
| } |
| if (delimiterAtEnd) { |
| buffer.append(delimiter); |
| } |
| return buffer.toString(); |
| } |
| |
| private static @Nullable String movePayloadTypesToFront( |
| List<String> preferredPayloadTypes, String mLine) { |
| // The format of the media description line should be: m=<media> <port> <proto> <fmt> ... |
| final List<String> origLineParts = Arrays.asList(mLine.split(" ")); |
| if (origLineParts.size() <= 3) { |
| Log.e(TAG, "Wrong SDP media description format: " + mLine); |
| return null; |
| } |
| final List<String> header = origLineParts.subList(0, 3); |
| final List<String> unpreferredPayloadTypes = |
| new ArrayList<>(origLineParts.subList(3, origLineParts.size())); |
| unpreferredPayloadTypes.removeAll(preferredPayloadTypes); |
| // Reconstruct the line with `preferredPayloadTypes` moved to the beginning of the payload |
| // types. |
| final List<String> newLineParts = new ArrayList<>(); |
| newLineParts.addAll(header); |
| newLineParts.addAll(preferredPayloadTypes); |
| newLineParts.addAll(unpreferredPayloadTypes); |
| return joinString(newLineParts, " ", false /* delimiterAtEnd */); |
| } |
| |
| private static String preferCodec(String sdp, String codec, boolean isAudio) { |
| final String[] lines = sdp.split("\r\n"); |
| final int mLineIndex = findMediaDescriptionLine(isAudio, lines); |
| if (mLineIndex == -1) { |
| Log.w(TAG, "No mediaDescription line, so can't prefer " + codec); |
| return sdp; |
| } |
| // A list with all the payload types with name `codec`. The payload types are integers in the |
| // range 96-127, but they are stored as strings here. |
| final List<String> codecPayloadTypes = new ArrayList<>(); |
| // a=rtpmap:<payload type> <encoding name>/<clock rate> [/<encoding parameters>] |
| final Pattern codecPattern = Pattern.compile("^a=rtpmap:(\\d+) " + codec + "(/\\d+)+[\r]?$"); |
| for (String line : lines) { |
| Matcher codecMatcher = codecPattern.matcher(line); |
| if (codecMatcher.matches()) { |
| codecPayloadTypes.add(codecMatcher.group(1)); |
| } |
| } |
| if (codecPayloadTypes.isEmpty()) { |
| Log.w(TAG, "No payload types with name " + codec); |
| return sdp; |
| } |
| |
| final String newMLine = movePayloadTypesToFront(codecPayloadTypes, lines[mLineIndex]); |
| if (newMLine == null) { |
| return sdp; |
| } |
| Log.d(TAG, "Change media description from: " + lines[mLineIndex] + " to " + newMLine); |
| lines[mLineIndex] = newMLine; |
| return joinString(Arrays.asList(lines), "\r\n", true /* delimiterAtEnd */); |
| } |
| |
| private void drainCandidates() { |
| if (queuedRemoteCandidates != null) { |
| Log.d(TAG, "Add " + queuedRemoteCandidates.size() + " remote candidates"); |
| for (IceCandidate candidate : queuedRemoteCandidates) { |
| peerConnection.addIceCandidate(candidate, new AddIceObserver() { |
| @Override |
| public void onAddSuccess() { |
| Log.d(TAG, "Candidate " + candidate + " successfully added."); |
| } |
| @Override |
| public void onAddFailure(String error) { |
| Log.d(TAG, "Candidate " + candidate + " addition failed: " + error); |
| } |
| }); |
| } |
| queuedRemoteCandidates = null; |
| } |
| } |
| |
| private void switchCameraInternal() { |
| if (videoCapturer instanceof CameraVideoCapturer) { |
| if (!isVideoCallEnabled() || isError) { |
| Log.e(TAG, |
| "Failed to switch camera. Video: " + isVideoCallEnabled() + ". Error : " + isError); |
| return; // No video is sent or only one camera is available or error happened. |
| } |
| Log.d(TAG, "Switch camera"); |
| CameraVideoCapturer cameraVideoCapturer = (CameraVideoCapturer) videoCapturer; |
| cameraVideoCapturer.switchCamera(null); |
| } else { |
| Log.d(TAG, "Will not switch camera, video caputurer is not a camera"); |
| } |
| } |
| |
| public void switchCamera() { |
| executor.execute(this ::switchCameraInternal); |
| } |
| |
| public void changeCaptureFormat(final int width, final int height, final int framerate) { |
| executor.execute(() -> changeCaptureFormatInternal(width, height, framerate)); |
| } |
| |
| private void changeCaptureFormatInternal(int width, int height, int framerate) { |
| if (!isVideoCallEnabled() || isError || videoCapturer == null) { |
| Log.e(TAG, |
| "Failed to change capture format. Video: " + isVideoCallEnabled() |
| + ". Error : " + isError); |
| return; |
| } |
| Log.d(TAG, "changeCaptureFormat: " + width + "x" + height + "@" + framerate); |
| videoSource.adaptOutputFormat(width, height, framerate); |
| } |
| |
| // Implementation detail: observe ICE & stream changes and react accordingly. |
| private class PCObserver implements PeerConnection.Observer { |
| @Override |
| public void onIceCandidate(final IceCandidate candidate) { |
| executor.execute(() -> events.onIceCandidate(candidate)); |
| } |
| |
| @Override |
| public void onIceCandidatesRemoved(final IceCandidate[] candidates) { |
| executor.execute(() -> events.onIceCandidatesRemoved(candidates)); |
| } |
| |
| @Override |
| public void onSignalingChange(PeerConnection.SignalingState newState) { |
| Log.d(TAG, "SignalingState: " + newState); |
| } |
| |
| @Override |
| public void onIceConnectionChange(final PeerConnection.IceConnectionState newState) { |
| executor.execute(() -> { |
| Log.d(TAG, "IceConnectionState: " + newState); |
| if (newState == IceConnectionState.CONNECTED) { |
| events.onIceConnected(); |
| } else if (newState == IceConnectionState.DISCONNECTED) { |
| events.onIceDisconnected(); |
| } else if (newState == IceConnectionState.FAILED) { |
| reportError("ICE connection failed."); |
| } |
| }); |
| } |
| |
| @Override |
| public void onConnectionChange(final PeerConnection.PeerConnectionState newState) { |
| executor.execute(() -> { |
| Log.d(TAG, "PeerConnectionState: " + newState); |
| if (newState == PeerConnectionState.CONNECTED) { |
| events.onConnected(); |
| } else if (newState == PeerConnectionState.DISCONNECTED) { |
| events.onDisconnected(); |
| } else if (newState == PeerConnectionState.FAILED) { |
| reportError("DTLS connection failed."); |
| } |
| }); |
| } |
| |
| @Override |
| public void onIceGatheringChange(PeerConnection.IceGatheringState newState) { |
| Log.d(TAG, "IceGatheringState: " + newState); |
| } |
| |
| @Override |
| public void onIceConnectionReceivingChange(boolean receiving) { |
| Log.d(TAG, "IceConnectionReceiving changed to " + receiving); |
| } |
| |
| @Override |
| public void onSelectedCandidatePairChanged(CandidatePairChangeEvent event) { |
| Log.d(TAG, "Selected candidate pair changed because: " + event); |
| } |
| |
| @Override |
| public void onAddStream(final MediaStream stream) {} |
| |
| @Override |
| public void onRemoveStream(final MediaStream stream) {} |
| |
| @Override |
| public void onDataChannel(final DataChannel dc) { |
| Log.d(TAG, "New Data channel " + dc.label()); |
| |
| if (!dataChannelEnabled) |
| return; |
| |
| dc.registerObserver(new DataChannel.Observer() { |
| @Override |
| public void onBufferedAmountChange(long previousAmount) { |
| Log.d(TAG, "Data channel buffered amount changed: " + dc.label() + ": " + dc.state()); |
| } |
| |
| @Override |
| public void onStateChange() { |
| Log.d(TAG, "Data channel state changed: " + dc.label() + ": " + dc.state()); |
| } |
| |
| @Override |
| public void onMessage(final DataChannel.Buffer buffer) { |
| if (buffer.binary) { |
| Log.d(TAG, "Received binary msg over " + dc); |
| return; |
| } |
| ByteBuffer data = buffer.data; |
| final byte[] bytes = new byte[data.capacity()]; |
| data.get(bytes); |
| String strData = new String(bytes, Charset.forName("UTF-8")); |
| Log.d(TAG, "Got msg: " + strData + " over " + dc); |
| } |
| }); |
| } |
| |
| @Override |
| public void onRenegotiationNeeded() { |
| // No need to do anything; AppRTC follows a pre-agreed-upon |
| // signaling/negotiation protocol. |
| } |
| |
| @Override |
| public void onAddTrack(final RtpReceiver receiver, final MediaStream[] mediaStreams) {} |
| |
| @Override |
| public void onRemoveTrack(final RtpReceiver receiver) {} |
| } |
| |
| // Implementation detail: handle offer creation/signaling and answer setting, |
| // as well as adding remote ICE candidates once the answer SDP is set. |
| private class SDPObserver implements SdpObserver { |
| @Override |
| public void onCreateSuccess(final SessionDescription desc) { |
| if (localDescription != null) { |
| reportError("Multiple SDP create."); |
| return; |
| } |
| String sdp = desc.description; |
| if (preferIsac) { |
| sdp = preferCodec(sdp, AUDIO_CODEC_ISAC, true); |
| } |
| if (isVideoCallEnabled()) { |
| sdp = preferCodec(sdp, getSdpVideoCodecName(peerConnectionParameters), false); |
| } |
| final SessionDescription newDesc = new SessionDescription(desc.type, sdp); |
| localDescription = newDesc; |
| executor.execute(() -> { |
| if (peerConnection != null && !isError) { |
| Log.d(TAG, "Set local SDP from " + desc.type); |
| peerConnection.setLocalDescription(sdpObserver, newDesc); |
| } |
| }); |
| } |
| |
| @Override |
| public void onSetSuccess() { |
| executor.execute(() -> { |
| if (peerConnection == null || isError) { |
| return; |
| } |
| if (isInitiator) { |
| // For offering peer connection we first create offer and set |
| // local SDP, then after receiving answer set remote SDP. |
| if (peerConnection.getRemoteDescription() == null) { |
| // We've just set our local SDP so time to send it. |
| Log.d(TAG, "Local SDP set succesfully"); |
| events.onLocalDescription(localDescription); |
| } else { |
| // We've just set remote description, so drain remote |
| // and send local ICE candidates. |
| Log.d(TAG, "Remote SDP set succesfully"); |
| drainCandidates(); |
| } |
| } else { |
| // For answering peer connection we set remote SDP and then |
| // create answer and set local SDP. |
| if (peerConnection.getLocalDescription() != null) { |
| // We've just set our local SDP so time to send it, drain |
| // remote and send local ICE candidates. |
| Log.d(TAG, "Local SDP set succesfully"); |
| events.onLocalDescription(localDescription); |
| drainCandidates(); |
| } else { |
| // We've just set remote SDP - do nothing for now - |
| // answer will be created soon. |
| Log.d(TAG, "Remote SDP set succesfully"); |
| } |
| } |
| }); |
| } |
| |
| @Override |
| public void onCreateFailure(final String error) { |
| reportError("createSDP error: " + error); |
| } |
| |
| @Override |
| public void onSetFailure(final String error) { |
| reportError("setSDP error: " + error); |
| } |
| } |
| } |
