diff options
| author | Torne (Richard Coles) <torne@google.com> | 2014-11-06 11:12:37 +0000 |
|---|---|---|
| committer | Torne (Richard Coles) <torne@google.com> | 2014-11-06 11:12:37 +0000 |
| commit | ce708afc0470244d9ec2e52b28564c62a0f6cd01 (patch) | |
| tree | 426c61ded5c82b636ab13c887584ccbde01b5910 | |
| parent | b50a456d795434286a7555cefbc851360238716e (diff) | |
| parent | c55e14ee6a9f788b3f4967bcd44bcf2eab26f1df (diff) | |
| download | talk-ce708afc0470244d9ec2e52b28564c62a0f6cd01.tar.gz | |
Merge from Chromium at DEPS revision db3f05efe0f9
This commit was generated by merge_to_master.py.
Change-Id: Ied615f07e5617f89603f29f94846edd58c32a3a9
339 files changed, 3493 insertions, 2335 deletions
diff --git a/PRESUBMIT.py b/PRESUBMIT.py index a7b4851..9e7b5da 100644 --- a/PRESUBMIT.py +++ b/PRESUBMIT.py @@ -23,9 +23,6 @@ # OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF # ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. -# List of files that should not be committed to -DO_NOT_SUBMIT_FILES = ["talk/media/webrtc/webrtcvideoengine.cc"] - def _LicenseHeader(input_api): """Returns the license header regexp.""" # Accept any year number from start of project to the current year @@ -79,26 +76,11 @@ def _LicenseHeader(input_api): } return license_header -def _ProtectedFiles(input_api, output_api): - results = [] - changed_files = [] - for f in input_api.AffectedFiles(): - changed_files.append(f.LocalPath()) - bad_files = list(set(DO_NOT_SUBMIT_FILES) & set(changed_files)) - if bad_files: - error_type = output_api.PresubmitError - results.append(error_type( - 'The following affected files are only allowed to be updated when ' - 'importing libjingle', - bad_files)) - return results - def _CommonChecks(input_api, output_api): """Checks common to both upload and commit.""" results = [] results.extend(input_api.canned_checks.CheckLicense( input_api, output_api, _LicenseHeader(input_api))) - results.extend(_ProtectedFiles(input_api, output_api)) return results def CheckChangeOnUpload(input_api, output_api): diff --git a/app/webrtc/fakeportallocatorfactory.h b/app/webrtc/fakeportallocatorfactory.h index eee98b0..bfdc56b 100644 --- a/app/webrtc/fakeportallocatorfactory.h +++ b/app/webrtc/fakeportallocatorfactory.h @@ -32,7 +32,7 @@ #define TALK_APP_WEBRTC_FAKEPORTALLOCATORFACTORY_H_ #include "talk/app/webrtc/peerconnectioninterface.h" -#include "talk/p2p/client/fakeportallocator.h" +#include "webrtc/p2p/client/fakeportallocator.h" namespace webrtc { diff --git a/app/webrtc/java/android/org/webrtc/VideoRendererGui.java b/app/webrtc/java/android/org/webrtc/VideoRendererGui.java index e0542ed..dd1692c 100644 --- a/app/webrtc/java/android/org/webrtc/VideoRendererGui.java +++ b/app/webrtc/java/android/org/webrtc/VideoRendererGui.java @@ -80,6 +80,11 @@ public class VideoRendererGui implements GLSurfaceView.Renderer { // aspect ratio is changed if necessary. public static enum ScalingType { SCALE_ASPECT_FIT, SCALE_ASPECT_FILL, SCALE_FILL }; + private static final int EGL14_SDK_VERSION = + android.os.Build.VERSION_CODES.JELLY_BEAN_MR1; + // Current SDK version. + private static final int CURRENT_SDK_VERSION = + android.os.Build.VERSION.SDK_INT; private final String VERTEX_SHADER_STRING = "varying vec2 interp_tc;\n" + @@ -248,6 +253,8 @@ public class VideoRendererGui implements GLSurfaceView.Renderer { private FloatBuffer textureCoords; // Flag if texture vertices or coordinates update is needed. private boolean updateTextureProperties; + // Texture properties update lock. + private final Object updateTextureLock = new Object(); // Viewport dimensions. private int screenWidth; private int screenHeight; @@ -314,36 +321,49 @@ public class VideoRendererGui implements GLSurfaceView.Renderer { scalingType == ScalingType.SCALE_FILL) { return; } - // Re - calculate texture vertices to preserve video aspect ratio. - float texRight = this.texRight; - float texLeft = this.texLeft; - float texTop = this.texTop; - float texBottom = this.texBottom; - float displayWidth = (texRight - texLeft) * screenWidth / 2; - float displayHeight = (texTop - texBottom) * screenHeight / 2; - Log.d(TAG, "ID: " + id + ". Display: " + displayWidth + - " x " + displayHeight + ". Video: " + videoWidth + - " x " + videoHeight); - if (displayWidth > 1 && displayHeight > 1 && - videoWidth > 1 && videoHeight > 1) { - float displayAspectRatio = displayWidth / displayHeight; - float videoAspectRatio = (float)videoWidth / videoHeight; - if (scalingType == ScalingType.SCALE_ASPECT_FIT) { - // Need to re-adjust vertices width or height to match video AR. - if (displayAspectRatio > videoAspectRatio) { - float deltaX = (displayWidth - videoAspectRatio * displayHeight) / - instance.screenWidth; - texRight -= deltaX; - texLeft += deltaX; - } else { - float deltaY = (displayHeight - displayWidth / videoAspectRatio) / - instance.screenHeight; - texTop -= deltaY; - texBottom += deltaY; + synchronized(updateTextureLock) { + // Re - calculate texture vertices to preserve video aspect ratio. + float texRight = this.texRight; + float texLeft = this.texLeft; + float texTop = this.texTop; + float texBottom = this.texBottom; + float texOffsetU = 0; + float texOffsetV = 0; + float displayWidth = (texRight - texLeft) * screenWidth / 2; + float displayHeight = (texTop - texBottom) * screenHeight / 2; + Log.d(TAG, "ID: " + id + ". Display: " + displayWidth + + " x " + displayHeight + ". Video: " + videoWidth + + " x " + videoHeight); + if (displayWidth > 1 && displayHeight > 1 && + videoWidth > 1 && videoHeight > 1) { + float displayAspectRatio = displayWidth / displayHeight; + float videoAspectRatio = (float)videoWidth / videoHeight; + if (scalingType == ScalingType.SCALE_ASPECT_FIT) { + // Need to re-adjust vertices width or height to match video AR. + if (displayAspectRatio > videoAspectRatio) { + float deltaX = (displayWidth - videoAspectRatio * displayHeight) / + instance.screenWidth; + texRight -= deltaX; + texLeft += deltaX; + } else { + float deltaY = (displayHeight - displayWidth / videoAspectRatio) / + instance.screenHeight; + texTop -= deltaY; + texBottom += deltaY; + } + } + if (scalingType == ScalingType.SCALE_ASPECT_FILL) { + // Need to re-adjust UV coordinates to match display AR. + if (displayAspectRatio > videoAspectRatio) { + texOffsetV = (1.0f - videoAspectRatio / displayAspectRatio) / + 2.0f; + } else { + texOffsetU = (1.0f - displayAspectRatio / videoAspectRatio) / + 2.0f; + } } Log.d(TAG, " Texture vertices: (" + texLeft + "," + texBottom + ") - (" + texRight + "," + texTop + ")"); - // Re-allocate vertices buffer to adjust to video aspect ratio. float textureVeticesFloat[] = new float[] { texLeft, texTop, texLeft, texBottom, @@ -351,18 +371,8 @@ public class VideoRendererGui implements GLSurfaceView.Renderer { texRight, texBottom }; textureVertices = directNativeFloatBuffer(textureVeticesFloat); - } - if (scalingType == ScalingType.SCALE_ASPECT_FILL) { - float texOffsetU = 0; - float texOffsetV = 0; - // Need to re-adjust UV coordinates to match display AR. - if (displayAspectRatio > videoAspectRatio) { - texOffsetV = (1.0f - videoAspectRatio / displayAspectRatio) / 2.0f; - } else { - texOffsetU = (1.0f - displayAspectRatio / videoAspectRatio) / 2.0f; - } + Log.d(TAG, " Texture UV offsets: " + texOffsetU + ", " + texOffsetV); - // Re-allocate coordinates buffer to adjust to display aspect ratio. float textureCoordinatesFloat[] = new float[] { texOffsetU, texOffsetV, // left top texOffsetU, 1.0f - texOffsetV, // left bottom @@ -371,8 +381,8 @@ public class VideoRendererGui implements GLSurfaceView.Renderer { }; textureCoords = directNativeFloatBuffer(textureCoordinatesFloat); } + updateTextureProperties = false; } - updateTextureProperties = false; } private void draw() { @@ -485,18 +495,23 @@ public class VideoRendererGui implements GLSurfaceView.Renderer { } public void setScreenSize(final int screenWidth, final int screenHeight) { - this.screenWidth = screenWidth; - this.screenHeight = screenHeight; - updateTextureProperties = true; + synchronized(updateTextureLock) { + this.screenWidth = screenWidth; + this.screenHeight = screenHeight; + updateTextureProperties = true; + } } public void setPosition(int x, int y, int width, int height, ScalingType scalingType) { - texLeft = (x - 50) / 50.0f; - texTop = (50 - y) / 50.0f; - texRight = Math.min(1.0f, (x + width - 50) / 50.0f); - texBottom = Math.max(-1.0f, (50 - y - height) / 50.0f); - updateTextureProperties = true; + synchronized(updateTextureLock) { + texLeft = (x - 50) / 50.0f; + texTop = (50 - y) / 50.0f; + texRight = Math.min(1.0f, (x + width - 50) / 50.0f); + texBottom = Math.max(-1.0f, (50 - y - height) / 50.0f); + this.scalingType = scalingType; + updateTextureProperties = true; + } } @Override @@ -666,8 +681,10 @@ public class VideoRendererGui implements GLSurfaceView.Renderer { public void onSurfaceCreated(GL10 unused, EGLConfig config) { Log.d(TAG, "VideoRendererGui.onSurfaceCreated"); // Store render EGL context - eglContext = EGL14.eglGetCurrentContext(); - Log.d(TAG, "VideoRendererGui EGL Context: " + eglContext); + if (CURRENT_SDK_VERSION >= EGL14_SDK_VERSION) { + eglContext = EGL14.eglGetCurrentContext(); + Log.d(TAG, "VideoRendererGui EGL Context: " + eglContext); + } // Create YUV and OES programs. yuvProgram = createProgram(VERTEX_SHADER_STRING, @@ -683,7 +700,7 @@ public class VideoRendererGui implements GLSurfaceView.Renderer { onSurfaceCreatedCalled = true; } checkNoGLES2Error(); - GLES20.glClearColor(0.0f, 0.0f, 0.1f, 1.0f); + GLES20.glClearColor(0.15f, 0.15f, 0.15f, 1.0f); } @Override diff --git a/app/webrtc/java/jni/peerconnection_jni.cc b/app/webrtc/java/jni/peerconnection_jni.cc index d2b1639..fb36cf8 100644 --- a/app/webrtc/java/jni/peerconnection_jni.cc +++ b/app/webrtc/java/jni/peerconnection_jni.cc @@ -273,11 +273,19 @@ class ClassReferenceHolder { LoadClass(jni, "org/webrtc/IceCandidate"); #if defined(ANDROID) && !defined(WEBRTC_CHROMIUM_BUILD) LoadClass(jni, "android/graphics/SurfaceTexture"); - LoadClass(jni, "android/opengl/EGLContext"); LoadClass(jni, "org/webrtc/MediaCodecVideoEncoder"); LoadClass(jni, "org/webrtc/MediaCodecVideoEncoder$OutputBufferInfo"); LoadClass(jni, "org/webrtc/MediaCodecVideoDecoder"); LoadClass(jni, "org/webrtc/MediaCodecVideoDecoder$DecoderOutputBufferInfo"); + jclass j_decoder_class = GetClass("org/webrtc/MediaCodecVideoDecoder"); + jmethodID j_is_egl14_supported_method = jni->GetStaticMethodID( + j_decoder_class, "isEGL14Supported", "()Z"); + bool is_egl14_supported = jni->CallStaticBooleanMethod( + j_decoder_class, j_is_egl14_supported_method); + CHECK_EXCEPTION(jni); + if (is_egl14_supported) { + LoadClass(jni, "android/opengl/EGLContext"); + } #endif LoadClass(jni, "org/webrtc/MediaSource$State"); LoadClass(jni, "org/webrtc/MediaStream"); @@ -578,13 +586,6 @@ class PCOJava : public PeerConnectionObserver { CHECK_EXCEPTION(jni()) << "error during CallVoidMethod"; } - virtual void OnError() OVERRIDE { - ScopedLocalRefFrame local_ref_frame(jni()); - jmethodID m = GetMethodID(jni(), *j_observer_class_, "onError", "()V"); - jni()->CallVoidMethod(*j_observer_global_, m); - CHECK_EXCEPTION(jni()) << "error during CallVoidMethod"; - } - virtual void OnSignalingChange( PeerConnectionInterface::SignalingState new_state) OVERRIDE { ScopedLocalRefFrame local_ref_frame(jni()); @@ -3136,12 +3137,9 @@ JOW(jboolean, PeerConnection_nativeAddIceCandidate)( } JOW(jboolean, PeerConnection_nativeAddLocalStream)( - JNIEnv* jni, jobject j_pc, jlong native_stream, jobject j_constraints) { - scoped_ptr<ConstraintsWrapper> constraints( - new ConstraintsWrapper(jni, j_constraints)); + JNIEnv* jni, jobject j_pc, jlong native_stream) { return ExtractNativePC(jni, j_pc)->AddStream( - reinterpret_cast<MediaStreamInterface*>(native_stream), - constraints.get()); + reinterpret_cast<MediaStreamInterface*>(native_stream)); } JOW(void, PeerConnection_nativeRemoveLocalStream)( diff --git a/app/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java b/app/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java index 96d8b1b..f93f02a 100644 --- a/app/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java +++ b/app/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java @@ -95,6 +95,10 @@ class MediaCodecVideoDecoder { private EGLDisplay eglDisplay = EGL14.EGL_NO_DISPLAY; private EGLContext eglContext = EGL14.EGL_NO_CONTEXT; private EGLSurface eglSurface = EGL14.EGL_NO_SURFACE; + private static final int EGL14_SDK_VERSION = + android.os.Build.VERSION_CODES.JELLY_BEAN_MR1; + private static final int CURRENT_SDK_VERSION = + android.os.Build.VERSION.SDK_INT; private MediaCodecVideoDecoder() { } @@ -166,6 +170,11 @@ class MediaCodecVideoDecoder { return null; // No HW VP8 decoder. } + private static boolean isEGL14Supported() { + Log.d(TAG, "SDK version: " + CURRENT_SDK_VERSION); + return (CURRENT_SDK_VERSION >= EGL14_SDK_VERSION); + } + private static boolean isPlatformSupported() { return findVp8Decoder(false) != null; } diff --git a/app/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java b/app/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java index a2d6168..4c02891 100644 --- a/app/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java +++ b/app/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java @@ -43,7 +43,7 @@ import java.nio.ByteBuffer; // This class is an implementation detail of the Java PeerConnection API. // MediaCodec is thread-hostile so this class must be operated on a single // thread. -class MediaCodecVideoEncoder { +public class MediaCodecVideoEncoder { // This class is constructed, operated, and destroyed by its C++ incarnation, // so the class and its methods have non-public visibility. The API this // class exposes aims to mimic the webrtc::VideoEncoder API as closely as @@ -140,7 +140,7 @@ class MediaCodecVideoEncoder { return null; // No HW VP8 encoder. } - private static boolean isPlatformSupported() { + public static boolean isPlatformSupported() { return findVp8HwEncoder() != null; } diff --git a/app/webrtc/java/src/org/webrtc/PeerConnection.java b/app/webrtc/java/src/org/webrtc/PeerConnection.java index c2617de..3aef6ff 100644 --- a/app/webrtc/java/src/org/webrtc/PeerConnection.java +++ b/app/webrtc/java/src/org/webrtc/PeerConnection.java @@ -71,9 +71,6 @@ public class PeerConnection { /** Triggered when a new ICE candidate has been found. */ public void onIceCandidate(IceCandidate candidate); - /** Triggered on any error. */ - public void onError(); - /** Triggered when media is received on a new stream from remote peer. */ public void onAddStream(MediaStream stream); @@ -147,9 +144,8 @@ public class PeerConnection { candidate.sdpMid, candidate.sdpMLineIndex, candidate.sdp); } - public boolean addStream( - MediaStream stream, MediaConstraints constraints) { - boolean ret = nativeAddLocalStream(stream.nativeStream, constraints); + public boolean addStream(MediaStream stream) { + boolean ret = nativeAddLocalStream(stream.nativeStream); if (!ret) { return false; } @@ -194,8 +190,7 @@ public class PeerConnection { private native boolean nativeAddIceCandidate( String sdpMid, int sdpMLineIndex, String iceCandidateSdp); - private native boolean nativeAddLocalStream( - long nativeStream, MediaConstraints constraints); + private native boolean nativeAddLocalStream(long nativeStream); private native void nativeRemoveLocalStream(long nativeStream); diff --git a/app/webrtc/javatests/src/org/webrtc/PeerConnectionTest.java b/app/webrtc/javatests/src/org/webrtc/PeerConnectionTest.java index 240e996..048d92b 100644 --- a/app/webrtc/javatests/src/org/webrtc/PeerConnectionTest.java +++ b/app/webrtc/javatests/src/org/webrtc/PeerConnectionTest.java @@ -117,11 +117,6 @@ public class PeerConnectionTest extends TestCase { ++expectedErrors; } - @Override - public synchronized void onError() { - assertTrue(--expectedErrors >= 0); - } - public synchronized void expectSetSize() { if (RENDER_TO_GUI) { // When new frames are delivered to the GUI renderer we don't get @@ -489,7 +484,7 @@ public class PeerConnectionTest extends TestCase { lMS.addTrack(videoTrack); lMS.addTrack(factory.createAudioTrack( audioTrackId, factory.createAudioSource(new MediaConstraints()))); - pc.addStream(lMS, new MediaConstraints()); + pc.addStream(lMS); return new WeakReference<MediaStream>(lMS); } diff --git a/app/webrtc/jsepicecandidate.h b/app/webrtc/jsepicecandidate.h index 7be420c..71ed2c3 100644 --- a/app/webrtc/jsepicecandidate.h +++ b/app/webrtc/jsepicecandidate.h @@ -33,7 +33,7 @@ #include <string> #include "talk/app/webrtc/jsep.h" -#include "talk/p2p/base/candidate.h" +#include "webrtc/p2p/base/candidate.h" #include "webrtc/base/constructormagic.h" namespace webrtc { diff --git a/app/webrtc/jsepsessiondescription_unittest.cc b/app/webrtc/jsepsessiondescription_unittest.cc index 769e34a..cf992c0 100644 --- a/app/webrtc/jsepsessiondescription_unittest.cc +++ b/app/webrtc/jsepsessiondescription_unittest.cc @@ -29,9 +29,9 @@ #include "talk/app/webrtc/jsepicecandidate.h" #include "talk/app/webrtc/jsepsessiondescription.h" -#include "talk/p2p/base/candidate.h" -#include "talk/p2p/base/constants.h" -#include "talk/p2p/base/sessiondescription.h" +#include "webrtc/p2p/base/candidate.h" +#include "webrtc/p2p/base/constants.h" +#include "webrtc/p2p/base/sessiondescription.h" #include "talk/session/media/mediasession.h" #include "webrtc/base/gunit.h" #include "webrtc/base/helpers.h" diff --git a/app/webrtc/mediastreamsignaling_unittest.cc b/app/webrtc/mediastreamsignaling_unittest.cc index 038c67d..d7b9734 100644 --- a/app/webrtc/mediastreamsignaling_unittest.cc +++ b/app/webrtc/mediastreamsignaling_unittest.cc @@ -38,8 +38,8 @@ #include "talk/app/webrtc/videotrack.h" #include "talk/media/base/fakemediaengine.h" #include "talk/media/devices/fakedevicemanager.h" -#include "talk/p2p/base/constants.h" -#include "talk/p2p/base/sessiondescription.h" +#include "webrtc/p2p/base/constants.h" +#include "webrtc/p2p/base/sessiondescription.h" #include "talk/session/media/channelmanager.h" #include "webrtc/base/gunit.h" #include "webrtc/base/scoped_ptr.h" diff --git a/app/webrtc/objc/RTCPeerConnection.mm b/app/webrtc/objc/RTCPeerConnection.mm index 72ba373..925de73 100644 --- a/app/webrtc/objc/RTCPeerConnection.mm +++ b/app/webrtc/objc/RTCPeerConnection.mm @@ -151,10 +151,8 @@ class RTCStatsObserver : public StatsObserver { return self.peerConnection->AddIceCandidate(iceCandidate.get()); } -- (BOOL)addStream:(RTCMediaStream*)stream - constraints:(RTCMediaConstraints*)constraints { - BOOL ret = self.peerConnection->AddStream(stream.mediaStream, - constraints.constraints); +- (BOOL)addStream:(RTCMediaStream*)stream { + BOOL ret = self.peerConnection->AddStream(stream.mediaStream); if (!ret) { return NO; } diff --git a/app/webrtc/objc/RTCPeerConnectionObserver.h b/app/webrtc/objc/RTCPeerConnectionObserver.h index f66b567..8378ff8 100644 --- a/app/webrtc/objc/RTCPeerConnectionObserver.h +++ b/app/webrtc/objc/RTCPeerConnectionObserver.h @@ -41,8 +41,6 @@ class RTCPeerConnectionObserver : public PeerConnectionObserver { RTCPeerConnectionObserver(RTCPeerConnection* peerConnection); virtual ~RTCPeerConnectionObserver(); - virtual void OnError() OVERRIDE; - // Triggered when the SignalingState changed. virtual void OnSignalingChange( PeerConnectionInterface::SignalingState new_state) OVERRIDE; diff --git a/app/webrtc/objc/RTCPeerConnectionObserver.mm b/app/webrtc/objc/RTCPeerConnectionObserver.mm index a0206e5..f4cab7f 100644 --- a/app/webrtc/objc/RTCPeerConnectionObserver.mm +++ b/app/webrtc/objc/RTCPeerConnectionObserver.mm @@ -46,10 +46,6 @@ RTCPeerConnectionObserver::RTCPeerConnectionObserver( RTCPeerConnectionObserver::~RTCPeerConnectionObserver() { } -void RTCPeerConnectionObserver::OnError() { - [_peerConnection.delegate peerConnectionOnError:_peerConnection]; -} - void RTCPeerConnectionObserver::OnSignalingChange( PeerConnectionInterface::SignalingState new_state) { RTCSignalingState state = diff --git a/app/webrtc/objc/public/RTCPeerConnection.h b/app/webrtc/objc/public/RTCPeerConnection.h index 32a9830..6d47f77 100644 --- a/app/webrtc/objc/public/RTCPeerConnection.h +++ b/app/webrtc/objc/public/RTCPeerConnection.h @@ -64,8 +64,7 @@ // Add a new MediaStream to be sent on this PeerConnection. // Note that a SessionDescription negotiation is needed before the // remote peer can receive the stream. -- (BOOL)addStream:(RTCMediaStream *)stream - constraints:(RTCMediaConstraints *)constraints; +- (BOOL)addStream:(RTCMediaStream *)stream; // Remove a MediaStream from this PeerConnection. // Note that a SessionDescription negotiation is need before the diff --git a/app/webrtc/objc/public/RTCPeerConnectionDelegate.h b/app/webrtc/objc/public/RTCPeerConnectionDelegate.h index 4b177d5..ee6ec7a 100644 --- a/app/webrtc/objc/public/RTCPeerConnectionDelegate.h +++ b/app/webrtc/objc/public/RTCPeerConnectionDelegate.h @@ -38,9 +38,6 @@ // implemented to get messages from PeerConnection. @protocol RTCPeerConnectionDelegate<NSObject> -// Triggered when there is an error. -- (void)peerConnectionOnError:(RTCPeerConnection *)peerConnection; - // Triggered when the SignalingState changed. - (void)peerConnection:(RTCPeerConnection *)peerConnection signalingStateChanged:(RTCSignalingState)stateChanged; diff --git a/app/webrtc/objctests/RTCPeerConnectionSyncObserver.m b/app/webrtc/objctests/RTCPeerConnectionSyncObserver.m index c3f898a..fbcf217 100644 --- a/app/webrtc/objctests/RTCPeerConnectionSyncObserver.m +++ b/app/webrtc/objctests/RTCPeerConnectionSyncObserver.m @@ -151,11 +151,6 @@ #pragma mark - RTCPeerConnectionDelegate methods -- (void)peerConnectionOnError:(RTCPeerConnection*)peerConnection { - NSLog(@"RTCPeerConnectionDelegate::onError"); - NSAssert(--_expectedErrors >= 0, @"Unexpected error"); -} - - (void)peerConnection:(RTCPeerConnection*)peerConnection signalingStateChanged:(RTCSignalingState)stateChanged { int expectedState = [self popFirstElementAsInt:_expectedSignalingChanges]; diff --git a/app/webrtc/objctests/RTCPeerConnectionTest.mm b/app/webrtc/objctests/RTCPeerConnectionTest.mm index 909503a..6c5950b 100644 --- a/app/webrtc/objctests/RTCPeerConnectionTest.mm +++ b/app/webrtc/objctests/RTCPeerConnectionTest.mm @@ -89,8 +89,7 @@ [localMediaStream addVideoTrack:videoTrack]; RTCAudioTrack* audioTrack = [factory audioTrackWithID:audioTrackID]; [localMediaStream addAudioTrack:audioTrack]; - RTCMediaConstraints* constraints = [[RTCMediaConstraints alloc] init]; - [pc addStream:localMediaStream constraints:constraints]; + [pc addStream:localMediaStream]; return localMediaStream; } diff --git a/app/webrtc/peerconnection.cc b/app/webrtc/peerconnection.cc index d939f98..b64caf7 100644 --- a/app/webrtc/peerconnection.cc +++ b/app/webrtc/peerconnection.cc @@ -35,7 +35,7 @@ #include "talk/app/webrtc/mediaconstraintsinterface.h" #include "talk/app/webrtc/mediastreamhandler.h" #include "talk/app/webrtc/streamcollection.h" -#include "talk/p2p/client/basicportallocator.h" +#include "webrtc/p2p/client/basicportallocator.h" #include "talk/session/media/channelmanager.h" #include "webrtc/base/logging.h" #include "webrtc/base/stringencode.h" @@ -350,9 +350,10 @@ bool PeerConnection::DoInitialize( // To handle both internal and externally created port allocator, we will // enable BUNDLE here. - int portallocator_flags = cricket::PORTALLOCATOR_ENABLE_BUNDLE | - cricket::PORTALLOCATOR_ENABLE_SHARED_UFRAG | - cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET; + int portallocator_flags = port_allocator_->flags(); + portallocator_flags |= cricket::PORTALLOCATOR_ENABLE_BUNDLE | + cricket::PORTALLOCATOR_ENABLE_SHARED_UFRAG | + cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET; bool value; // If IPv6 flag was specified, we'll not override it by experiment. if (FindConstraint( @@ -403,8 +404,7 @@ PeerConnection::remote_streams() { return mediastream_signaling_->remote_streams(); } -bool PeerConnection::AddStream(MediaStreamInterface* local_stream, - const MediaConstraintsInterface* constraints) { +bool PeerConnection::AddStream(MediaStreamInterface* local_stream) { if (IsClosed()) { return false; } @@ -412,7 +412,6 @@ bool PeerConnection::AddStream(MediaStreamInterface* local_stream, local_stream)) return false; - // TODO(perkj): Implement support for MediaConstraints in AddStream. if (!mediastream_signaling_->AddLocalStream(local_stream)) { return false; } @@ -421,6 +420,11 @@ bool PeerConnection::AddStream(MediaStreamInterface* local_stream, return true; } +bool PeerConnection::AddStream(MediaStreamInterface* local_stream, + const MediaConstraintsInterface* constraints) { + return AddStream(local_stream); +} + void PeerConnection::RemoveStream(MediaStreamInterface* local_stream) { mediastream_signaling_->RemoveLocalStream(local_stream); if (IsClosed()) { diff --git a/app/webrtc/peerconnection.h b/app/webrtc/peerconnection.h index fb03802..355211c 100644 --- a/app/webrtc/peerconnection.h +++ b/app/webrtc/peerconnection.h @@ -65,6 +65,7 @@ class PeerConnection : public PeerConnectionInterface, PeerConnectionObserver* observer); virtual rtc::scoped_refptr<StreamCollectionInterface> local_streams(); virtual rtc::scoped_refptr<StreamCollectionInterface> remote_streams(); + virtual bool AddStream(MediaStreamInterface* local_stream); virtual bool AddStream(MediaStreamInterface* local_stream, const MediaConstraintsInterface* constraints); virtual void RemoveStream(MediaStreamInterface* local_stream); diff --git a/app/webrtc/peerconnection_unittest.cc b/app/webrtc/peerconnection_unittest.cc index da75e9d..c250eea 100644 --- a/app/webrtc/peerconnection_unittest.cc +++ b/app/webrtc/peerconnection_unittest.cc @@ -46,8 +46,8 @@ #include "talk/app/webrtc/test/mockpeerconnectionobservers.h" #include "talk/app/webrtc/videosourceinterface.h" #include "talk/media/webrtc/fakewebrtcvideoengine.h" -#include "talk/p2p/base/constants.h" -#include "talk/p2p/base/sessiondescription.h" +#include "webrtc/p2p/base/constants.h" +#include "webrtc/p2p/base/sessiondescription.h" #include "talk/session/media/mediasession.h" #include "webrtc/base/gunit.h" #include "webrtc/base/scoped_ptr.h" @@ -179,7 +179,7 @@ class PeerConnectionTestClientBase stream->AddTrack(CreateLocalVideoTrack(stream_label)); } - EXPECT_TRUE(peer_connection_->AddStream(stream, NULL)); + EXPECT_TRUE(peer_connection_->AddStream(stream)); } size_t NumberOfLocalMediaStreams() { @@ -426,7 +426,6 @@ class PeerConnectionTestClientBase } // PeerConnectionObserver callbacks. - virtual void OnError() {} virtual void OnMessage(const std::string&) {} virtual void OnSignalingMessage(const std::string& /*msg*/) {} virtual void OnSignalingChange( diff --git a/app/webrtc/peerconnectionendtoend_unittest.cc b/app/webrtc/peerconnectionendtoend_unittest.cc index 4db3fe3..d743182 100644 --- a/app/webrtc/peerconnectionendtoend_unittest.cc +++ b/app/webrtc/peerconnectionendtoend_unittest.cc @@ -320,7 +320,14 @@ TEST_F(PeerConnectionEndToEndTest, CreateDataChannelBeforeNegotiate) { // Verifies that a DataChannel created after the negotiation can transition to // "OPEN" and transfer data. -TEST_F(PeerConnectionEndToEndTest, CreateDataChannelAfterNegotiate) { +#if defined(MEMORY_SANITIZER) +// Fails under MemorySanitizer: +// See https://code.google.com/p/webrtc/issues/detail?id=3980. +#define MAYBE_CreateDataChannelAfterNegotiate DISABLED_CreateDataChannelAfterNegotiate +#else +#define MAYBE_CreateDataChannelAfterNegotiate CreateDataChannelAfterNegotiate +#endif +TEST_F(PeerConnectionEndToEndTest, MAYBE_CreateDataChannelAfterNegotiate) { MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); CreatePcs(); diff --git a/app/webrtc/peerconnectionfactory_unittest.cc b/app/webrtc/peerconnectionfactory_unittest.cc index 5995c46..e687b8b 100644 --- a/app/webrtc/peerconnectionfactory_unittest.cc +++ b/app/webrtc/peerconnectionfactory_unittest.cc @@ -40,6 +40,7 @@ #include "webrtc/base/thread.h" using webrtc::FakeVideoTrackRenderer; +using webrtc::DataChannelInterface; using webrtc::MediaStreamInterface; using webrtc::PeerConnectionFactoryInterface; using webrtc::PeerConnectionInterface; @@ -83,13 +84,13 @@ static const char kTurnIceServerWithIPv6Address[] = class NullPeerConnectionObserver : public PeerConnectionObserver { public: - virtual void OnError() {} virtual void OnMessage(const std::string& msg) {} virtual void OnSignalingMessage(const std::string& msg) {} virtual void OnSignalingChange( PeerConnectionInterface::SignalingState new_state) {} virtual void OnAddStream(MediaStreamInterface* stream) {} virtual void OnRemoveStream(MediaStreamInterface* stream) {} + virtual void OnDataChannel(DataChannelInterface* data_channel) {} virtual void OnRenegotiationNeeded() {} virtual void OnIceConnectionChange( PeerConnectionInterface::IceConnectionState new_state) {} diff --git a/app/webrtc/peerconnectioninterface.h b/app/webrtc/peerconnectioninterface.h index 6ef4847..68b7879 100644 --- a/app/webrtc/peerconnectioninterface.h +++ b/app/webrtc/peerconnectioninterface.h @@ -255,8 +255,15 @@ class PeerConnectionInterface : public rtc::RefCountInterface { // Add a new MediaStream to be sent on this PeerConnection. // Note that a SessionDescription negotiation is needed before the // remote peer can receive the stream. + // TODO(perkj): Make pure virtual once Chrome mocks have implemented. + virtual bool AddStream(MediaStreamInterface* stream) { return false;} + + // Deprecated: + // TODO(perkj): Remove once its not used by Chrome. virtual bool AddStream(MediaStreamInterface* stream, - const MediaConstraintsInterface* constraints) = 0; + const MediaConstraintsInterface* constraints) { + return false; + } // Remove a MediaStream from this PeerConnection. // Note that a SessionDescription negotiation is need before the @@ -344,7 +351,9 @@ class PeerConnectionObserver { kIceState, }; - virtual void OnError() = 0; + // Deprecated. + // TODO(perkj): Remove once its not used by Chrome. + virtual void OnError() {} // Triggered when the SignalingState changed. virtual void OnSignalingChange( @@ -361,8 +370,7 @@ class PeerConnectionObserver { virtual void OnRemoveStream(MediaStreamInterface* stream) = 0; // Triggered when a remote peer open a data channel. - // TODO(perkj): Make pure virtual. - virtual void OnDataChannel(DataChannelInterface* data_channel) {} + virtual void OnDataChannel(DataChannelInterface* data_channel) = 0; // Triggered when renegotiation is needed, for example the ICE has restarted. virtual void OnRenegotiationNeeded() = 0; diff --git a/app/webrtc/peerconnectioninterface_unittest.cc b/app/webrtc/peerconnectioninterface_unittest.cc index bf60673..3be6280 100644 --- a/app/webrtc/peerconnectioninterface_unittest.cc +++ b/app/webrtc/peerconnectioninterface_unittest.cc @@ -132,7 +132,6 @@ class MockPeerConnectionObserver : public PeerConnectionObserver { state_ = pc_->signaling_state(); } } - virtual void OnError() {} virtual void OnSignalingChange( PeerConnectionInterface::SignalingState new_state) { EXPECT_EQ(pc_->signaling_state(), new_state); @@ -320,7 +319,7 @@ class PeerConnectionInterfaceTest : public testing::Test { scoped_refptr<VideoTrackInterface> video_track( pc_factory_->CreateVideoTrack(label + "v0", video_source)); stream->AddTrack(video_track.get()); - EXPECT_TRUE(pc_->AddStream(stream, NULL)); + EXPECT_TRUE(pc_->AddStream(stream)); EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout); observer_.renegotiation_needed_ = false; } @@ -332,7 +331,7 @@ class PeerConnectionInterfaceTest : public testing::Test { scoped_refptr<AudioTrackInterface> audio_track( pc_factory_->CreateAudioTrack(label + "a0", NULL)); stream->AddTrack(audio_track.get()); - EXPECT_TRUE(pc_->AddStream(stream, NULL)); + EXPECT_TRUE(pc_->AddStream(stream)); EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout); observer_.renegotiation_needed_ = false; } @@ -350,7 +349,7 @@ class PeerConnectionInterfaceTest : public testing::Test { scoped_refptr<VideoTrackInterface> video_track( pc_factory_->CreateVideoTrack(video_track_label, NULL)); stream->AddTrack(video_track.get()); - EXPECT_TRUE(pc_->AddStream(stream, NULL)); + EXPECT_TRUE(pc_->AddStream(stream)); EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout); observer_.renegotiation_needed_ = false; } @@ -574,7 +573,7 @@ TEST_F(PeerConnectionInterfaceTest, AddStreams) { pc_factory_->CreateAudioTrack( kStreamLabel3, static_cast<AudioSourceInterface*>(NULL))); stream->AddTrack(audio_track.get()); - EXPECT_TRUE(pc_->AddStream(stream, NULL)); + EXPECT_TRUE(pc_->AddStream(stream)); EXPECT_EQ(3u, pc_->local_streams()->count()); // Remove the third stream. @@ -1180,7 +1179,7 @@ TEST_F(PeerConnectionInterfaceTest, CloseAndTestMethods) { pc_->Close(); pc_->RemoveStream(local_stream); - EXPECT_FALSE(pc_->AddStream(local_stream, NULL)); + EXPECT_FALSE(pc_->AddStream(local_stream)); ASSERT_FALSE(local_stream->GetAudioTracks().empty()); rtc::scoped_refptr<webrtc::DtmfSenderInterface> dtmf_sender( diff --git a/app/webrtc/peerconnectionproxy.h b/app/webrtc/peerconnectionproxy.h index ed26eb8..571c676 100644 --- a/app/webrtc/peerconnectionproxy.h +++ b/app/webrtc/peerconnectionproxy.h @@ -39,8 +39,9 @@ BEGIN_PROXY_MAP(PeerConnection) local_streams) PROXY_METHOD0(rtc::scoped_refptr<StreamCollectionInterface>, remote_streams) + PROXY_METHOD1(bool, AddStream, MediaStreamInterface*) PROXY_METHOD2(bool, AddStream, MediaStreamInterface*, - const MediaConstraintsInterface*) + const MediaConstraintsInterface*); PROXY_METHOD1(void, RemoveStream, MediaStreamInterface*) PROXY_METHOD1(rtc::scoped_refptr<DtmfSenderInterface>, CreateDtmfSender, AudioTrackInterface*) diff --git a/app/webrtc/portallocatorfactory.cc b/app/webrtc/portallocatorfactory.cc index 946f8ad..da4c706 100644 --- a/app/webrtc/portallocatorfactory.cc +++ b/app/webrtc/portallocatorfactory.cc @@ -27,8 +27,8 @@ #include "talk/app/webrtc/portallocatorfactory.h" -#include "talk/p2p/base/basicpacketsocketfactory.h" -#include "talk/p2p/client/basicportallocator.h" +#include "webrtc/p2p/base/basicpacketsocketfactory.h" +#include "webrtc/p2p/client/basicportallocator.h" #include "webrtc/base/logging.h" #include "webrtc/base/network.h" #include "webrtc/base/thread.h" diff --git a/app/webrtc/proxy.h b/app/webrtc/proxy.h index 0c21ef9..134f580 100644 --- a/app/webrtc/proxy.h +++ b/app/webrtc/proxy.h @@ -55,6 +55,7 @@ #ifndef TALK_APP_WEBRTC_PROXY_H_ #define TALK_APP_WEBRTC_PROXY_H_ +#include "webrtc/base/event.h" #include "webrtc/base/thread.h" namespace webrtc { @@ -92,6 +93,34 @@ class ReturnType<void> { void value() {} }; +namespace internal { + +class SynchronousMethodCall + : public rtc::MessageData, + public rtc::MessageHandler { + public: + SynchronousMethodCall(rtc::MessageHandler* proxy) + : e_(), proxy_(proxy) {} + ~SynchronousMethodCall() {} + + void Invoke(rtc::Thread* t) { + if (t->IsCurrent()) { + proxy_->OnMessage(NULL); + } else { + e_.reset(new rtc::Event(false, false)); + t->Post(this, 0); + e_->Wait(rtc::kForever); + } + } + + private: + void OnMessage(rtc::Message*) { proxy_->OnMessage(NULL); e_->Set(); } + rtc::scoped_ptr<rtc::Event> e_; + rtc::MessageHandler* proxy_; +}; + +} // internal + template <typename C, typename R> class MethodCall0 : public rtc::Message, public rtc::MessageHandler { @@ -100,12 +129,12 @@ class MethodCall0 : public rtc::Message, MethodCall0(C* c, Method m) : c_(c), m_(m) {} R Marshal(rtc::Thread* t) { - t->Send(this, 0); + internal::SynchronousMethodCall(this).Invoke(t); return r_.value(); } private: - void OnMessage(rtc::Message*) { r_.Invoke(c_, m_);} + void OnMessage(rtc::Message*) { r_.Invoke(c_, m_); } C* c_; Method m_; @@ -120,7 +149,7 @@ class ConstMethodCall0 : public rtc::Message, ConstMethodCall0(C* c, Method m) : c_(c), m_(m) {} R Marshal(rtc::Thread* t) { - t->Send(this, 0); + internal::SynchronousMethodCall(this).Invoke(t); return r_.value(); } @@ -140,7 +169,7 @@ class MethodCall1 : public rtc::Message, MethodCall1(C* c, Method m, T1 a1) : c_(c), m_(m), a1_(a1) {} R Marshal(rtc::Thread* t) { - t->Send(this, 0); + internal::SynchronousMethodCall(this).Invoke(t); return r_.value(); } @@ -161,7 +190,7 @@ class ConstMethodCall1 : public rtc::Message, ConstMethodCall1(C* c, Method m, T1 a1) : c_(c), m_(m), a1_(a1) {} R Marshal(rtc::Thread* t) { - t->Send(this, 0); + internal::SynchronousMethodCall(this).Invoke(t); return r_.value(); } @@ -182,7 +211,7 @@ class MethodCall2 : public rtc::Message, MethodCall2(C* c, Method m, T1 a1, T2 a2) : c_(c), m_(m), a1_(a1), a2_(a2) {} R Marshal(rtc::Thread* t) { - t->Send(this, 0); + internal::SynchronousMethodCall(this).Invoke(t); return r_.value(); } @@ -205,7 +234,7 @@ class MethodCall3 : public rtc::Message, : c_(c), m_(m), a1_(a1), a2_(a2), a3_(a3) {} R Marshal(rtc::Thread* t) { - t->Send(this, 0); + internal::SynchronousMethodCall(this).Invoke(t); return r_.value(); } diff --git a/app/webrtc/statscollector.cc b/app/webrtc/statscollector.cc index 5271827..fefb0ad 100644 --- a/app/webrtc/statscollector.cc +++ b/app/webrtc/statscollector.cc @@ -89,8 +89,6 @@ const char StatsReport::kStatsValueNameEchoReturnLoss[] = const char StatsReport::kStatsValueNameEchoReturnLossEnhancement[] = "googEchoCancellationReturnLossEnhancement"; -const char StatsReport::kStatsValueNameEncodeRelStdDev[] = - "googEncodeRelStdDev"; const char StatsReport::kStatsValueNameEncodeUsagePercent[] = "googEncodeUsagePercent"; const char StatsReport::kStatsValueNameExpandRate[] = "googExpandRate"; @@ -459,8 +457,6 @@ void ExtractStats(const cricket::VideoSenderInfo& info, StatsReport* report) { info.capture_queue_delay_ms_per_s); report->AddValue(StatsReport::kStatsValueNameEncodeUsagePercent, info.encode_usage_percent); - report->AddValue(StatsReport::kStatsValueNameEncodeRelStdDev, - info.encode_rsd); } void ExtractStats(const cricket::BandwidthEstimationInfo& info, diff --git a/app/webrtc/statscollector_unittest.cc b/app/webrtc/statscollector_unittest.cc index 2e55af9..c573a88 100644 --- a/app/webrtc/statscollector_unittest.cc +++ b/app/webrtc/statscollector_unittest.cc @@ -35,7 +35,7 @@ #include "talk/app/webrtc/videotrack.h" #include "talk/media/base/fakemediaengine.h" #include "talk/media/devices/fakedevicemanager.h" -#include "talk/p2p/base/fakesession.h" +#include "webrtc/p2p/base/fakesession.h" #include "talk/session/media/channelmanager.h" #include "testing/gmock/include/gmock/gmock.h" #include "testing/gtest/include/gtest/gtest.h" diff --git a/app/webrtc/statstypes.h b/app/webrtc/statstypes.h index 656b83c..8a1530a 100644 --- a/app/webrtc/statstypes.h +++ b/app/webrtc/statstypes.h @@ -199,7 +199,6 @@ class StatsReport { // Internal StatsValue names static const char kStatsValueNameAvgEncodeMs[]; - static const char kStatsValueNameEncodeRelStdDev[]; static const char kStatsValueNameEncodeUsagePercent[]; static const char kStatsValueNameCaptureJitterMs[]; static const char kStatsValueNameCaptureQueueDelayMsPerS[]; diff --git a/app/webrtc/test/peerconnectiontestwrapper.cc b/app/webrtc/test/peerconnectiontestwrapper.cc index 24932b8..e3b8015 100644 --- a/app/webrtc/test/peerconnectiontestwrapper.cc +++ b/app/webrtc/test/peerconnectiontestwrapper.cc @@ -253,7 +253,7 @@ void PeerConnectionTestWrapper::GetAndAddUserMedia( bool video, const webrtc::FakeConstraints& video_constraints) { rtc::scoped_refptr<webrtc::MediaStreamInterface> stream = GetUserMedia(audio, audio_constraints, video, video_constraints); - EXPECT_TRUE(peer_connection_->AddStream(stream, NULL)); + EXPECT_TRUE(peer_connection_->AddStream(stream)); } rtc::scoped_refptr<webrtc::MediaStreamInterface> diff --git a/app/webrtc/test/peerconnectiontestwrapper.h b/app/webrtc/test/peerconnectiontestwrapper.h index d4a0e4e..d8299ec 100644 --- a/app/webrtc/test/peerconnectiontestwrapper.h +++ b/app/webrtc/test/peerconnectiontestwrapper.h @@ -57,7 +57,6 @@ class PeerConnectionTestWrapper const webrtc::DataChannelInit& init); // Implements PeerConnectionObserver. - virtual void OnError() {} virtual void OnSignalingChange( webrtc::PeerConnectionInterface::SignalingState new_state) {} virtual void OnStateChange( diff --git a/app/webrtc/webrtcsdp.cc b/app/webrtc/webrtcsdp.cc index 5887409..23b8f3d 100644 --- a/app/webrtc/webrtcsdp.cc +++ b/app/webrtc/webrtcsdp.cc @@ -39,9 +39,9 @@ #include "talk/media/base/constants.h" #include "talk/media/base/cryptoparams.h" #include "talk/media/sctp/sctpdataengine.h" -#include "talk/p2p/base/candidate.h" -#include "talk/p2p/base/constants.h" -#include "talk/p2p/base/port.h" +#include "webrtc/p2p/base/candidate.h" +#include "webrtc/p2p/base/constants.h" +#include "webrtc/p2p/base/port.h" #include "talk/session/media/mediasession.h" #include "talk/session/media/mediasessionclient.h" #include "webrtc/base/common.h" @@ -197,13 +197,17 @@ static const char kSessionName[] = "s=-"; static const char kTimeDescription[] = "t=0 0"; static const char kAttrGroup[] = "a=group:BUNDLE"; static const char kConnectionNettype[] = "IN"; -static const char kConnectionAddrtype[] = "IP4"; +static const char kConnectionIpv4Addrtype[] = "IP4"; +static const char kConnectionIpv6Addrtype[] = "IP6"; static const char kMediaTypeVideo[] = "video"; static const char kMediaTypeAudio[] = "audio"; static const char kMediaTypeData[] = "application"; static const char kMediaPortRejected[] = "0"; -static const char kDefaultAddress[] = "0.0.0.0"; -static const char kDefaultPort[] = "1"; +// draft-ietf-mmusic-trickle-ice-01 +// When no candidates have been gathered, set the connection +// address to IP6 ::. +static const char kDummyAddress[] = "::"; +static const char kDummyPort[] = "9"; // RFC 3556 static const char kApplicationSpecificMaximum[] = "AS"; @@ -667,10 +671,13 @@ static int GetCandidatePreferenceFromType(const std::string& type) { // The value of |component_id| currently supported are 1 (RTP) and 2 (RTCP). // TODO: Decide the default destination in webrtcsession and // pass it down via SessionDescription. -static bool GetDefaultDestination(const std::vector<Candidate>& candidates, - int component_id, std::string* port, std::string* ip) { - *port = kDefaultPort; - *ip = kDefaultAddress; +static void GetDefaultDestination( + const std::vector<Candidate>& candidates, + int component_id, std::string* port, + std::string* ip, std::string* addr_type) { + *addr_type = kConnectionIpv6Addrtype; + *port = kDummyPort; + *ip = kDummyAddress; int current_preference = kPreferenceUnknown; for (std::vector<Candidate>::const_iterator it = candidates.begin(); it != candidates.end(); ++it) { @@ -685,8 +692,13 @@ static bool GetDefaultDestination(const std::vector<Candidate>& candidates, current_preference = preference; *port = it->address().PortAsString(); *ip = it->address().ipaddr().ToString(); + int family = it->address().ipaddr().family(); + if (family == AF_INET) { + addr_type->assign(kConnectionIpv4Addrtype); + } else if (family == AF_INET6) { + addr_type->assign(kConnectionIpv6Addrtype); + } } - return true; } // Update |mline|'s default destination and append a c line after it. @@ -705,55 +717,52 @@ static void UpdateMediaDefaultDestination( } std::ostringstream os; - std::string rtp_port, rtp_ip; - if (GetDefaultDestination(candidates, ICE_CANDIDATE_COMPONENT_RTP, - &rtp_port, &rtp_ip)) { - // Found default RTP candidate. - // RFC 5245 - // The default candidates are added to the SDP as the default - // destination for media. For streams based on RTP, this is done by - // placing the IP address and port of the RTP candidate into the c and m - // lines, respectively. - - // Update the port in the m line. - // If this is a m-line with port equal to 0, we don't change it. - if (fields[1] != kMediaPortRejected) { - new_lines.replace(fields[0].size() + 1, - fields[1].size(), - rtp_port); - } - // Add the c line. - // RFC 4566 - // c=<nettype> <addrtype> <connection-address> - InitLine(kLineTypeConnection, kConnectionNettype, &os); - os << " " << kConnectionAddrtype << " " << rtp_ip; - AddLine(os.str(), &new_lines); - } + std::string rtp_port, rtp_ip, addr_type; + GetDefaultDestination(candidates, ICE_CANDIDATE_COMPONENT_RTP, + &rtp_port, &rtp_ip, &addr_type); + // Found default RTP candidate. + // RFC 5245 + // The default candidates are added to the SDP as the default + // destination for media. For streams based on RTP, this is done by + // placing the IP address and port of the RTP candidate into the c and m + // lines, respectively. + // Update the port in the m line. + // If this is a m-line with port equal to 0, we don't change it. + if (fields[1] != kMediaPortRejected) { + new_lines.replace(fields[0].size() + 1, + fields[1].size(), + rtp_port); + } + // Add the c line. + // RFC 4566 + // c=<nettype> <addrtype> <connection-address> + InitLine(kLineTypeConnection, kConnectionNettype, &os); + os << " " << addr_type << " " << rtp_ip; + AddLine(os.str(), &new_lines); message->append(new_lines); } // Gets "a=rtcp" line if found default RTCP candidate from |candidates|. static std::string GetRtcpLine(const std::vector<Candidate>& candidates) { - std::string rtcp_line, rtcp_port, rtcp_ip; - if (GetDefaultDestination(candidates, ICE_CANDIDATE_COMPONENT_RTCP, - &rtcp_port, &rtcp_ip)) { - // Found default RTCP candidate. - // RFC 5245 - // If the agent is utilizing RTCP, it MUST encode the RTCP candidate - // using the a=rtcp attribute as defined in RFC 3605. + std::string rtcp_line, rtcp_port, rtcp_ip, addr_type; + GetDefaultDestination(candidates, ICE_CANDIDATE_COMPONENT_RTCP, + &rtcp_port, &rtcp_ip, &addr_type); + // Found default RTCP candidate. + // RFC 5245 + // If the agent is utilizing RTCP, it MUST encode the RTCP candidate + // using the a=rtcp attribute as defined in RFC 3605. - // RFC 3605 - // rtcp-attribute = "a=rtcp:" port [nettype space addrtype space - // connection-address] CRLF - std::ostringstream os; - InitAttrLine(kAttributeRtcp, &os); - os << kSdpDelimiterColon - << rtcp_port << " " - << kConnectionNettype << " " - << kConnectionAddrtype << " " - << rtcp_ip; - rtcp_line = os.str(); - } + // RFC 3605 + // rtcp-attribute = "a=rtcp:" port [nettype space addrtype space + // connection-address] CRLF + std::ostringstream os; + InitAttrLine(kAttributeRtcp, &os); + os << kSdpDelimiterColon + << rtcp_port << " " + << kConnectionNettype << " " + << addr_type << " " + << rtcp_ip; + rtcp_line = os.str(); return rtcp_line; } @@ -1243,7 +1252,7 @@ void BuildMediaDescription(const ContentInfo* content_info, // To reject an offered stream, the port number in the corresponding stream in // the answer MUST be set to zero. const std::string port = content_info->rejected ? - kMediaPortRejected : kDefaultPort; + kMediaPortRejected : kDummyPort; rtc::SSLFingerprint* fp = (transport_info) ? transport_info->description.identity_fingerprint.get() : NULL; diff --git a/app/webrtc/webrtcsdp_unittest.cc b/app/webrtc/webrtcsdp_unittest.cc index 560d5da..ea590db 100644 --- a/app/webrtc/webrtcsdp_unittest.cc +++ b/app/webrtc/webrtcsdp_unittest.cc @@ -32,7 +32,7 @@ #include "talk/app/webrtc/jsepsessiondescription.h" #include "talk/app/webrtc/webrtcsdp.h" #include "talk/media/base/constants.h" -#include "talk/p2p/base/constants.h" +#include "webrtc/p2p/base/constants.h" #include "talk/session/media/mediasession.h" #include "webrtc/base/gunit.h" #include "webrtc/base/logging.h" @@ -212,9 +212,9 @@ static const char kSdpString[] = "s=-\r\n" "t=0 0\r\n" "a=msid-semantic: WMS local_stream_1 local_stream_2\r\n" - "m=audio 1 RTP/SAVPF 111 103 104\r\n" - "c=IN IP4 0.0.0.0\r\n" - "a=rtcp:1 IN IP4 0.0.0.0\r\n" + "m=audio 9 RTP/SAVPF 111 103 104\r\n" + "c=IN IP6 ::\r\n" + "a=rtcp:9 IN IP6 ::\r\n" "a=ice-ufrag:ufrag_voice\r\na=ice-pwd:pwd_voice\r\n" "a=mid:audio_content_name\r\n" "a=sendrecv\r\n" @@ -233,9 +233,9 @@ static const char kSdpString[] = "a=ssrc:4 msid:local_stream_2 audio_track_id_2\r\n" "a=ssrc:4 mslabel:local_stream_2\r\n" "a=ssrc:4 label:audio_track_id_2\r\n" - "m=video 1 RTP/SAVPF 120\r\n" - "c=IN IP4 0.0.0.0\r\n" - "a=rtcp:1 IN IP4 0.0.0.0\r\n" + "m=video 9 RTP/SAVPF 120\r\n" + "c=IN IP6 ::\r\n" + "a=rtcp:9 IN IP6 ::\r\n" "a=ice-ufrag:ufrag_video\r\na=ice-pwd:pwd_video\r\n" "a=mid:video_content_name\r\n" "a=sendrecv\r\n" @@ -261,9 +261,9 @@ static const char kSdpString[] = "a=ssrc:6 label:video_track_id_3\r\n"; static const char kSdpRtpDataChannelString[] = - "m=application 1 RTP/SAVPF 101\r\n" - "c=IN IP4 0.0.0.0\r\n" - "a=rtcp:1 IN IP4 0.0.0.0\r\n" + "m=application 9 RTP/SAVPF 101\r\n" + "c=IN IP6 ::\r\n" + "a=rtcp:9 IN IP6 ::\r\n" "a=ice-ufrag:ufrag_data\r\n" "a=ice-pwd:pwd_data\r\n" "a=mid:data_content_name\r\n" @@ -277,8 +277,8 @@ static const char kSdpRtpDataChannelString[] = "a=ssrc:10 label:data_channeld0\r\n"; static const char kSdpSctpDataChannelString[] = - "m=application 1 DTLS/SCTP 5000\r\n" - "c=IN IP4 0.0.0.0\r\n" + "m=application 9 DTLS/SCTP 5000\r\n" + "c=IN IP6 ::\r\n" "a=ice-ufrag:ufrag_data\r\n" "a=ice-pwd:pwd_data\r\n" "a=mid:data_content_name\r\n" @@ -286,10 +286,10 @@ static const char kSdpSctpDataChannelString[] = // draft-ietf-mmusic-sctp-sdp-07 static const char kSdpSctpDataChannelStringWithSctpPort[] = - "m=application 1 DTLS/SCTP webrtc-datachannel\r\n" + "m=application 9 DTLS/SCTP webrtc-datachannel\r\n" "a=fmtp:webrtc-datachannel max-message-size=100000\r\n" "a=sctp-port 5000\r\n" - "c=IN IP4 0.0.0.0\r\n" + "c=IN IP6 ::\r\n" "a=ice-ufrag:ufrag_data\r\n" "a=ice-pwd:pwd_data\r\n" "a=mid:data_content_name\r\n"; @@ -315,11 +315,11 @@ static const char kSdpConferenceString[] = "s=-\r\n" "t=0 0\r\n" "a=msid-semantic: WMS\r\n" - "m=audio 1 RTP/SAVPF 111 103 104\r\n" - "c=IN IP4 0.0.0.0\r\n" + "m=audio 9 RTP/SAVPF 111 103 104\r\n" + "c=IN IP6 ::\r\n" "a=x-google-flag:conference\r\n" - "m=video 1 RTP/SAVPF 120\r\n" - "c=IN IP4 0.0.0.0\r\n" + "m=video 9 RTP/SAVPF 120\r\n" + "c=IN IP6 ::\r\n" "a=x-google-flag:conference\r\n"; static const char kSdpSessionString[] = @@ -330,9 +330,9 @@ static const char kSdpSessionString[] = "a=msid-semantic: WMS local_stream\r\n"; static const char kSdpAudioString[] = - "m=audio 1 RTP/SAVPF 111\r\n" - "c=IN IP4 0.0.0.0\r\n" - "a=rtcp:1 IN IP4 0.0.0.0\r\n" + "m=audio 9 RTP/SAVPF 111\r\n" + "c=IN IP6 ::\r\n" + "a=rtcp:9 IN IP6 ::\r\n" "a=ice-ufrag:ufrag_voice\r\na=ice-pwd:pwd_voice\r\n" "a=mid:audio_content_name\r\n" "a=sendrecv\r\n" @@ -343,9 +343,9 @@ static const char kSdpAudioString[] = "a=ssrc:1 label:audio_track_id_1\r\n"; static const char kSdpVideoString[] = - "m=video 1 RTP/SAVPF 120\r\n" - "c=IN IP4 0.0.0.0\r\n" - "a=rtcp:1 IN IP4 0.0.0.0\r\n" + "m=video 9 RTP/SAVPF 120\r\n" + "c=IN IP6 ::\r\n" + "a=rtcp:9 IN IP6 ::\r\n" "a=ice-ufrag:ufrag_video\r\na=ice-pwd:pwd_video\r\n" "a=mid:video_content_name\r\n" "a=sendrecv\r\n" @@ -1195,7 +1195,7 @@ class WebRtcSdpTest : public testing::Test { // description. "a=msid-semantic: WMS\r\n" // Pl type 111 preferred. - "m=audio 1 RTP/SAVPF 111 104 103 102\r\n" + "m=audio 9 RTP/SAVPF 111 104 103 102\r\n" // Pltype 111 listed before 103 and 104 in the map. "a=rtpmap:111 opus/48000/2\r\n" // Pltype 103 listed before 104. @@ -1217,7 +1217,7 @@ class WebRtcSdpTest : public testing::Test { os.clear(); os.str(""); // Pl type 100 preferred. - os << "m=video 1 RTP/SAVPF 99 95\r\n" + os << "m=video 9 RTP/SAVPF 99 95\r\n" << "a=rtpmap:99 VP8/90000\r\n" << "a=rtpmap:95 RTX/90000\r\n" << "a=fmtp:95 apt=99;rtx-time=1000\r\n"; @@ -1279,7 +1279,7 @@ class WebRtcSdpTest : public testing::Test { // this parser, and will be added to the SDP when serializing a session // description. "a=msid-semantic: WMS\r\n" - "m=audio 1 RTP/SAVPF 111\r\n" + "m=audio 9 RTP/SAVPF 111\r\n" "a=rtpmap:111 opus/48000/2\r\n" "a=rtcp-fb:111 nack\r\n" "m=video 3457 RTP/SAVPF 101\r\n" @@ -1596,7 +1596,7 @@ TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithDataChannelAndBandwidth) { // TODO(pthatcher): We need to temporarily allow the SDP to control // this for backwards-compatibility. Once we don't need that any // more, remove this. - InjectAfter("m=application 1 RTP/SAVPF 101\r\nc=IN IP4 0.0.0.0\r\n", + InjectAfter("m=application 9 RTP/SAVPF 101\r\nc=IN IP6 ::\r\n", "b=AS:100\r\n", &expected_sdp); EXPECT_EQ(expected_sdp, message); @@ -2308,7 +2308,7 @@ TEST_F(WebRtcSdpTest, DeserializeSdpWithReorderedPltypes) { "o=- 18446744069414584320 18446462598732840960 IN IP4 127.0.0.1\r\n" "s=-\r\n" "t=0 0\r\n" - "m=audio 1 RTP/SAVPF 104 103\r\n" // Pl type 104 preferred. + "m=audio 9 RTP/SAVPF 104 103\r\n" // Pl type 104 preferred. "a=rtpmap:111 opus/48000/2\r\n" // Pltype 111 listed before 103 and 104 // in the map. "a=rtpmap:103 ISAC/16000\r\n" // Pltype 103 listed before 104 in the map. diff --git a/app/webrtc/webrtcsession.h b/app/webrtc/webrtcsession.h index 86ae435..25e9646 100644 --- a/app/webrtc/webrtcsession.h +++ b/app/webrtc/webrtcsession.h @@ -36,7 +36,7 @@ #include "talk/app/webrtc/peerconnectioninterface.h" #include "talk/app/webrtc/statstypes.h" #include "talk/media/base/mediachannel.h" -#include "talk/p2p/base/session.h" +#include "webrtc/p2p/base/session.h" #include "talk/session/media/mediasession.h" #include "webrtc/base/sigslot.h" #include "webrtc/base/thread.h" diff --git a/app/webrtc/webrtcsession_unittest.cc b/app/webrtc/webrtcsession_unittest.cc index d3480a0..e79001a 100644 --- a/app/webrtc/webrtcsession_unittest.cc +++ b/app/webrtc/webrtcsession_unittest.cc @@ -40,10 +40,10 @@ #include "talk/media/base/fakevideorenderer.h" #include "talk/media/base/mediachannel.h" #include "talk/media/devices/fakedevicemanager.h" -#include "talk/p2p/base/stunserver.h" -#include "talk/p2p/base/teststunserver.h" -#include "talk/p2p/base/testturnserver.h" -#include "talk/p2p/client/basicportallocator.h" +#include "webrtc/p2p/base/stunserver.h" +#include "webrtc/p2p/base/teststunserver.h" +#include "webrtc/p2p/base/testturnserver.h" +#include "webrtc/p2p/client/basicportallocator.h" #include "talk/session/media/channelmanager.h" #include "talk/session/media/mediasession.h" #include "webrtc/base/fakenetwork.h" diff --git a/app/webrtc/webrtcsessiondescriptionfactory.h b/app/webrtc/webrtcsessiondescriptionfactory.h index b870856..31f72fe 100644 --- a/app/webrtc/webrtcsessiondescriptionfactory.h +++ b/app/webrtc/webrtcsessiondescriptionfactory.h @@ -29,7 +29,7 @@ #define TALK_APP_WEBRTC_WEBRTCSESSIONDESCRIPTIONFACTORY_H_ #include "talk/app/webrtc/peerconnectioninterface.h" -#include "talk/p2p/base/transportdescriptionfactory.h" +#include "webrtc/p2p/base/transportdescriptionfactory.h" #include "talk/session/media/mediasession.h" #include "webrtc/base/messagehandler.h" diff --git a/build/common.gypi b/build/common.gypi index 7ee4224..d682381 100644 --- a/build/common.gypi +++ b/build/common.gypi @@ -39,9 +39,7 @@ 'libpeer_target_type%': 'static_library', 'conditions': [ ['OS=="android" or OS=="linux"', { - # TODO(henrike): make sure waterfall bots have $JAVA_HOME configured - # properly and remove the default value below. See issue 2113. - 'java_home%': '<!(python -c "import os; dir=os.getenv(\'JAVA_HOME\', \'/usr/lib/jvm/java-6-sun\'); assert os.path.exists(os.path.join(dir, \'include/jni.h\')), \'Point \\$JAVA_HOME or the java_home gyp variable to a directory containing include/jni.h!\'; print dir")', + 'java_home%': '<!(python -c "import os; dir=os.getenv(\'JAVA_HOME\', \'/usr/lib/jvm/java-7-openjdk-amd64\'); assert os.path.exists(os.path.join(dir, \'include/jni.h\')), \'Point \\$JAVA_HOME or the java_home gyp variable to a directory containing include/jni.h!\'; print dir")', }], ], }, diff --git a/build/isolate.gypi b/build/isolate.gypi index 6f884cf..df28525 100644 --- a/build/isolate.gypi +++ b/build/isolate.gypi @@ -81,18 +81,6 @@ # Files that are known to be involved in this step. '<(DEPTH)/tools/swarming_client/isolate.py', '<(DEPTH)/tools/swarming_client/run_isolated.py', - - # Disable file tracking by the build driver for now. This means the - # project must have the proper build-time dependency for their runtime - # dependency. This improves the runtime of the build driver since it - # doesn't have to stat() all these files. - # - # More importantly, it means that even if a isolate_dependency_tracked - # file is missing, for example if a file was deleted and the .isolate - # file was not updated, that won't break the build, especially in the - # case where foo_tests_run is not built! This should be reenabled once - # the switch-over to running tests on Swarm is completed. - #'<@(isolate_dependency_tracked)', ], 'outputs': [ '<(PRODUCT_DIR)/<(RULE_INPUT_ROOT).isolated', diff --git a/examples/android/AndroidManifest.xml b/examples/android/AndroidManifest.xml index f898641..30fd46c 100644 --- a/examples/android/AndroidManifest.xml +++ b/examples/android/AndroidManifest.xml @@ -4,38 +4,46 @@ android:versionCode="1" android:versionName="1.0"> - <uses-feature android:name="android.hardware.camera" /> - <uses-feature android:name="android.hardware.camera.autofocus" /> - <uses-feature android:glEsVersion="0x00020000" android:required="true"></uses-feature> - <uses-sdk android:minSdkVersion="13" android:targetSdkVersion="17" /> + <uses-feature android:name="android.hardware.camera" /> + <uses-feature android:name="android.hardware.camera.autofocus" /> + <uses-feature android:glEsVersion="0x00020000" android:required="true" /> + <uses-sdk android:minSdkVersion="13" android:targetSdkVersion="19" /> - <uses-permission android:name="android.permission.CAMERA"></uses-permission> - <uses-permission android:name="android.permission.MODIFY_AUDIO_SETTINGS" /> - <uses-permission android:name="android.permission.RECORD_AUDIO" /> - <uses-permission android:name="android.permission.INTERNET" /> - <uses-permission android:name="android.permission.WRITE_EXTERNAL_STORAGE"/> + <uses-permission android:name="android.permission.CAMERA" /> + <uses-permission android:name="android.permission.MODIFY_AUDIO_SETTINGS" /> + <uses-permission android:name="android.permission.RECORD_AUDIO" /> + <uses-permission android:name="android.permission.INTERNET" /> + <uses-permission android:name="android.permission.WRITE_EXTERNAL_STORAGE"/> - <application android:label="@string/app_name" - android:icon="@drawable/ic_launcher" - android:debuggable="true" - android:allowBackup="false"> - <activity android:name="AppRTCDemoActivity" - android:label="@string/app_name" - android:screenOrientation="fullUser" - android:configChanges="orientation|screenSize" - android:theme="@android:style/Theme.Black.NoTitleBar.Fullscreen"> - <intent-filter> - <action android:name="android.intent.action.MAIN" /> - <category android:name="android.intent.category.LAUNCHER" /> - </intent-filter> + <application android:label="@string/app_name" + android:icon="@drawable/ic_launcher" + android:allowBackup="false"> - <intent-filter> - <action android:name="android.intent.action.VIEW" /> - <category android:name="android.intent.category.DEFAULT" /> - <category android:name="android.intent.category.BROWSABLE" /> - <data android:scheme="https" android:host="apprtc.appspot.com" /> - <data android:scheme="http" android:host="apprtc.appspot.com" /> - </intent-filter> - </activity> - </application> + <activity android:name="ConnectActivity" + android:label="@string/app_name"> + <intent-filter> + <action android:name="android.intent.action.MAIN"/> + <category android:name="android.intent.category.LAUNCHER"/> + </intent-filter> + + <intent-filter> + <action android:name="android.intent.action.VIEW"/> + <category android:name="android.intent.category.DEFAULT"/> + <category android:name="android.intent.category.BROWSABLE"/> + <data android:scheme="https" android:host="apprtc.appspot.com"/> + <data android:scheme="http" android:host="apprtc.appspot.com"/> + </intent-filter> + </activity> + + <activity android:name="SettingsActivity" + android:label="@string/settings_name"> + </activity> + + <activity android:name="AppRTCDemoActivity" + android:label="@string/app_name" + android:screenOrientation="fullUser" + android:configChanges="orientation|screenSize" + android:theme="@android:style/Theme.Black.NoTitleBar.Fullscreen"> + </activity> + </application> </manifest> diff --git a/examples/android/project.properties b/examples/android/project.properties index 8459f9b..47b7078 100644 --- a/examples/android/project.properties +++ b/examples/android/project.properties @@ -11,6 +11,6 @@ #proguard.config=${sdk.dir}/tools/proguard/proguard-android.txt:proguard-project.txt # Project target. -target=android-19 +target=android-21 java.compilerargs=-Xlint:all -Werror diff --git a/examples/android/res/drawable-hdpi/disconnect.png b/examples/android/res/drawable-hdpi/disconnect.png Binary files differnew file mode 100644 index 0000000..be36174 --- /dev/null +++ b/examples/android/res/drawable-hdpi/disconnect.png diff --git a/examples/android/res/drawable-hdpi/ic_action_full_screen.png b/examples/android/res/drawable-hdpi/ic_action_full_screen.png Binary files differnew file mode 100644 index 0000000..22f30d3 --- /dev/null +++ b/examples/android/res/drawable-hdpi/ic_action_full_screen.png diff --git a/examples/android/res/drawable-hdpi/ic_action_return_from_full_screen.png b/examples/android/res/drawable-hdpi/ic_action_return_from_full_screen.png Binary files differnew file mode 100644 index 0000000..d9436e5 --- /dev/null +++ b/examples/android/res/drawable-hdpi/ic_action_return_from_full_screen.png diff --git a/examples/android/res/drawable-hdpi/ic_launcher.png b/examples/android/res/drawable-hdpi/ic_launcher.png Binary files differindex f3e9d12..f01a31a 100644 --- a/examples/android/res/drawable-hdpi/ic_launcher.png +++ b/examples/android/res/drawable-hdpi/ic_launcher.png diff --git a/examples/android/res/drawable-hdpi/ic_loopback_call.png b/examples/android/res/drawable-hdpi/ic_loopback_call.png Binary files differnew file mode 100644 index 0000000..3931185 --- /dev/null +++ b/examples/android/res/drawable-hdpi/ic_loopback_call.png diff --git a/examples/android/res/drawable-ldpi/disconnect.png b/examples/android/res/drawable-ldpi/disconnect.png Binary files differnew file mode 100644 index 0000000..be36174 --- /dev/null +++ b/examples/android/res/drawable-ldpi/disconnect.png diff --git a/examples/android/res/drawable-ldpi/ic_action_full_screen.png b/examples/android/res/drawable-ldpi/ic_action_full_screen.png Binary files differnew file mode 100644 index 0000000..e4a9ff0 --- /dev/null +++ b/examples/android/res/drawable-ldpi/ic_action_full_screen.png diff --git a/examples/android/res/drawable-ldpi/ic_action_return_from_full_screen.png b/examples/android/res/drawable-ldpi/ic_action_return_from_full_screen.png Binary files differnew file mode 100644 index 0000000..f5c80f0 --- /dev/null +++ b/examples/android/res/drawable-ldpi/ic_action_return_from_full_screen.png diff --git a/examples/android/res/drawable-ldpi/ic_loopback_call.png b/examples/android/res/drawable-ldpi/ic_loopback_call.png Binary files differnew file mode 100644 index 0000000..3931185 --- /dev/null +++ b/examples/android/res/drawable-ldpi/ic_loopback_call.png diff --git a/examples/android/res/drawable-mdpi/disconnect.png b/examples/android/res/drawable-mdpi/disconnect.png Binary files differnew file mode 100644 index 0000000..be36174 --- /dev/null +++ b/examples/android/res/drawable-mdpi/disconnect.png diff --git a/examples/android/res/drawable-mdpi/ic_action_full_screen.png b/examples/android/res/drawable-mdpi/ic_action_full_screen.png Binary files differnew file mode 100644 index 0000000..e4a9ff0 --- /dev/null +++ b/examples/android/res/drawable-mdpi/ic_action_full_screen.png diff --git a/examples/android/res/drawable-mdpi/ic_action_return_from_full_screen.png b/examples/android/res/drawable-mdpi/ic_action_return_from_full_screen.png Binary files differnew file mode 100644 index 0000000..f5c80f0 --- /dev/null +++ b/examples/android/res/drawable-mdpi/ic_action_return_from_full_screen.png diff --git a/examples/android/res/drawable-mdpi/ic_launcher.png b/examples/android/res/drawable-mdpi/ic_launcher.png Binary files differindex 9709a1e..b8b4b0e 100644 --- a/examples/android/res/drawable-mdpi/ic_launcher.png +++ b/examples/android/res/drawable-mdpi/ic_launcher.png diff --git a/examples/android/res/drawable-mdpi/ic_loopback_call.png b/examples/android/res/drawable-mdpi/ic_loopback_call.png Binary files differnew file mode 100644 index 0000000..3931185 --- /dev/null +++ b/examples/android/res/drawable-mdpi/ic_loopback_call.png diff --git a/examples/android/res/drawable-xhdpi/disconnect.png b/examples/android/res/drawable-xhdpi/disconnect.png Binary files differnew file mode 100644 index 0000000..be36174 --- /dev/null +++ b/examples/android/res/drawable-xhdpi/disconnect.png diff --git a/examples/android/res/drawable-xhdpi/ic_action_full_screen.png b/examples/android/res/drawable-xhdpi/ic_action_full_screen.png Binary files differnew file mode 100644 index 0000000..6d90c07 --- /dev/null +++ b/examples/android/res/drawable-xhdpi/ic_action_full_screen.png diff --git a/examples/android/res/drawable-xhdpi/ic_action_return_from_full_screen.png b/examples/android/res/drawable-xhdpi/ic_action_return_from_full_screen.png Binary files differnew file mode 100644 index 0000000..a773b34 --- /dev/null +++ b/examples/android/res/drawable-xhdpi/ic_action_return_from_full_screen.png diff --git a/examples/android/res/drawable-xhdpi/ic_launcher.png b/examples/android/res/drawable-xhdpi/ic_launcher.png Binary files differindex db2c4f6..a3cd458 100644 --- a/examples/android/res/drawable-xhdpi/ic_launcher.png +++ b/examples/android/res/drawable-xhdpi/ic_launcher.png diff --git a/examples/android/res/drawable-xhdpi/ic_loopback_call.png b/examples/android/res/drawable-xhdpi/ic_loopback_call.png Binary files differnew file mode 100644 index 0000000..3931185 --- /dev/null +++ b/examples/android/res/drawable-xhdpi/ic_loopback_call.png diff --git a/examples/android/res/layout/activity_connect.xml b/examples/android/res/layout/activity_connect.xml new file mode 100644 index 0000000..5ca0f19 --- /dev/null +++ b/examples/android/res/layout/activity_connect.xml @@ -0,0 +1,72 @@ +<?xml version="1.0" encoding="utf-8"?> +<LinearLayout + xmlns:android="http://schemas.android.com/apk/res/android" + android:layout_width="match_parent" + android:layout_height="match_parent" + android:orientation="vertical" + android:weightSum="1" + android:layout_margin="8dp" + android:layout_centerHorizontal="true"> + + <LinearLayout + android:layout_width="match_parent" + android:layout_height="wrap_content" + android:orientation="horizontal" > + <ImageButton + android:id="@+id/add_room_button" + android:background="@android:drawable/ic_menu_add" + android:contentDescription="@string/add_room_description" + android:layout_marginRight="20dp" + android:layout_width="48dp" + android:layout_height="48dp"/> + <ImageButton + android:id="@+id/remove_room_button" + android:background="@android:drawable/ic_delete" + android:contentDescription="@string/remove_room_description" + android:layout_marginRight="20dp" + android:layout_width="48dp" + android:layout_height="48dp"/> + <ImageButton + android:id="@+id/connect_button" + android:background="@android:drawable/sym_action_call" + android:contentDescription="@string/connect_description" + android:layout_marginRight="20dp" + android:layout_width="48dp" + android:layout_height="48dp"/> + <ImageButton + android:id="@+id/connect_loopback_button" + android:background="@drawable/ic_loopback_call" + android:contentDescription="@string/connect_loopback_description" + android:layout_width="48dp" + android:layout_height="48dp"/> + </LinearLayout> + <TextView + android:id="@+id/room_edittext_description" + android:layout_width="fill_parent" + android:layout_height="wrap_content" + android:layout_margin="5dp" + android:text="@string/room_description"/> + <EditText + android:id="@+id/room_edittext" + android:layout_width="match_parent" + android:layout_height="wrap_content" + android:singleLine="true" + android:imeOptions="actionDone"/> + <TextView + android:id="@+id/room_listview_description" + android:layout_width="fill_parent" + android:layout_height="wrap_content" + android:layout_marginTop="5dp" + android:lines="1" + android:maxLines="1" + android:textAppearance="?android:attr/textAppearanceMedium" + android:text="@string/room_names"/> + <ListView + android:id="@+id/room_listview" + android:layout_width="fill_parent" + android:layout_height="wrap_content" + android:choiceMode="singleChoice" + android:listSelector="@android:color/darker_gray" + android:drawSelectorOnTop="false" /> + +</LinearLayout>
\ No newline at end of file diff --git a/examples/android/res/layout/activity_fullscreen.xml b/examples/android/res/layout/activity_fullscreen.xml new file mode 100644 index 0000000..fc9ee9e --- /dev/null +++ b/examples/android/res/layout/activity_fullscreen.xml @@ -0,0 +1,33 @@ +<?xml version="1.0" encoding="utf-8"?> + +<RelativeLayout + xmlns:android="http://schemas.android.com/apk/res/android" + xmlns:tools="http://schemas.android.com/tools" + android:layout_width="match_parent" + android:layout_height="match_parent" + android:orientation="vertical"> + + <android.opengl.GLSurfaceView + android:id="@+id/glview" + android:layout_width="match_parent" + android:layout_height="match_parent" /> + + <TextView + android:id="@+id/room_name" + android:layout_width="wrap_content" + android:layout_height="wrap_content" + android:layout_centerHorizontal="true" + android:layout_above="@+id/menubar_fragment" + android:textSize="24sp" + android:layout_margin="8dp"/> + <fragment + android:name="org.appspot.apprtc.AppRTCDemoActivity$MenuBarFragment" + android:id="@+id/menubar_fragment" + android:layout_centerHorizontal="true" + android:layout_width="wrap_content" + android:layout_height="wrap_content" + android:layout_alignParentBottom="true" + android:layout_marginBottom="32dp" + tools:layout="@layout/fragment_menubar"/> + +</RelativeLayout> diff --git a/examples/android/res/layout/fragment_menubar.xml b/examples/android/res/layout/fragment_menubar.xml new file mode 100644 index 0000000..77dc819 --- /dev/null +++ b/examples/android/res/layout/fragment_menubar.xml @@ -0,0 +1,45 @@ +<?xml version="1.0" encoding="utf-8"?> + +<LinearLayout xmlns:android="http://schemas.android.com/apk/res/android" + xmlns:tools="http://schemas.android.com/tools" + tools:context="org.appspot.apprtc.AppRTCDemoActivity$MenuBarFragment" + android:id="@+id/menubar" + android:orientation="horizontal" + android:layout_width="wrap_content" + android:layout_height="match_parent" + android:layout_gravity="center_vertical|center_horizontal"> + + <ImageButton + android:id="@+id/button_disconnect" + android:background="@drawable/disconnect" + android:contentDescription="@string/disconnect_call" + android:layout_marginRight="16dp" + android:layout_width="48dp" + android:layout_height="48dp"/> + + <!-- TODO(kjellander): Add audio and video mute buttons. --> + + <ImageButton + android:id="@+id/button_switch_camera" + android:background="@android:drawable/ic_menu_camera" + android:contentDescription="@string/switch_camera" + android:layout_marginRight="8dp" + android:layout_width="48dp" + android:layout_height="48dp"/> + + <ImageButton + android:id="@+id/button_toggle_debug" + android:background="@android:drawable/ic_menu_info_details" + android:contentDescription="@string/disconnect_call" + android:layout_marginRight="8dp" + android:layout_width="48dp" + android:layout_height="48dp"/> + + <ImageButton + android:id="@+id/button_scaling_mode" + android:background="@drawable/ic_action_return_from_full_screen" + android:contentDescription="@string/disconnect_call" + android:layout_width="48dp" + android:layout_height="48dp"/> + +</LinearLayout> diff --git a/examples/android/res/menu/connect_menu.xml b/examples/android/res/menu/connect_menu.xml new file mode 100644 index 0000000..d9f9486 --- /dev/null +++ b/examples/android/res/menu/connect_menu.xml @@ -0,0 +1,8 @@ +<menu xmlns:android="http://schemas.android.com/apk/res/android" > + <item + android:id="@+id/action_settings" + android:orderInCategory="100" + android:icon="@android:drawable/ic_menu_preferences" + android:showAsAction="ifRoom" + android:title="@string/action_settings"/> +</menu> diff --git a/examples/android/res/values/arrays.xml b/examples/android/res/values/arrays.xml new file mode 100644 index 0000000..3127a85 --- /dev/null +++ b/examples/android/res/values/arrays.xml @@ -0,0 +1,22 @@ +<?xml version="1.0" encoding="utf-8"?> +<resources> + <string-array name="videoResolutions"> + <item>Default</item> + <item>HD (1280 x 720)</item> + <item>VGA (640 x 480)</item> + <item>QVGA (320 x 240)</item> + </string-array> + <string-array name="videoResolutionsValues"> + <item>Default</item> + <item>1280 x 720</item> + <item>640 x 480</item> + <item>320 x 240</item> + </string-array> + + <string-array name="cameraFps"> + <item>Default</item> + <item>30 fps</item> + <item>15 fps</item> + </string-array> + +</resources> diff --git a/examples/android/res/values/strings.xml b/examples/android/res/values/strings.xml index bac765a..2a1d64a 100644 --- a/examples/android/res/values/strings.xml +++ b/examples/android/res/values/strings.xml @@ -1,4 +1,47 @@ <?xml version="1.0" encoding="utf-8"?> <resources> - <string name="app_name">AppRTC</string> + <string name="app_name" translatable="no">AppRTC</string> + <string name="settings_name" translatable="no">AppRTC Settings</string> + <string name="disconnect_call">Disconnect Call</string> + <string name="room_names">Room names:</string> + <string name="room_description"> + Please enter a room name. Room names are shared with everyone, so think + of something unique and send it to a friend. + </string> + <string name="connect_text">Connect</string> + <string name="invalid_url_title">Invalid URL</string> + <string name="invalid_url_text">The URL or room name you entered resulted in an invalid URL: %1$s + </string> + <string name="channel_error_title">Connection error</string> + <string name="connecting_to">Connecting to: %1$s</string> + <string name="missing_url">FATAL ERROR: Missing URL to connect to.</string> + <string name="ok">OK</string> + <string name="switch_camera">Switch front/back camera</string> + <string name="action_settings">Settings</string> + <string name="add_room_description">Add new room to the list</string> + <string name="remove_room_description">Remove room from the list</string> + <string name="connect_description">Connect to the room</string> + <string name="connect_loopback_description">Loopback connection</string> + + <!-- Settings strings. --> + <string name="pref_room_key">room_preference</string> + <string name="pref_room_list_key">room_list_preference</string> + + <string name="pref_url_key">url_preference</string> + <string name="pref_url_title">Connection URL:</string> + <string name="pref_url_summary">AppRTC connection server URL.</string> + <string name="pref_url_dlg">Enter AppRTC connection server URL.</string> + <string name="pref_url_default">https://apprtc.appspot.com</string> + + <string name="pref_resolution_key">resolution_preference</string> + <string name="pref_resolution_title">Video resolution.</string> + <string name="pref_resolution_summary">Video resolution.</string> + <string name="pref_resolution_dlg">Enter AppRTC local video resolution.</string> + <string name="pref_resolution_default">Default</string> + + <string name="pref_fps_key">fps_preference</string> + <string name="pref_fps_title">Camera fps.</string> + <string name="pref_fps_summary">Camera fps.</string> + <string name="pref_fps_dlg">Enter local camera fps.</string> + <string name="pref_fps_default">Default</string> </resources> diff --git a/examples/android/res/xml/preferences.xml b/examples/android/res/xml/preferences.xml new file mode 100644 index 0000000..f3f91d8 --- /dev/null +++ b/examples/android/res/xml/preferences.xml @@ -0,0 +1,26 @@ +<?xml version="1.0" encoding="utf-8"?> +<PreferenceScreen xmlns:android="http://schemas.android.com/apk/res/android"> + <EditTextPreference + android:key="@string/pref_url_key" + android:title="@string/pref_url_title" + android:summary="@string/pref_url_summary" + android:defaultValue="@string/pref_url_default" + android:inputType="textWebEmailAddress" + android:dialogTitle="@string/pref_url_dlg" /> + <ListPreference + android:key="@string/pref_resolution_key" + android:title="@string/pref_resolution_title" + android:summary="@string/pref_resolution_summary" + android:defaultValue="@string/pref_resolution_default" + android:dialogTitle="@string/pref_resolution_dlg" + android:entries="@array/videoResolutions" + android:entryValues="@array/videoResolutionsValues" /> + <ListPreference + android:key="@string/pref_fps_key" + android:title="@string/pref_fps_title" + android:summary="@string/pref_fps_summary" + android:defaultValue="@string/pref_fps_default" + android:dialogTitle="@string/pref_fps_dlg" + android:entries="@array/cameraFps" + android:entryValues="@array/cameraFps" /> +</PreferenceScreen>
\ No newline at end of file diff --git a/examples/android/src/org/appspot/apprtc/AppRTCClient.java b/examples/android/src/org/appspot/apprtc/AppRTCClient.java index 4728e51..5c34fca 100644 --- a/examples/android/src/org/appspot/apprtc/AppRTCClient.java +++ b/examples/android/src/org/appspot/apprtc/AppRTCClient.java @@ -80,7 +80,9 @@ public interface AppRTCClient { } /** - * Signaling callbacks. + * Callback interface for messages delivered on signalling channel. + * + * Methods are guaranteed to be invoked on the UI thread of |activity|. */ public static interface AppRTCSignalingEvents { /** diff --git a/examples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java b/examples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java index 4718a01..ad0e2d5 100644 --- a/examples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java +++ b/examples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java @@ -29,19 +29,24 @@ package org.appspot.apprtc; import android.app.Activity; import android.app.AlertDialog; +import android.app.Fragment; import android.content.DialogInterface; import android.content.Intent; -import android.content.res.Configuration; +import android.content.pm.ActivityInfo; import android.graphics.Color; -import android.graphics.Point; import android.media.AudioManager; +import android.net.Uri; +import android.opengl.GLSurfaceView; import android.os.Bundle; import android.util.Log; import android.util.TypedValue; +import android.view.LayoutInflater; import android.view.View; +import android.view.ViewGroup; import android.view.ViewGroup.LayoutParams; +import android.view.Window; import android.view.WindowManager; -import android.widget.EditText; +import android.widget.ImageButton; import android.widget.TextView; import android.widget.Toast; @@ -53,6 +58,7 @@ import org.webrtc.StatsObserver; import org.webrtc.StatsReport; import org.webrtc.VideoRenderer; import org.webrtc.VideoRendererGui; +import org.webrtc.VideoRendererGui.ScalingType; /** * Main Activity of the AppRTCDemo Android app demonstrating interoperability @@ -66,43 +72,112 @@ public class AppRTCDemoActivity extends Activity private PeerConnectionClient pc; private AppRTCClient appRtcClient = new GAERTCClient(this, this); private AppRTCSignalingParameters appRtcParameters; - private AppRTCGLView vsv; + private View rootView; + private View menuBar; + private GLSurfaceView videoView; private VideoRenderer.Callbacks localRender; private VideoRenderer.Callbacks remoteRender; + private ScalingType scalingType; private Toast logToast; private final LayoutParams hudLayout = new LayoutParams(LayoutParams.WRAP_CONTENT, LayoutParams.WRAP_CONTENT); private TextView hudView; - // Synchronize on quit[0] to avoid teardown-related crashes. - private final Boolean[] quit = new Boolean[] { false }; + private TextView roomName; + private ImageButton videoScalingButton; + private boolean iceConnected; @Override public void onCreate(Bundle savedInstanceState) { super.onCreate(savedInstanceState); + // Set window styles for fullscreen-window size. Needs to be done before + // adding content. + requestWindowFeature(Window.FEATURE_NO_TITLE); + getWindow().addFlags(WindowManager.LayoutParams.FLAG_FULLSCREEN); + getWindow().addFlags(WindowManager.LayoutParams.FLAG_KEEP_SCREEN_ON); + getWindow().getDecorView().setSystemUiVisibility( + View.SYSTEM_UI_FLAG_HIDE_NAVIGATION | + View.SYSTEM_UI_FLAG_FULLSCREEN | + View.SYSTEM_UI_FLAG_IMMERSIVE_STICKY); + + setContentView(R.layout.activity_fullscreen); + Thread.setDefaultUncaughtExceptionHandler( new UnhandledExceptionHandler(this)); + iceConnected = false; - getWindow().addFlags(WindowManager.LayoutParams.FLAG_FULLSCREEN); - getWindow().addFlags(WindowManager.LayoutParams.FLAG_KEEP_SCREEN_ON); + rootView = findViewById(android.R.id.content); + menuBar = findViewById(R.id.menubar_fragment); + roomName = (TextView) findViewById(R.id.room_name); + videoView = (GLSurfaceView) findViewById(R.id.glview); + + VideoRendererGui.setView(videoView); + scalingType = ScalingType.SCALE_ASPECT_FILL; + remoteRender = VideoRendererGui.create(0, 0, 100, 100, scalingType); + localRender = VideoRendererGui.create(0, 0, 100, 100, scalingType); - Point displaySize = new Point(); - getWindowManager().getDefaultDisplay().getRealSize(displaySize); + videoView.setOnClickListener( + new View.OnClickListener() { + @Override + public void onClick(View view) { + int visibility = menuBar.getVisibility() == View.VISIBLE + ? View.INVISIBLE : View.VISIBLE; + menuBar.setVisibility(visibility); + roomName.setVisibility(visibility); + if (visibility == View.VISIBLE) { + menuBar.bringToFront(); + roomName.bringToFront(); + rootView.invalidate(); + } + } + }); - vsv = new AppRTCGLView(this, displaySize); - VideoRendererGui.setView(vsv); - remoteRender = VideoRendererGui.create(0, 0, 100, 100, - VideoRendererGui.ScalingType.SCALE_ASPECT_FIT); - localRender = VideoRendererGui.create(0, 0, 100, 100, - VideoRendererGui.ScalingType.SCALE_ASPECT_FIT); + ((ImageButton) findViewById(R.id.button_disconnect)).setOnClickListener( + new View.OnClickListener() { + @Override + public void onClick(View view) { + logAndToast("Disconnecting call."); + disconnect(); + } + }); - vsv.setOnClickListener(new View.OnClickListener() { - @Override public void onClick(View v) { - toggleHUD(); - } - }); - setContentView(vsv); - logAndToast("Tap the screen to toggle stats visibility"); + ((ImageButton) findViewById(R.id.button_switch_camera)).setOnClickListener( + new View.OnClickListener() { + @Override + public void onClick(View view) { + if (pc != null) { + pc.switchCamera(); + } + } + }); + + ((ImageButton) findViewById(R.id.button_toggle_debug)).setOnClickListener( + new View.OnClickListener() { + @Override + public void onClick(View view) { + int visibility = hudView.getVisibility() == View.VISIBLE + ? View.INVISIBLE : View.VISIBLE; + hudView.setVisibility(visibility); + } + }); + + videoScalingButton = (ImageButton) findViewById(R.id.button_scaling_mode); + videoScalingButton.setOnClickListener( + new View.OnClickListener() { + @Override + public void onClick(View view) { + if (scalingType == ScalingType.SCALE_ASPECT_FILL) { + videoScalingButton.setBackgroundResource( + R.drawable.ic_action_full_screen); + scalingType = ScalingType.SCALE_ASPECT_FIT; + } else { + videoScalingButton.setBackgroundResource( + R.drawable.ic_action_return_from_full_screen); + scalingType = ScalingType.SCALE_ASPECT_FILL; + } + updateVideoView(); + } + }); hudView = new TextView(this); hudView.setTextColor(Color.BLACK); @@ -123,17 +198,44 @@ public class AppRTCDemoActivity extends Activity audioManager.setSpeakerphoneOn(!isWiredHeadsetOn); final Intent intent = getIntent(); - if ("android.intent.action.VIEW".equals(intent.getAction())) { - connectToRoom(intent.getData().toString()); - return; + Uri url = intent.getData(); + if (url != null) { + String room = url.getQueryParameter("r"); + String loopback = url.getQueryParameter("debug"); + if ((room != null && !room.equals("")) || + (loopback != null && loopback.equals("loopback"))) { + logAndToast(getString(R.string.connecting_to, url)); + appRtcClient.connectToRoom(url.toString()); + if (room != null && !room.equals("")) { + roomName.setText(room); + } else { + roomName.setText("loopback"); + } + } else { + logAndToast("Empty or missing room name!"); + finish(); + } + } else { + logAndToast(getString(R.string.missing_url)); + Log.wtf(TAG, "Didn't get any URL in intent!"); + finish(); + } + } + + public static class MenuBarFragment extends Fragment { + @Override + public View onCreateView( + LayoutInflater inflater, + ViewGroup container, + Bundle savedInstanceState) { + return inflater.inflate(R.layout.fragment_menubar, container, false); } - showGetRoomUI(); } @Override public void onPause() { super.onPause(); - vsv.onPause(); + videoView.onPause(); if (pc != null) { pc.stopVideoSource(); } @@ -142,56 +244,25 @@ public class AppRTCDemoActivity extends Activity @Override public void onResume() { super.onResume(); - vsv.onResume(); + videoView.onResume(); if (pc != null) { pc.startVideoSource(); } } @Override - public void onConfigurationChanged (Configuration newConfig) { - Point displaySize = new Point(); - getWindowManager().getDefaultDisplay().getSize(displaySize); - vsv.updateDisplaySize(displaySize); - super.onConfigurationChanged(newConfig); - } - - @Override protected void onDestroy() { - disconnectAndExit(); + disconnect(); super.onDestroy(); } - private void showGetRoomUI() { - final EditText roomInput = new EditText(this); - roomInput.setText("https://apprtc.appspot.com/?r="); - roomInput.setSelection(roomInput.getText().length()); - DialogInterface.OnClickListener listener = - new DialogInterface.OnClickListener() { - @Override - public void onClick(DialogInterface dialog, int which) { - abortUnless(which == DialogInterface.BUTTON_POSITIVE, "lolwat?"); - dialog.dismiss(); - connectToRoom(roomInput.getText().toString()); - } - }; - AlertDialog.Builder builder = new AlertDialog.Builder(this); - builder - .setMessage("Enter room URL").setView(roomInput) - .setPositiveButton("Go!", listener).show(); - } - - private void connectToRoom(String roomUrl) { - logAndToast("Connecting to room..."); - appRtcClient.connectToRoom(roomUrl); - } - - // Toggle visibility of the heads-up display. - private void toggleHUD() { - if (hudView.getVisibility() == View.VISIBLE) { - hudView.setVisibility(View.INVISIBLE); + private void updateVideoView() { + VideoRendererGui.update(remoteRender, 0, 0, 100, 100, scalingType); + if (iceConnected) { + VideoRendererGui.update(localRender, 70, 70, 28, 28, + ScalingType.SCALE_ASPECT_FIT); } else { - hudView.setVisibility(View.VISIBLE); + VideoRendererGui.update(localRender, 0, 0, 100, 100, scalingType); } } @@ -243,22 +314,29 @@ public class AppRTCDemoActivity extends Activity } // Disconnect from remote resources, dispose of local resources, and exit. - private void disconnectAndExit() { - synchronized (quit[0]) { - if (quit[0]) { - return; - } - quit[0] = true; - if (pc != null) { - pc.close(); - pc = null; - } - if (appRtcClient != null) { - appRtcClient.disconnect(); - appRtcClient = null; - } - finish(); + private void disconnect() { + if (appRtcClient != null) { + appRtcClient.disconnect(); + appRtcClient = null; + } + if (pc != null) { + pc.close(); + pc = null; } + finish(); + } + + private void disconnectWithMessage(String errorMessage) { + new AlertDialog.Builder(this) + .setTitle(getText(R.string.channel_error_title)) + .setMessage(errorMessage) + .setCancelable(false) + .setNeutralButton(R.string.ok, new DialogInterface.OnClickListener() { + public void onClick(DialogInterface dialog, int id) { + dialog.cancel(); + disconnect(); + } + }).create().show(); } // Poor-man's assert(): die with |msg| unless |condition| is true. @@ -289,40 +367,42 @@ public class AppRTCDemoActivity extends Activity logAndToast("Creating peer connection..."); pc = new PeerConnectionClient( this, localRender, remoteRender, appRtcParameters, this); + if (pc.isHDVideo()) { + setRequestedOrientation(ActivityInfo.SCREEN_ORIENTATION_LANDSCAPE); + } else { + setRequestedOrientation(ActivityInfo.SCREEN_ORIENTATION_UNSPECIFIED); + } { - final PeerConnectionClient finalPC = pc; final Runnable repeatedStatsLogger = new Runnable() { - public void run() { - synchronized (quit[0]) { - if (quit[0]) { - return; - } - final Runnable runnableThis = this; - if (hudView.getVisibility() == View.INVISIBLE) { - vsv.postDelayed(runnableThis, 1000); - return; - } - boolean success = finalPC.getStats(new StatsObserver() { - public void onComplete(final StatsReport[] reports) { - runOnUiThread(new Runnable() { - public void run() { - updateHUD(reports); - } - }); - for (StatsReport report : reports) { - Log.d(TAG, "Stats: " + report.toString()); - } - vsv.postDelayed(runnableThis, 1000); + public void run() { + if (pc == null) { + return; + } + final Runnable runnableThis = this; + if (hudView.getVisibility() == View.INVISIBLE) { + videoView.postDelayed(runnableThis, 1000); + return; + } + boolean success = pc.getStats(new StatsObserver() { + public void onComplete(final StatsReport[] reports) { + runOnUiThread(new Runnable() { + public void run() { + updateHUD(reports); + } + }); + for (StatsReport report : reports) { + Log.d(TAG, "Stats: " + report.toString()); } - }, null); - if (!success) { - throw new RuntimeException("getStats() return false!"); - } + videoView.postDelayed(runnableThis, 1000); + } + }, null); + if (!success) { + throw new RuntimeException("getStats() return false!"); } } - }; - vsv.postDelayed(repeatedStatsLogger, 1000); + }; + videoView.postDelayed(repeatedStatsLogger, 1000); } logAndToast("Waiting for remote connection..."); @@ -330,6 +410,9 @@ public class AppRTCDemoActivity extends Activity @Override public void onChannelOpen() { + if (pc == null) { + return; + } if (appRtcParameters.initiator) { logAndToast("Creating OFFER..."); // Create offer. Offer SDP will be sent to answering client in @@ -340,6 +423,9 @@ public class AppRTCDemoActivity extends Activity @Override public void onRemoteDescription(final SessionDescription sdp) { + if (pc == null) { + return; + } logAndToast("Received remote " + sdp.type + " ..."); pc.setRemoteDescription(sdp); if (!appRtcParameters.initiator) { @@ -352,19 +438,20 @@ public class AppRTCDemoActivity extends Activity @Override public void onRemoteIceCandidate(final IceCandidate candidate) { - pc.addRemoteIceCandidate(candidate); + if (pc != null) { + pc.addRemoteIceCandidate(candidate); + } } @Override public void onChannelClose() { logAndToast("Remote end hung up; dropping PeerConnection"); - disconnectAndExit(); + disconnect(); } @Override public void onChannelError(int code, String description) { - logAndToast("Channel error: " + code + ". " + description); - disconnectAndExit(); + disconnectWithMessage(description); } // -----Implementation of PeerConnectionClient.PeerConnectionEvents.--------- @@ -372,19 +459,35 @@ public class AppRTCDemoActivity extends Activity // All callbacks are invoked from UI thread. @Override public void onLocalDescription(final SessionDescription sdp) { - logAndToast("Sending " + sdp.type + " ..."); - appRtcClient.sendLocalDescription(sdp); + if (appRtcClient != null) { + logAndToast("Sending " + sdp.type + " ..."); + appRtcClient.sendLocalDescription(sdp); + } } @Override public void onIceCandidate(final IceCandidate candidate) { - appRtcClient.sendLocalIceCandidate(candidate); + if (appRtcClient != null) { + appRtcClient.sendLocalIceCandidate(candidate); + } } @Override public void onIceConnected() { logAndToast("ICE connected"); - VideoRendererGui.update(localRender, 70, 70, 28, 28, - VideoRendererGui.ScalingType.SCALE_ASPECT_FIT); + iceConnected = true; + updateVideoView(); + } + + @Override + public void onIceDisconnected() { + logAndToast("ICE disconnected"); + disconnect(); } + + @Override + public void onPeerConnectionError(String description) { + disconnectWithMessage(description); + } + } diff --git a/examples/android/src/org/appspot/apprtc/ConnectActivity.java b/examples/android/src/org/appspot/apprtc/ConnectActivity.java new file mode 100644 index 0000000..8f00c1c --- /dev/null +++ b/examples/android/src/org/appspot/apprtc/ConnectActivity.java @@ -0,0 +1,322 @@ +/* + * libjingle + * Copyright 2014, Google Inc. + * + * Redistribution and use in source and binary forms, with or without + * modification, are permitted provided that the following conditions are met: + * + * 1. Redistributions of source code must retain the above copyright notice, + * this list of conditions and the following disclaimer. + * 2. Redistributions in binary form must reproduce the above copyright notice, + * this list of conditions and the following disclaimer in the documentation + * and/or other materials provided with the distribution. + * 3. The name of the author may not be used to endorse or promote products + * derived from this software without specific prior written permission. + * + * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED + * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF + * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO + * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, + * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; + * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, + * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR + * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF + * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. + */ + +package org.appspot.apprtc; + +import android.app.Activity; +import android.app.AlertDialog; +import android.content.DialogInterface; +import android.content.Intent; +import android.content.SharedPreferences; +import android.net.Uri; +import android.os.Bundle; +import android.preference.PreferenceManager; +import android.util.Log; +import android.view.KeyEvent; +import android.view.Menu; +import android.view.MenuItem; +import android.view.View; +import android.view.View.OnClickListener; +import android.view.inputmethod.EditorInfo; +import android.webkit.URLUtil; +import android.widget.AdapterView; +import android.widget.ArrayAdapter; +import android.widget.EditText; +import android.widget.ImageButton; +import android.widget.ListView; +import android.widget.TextView; + +import java.util.ArrayList; + +import org.json.JSONArray; +import org.json.JSONException; +import org.webrtc.MediaCodecVideoEncoder; + + +/** + * Handles the initial setup where the user selects which room to join. + */ +public class ConnectActivity extends Activity { + + private static final String TAG = "ConnectActivity"; + private ImageButton addRoomButton; + private ImageButton removeRoomButton; + private ImageButton connectButton; + private ImageButton connectLoopbackButton; + private EditText roomEditText; + private ListView roomListView; + private SharedPreferences sharedPref; + private String keyprefUrl; + private String keyprefResolution; + private String keyprefFps; + private String keyprefRoom; + private String keyprefRoomList; + private ArrayList<String> roomList; + private ArrayAdapter<String> adapter; + + @Override + public void onCreate(Bundle savedInstanceState) { + super.onCreate(savedInstanceState); + + // Get setting keys. + PreferenceManager.setDefaultValues(this, R.xml.preferences, false); + sharedPref = PreferenceManager.getDefaultSharedPreferences(this); + keyprefUrl = getString(R.string.pref_url_key); + keyprefResolution = getString(R.string.pref_resolution_key); + keyprefFps = getString(R.string.pref_fps_key); + keyprefRoom = getString(R.string.pref_room_key); + keyprefRoomList = getString(R.string.pref_room_list_key); + + // If an implicit VIEW intent is launching the app, go directly to that URL. + final Intent intent = getIntent(); + if ("android.intent.action.VIEW".equals(intent.getAction())) { + connectToRoom(intent.getData().toString()); + return; + } + + setContentView(R.layout.activity_connect); + + roomEditText = (EditText) findViewById(R.id.room_edittext); + roomEditText.setOnEditorActionListener( + new TextView.OnEditorActionListener() { + @Override + public boolean onEditorAction( + TextView textView, int i, KeyEvent keyEvent) { + if (i == EditorInfo.IME_ACTION_DONE) { + addRoomButton.performClick(); + return true; + } + return false; + } + }); + roomEditText.requestFocus(); + + roomListView = (ListView) findViewById(R.id.room_listview); + roomListView.setChoiceMode(ListView.CHOICE_MODE_SINGLE); + + addRoomButton = (ImageButton) findViewById(R.id.add_room_button); + addRoomButton.setOnClickListener(addRoomListener); + removeRoomButton = (ImageButton) findViewById(R.id.remove_room_button); + removeRoomButton.setOnClickListener(removeRoomListener); + connectButton = (ImageButton) findViewById(R.id.connect_button); + connectButton.setOnClickListener(connectListener); + connectLoopbackButton = + (ImageButton) findViewById(R.id.connect_loopback_button); + connectLoopbackButton.setOnClickListener(connectListener); + } + + @Override + public boolean onCreateOptionsMenu(Menu menu) { + getMenuInflater().inflate(R.menu.connect_menu, menu); + return true; + } + + @Override + public boolean onOptionsItemSelected(MenuItem item) { + // Handle presses on the action bar items. + if (item.getItemId() == R.id.action_settings) { + Intent intent = new Intent(this, SettingsActivity.class); + startActivity(intent); + return true; + } else { + return super.onOptionsItemSelected(item); + } + } + + @Override + public void onPause() { + super.onPause(); + String room = roomEditText.getText().toString(); + String roomListJson = new JSONArray(roomList).toString(); + SharedPreferences.Editor editor = sharedPref.edit(); + editor.putString(keyprefRoom, room); + editor.putString(keyprefRoomList, roomListJson); + editor.commit(); + } + + @Override + public void onResume() { + super.onResume(); + String room = sharedPref.getString(keyprefRoom, ""); + roomEditText.setText(room); + roomList = new ArrayList<String>(); + String roomListJson = sharedPref.getString(keyprefRoomList, null); + if (roomListJson != null) { + try { + JSONArray jsonArray = new JSONArray(roomListJson); + for (int i = 0; i < jsonArray.length(); i++) { + roomList.add(jsonArray.get(i).toString()); + } + } catch (JSONException e) { + Log.e(TAG, "Failed to load room list: " + e.toString()); + } + } + adapter = new ArrayAdapter<String>( + this, android.R.layout.simple_list_item_1, roomList); + roomListView.setAdapter(adapter); + if (adapter.getCount() > 0) { + roomListView.requestFocus(); + roomListView.setItemChecked(0, true); + } + } + + private final OnClickListener connectListener = new OnClickListener() { + @Override + public void onClick(View view) { + boolean loopback = false; + if (view.getId() == R.id.connect_loopback_button) { + loopback = true; + } + String url = sharedPref.getString(keyprefUrl, + getString(R.string.pref_url_default)); + if (loopback) { + url += "/?debug=loopback"; + } else { + String roomName = getSelectedItem(); + if (roomName == null) { + roomName = roomEditText.getText().toString(); + } + url += "/?r=" + roomName; + } + String parametersResolution = null; + String parametersFps = null; + // Add video resolution constraints. + String resolution = sharedPref.getString(keyprefResolution, + getString(R.string.pref_resolution_default)); + String[] dimensions = resolution.split("[ x]+"); + if (dimensions.length == 2) { + try { + int maxWidth = Integer.parseInt(dimensions[0]); + int maxHeight = Integer.parseInt(dimensions[1]); + if (maxWidth > 0 && maxHeight > 0) { + parametersResolution = "minHeight=" + maxHeight + ",maxHeight=" + + maxHeight + ",minWidth=" + maxWidth + ",maxWidth=" + maxWidth; + } + } catch (NumberFormatException e) { + Log.e(TAG, "Wrong video resolution setting: " + resolution); + } + } + // Add camera fps constraints. + String fps = sharedPref.getString(keyprefFps, + getString(R.string.pref_fps_default)); + String[] fpsValues = fps.split("[ x]+"); + if (fpsValues.length == 2) { + try { + int cameraFps = Integer.parseInt(fpsValues[0]); + if (cameraFps > 0) { + parametersFps = "minFrameRate=" + cameraFps + + ",maxFrameRate=" + cameraFps; + } + } catch (NumberFormatException e) { + Log.e(TAG, "Wrong camera fps setting: " + fps); + } + } + // Modify connection URL. + if (parametersResolution != null || parametersFps != null) { + url += "&video="; + if (parametersResolution != null) { + url += parametersResolution; + if (parametersFps != null) { + url += ","; + } + } + if (parametersFps != null) { + url += parametersFps; + } + } else { + if (MediaCodecVideoEncoder.isPlatformSupported()) { + url += "&hd=true"; + } + } + // TODO(kjellander): Add support for custom parameters to the URL. + connectToRoom(url); + } + }; + + private void connectToRoom(String roomUrl) { + if (validateUrl(roomUrl)) { + Uri url = Uri.parse(roomUrl); + Intent intent = new Intent(this, AppRTCDemoActivity.class); + intent.setData(url); + startActivity(intent); + } + } + + private boolean validateUrl(String url) { + if (URLUtil.isHttpsUrl(url) || URLUtil.isHttpUrl(url)) + return true; + + new AlertDialog.Builder(this) + .setTitle(getText(R.string.invalid_url_title)) + .setMessage(getString(R.string.invalid_url_text, url)) + .setCancelable(false) + .setNeutralButton(R.string.ok, new DialogInterface.OnClickListener() { + public void onClick(DialogInterface dialog, int id) { + dialog.cancel(); + } + }).create().show(); + return false; + } + + private final OnClickListener addRoomListener = new OnClickListener() { + @Override + public void onClick(View view) { + String newRoom = roomEditText.getText().toString(); + if (newRoom.length() > 0 && !roomList.contains(newRoom)) { + adapter.add(newRoom); + adapter.notifyDataSetChanged(); + } + } + }; + + private final OnClickListener removeRoomListener = new OnClickListener() { + @Override + public void onClick(View view) { + String selectedRoom = getSelectedItem(); + if (selectedRoom != null) { + adapter.remove(selectedRoom); + adapter.notifyDataSetChanged(); + } + } + }; + + private String getSelectedItem() { + int position = AdapterView.INVALID_POSITION; + if (roomListView.getCheckedItemCount() > 0 && adapter.getCount() > 0) { + position = roomListView.getCheckedItemPosition(); + if (position >= adapter.getCount()) { + position = AdapterView.INVALID_POSITION; + } + } + if (position != AdapterView.INVALID_POSITION) { + return adapter.getItem(position); + } else { + return null; + } + } + +} diff --git a/examples/android/src/org/appspot/apprtc/GAEChannelClient.java b/examples/android/src/org/appspot/apprtc/GAEChannelClient.java index bcc06ab..5fd0a54 100644 --- a/examples/android/src/org/appspot/apprtc/GAEChannelClient.java +++ b/examples/android/src/org/appspot/apprtc/GAEChannelClient.java @@ -51,15 +51,12 @@ public class GAEChannelClient { /** * Callback interface for messages delivered on the Google AppEngine channel. - * - * Methods are guaranteed to be invoked on the UI thread of |activity| passed - * to GAEChannelClient's constructor. */ public interface GAEMessageHandler { public void onOpen(); - public void onMessage(String data); + public void onMessage(final String data); public void onClose(); - public void onError(int code, String description); + public void onError(final int code, final String description); } /** Asynchronously open an AppEngine channel. */ @@ -83,8 +80,7 @@ public class GAEChannelClient { ", desc: " + description); } }); - proxyingMessageHandler = - new ProxyingMessageHandler(activity, handler, token); + proxyingMessageHandler = new ProxyingMessageHandler(handler, token); webView.addJavascriptInterface( proxyingMessageHandler, "androidMessageHandler"); webView.loadUrl("file:///android_asset/channel.html"); @@ -102,72 +98,52 @@ public class GAEChannelClient { } // Helper class for proxying callbacks from the Java<->JS interaction - // (private, background) thread to the Activity's UI thread. + // (private, background) thread. private static class ProxyingMessageHandler { - private final Activity activity; private final GAEMessageHandler handler; - private final boolean[] disconnected = { false }; + private boolean disconnected = false; private final String token; - public - ProxyingMessageHandler(Activity activity, GAEMessageHandler handler, - String token) { - this.activity = activity; + public ProxyingMessageHandler(GAEMessageHandler handler, String token) { this.handler = handler; this.token = token; } public void disconnect() { - disconnected[0] = true; + disconnected = true; } - private boolean disconnected() { - return disconnected[0]; - } - - @JavascriptInterface public String getToken() { + @JavascriptInterface + public String getToken() { return token; } - @JavascriptInterface public void onOpen() { - activity.runOnUiThread(new Runnable() { - public void run() { - if (!disconnected()) { - handler.onOpen(); - } - } - }); + @JavascriptInterface + public void onOpen() { + if (!disconnected) { + handler.onOpen(); + } } - @JavascriptInterface public void onMessage(final String data) { - activity.runOnUiThread(new Runnable() { - public void run() { - if (!disconnected()) { - handler.onMessage(data); - } - } - }); + @JavascriptInterface + public void onMessage(final String data) { + if (!disconnected) { + handler.onMessage(data); + } } - @JavascriptInterface public void onClose() { - activity.runOnUiThread(new Runnable() { - public void run() { - if (!disconnected()) { - handler.onClose(); - } - } - }); + @JavascriptInterface + public void onClose() { + if (!disconnected) { + handler.onClose(); + } } - @JavascriptInterface public void onError( - final int code, final String description) { - activity.runOnUiThread(new Runnable() { - public void run() { - if (!disconnected()) { - handler.onError(code, description); - } - } - }); + @JavascriptInterface + public void onError(final int code, final String description) { + if (!disconnected) { + handler.onError(code, description); + } } } } diff --git a/examples/android/src/org/appspot/apprtc/GAERTCClient.java b/examples/android/src/org/appspot/apprtc/GAERTCClient.java index 1d1f817..c3d9564 100644 --- a/examples/android/src/org/appspot/apprtc/GAERTCClient.java +++ b/examples/android/src/org/appspot/apprtc/GAERTCClient.java @@ -29,7 +29,6 @@ package org.appspot.apprtc; import android.app.Activity; import android.os.AsyncTask; import android.util.Log; -import android.webkit.JavascriptInterface; import org.json.JSONArray; import org.json.JSONException; @@ -41,7 +40,6 @@ import org.webrtc.SessionDescription; import java.io.IOException; import java.io.InputStream; -import java.net.HttpURLConnection; import java.net.URL; import java.net.URLConnection; import java.util.LinkedList; @@ -83,11 +81,6 @@ public class GAERTCClient implements AppRTCClient { */ @Override public void connectToRoom(String url) { - while (url.indexOf('?') < 0) { - // Keep redirecting until we get a room number. - (new RedirectResolver()).execute(url); - return; // RedirectResolver above calls us back with the next URL. - } (new RoomParameterGetter()).execute(url); } @@ -97,6 +90,7 @@ public class GAERTCClient implements AppRTCClient { @Override public void disconnect() { if (channelClient != null) { + Log.d(TAG, "Closing GAE Channel."); sendMessage("{\"type\": \"bye\"}"); channelClient.close(); channelClient = null; @@ -151,68 +145,36 @@ public class GAERTCClient implements AppRTCClient { } } - // Load the given URL and return the value of the Location header of the - // resulting 302 response. If the result is not a 302, throws. - private class RedirectResolver extends AsyncTask<String, Void, String> { - @Override - protected String doInBackground(String... urls) { - if (urls.length != 1) { - throw new RuntimeException("Must be called with a single URL"); - } - try { - return followRedirect(urls[0]); - } catch (IOException e) { - throw new RuntimeException(e); - } - } - - @Override - protected void onPostExecute(String url) { - connectToRoom(url); - } - - private String followRedirect(String url) throws IOException { - HttpURLConnection connection = (HttpURLConnection) - new URL(url).openConnection(); - connection.setInstanceFollowRedirects(false); - int code = connection.getResponseCode(); - if (code != HttpURLConnection.HTTP_MOVED_TEMP) { - throw new IOException("Unexpected response: " + code + " for " + url + - ", with contents: " + drainStream(connection.getInputStream())); - } - int n = 0; - String name, value; - while ((name = connection.getHeaderFieldKey(n)) != null) { - value = connection.getHeaderField(n); - if (name.equals("Location")) { - return value; - } - ++n; - } - throw new IOException("Didn't find Location header!"); - } - } - // AsyncTask that converts an AppRTC room URL into the set of signaling // parameters to use with that room. private class RoomParameterGetter extends AsyncTask<String, Void, AppRTCSignalingParameters> { + private Exception exception = null; + @Override protected AppRTCSignalingParameters doInBackground(String... urls) { if (urls.length != 1) { - throw new RuntimeException("Must be called with a single URL"); + exception = new RuntimeException("Must be called with a single URL"); + return null; } try { + exception = null; return getParametersForRoomUrl(urls[0]); } catch (JSONException e) { - throw new RuntimeException(e); + exception = e; } catch (IOException e) { - throw new RuntimeException(e); + exception = e; } + return null; } @Override protected void onPostExecute(AppRTCSignalingParameters params) { + if (exception != null) { + Log.e(TAG, "Room connection error: " + exception.toString()); + events.onChannelError(0, exception.getMessage()); + return; + } channelClient = new GAEChannelClient(activity, channelToken, gaeHandler); synchronized (sendQueue) { @@ -445,42 +407,67 @@ public class GAERTCClient implements AppRTCClient { // Implementation detail: handler for receiving GAE messages and dispatching // them appropriately. private class GAEHandler implements GAEChannelClient.GAEMessageHandler { - @JavascriptInterface public void onOpen() { - events.onChannelOpen(); + private boolean channelOpen = false; + + public void onOpen() { + activity.runOnUiThread(new Runnable() { + public void run() { + events.onChannelOpen(); + channelOpen = true; + } + }); } - @JavascriptInterface public void onMessage(String msg) { + public void onMessage(final String msg) { Log.d(TAG, "RECEIVE: " + msg); - try { - JSONObject json = new JSONObject(msg); - String type = (String) json.get("type"); - if (type.equals("candidate")) { - IceCandidate candidate = new IceCandidate( - (String) json.get("id"), - json.getInt("label"), - (String) json.get("candidate")); - events.onRemoteIceCandidate(candidate); - } else if (type.equals("answer") || type.equals("offer")) { - SessionDescription sdp = new SessionDescription( - SessionDescription.Type.fromCanonicalForm(type), - (String)json.get("sdp")); - events.onRemoteDescription(sdp); - } else if (type.equals("bye")) { - events.onChannelClose(); - } else { - throw new RuntimeException("Unexpected message: " + msg); + activity.runOnUiThread(new Runnable() { + public void run() { + if (!channelOpen) { + return; + } + try { + JSONObject json = new JSONObject(msg); + String type = (String) json.get("type"); + if (type.equals("candidate")) { + IceCandidate candidate = new IceCandidate( + (String) json.get("id"), + json.getInt("label"), + (String) json.get("candidate")); + events.onRemoteIceCandidate(candidate); + } else if (type.equals("answer") || type.equals("offer")) { + SessionDescription sdp = new SessionDescription( + SessionDescription.Type.fromCanonicalForm(type), + (String)json.get("sdp")); + events.onRemoteDescription(sdp); + } else if (type.equals("bye")) { + events.onChannelClose(); + } else { + events.onChannelError(1, "Unexpected channel message: " + msg); + } + } catch (JSONException e) { + events.onChannelError(1, "Channel message JSON parsing error: " + + e.toString()); + } } - } catch (JSONException e) { - throw new RuntimeException(e); - } + }); } - @JavascriptInterface public void onClose() { - events.onChannelClose(); + public void onClose() { + activity.runOnUiThread(new Runnable() { + public void run() { + events.onChannelClose(); + channelOpen = false; + } + }); } - @JavascriptInterface public void onError(int code, String description) { - events.onChannelError(code, description); + public void onError(final int code, final String description) { + activity.runOnUiThread(new Runnable() { + public void run() { + events.onChannelError(code, description); + channelOpen = false; + } + }); } } diff --git a/examples/android/src/org/appspot/apprtc/PeerConnectionClient.java b/examples/android/src/org/appspot/apprtc/PeerConnectionClient.java index 0376000..9c917bb 100644 --- a/examples/android/src/org/appspot/apprtc/PeerConnectionClient.java +++ b/examples/android/src/org/appspot/apprtc/PeerConnectionClient.java @@ -37,6 +37,7 @@ import org.webrtc.MediaConstraints; import org.webrtc.MediaStream; import org.webrtc.MediaStreamTrack; import org.webrtc.PeerConnection; +import org.webrtc.MediaConstraints.KeyValuePair; import org.webrtc.PeerConnection.IceConnectionState; import org.webrtc.PeerConnectionFactory; import org.webrtc.SdpObserver; @@ -60,13 +61,17 @@ public class PeerConnectionClient { private boolean videoSourceStopped; private final PCObserver pcObserver = new PCObserver(); private final SDPObserver sdpObserver = new SDPObserver(); + private final VideoRenderer.Callbacks localRender; private final VideoRenderer.Callbacks remoteRender; private LinkedList<IceCandidate> queuedRemoteCandidates = new LinkedList<IceCandidate>(); private MediaConstraints sdpMediaConstraints; + private MediaConstraints videoConstraints; private PeerConnectionEvents events; private boolean isInitiator; + private boolean useFrontFacingCamera = true; private SessionDescription localSdp = null; // either offer or answer SDP + private MediaStream videoMediaStream = null; public PeerConnectionClient( Activity activity, @@ -75,6 +80,7 @@ public class PeerConnectionClient { AppRTCSignalingParameters appRtcParameters, PeerConnectionEvents events) { this.activity = activity; + this.localRender = localRender; this.remoteRender = remoteRender; this.events = events; isInitiator = appRtcParameters.initiator; @@ -84,9 +90,9 @@ public class PeerConnectionClient { "OfferToReceiveAudio", "true")); sdpMediaConstraints.mandatory.add(new MediaConstraints.KeyValuePair( "OfferToReceiveVideo", "true")); + videoConstraints = appRtcParameters.videoConstraints; factory = new PeerConnectionFactory(); - MediaConstraints pcConstraints = appRtcParameters.pcConstraints; pcConstraints.optional.add( new MediaConstraints.KeyValuePair("RtpDataChannels", "true")); @@ -101,23 +107,47 @@ public class PeerConnectionClient { // EnumSet.of(Logging.TraceLevel.TRACE_ALL), // Logging.Severity.LS_SENSITIVE); - Log.d(TAG, "Creating local video source"); - MediaStream lMS = factory.createLocalMediaStream("ARDAMS"); - if (appRtcParameters.videoConstraints != null) { - VideoCapturer capturer = getVideoCapturer(); - videoSource = factory.createVideoSource( - capturer, appRtcParameters.videoConstraints); - VideoTrack videoTrack = - factory.createVideoTrack("ARDAMSv0", videoSource); - videoTrack.addRenderer(new VideoRenderer(localRender)); - lMS.addTrack(videoTrack); + if (videoConstraints != null) { + videoMediaStream = factory.createLocalMediaStream("ARDAMSVideo"); + videoMediaStream.addTrack(createVideoTrack(useFrontFacingCamera)); + pc.addStream(videoMediaStream); } + if (appRtcParameters.audioConstraints != null) { + MediaStream lMS = factory.createLocalMediaStream("ARDAMSAudio"); lMS.addTrack(factory.createAudioTrack( "ARDAMSa0", factory.createAudioSource(appRtcParameters.audioConstraints))); + pc.addStream(lMS); + } + } + + public boolean isHDVideo() { + if (videoConstraints == null) { + return false; + } + int minWidth = 0; + int minHeight = 0; + for (KeyValuePair keyValuePair : videoConstraints.mandatory) { + if (keyValuePair.getKey().equals("minWidth")) { + try { + minWidth = Integer.parseInt(keyValuePair.getValue()); + } catch (NumberFormatException e) { + Log.e(TAG, "Can not parse video width from video constraints"); + } + } else if (keyValuePair.getKey().equals("minHeight")) { + try { + minHeight = Integer.parseInt(keyValuePair.getValue()); + } catch (NumberFormatException e) { + Log.e(TAG, "Can not parse video height from video constraints"); + } + } + } + if (minWidth * minHeight >= 1280 * 720) { + return true; + } else { + return false; } - pc.addStream(lMS, new MediaConstraints()); } public boolean getStats(StatsObserver observer, MediaStreamTrack track) { @@ -125,33 +155,58 @@ public class PeerConnectionClient { } public void createOffer() { - isInitiator = true; - pc.createOffer(sdpObserver, sdpMediaConstraints); + activity.runOnUiThread(new Runnable() { + public void run() { + if (pc != null) { + isInitiator = true; + pc.createOffer(sdpObserver, sdpMediaConstraints); + } + } + }); } public void createAnswer() { - isInitiator = false; - pc.createAnswer(sdpObserver, sdpMediaConstraints); + activity.runOnUiThread(new Runnable() { + public void run() { + if (pc != null) { + isInitiator = false; + pc.createAnswer(sdpObserver, sdpMediaConstraints); + } + } + }); } - public void addRemoteIceCandidate(IceCandidate candidate) { - if (queuedRemoteCandidates != null) { - queuedRemoteCandidates.add(candidate); - } else { - pc.addIceCandidate(candidate); - } + public void addRemoteIceCandidate(final IceCandidate candidate) { + activity.runOnUiThread(new Runnable() { + public void run() { + if (pc != null) { + if (queuedRemoteCandidates != null) { + queuedRemoteCandidates.add(candidate); + } else { + pc.addIceCandidate(candidate); + } + } + } + }); } - public void setRemoteDescription(SessionDescription sdp) { - SessionDescription sdpISAC = new SessionDescription( - sdp.type, preferISAC(sdp.description)); - Log.d(TAG, "Set remote SDP"); - pc.setRemoteDescription(sdpObserver, sdpISAC); + public void setRemoteDescription(final SessionDescription sdp) { + activity.runOnUiThread(new Runnable() { + public void run() { + if (pc != null) { + SessionDescription sdpISAC = new SessionDescription( + sdp.type, preferISAC(sdp.description)); + Log.d(TAG, "Set remote SDP"); + pc.setRemoteDescription(sdpObserver, sdpISAC); + } + } + }); } public void stopVideoSource() { if (videoSource != null) { + Log.d(TAG, "Stop video source."); videoSource.stop(); videoSourceStopped = true; } @@ -159,24 +214,30 @@ public class PeerConnectionClient { public void startVideoSource() { if (videoSource != null && videoSourceStopped) { + Log.d(TAG, "Restart video source."); videoSource.restart(); videoSourceStopped = false; } } public void close() { - if (pc != null) { - pc.dispose(); - pc = null; - } - if (videoSource != null) { - videoSource.dispose(); - videoSource = null; - } - if (factory != null) { - factory.dispose(); - factory = null; - } + activity.runOnUiThread(new Runnable() { + public void run() { + Log.d(TAG, "Closing peer connection."); + if (pc != null) { + pc.dispose(); + pc = null; + } + if (videoSource != null) { + videoSource.dispose(); + videoSource = null; + } + if (factory != null) { + factory.dispose(); + factory = null; + } + } + }); } /** @@ -198,15 +259,38 @@ public class PeerConnectionClient { * CONNECTED). */ public void onIceConnected(); + + /** + * Callback fired once connection is closed (IceConnectionState is + * DISCONNECTED). + */ + public void onIceDisconnected(); + + /** + * Callback fired once peer connection error happened. + */ + public void onPeerConnectionError(String description); + } + + private void reportError(final String errorMessage) { + activity.runOnUiThread(new Runnable() { + public void run() { + events.onPeerConnectionError(errorMessage); + } + }); } // Cycle through likely device names for the camera and return the first // capturer that works, or crash if none do. - private VideoCapturer getVideoCapturer() { + private VideoCapturer getVideoCapturer(boolean useFrontFacing) { String[] cameraFacing = { "front", "back" }; - int[] cameraIndex = { 0, 1 }; - int[] cameraOrientation = { 0, 90, 180, 270 }; + if (!useFrontFacing) { + cameraFacing[0] = "back"; + cameraFacing[1] = "front"; + } for (String facing : cameraFacing) { + int[] cameraIndex = { 0, 1 }; + int[] cameraOrientation = { 0, 90, 180, 270 }; for (int index : cameraIndex) { for (int orientation : cameraOrientation) { String name = "Camera " + index + ", Facing " + facing + @@ -219,13 +303,30 @@ public class PeerConnectionClient { } } } - throw new RuntimeException("Failed to open capturer"); + reportError("Failed to open capturer"); + return null; + } + + private VideoTrack createVideoTrack(boolean frontFacing) { + VideoCapturer capturer = getVideoCapturer(frontFacing); + if (videoSource != null) { + videoSource.stop(); + videoSource.dispose(); + } + + videoSource = factory.createVideoSource( + capturer, videoConstraints); + String trackExtension = frontFacing ? "frontFacing" : "backFacing"; + VideoTrack videoTrack = + factory.createVideoTrack("ARDAMSv0" + trackExtension, videoSource); + videoTrack.addRenderer(new VideoRenderer(localRender)); + return videoTrack; } // Poor-man's assert(): die with |msg| unless |condition| is true. - private static void abortUnless(boolean condition, String msg) { + private void abortUnless(boolean condition, String msg) { if (!condition) { - throw new RuntimeException(msg); + reportError(msg); } } @@ -285,24 +386,58 @@ public class PeerConnectionClient { queuedRemoteCandidates = null; } + public void switchCamera() { + if (videoConstraints == null) + return; // No video is sent. + + if (pc.signalingState() != PeerConnection.SignalingState.STABLE) { + Log.e(TAG, "Switching camera during negotiation is not handled."); + return; + } + + pc.removeStream(videoMediaStream); + VideoTrack currentTrack = videoMediaStream.videoTracks.get(0); + videoMediaStream.removeTrack(currentTrack); + + String trackId = currentTrack.id(); + // On Android, there can only be one camera open at the time and we + // need to release our implicit references to the videoSource before the + // PeerConnectionFactory is released. Since createVideoTrack creates a new + // videoSource and frees the old one, we need to release the track here. + currentTrack.dispose(); + + useFrontFacingCamera = !useFrontFacingCamera; + VideoTrack newTrack = createVideoTrack(useFrontFacingCamera); + videoMediaStream.addTrack(newTrack); + pc.addStream(videoMediaStream); + + SessionDescription remoteDesc = pc.getRemoteDescription(); + if (localSdp == null || remoteDesc == null) { + Log.d(TAG, "Switching camera before the negotiation started."); + return; + } + + localSdp = new SessionDescription(localSdp.type, + localSdp.description.replaceAll(trackId, newTrack.id())); + + if (isInitiator) { + pc.setLocalDescription(new SwitchCameraSdbObserver(), localSdp); + pc.setRemoteDescription(new SwitchCameraSdbObserver(), remoteDesc); + } else { + pc.setRemoteDescription(new SwitchCameraSdbObserver(), remoteDesc); + pc.setLocalDescription(new SwitchCameraSdbObserver(), localSdp); + } + } + // Implementation detail: observe ICE & stream changes and react accordingly. private class PCObserver implements PeerConnection.Observer { @Override public void onIceCandidate(final IceCandidate candidate){ activity.runOnUiThread(new Runnable() { - public void run() { - events.onIceCandidate(candidate); - } - }); - } - - @Override - public void onError(){ - activity.runOnUiThread(new Runnable() { - public void run() { - throw new RuntimeException("PeerConnection error!"); - } - }); + public void run() { + events.onIceCandidate(candidate); + } + }); } @Override @@ -321,47 +456,50 @@ public class PeerConnectionClient { events.onIceConnected(); } }); + } else if (newState == IceConnectionState.DISCONNECTED) { + activity.runOnUiThread(new Runnable() { + public void run() { + events.onIceDisconnected(); + } + }); + } else if (newState == IceConnectionState.FAILED) { + reportError("ICE connection failed."); } } @Override public void onIceGatheringChange( - PeerConnection.IceGatheringState newState) { + PeerConnection.IceGatheringState newState) { } @Override public void onAddStream(final MediaStream stream){ activity.runOnUiThread(new Runnable() { - public void run() { - abortUnless(stream.audioTracks.size() <= 1 && - stream.videoTracks.size() <= 1, - "Weird-looking stream: " + stream); - if (stream.videoTracks.size() == 1) { - stream.videoTracks.get(0).addRenderer( - new VideoRenderer(remoteRender)); - } + public void run() { + abortUnless(stream.audioTracks.size() <= 1 && + stream.videoTracks.size() <= 1, + "Weird-looking stream: " + stream); + if (stream.videoTracks.size() == 1) { + stream.videoTracks.get(0).addRenderer( + new VideoRenderer(remoteRender)); } - }); + } + }); } @Override public void onRemoveStream(final MediaStream stream){ activity.runOnUiThread(new Runnable() { - public void run() { - stream.videoTracks.get(0).dispose(); - } - }); + public void run() { + stream.videoTracks.get(0).dispose(); + } + }); } @Override public void onDataChannel(final DataChannel dc) { - activity.runOnUiThread(new Runnable() { - public void run() { - throw new RuntimeException( - "AppRTC doesn't use data channels, but got: " + dc.label() + - " anyway!"); - } - }); + reportError("AppRTC doesn't use data channels, but got: " + dc.label() + + " anyway!"); } @Override @@ -381,65 +519,82 @@ public class PeerConnectionClient { origSdp.type, preferISAC(origSdp.description)); localSdp = sdp; activity.runOnUiThread(new Runnable() { - public void run() { + public void run() { + if (pc != null) { Log.d(TAG, "Set local SDP from " + sdp.type); pc.setLocalDescription(sdpObserver, sdp); } - }); + } + }); } @Override public void onSetSuccess() { activity.runOnUiThread(new Runnable() { - public void run() { - if (isInitiator) { - // For offering peer connection we first create offer and set - // local SDP, then after receiving answer set remote SDP. - if (pc.getRemoteDescription() == null) { - // We've just set our local SDP so time to send it. - Log.d(TAG, "Local SDP set succesfully"); - events.onLocalDescription(localSdp); - } else { - // We've just set remote description, - // so drain remote ICE candidates. - Log.d(TAG, "Remote SDP set succesfully"); - drainRemoteCandidates(); - } + public void run() { + if (pc == null) { + return; + } + if (isInitiator) { + // For offering peer connection we first create offer and set + // local SDP, then after receiving answer set remote SDP. + if (pc.getRemoteDescription() == null) { + // We've just set our local SDP so time to send it. + Log.d(TAG, "Local SDP set succesfully"); + events.onLocalDescription(localSdp); + } else { + // We've just set remote description, + // so drain remote ICE candidates. + Log.d(TAG, "Remote SDP set succesfully"); + drainRemoteCandidates(); + } + } else { + // For answering peer connection we set remote SDP and then + // create answer and set local SDP. + if (pc.getLocalDescription() != null) { + // We've just set our local SDP so time to send it and drain + // remote ICE candidates. + Log.d(TAG, "Local SDP set succesfully"); + events.onLocalDescription(localSdp); + drainRemoteCandidates(); } else { - // For answering peer connection we set remote SDP and then - // create answer and set local SDP. - if (pc.getLocalDescription() != null) { - // We've just set our local SDP so time to send it and drain - // remote ICE candidates. - Log.d(TAG, "Local SDP set succesfully"); - events.onLocalDescription(localSdp); - drainRemoteCandidates(); - } else { - // We've just set remote SDP - do nothing for now - - // answer will be created soon. - Log.d(TAG, "Remote SDP set succesfully"); - } + // 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) { - activity.runOnUiThread(new Runnable() { - public void run() { - throw new RuntimeException("createSDP error: " + error); - } - }); + reportError("createSDP error: " + error); } @Override public void onSetFailure(final String error) { - activity.runOnUiThread(new Runnable() { - public void run() { - throw new RuntimeException("setSDP error: " + error); - } - }); + reportError("setSDP error: " + error); + } + } + + private class SwitchCameraSdbObserver implements SdpObserver { + @Override + public void onCreateSuccess(SessionDescription sdp) { + } + + @Override + public void onSetSuccess() { + Log.d(TAG, "Camera switch SDP set succesfully"); + } + + @Override + public void onCreateFailure(final String error) { + } + + @Override + public void onSetFailure(final String error) { + reportError("setSDP error while switching camera: " + error); } } } diff --git a/examples/android/src/org/appspot/apprtc/SettingsActivity.java b/examples/android/src/org/appspot/apprtc/SettingsActivity.java new file mode 100644 index 0000000..2354ceb --- /dev/null +++ b/examples/android/src/org/appspot/apprtc/SettingsActivity.java @@ -0,0 +1,92 @@ +/* + * libjingle + * Copyright 2014, Google Inc. + * + * Redistribution and use in source and binary forms, with or without + * modification, are permitted provided that the following conditions are met: + * + * 1. Redistributions of source code must retain the above copyright notice, + * this list of conditions and the following disclaimer. + * 2. Redistributions in binary form must reproduce the above copyright notice, + * this list of conditions and the following disclaimer in the documentation + * and/or other materials provided with the distribution. + * 3. The name of the author may not be used to endorse or promote products + * derived from this software without specific prior written permission. + * + * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED + * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF + * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO + * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, + * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; + * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, + * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR + * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF + * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. + */ + +package org.appspot.apprtc; + +import android.app.Activity; +import android.content.SharedPreferences; +import android.content.SharedPreferences.OnSharedPreferenceChangeListener; +import android.os.Bundle; +import android.preference.Preference; + +public class SettingsActivity extends Activity + implements OnSharedPreferenceChangeListener{ + private SettingsFragment settingsFragment; + private String keyprefUrl; + private String keyprefResolution; + private String keyprefFps; + + @Override + protected void onCreate(Bundle savedInstanceState) { + super.onCreate(savedInstanceState); + keyprefUrl = getString(R.string.pref_url_key); + keyprefResolution = getString(R.string.pref_resolution_key); + keyprefFps = getString(R.string.pref_fps_key); + + // Display the fragment as the main content. + settingsFragment = new SettingsFragment(); + getFragmentManager().beginTransaction() + .replace(android.R.id.content, settingsFragment) + .commit(); + } + + @Override + protected void onResume() { + super.onResume(); + // Set summary to be the user-description for the selected value + SharedPreferences sharedPreferences = + settingsFragment.getPreferenceScreen().getSharedPreferences(); + sharedPreferences.registerOnSharedPreferenceChangeListener(this); + updateSummary(sharedPreferences, keyprefUrl); + updateSummary(sharedPreferences, keyprefResolution); + updateSummary(sharedPreferences, keyprefFps); + } + + @Override + protected void onPause() { + super.onPause(); + SharedPreferences sharedPreferences = + settingsFragment.getPreferenceScreen().getSharedPreferences(); + sharedPreferences.unregisterOnSharedPreferenceChangeListener(this); + } + + @Override + public void onSharedPreferenceChanged(SharedPreferences sharedPreferences, + String key) { + if (key.equals(keyprefUrl) || key.equals(keyprefResolution) || + key.equals(keyprefFps)) { + updateSummary(sharedPreferences, key); + } + } + + private void updateSummary(SharedPreferences sharedPreferences, String key) { + Preference updatedPref = settingsFragment.findPreference(key); + // Set summary to be the user-description for the selected value + updatedPref.setSummary(sharedPreferences.getString(key, "")); + } + +} diff --git a/examples/android/src/org/appspot/apprtc/AppRTCGLView.java b/examples/android/src/org/appspot/apprtc/SettingsFragment.java index e622ab9..ec4b704 100644 --- a/examples/android/src/org/appspot/apprtc/AppRTCGLView.java +++ b/examples/android/src/org/appspot/apprtc/SettingsFragment.java @@ -27,32 +27,15 @@ package org.appspot.apprtc; -import android.content.Context; -import android.graphics.Point; -import android.opengl.GLSurfaceView; +import android.os.Bundle; +import android.preference.PreferenceFragment; -public class AppRTCGLView extends GLSurfaceView { - private Point screenDimensions; - - public AppRTCGLView(Context c, Point screenDimensions) { - super(c); - this.screenDimensions = screenDimensions; - } - - public void updateDisplaySize(Point screenDimensions) { - this.screenDimensions = screenDimensions; - } - - @Override - protected void onMeasure(int unusedX, int unusedY) { - // Go big or go home! - setMeasuredDimension(screenDimensions.x, screenDimensions.y); - } +public class SettingsFragment extends PreferenceFragment { @Override - protected void onAttachedToWindow() { - super.onAttachedToWindow(); - setSystemUiVisibility(SYSTEM_UI_FLAG_HIDE_NAVIGATION | - SYSTEM_UI_FLAG_FULLSCREEN | SYSTEM_UI_FLAG_IMMERSIVE_STICKY); + public void onCreate(Bundle savedInstanceState) { + super.onCreate(savedInstanceState); + // Load the preferences from an XML resource + addPreferencesFromResource(R.xml.preferences); } } diff --git a/examples/call/call_main.cc b/examples/call/call_main.cc index cb309dc..638f6f9 100644 --- a/examples/call/call_main.cc +++ b/examples/call/call_main.cc @@ -41,13 +41,13 @@ #include "talk/examples/call/callclient.h" #include "talk/examples/call/console.h" #include "talk/examples/call/mediaenginefactory.h" -#include "talk/p2p/base/constants.h" +#include "webrtc/p2p/base/constants.h" #include "talk/session/media/mediasessionclient.h" #include "talk/session/media/srtpfilter.h" -#include "talk/xmpp/xmppauth.h" -#include "talk/xmpp/xmppclientsettings.h" -#include "talk/xmpp/xmpppump.h" -#include "talk/xmpp/xmppsocket.h" +#include "webrtc/libjingle/xmpp/xmppauth.h" +#include "webrtc/libjingle/xmpp/xmppclientsettings.h" +#include "webrtc/libjingle/xmpp/xmpppump.h" +#include "webrtc/libjingle/xmpp/xmppsocket.h" #include "webrtc/base/pathutils.h" #include "webrtc/base/ssladapter.h" #include "webrtc/base/stream.h" diff --git a/examples/call/callclient.cc b/examples/call/callclient.cc index b31e24a..866133e 100644 --- a/examples/call/callclient.cc +++ b/examples/call/callclient.cc @@ -53,17 +53,17 @@ #include "talk/media/base/videorenderer.h" #include "talk/media/devices/devicemanager.h" #include "talk/media/devices/videorendererfactory.h" -#include "talk/p2p/base/sessionmanager.h" -#include "talk/p2p/client/basicportallocator.h" -#include "talk/p2p/client/sessionmanagertask.h" +#include "webrtc/p2p/base/sessionmanager.h" +#include "webrtc/p2p/client/basicportallocator.h" +#include "webrtc/p2p/client/sessionmanagertask.h" #include "talk/session/media/mediamessages.h" #include "talk/session/media/mediasessionclient.h" -#include "talk/xmpp/constants.h" -#include "talk/xmpp/hangoutpubsubclient.h" -#include "talk/xmpp/mucroomconfigtask.h" -#include "talk/xmpp/mucroomlookuptask.h" -#include "talk/xmpp/pingtask.h" -#include "talk/xmpp/presenceouttask.h" +#include "webrtc/libjingle/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/hangoutpubsubclient.h" +#include "webrtc/libjingle/xmpp/mucroomconfigtask.h" +#include "webrtc/libjingle/xmpp/mucroomlookuptask.h" +#include "webrtc/libjingle/xmpp/pingtask.h" +#include "webrtc/libjingle/xmpp/presenceouttask.h" namespace { diff --git a/examples/call/callclient.h b/examples/call/callclient.h index 50c584b..075dc6e 100644 --- a/examples/call/callclient.h +++ b/examples/call/callclient.h @@ -34,12 +34,12 @@ #include "talk/examples/call/console.h" #include "talk/media/base/mediachannel.h" -#include "talk/p2p/base/session.h" +#include "webrtc/p2p/base/session.h" #include "talk/session/media/mediamessages.h" #include "talk/session/media/mediasessionclient.h" -#include "talk/xmpp/hangoutpubsubclient.h" -#include "talk/xmpp/presencestatus.h" -#include "talk/xmpp/xmppclient.h" +#include "webrtc/libjingle/xmpp/hangoutpubsubclient.h" +#include "webrtc/libjingle/xmpp/presencestatus.h" +#include "webrtc/libjingle/xmpp/xmppclient.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/base/sslidentity.h" diff --git a/examples/call/callclient_unittest.cc b/examples/call/callclient_unittest.cc index d11580e..256853a 100644 --- a/examples/call/callclient_unittest.cc +++ b/examples/call/callclient_unittest.cc @@ -30,7 +30,7 @@ #include "talk/examples/call/callclient.h" #include "talk/media/base/filemediaengine.h" #include "talk/media/base/mediaengine.h" -#include "talk/xmpp/xmppthread.h" +#include "webrtc/libjingle/xmpp/xmppthread.h" #include "webrtc/base/gunit.h" TEST(CallClientTest, CreateCallClientWithDefaultMediaEngine) { diff --git a/examples/call/friendinvitesendtask.cc b/examples/call/friendinvitesendtask.cc index dae1006..e2b8dde 100644 --- a/examples/call/friendinvitesendtask.cc +++ b/examples/call/friendinvitesendtask.cc @@ -26,7 +26,7 @@ */ #include "talk/examples/call/friendinvitesendtask.h" -#include "talk/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/constants.h" namespace buzz { diff --git a/examples/call/friendinvitesendtask.h b/examples/call/friendinvitesendtask.h index 625f077..2928452 100644 --- a/examples/call/friendinvitesendtask.h +++ b/examples/call/friendinvitesendtask.h @@ -28,8 +28,8 @@ #ifndef _FRIENDINVITESENDTASK_H_ #define _FRIENDINVITESENDTASK_H_ -#include "talk/xmpp/xmppengine.h" -#include "talk/xmpp/xmpptask.h" +#include "webrtc/libjingle/xmpp/xmppengine.h" +#include "webrtc/libjingle/xmpp/xmpptask.h" namespace buzz { diff --git a/examples/call/muc.h b/examples/call/muc.h index 0e937ca..9be192f 100644 --- a/examples/call/muc.h +++ b/examples/call/muc.h @@ -29,8 +29,8 @@ #define _MUC_H_ #include <map> -#include "talk/xmpp/jid.h" -#include "talk/xmpp/presencestatus.h" +#include "webrtc/libjingle/xmpp/jid.h" +#include "webrtc/libjingle/xmpp/presencestatus.h" namespace buzz { diff --git a/examples/call/mucinviterecvtask.cc b/examples/call/mucinviterecvtask.cc index e32456f..32de8fe 100644 --- a/examples/call/mucinviterecvtask.cc +++ b/examples/call/mucinviterecvtask.cc @@ -26,7 +26,7 @@ */ #include "talk/examples/call/mucinviterecvtask.h" -#include "talk/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/constants.h" namespace buzz { diff --git a/examples/call/mucinviterecvtask.h b/examples/call/mucinviterecvtask.h index ddfd6be..0466c94 100644 --- a/examples/call/mucinviterecvtask.h +++ b/examples/call/mucinviterecvtask.h @@ -30,8 +30,8 @@ #include <vector> -#include "talk/xmpp/xmppengine.h" -#include "talk/xmpp/xmpptask.h" +#include "webrtc/libjingle/xmpp/xmppengine.h" +#include "webrtc/libjingle/xmpp/xmpptask.h" #include "webrtc/base/sigslot.h" namespace buzz { diff --git a/examples/call/mucinvitesendtask.cc b/examples/call/mucinvitesendtask.cc index d648fef..2299b78 100644 --- a/examples/call/mucinvitesendtask.cc +++ b/examples/call/mucinvitesendtask.cc @@ -26,8 +26,8 @@ */ #include "talk/examples/call/mucinvitesendtask.h" -#include "talk/xmpp/constants.h" -#include "talk/xmpp/xmppclient.h" +#include "webrtc/libjingle/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/xmppclient.h" namespace buzz { diff --git a/examples/call/mucinvitesendtask.h b/examples/call/mucinvitesendtask.h index 3ae74c1..673efb0 100644 --- a/examples/call/mucinvitesendtask.h +++ b/examples/call/mucinvitesendtask.h @@ -29,8 +29,8 @@ #define _MUCINVITESENDTASK_H_ #include "talk/examples/call/muc.h" -#include "talk/xmpp/xmppengine.h" -#include "talk/xmpp/xmpptask.h" +#include "webrtc/libjingle/xmpp/xmppengine.h" +#include "webrtc/libjingle/xmpp/xmpptask.h" namespace buzz { diff --git a/examples/call/presencepushtask.cc b/examples/call/presencepushtask.cc index b598eb6..8d8dca8 100644 --- a/examples/call/presencepushtask.cc +++ b/examples/call/presencepushtask.cc @@ -28,7 +28,7 @@ #include "talk/examples/call/presencepushtask.h" #include "talk/examples/call/muc.h" -#include "talk/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/constants.h" #include "webrtc/base/stringencode.h" diff --git a/examples/call/presencepushtask.h b/examples/call/presencepushtask.h index c13a4f5..e3d3268 100644 --- a/examples/call/presencepushtask.h +++ b/examples/call/presencepushtask.h @@ -31,9 +31,9 @@ #include <vector> #include "talk/examples/call/callclient.h" -#include "talk/xmpp/presencestatus.h" -#include "talk/xmpp/xmppengine.h" -#include "talk/xmpp/xmpptask.h" +#include "webrtc/libjingle/xmpp/presencestatus.h" +#include "webrtc/libjingle/xmpp/xmppengine.h" +#include "webrtc/libjingle/xmpp/xmpptask.h" #include "webrtc/base/sigslot.h" namespace buzz { diff --git a/examples/login/login_main.cc b/examples/login/login_main.cc index bfe12af..581c986 100644 --- a/examples/login/login_main.cc +++ b/examples/login/login_main.cc @@ -29,10 +29,10 @@ #include <iostream> -#include "talk/xmpp/constants.h" -#include "talk/xmpp/xmppclientsettings.h" -#include "talk/xmpp/xmppengine.h" -#include "talk/xmpp/xmppthread.h" +#include "webrtc/libjingle/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/xmppclientsettings.h" +#include "webrtc/libjingle/xmpp/xmppengine.h" +#include "webrtc/libjingle/xmpp/xmppthread.h" #include "webrtc/base/thread.h" int main(int argc, char **argv) { diff --git a/examples/objc/AppRTCDemo/APPRTCConnectionManager.m b/examples/objc/AppRTCDemo/APPRTCConnectionManager.m index b411a62..9a39528 100644 --- a/examples/objc/AppRTCDemo/APPRTCConnectionManager.m +++ b/examples/objc/AppRTCDemo/APPRTCConnectionManager.m @@ -170,7 +170,7 @@ #endif [lms addAudioTrack:[self.peerConnectionFactory audioTrackWithID:@"ARDAMSa0"]]; - [self.peerConnection addStream:lms constraints:constraints]; + [self.peerConnection addStream:lms]; [self.logger logMessage:@"onICEServers - added local stream."]; } @@ -243,16 +243,6 @@ #pragma mark - RTCPeerConnectionDelegate -- (void)peerConnectionOnError:(RTCPeerConnection*)peerConnection { - dispatch_async(dispatch_get_main_queue(), ^{ - NSString* message = @"PeerConnection error"; - NSLog(@"%@", message); - NSAssert(NO, @"PeerConnection failed."); - [self.delegate connectionManager:self - didErrorWithMessage:message]; - }); -} - - (void)peerConnection:(RTCPeerConnection*)peerConnection signalingStateChanged:(RTCSignalingState)stateChanged { dispatch_async(dispatch_get_main_queue(), ^{ diff --git a/examples/peerconnection/client/conductor.cc b/examples/peerconnection/client/conductor.cc index f49aee6..e81f7fc 100644 --- a/examples/peerconnection/client/conductor.cc +++ b/examples/peerconnection/client/conductor.cc @@ -137,11 +137,6 @@ void Conductor::EnsureStreamingUI() { // PeerConnectionObserver implementation. // -void Conductor::OnError() { - LOG(LS_ERROR) << __FUNCTION__; - main_wnd_->QueueUIThreadCallback(PEER_CONNECTION_ERROR, NULL); -} - // Called when a remote stream is added void Conductor::OnAddStream(webrtc::MediaStreamInterface* stream) { LOG(INFO) << __FUNCTION__ << " " << stream->label(); @@ -373,7 +368,7 @@ void Conductor::AddStreams() { stream->AddTrack(audio_track); stream->AddTrack(video_track); - if (!peer_connection_->AddStream(stream, NULL)) { + if (!peer_connection_->AddStream(stream)) { LOG(LS_ERROR) << "Adding stream to PeerConnection failed"; } typedef std::pair<std::string, @@ -440,10 +435,6 @@ void Conductor::UIThreadCallback(int msg_id, void* data) { break; } - case PEER_CONNECTION_ERROR: - main_wnd_->MessageBox("Error", "an unknown error occurred", true); - break; - case NEW_STREAM_ADDED: { webrtc::MediaStreamInterface* stream = reinterpret_cast<webrtc::MediaStreamInterface*>( diff --git a/examples/peerconnection/client/conductor.h b/examples/peerconnection/client/conductor.h index 0aff531..3ef5253 100644 --- a/examples/peerconnection/client/conductor.h +++ b/examples/peerconnection/client/conductor.h @@ -58,7 +58,6 @@ class Conductor MEDIA_CHANNELS_INITIALIZED = 1, PEER_CONNECTION_CLOSED, SEND_MESSAGE_TO_PEER, - PEER_CONNECTION_ERROR, NEW_STREAM_ADDED, STREAM_REMOVED, }; @@ -80,11 +79,11 @@ class Conductor // // PeerConnectionObserver implementation. // - virtual void OnError(); virtual void OnStateChange( webrtc::PeerConnectionObserver::StateType state_changed) {} virtual void OnAddStream(webrtc::MediaStreamInterface* stream); virtual void OnRemoveStream(webrtc::MediaStreamInterface* stream); + virtual void OnDataChannel(webrtc::DataChannelInterface* channel) {} virtual void OnRenegotiationNeeded() {} virtual void OnIceChange() {} virtual void OnIceCandidate(const webrtc::IceCandidateInterface* candidate); diff --git a/examples/relayserver/relayserver_main.cc b/examples/relayserver/relayserver_main.cc index 1a4ab78..5a8bec3 100644 --- a/examples/relayserver/relayserver_main.cc +++ b/examples/relayserver/relayserver_main.cc @@ -27,7 +27,7 @@ #include <iostream> // NOLINT -#include "talk/p2p/base/relayserver.h" +#include "webrtc/p2p/base/relayserver.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/base/thread.h" diff --git a/examples/stunserver/stunserver_main.cc b/examples/stunserver/stunserver_main.cc index f800f31..3cbed91 100644 --- a/examples/stunserver/stunserver_main.cc +++ b/examples/stunserver/stunserver_main.cc @@ -31,7 +31,7 @@ #include <iostream> -#include "talk/p2p/base/stunserver.h" +#include "webrtc/p2p/base/stunserver.h" #include "webrtc/base/thread.h" using namespace cricket; diff --git a/examples/turnserver/turnserver_main.cc b/examples/turnserver/turnserver_main.cc index 607b8cf..692b8a2 100644 --- a/examples/turnserver/turnserver_main.cc +++ b/examples/turnserver/turnserver_main.cc @@ -27,8 +27,8 @@ #include <iostream> // NOLINT -#include "talk/p2p/base/basicpacketsocketfactory.h" -#include "talk/p2p/base/turnserver.h" +#include "webrtc/p2p/base/basicpacketsocketfactory.h" +#include "webrtc/p2p/base/turnserver.h" #include "webrtc/base/asyncudpsocket.h" #include "webrtc/base/optionsfile.h" #include "webrtc/base/stringencode.h" diff --git a/libjingle.gyp b/libjingle.gyp index 335a788..803eaa3 100755 --- a/libjingle.gyp +++ b/libjingle.gyp @@ -299,7 +299,7 @@ 'targets': [ { 'target_name': 'libjingle', - 'type': 'static_library', + 'type': 'none', 'dependencies': [ '<(DEPTH)/third_party/expat/expat.gyp:expat', '<(DEPTH)/third_party/jsoncpp/jsoncpp.gyp:jsoncpp', @@ -309,81 +309,6 @@ '<(DEPTH)/third_party/expat/expat.gyp:expat', '<(DEPTH)/third_party/jsoncpp/jsoncpp.gyp:jsoncpp', ], - 'sources': [ - 'xmpp/asyncsocket.h', - 'xmpp/chatroommodule.h', - 'xmpp/chatroommoduleimpl.cc', - 'xmpp/constants.cc', - 'xmpp/constants.h', - 'xmpp/discoitemsquerytask.cc', - 'xmpp/discoitemsquerytask.h', - 'xmpp/hangoutpubsubclient.cc', - 'xmpp/hangoutpubsubclient.h', - 'xmpp/iqtask.cc', - 'xmpp/iqtask.h', - 'xmpp/jid.cc', - 'xmpp/jid.h', - 'xmpp/module.h', - 'xmpp/moduleimpl.cc', - 'xmpp/moduleimpl.h', - 'xmpp/mucroomconfigtask.cc', - 'xmpp/mucroomconfigtask.h', - 'xmpp/mucroomdiscoverytask.cc', - 'xmpp/mucroomdiscoverytask.h', - 'xmpp/mucroomlookuptask.cc', - 'xmpp/mucroomlookuptask.h', - 'xmpp/mucroomuniquehangoutidtask.cc', - 'xmpp/mucroomuniquehangoutidtask.h', - 'xmpp/pingtask.cc', - 'xmpp/pingtask.h', - 'xmpp/plainsaslhandler.h', - 'xmpp/presenceouttask.cc', - 'xmpp/presenceouttask.h', - 'xmpp/presencereceivetask.cc', - 'xmpp/presencereceivetask.h', - 'xmpp/presencestatus.cc', - 'xmpp/presencestatus.h', - 'xmpp/prexmppauth.h', - 'xmpp/pubsub_task.cc', - 'xmpp/pubsub_task.h', - 'xmpp/pubsubclient.cc', - 'xmpp/pubsubclient.h', - 'xmpp/pubsubstateclient.cc', - 'xmpp/pubsubstateclient.h', - 'xmpp/pubsubtasks.cc', - 'xmpp/pubsubtasks.h', - 'xmpp/receivetask.cc', - 'xmpp/receivetask.h', - 'xmpp/rostermodule.h', - 'xmpp/rostermoduleimpl.cc', - 'xmpp/rostermoduleimpl.h', - 'xmpp/saslcookiemechanism.h', - 'xmpp/saslhandler.h', - 'xmpp/saslmechanism.cc', - 'xmpp/saslmechanism.h', - 'xmpp/saslplainmechanism.h', - 'xmpp/xmppauth.cc', - 'xmpp/xmppauth.h', - 'xmpp/xmppclient.cc', - 'xmpp/xmppclient.h', - 'xmpp/xmppclientsettings.h', - 'xmpp/xmppengine.h', - 'xmpp/xmppengineimpl.cc', - 'xmpp/xmppengineimpl.h', - 'xmpp/xmppengineimpl_iq.cc', - 'xmpp/xmpplogintask.cc', - 'xmpp/xmpplogintask.h', - 'xmpp/xmpppump.cc', - 'xmpp/xmpppump.h', - 'xmpp/xmppsocket.cc', - 'xmpp/xmppsocket.h', - 'xmpp/xmppstanzaparser.cc', - 'xmpp/xmppstanzaparser.h', - 'xmpp/xmpptask.cc', - 'xmpp/xmpptask.h', - 'xmpp/xmppthread.cc', - 'xmpp/xmppthread.h', - ], }, # target libjingle { 'target_name': 'libjingle_media', @@ -401,8 +326,10 @@ '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine', '<(webrtc_root)/sound/sound.gyp:rtc_sound', '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers', - '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:field_trial_default', + '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers_default', '<(webrtc_root)/libjingle/xmllite/xmllite.gyp:rtc_xmllite', + '<(webrtc_root)/libjingle/xmpp/xmpp.gyp:rtc_xmpp', + '<(webrtc_root)/p2p/p2p.gyp:rtc_p2p', 'libjingle', ], 'direct_dependent_settings': { @@ -628,90 +555,6 @@ ], }, 'sources': [ - 'p2p/base/asyncstuntcpsocket.cc', - 'p2p/base/asyncstuntcpsocket.h', - 'p2p/base/basicpacketsocketfactory.cc', - 'p2p/base/basicpacketsocketfactory.h', - 'p2p/base/candidate.h', - 'p2p/base/common.h', - 'p2p/base/constants.cc', - 'p2p/base/constants.h', - 'p2p/base/dtlstransportchannel.cc', - 'p2p/base/dtlstransportchannel.h', - 'p2p/base/p2ptransport.cc', - 'p2p/base/p2ptransport.h', - 'p2p/base/p2ptransportchannel.cc', - 'p2p/base/p2ptransportchannel.h', - 'p2p/base/packetsocketfactory.h', - 'p2p/base/parsing.cc', - 'p2p/base/parsing.h', - 'p2p/base/port.cc', - 'p2p/base/port.h', - 'p2p/base/portallocator.cc', - 'p2p/base/portallocator.h', - 'p2p/base/portallocatorsessionproxy.cc', - 'p2p/base/portallocatorsessionproxy.h', - 'p2p/base/portinterface.h', - 'p2p/base/portproxy.cc', - 'p2p/base/portproxy.h', - 'p2p/base/pseudotcp.cc', - 'p2p/base/pseudotcp.h', - 'p2p/base/rawtransport.cc', - 'p2p/base/rawtransport.h', - 'p2p/base/rawtransportchannel.cc', - 'p2p/base/rawtransportchannel.h', - 'p2p/base/relayport.cc', - 'p2p/base/relayport.h', - 'p2p/base/relayserver.cc', - 'p2p/base/relayserver.h', - 'p2p/base/session.cc', - 'p2p/base/session.h', - 'p2p/base/sessionclient.h', - 'p2p/base/sessiondescription.cc', - 'p2p/base/sessiondescription.h', - 'p2p/base/sessionid.h', - 'p2p/base/sessionmanager.cc', - 'p2p/base/sessionmanager.h', - 'p2p/base/sessionmessages.cc', - 'p2p/base/sessionmessages.h', - 'p2p/base/stun.cc', - 'p2p/base/stun.h', - 'p2p/base/stunport.cc', - 'p2p/base/stunport.h', - 'p2p/base/stunrequest.cc', - 'p2p/base/stunrequest.h', - 'p2p/base/stunserver.cc', - 'p2p/base/stunserver.h', - 'p2p/base/tcpport.cc', - 'p2p/base/tcpport.h', - 'p2p/base/transport.cc', - 'p2p/base/transport.h', - 'p2p/base/transportchannel.cc', - 'p2p/base/transportchannel.h', - 'p2p/base/transportchannelimpl.h', - 'p2p/base/transportchannelproxy.cc', - 'p2p/base/transportchannelproxy.h', - 'p2p/base/transportdescription.cc', - 'p2p/base/transportdescription.h', - 'p2p/base/transportdescriptionfactory.cc', - 'p2p/base/transportdescriptionfactory.h', - 'p2p/base/transportinfo.h', - 'p2p/base/turnport.cc', - 'p2p/base/turnport.h', - 'p2p/base/turnserver.cc', - 'p2p/base/turnserver.h', - 'p2p/base/udpport.h', - 'p2p/client/autoportallocator.h', - 'p2p/client/basicportallocator.cc', - 'p2p/client/basicportallocator.h', - 'p2p/client/connectivitychecker.cc', - 'p2p/client/connectivitychecker.h', - 'p2p/client/httpportallocator.cc', - 'p2p/client/httpportallocator.h', - 'p2p/client/sessionmanagertask.h', - 'p2p/client/sessionsendtask.h', - 'p2p/client/socketmonitor.cc', - 'p2p/client/socketmonitor.h', 'session/tunnel/pseudotcpchannel.cc', 'session/tunnel/pseudotcpchannel.h', 'session/tunnel/tunnelsessionclient.cc', diff --git a/libjingle_examples.gyp b/libjingle_examples.gyp index 085da39..d0f1747 100755 --- a/libjingle_examples.gyp +++ b/libjingle_examples.gyp @@ -39,8 +39,8 @@ 'libjingle.gyp:libjingle_p2p', ], 'sources': [ - 'xmpp/jingleinfotask.cc', - 'xmpp/jingleinfotask.h', + '<(webrtc_root)/libjingle/xmpp/jingleinfotask.cc', + '<(webrtc_root)/libjingle/xmpp/jingleinfotask.h', ], }, # target libjingle_xmpphelp { @@ -317,17 +317,41 @@ 'examples/android/build.xml', 'examples/android/jni/Android.mk', 'examples/android/project.properties', + 'examples/android/res/drawable-hdpi/disconnect.png', + 'examples/android/res/drawable-hdpi/ic_action_full_screen.png', + 'examples/android/res/drawable-hdpi/ic_action_return_from_full_screen.png', + 'examples/android/res/drawable-hdpi/ic_loopback_call.png', 'examples/android/res/drawable-hdpi/ic_launcher.png', + 'examples/android/res/drawable-ldpi/disconnect.png', + 'examples/android/res/drawable-ldpi/ic_action_full_screen.png', + 'examples/android/res/drawable-ldpi/ic_action_return_from_full_screen.png', + 'examples/android/res/drawable-ldpi/ic_loopback_call.png', 'examples/android/res/drawable-ldpi/ic_launcher.png', + 'examples/android/res/drawable-mdpi/disconnect.png', + 'examples/android/res/drawable-mdpi/ic_action_full_screen.png', + 'examples/android/res/drawable-mdpi/ic_action_return_from_full_screen.png', + 'examples/android/res/drawable-mdpi/ic_loopback_call.png', 'examples/android/res/drawable-mdpi/ic_launcher.png', + 'examples/android/res/drawable-xhdpi/disconnect.png', + 'examples/android/res/drawable-xhdpi/ic_action_full_screen.png', + 'examples/android/res/drawable-xhdpi/ic_action_return_from_full_screen.png', + 'examples/android/res/drawable-xhdpi/ic_loopback_call.png', 'examples/android/res/drawable-xhdpi/ic_launcher.png', + 'examples/android/res/layout/activity_connect.xml', + 'examples/android/res/layout/activity_fullscreen.xml', + 'examples/android/res/layout/fragment_menubar.xml', + 'examples/android/res/menu/connect_menu.xml', + 'examples/android/res/values/arrays.xml', 'examples/android/res/values/strings.xml', + 'examples/android/res/xml/preferences.xml', 'examples/android/src/org/appspot/apprtc/AppRTCClient.java', 'examples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java', - 'examples/android/src/org/appspot/apprtc/AppRTCGLView.java', + 'examples/android/src/org/appspot/apprtc/ConnectActivity.java', 'examples/android/src/org/appspot/apprtc/GAEChannelClient.java', 'examples/android/src/org/appspot/apprtc/GAERTCClient.java', 'examples/android/src/org/appspot/apprtc/PeerConnectionClient.java', + 'examples/android/src/org/appspot/apprtc/SettingsActivity.java', + 'examples/android/src/org/appspot/apprtc/SettingsFragment.java', 'examples/android/src/org/appspot/apprtc/UnhandledExceptionHandler.java', ], 'outputs': [ diff --git a/libjingle_media_unittest.isolate b/libjingle_media_unittest.isolate index 666478b..970a166 100644 --- a/libjingle_media_unittest.isolate +++ b/libjingle_media_unittest.isolate @@ -31,12 +31,9 @@ 'command': [ '<(PRODUCT_DIR)/libjingle_media_unittest<(EXECUTABLE_SUFFIX)', ], - 'isolate_dependency_tracked': [ - 'media/testdata/captured-320x240-2s-48.frames', + 'files': [ '<(PRODUCT_DIR)/libjingle_media_unittest<(EXECUTABLE_SUFFIX)', - ], - 'isolate_dependency_untracked': [ - '<(DEPTH)/tools/swarming_client/', + 'media/testdata/captured-320x240-2s-48.frames', ], }, }], diff --git a/libjingle_p2p_unittest.isolate b/libjingle_p2p_unittest.isolate index 9ff0d77..d9f0423 100644 --- a/libjingle_p2p_unittest.isolate +++ b/libjingle_p2p_unittest.isolate @@ -31,12 +31,9 @@ 'command': [ '<(PRODUCT_DIR)/libjingle_p2p_unittest<(EXECUTABLE_SUFFIX)', ], - 'isolate_dependency_tracked': [ + 'files': [ '<(PRODUCT_DIR)/libjingle_p2p_unittest<(EXECUTABLE_SUFFIX)', ], - 'isolate_dependency_untracked': [ - '<(DEPTH)/tools/swarming_client/', - ], }, }], ], diff --git a/libjingle_peerconnection_unittest.isolate b/libjingle_peerconnection_unittest.isolate index df07d4a..4bc8c43 100644 --- a/libjingle_peerconnection_unittest.isolate +++ b/libjingle_peerconnection_unittest.isolate @@ -31,12 +31,9 @@ 'command': [ '<(PRODUCT_DIR)/libjingle_peerconnection_unittest<(EXECUTABLE_SUFFIX)', ], - 'isolate_dependency_tracked': [ + 'files': [ '<(PRODUCT_DIR)/libjingle_peerconnection_unittest<(EXECUTABLE_SUFFIX)', ], - 'isolate_dependency_untracked': [ - '<(DEPTH)/tools/swarming_client/', - ], }, }], ], diff --git a/libjingle_sound_unittest.isolate b/libjingle_sound_unittest.isolate index 728e810..b534200 100644 --- a/libjingle_sound_unittest.isolate +++ b/libjingle_sound_unittest.isolate @@ -31,12 +31,9 @@ 'command': [ '<(PRODUCT_DIR)/libjingle_sound_unittest<(EXECUTABLE_SUFFIX)', ], - 'isolate_dependency_tracked': [ + 'files': [ '<(PRODUCT_DIR)/libjingle_sound_unittest<(EXECUTABLE_SUFFIX)', ], - 'isolate_dependency_untracked': [ - '<(DEPTH)/tools/swarming_client/', - ], }, }], ], diff --git a/libjingle_tests.gyp b/libjingle_tests.gyp index 44adec3..6a25c0f 100755 --- a/libjingle_tests.gyp +++ b/libjingle_tests.gyp @@ -81,21 +81,7 @@ 'libjingle_unittest_main', ], 'sources': [ - 'xmpp/fakexmppclient.h', - 'xmpp/hangoutpubsubclient_unittest.cc', - 'xmpp/jid_unittest.cc', - 'xmpp/mucroomconfigtask_unittest.cc', - 'xmpp/mucroomdiscoverytask_unittest.cc', - 'xmpp/mucroomlookuptask_unittest.cc', - 'xmpp/mucroomuniquehangoutidtask_unittest.cc', - 'xmpp/pingtask_unittest.cc', - 'xmpp/pubsubclient_unittest.cc', - 'xmpp/pubsubtasks_unittest.cc', - 'xmpp/util_unittest.cc', - 'xmpp/util_unittest.h', - 'xmpp/xmppengine_unittest.cc', - 'xmpp/xmpplogintask_unittest.cc', - 'xmpp/xmppstanzaparser_unittest.cc', + '<(DEPTH)/webrtc/test/testsupport/always_passing_unittest.cc', ], # sources }, # target libjingle_unittest { @@ -190,28 +176,6 @@ '<(DEPTH)/third_party/libsrtp/srtp', ], 'sources': [ - 'p2p/base/dtlstransportchannel_unittest.cc', - 'p2p/base/fakesession.h', - 'p2p/base/p2ptransportchannel_unittest.cc', - 'p2p/base/port_unittest.cc', - 'p2p/base/portallocatorsessionproxy_unittest.cc', - 'p2p/base/pseudotcp_unittest.cc', - 'p2p/base/relayport_unittest.cc', - 'p2p/base/relayserver_unittest.cc', - 'p2p/base/session_unittest.cc', - 'p2p/base/stun_unittest.cc', - 'p2p/base/stunport_unittest.cc', - 'p2p/base/stunrequest_unittest.cc', - 'p2p/base/stunserver_unittest.cc', - 'p2p/base/testrelayserver.h', - 'p2p/base/teststunserver.h', - 'p2p/base/testturnserver.h', - 'p2p/base/transport_unittest.cc', - 'p2p/base/transportdescriptionfactory_unittest.cc', - 'p2p/base/turnport_unittest.cc', - 'p2p/client/connectivitychecker_unittest.cc', - 'p2p/client/fakeportallocator.h', - 'p2p/client/portallocator_unittest.cc', 'session/media/bundlefilter_unittest.cc', 'session/media/channel_unittest.cc', 'session/media/channelmanager_unittest.cc', @@ -438,7 +402,6 @@ ], 'includes': [ 'build/isolate.gypi', - 'libjingle_media_unittest.isolate', ], 'sources': [ 'libjingle_media_unittest.isolate', @@ -452,7 +415,6 @@ ], 'includes': [ 'build/isolate.gypi', - 'libjingle_p2p_unittest.isolate', ], 'sources': [ 'libjingle_p2p_unittest.isolate', @@ -466,26 +428,11 @@ ], 'includes': [ 'build/isolate.gypi', - 'libjingle_peerconnection_unittest.isolate', ], 'sources': [ 'libjingle_peerconnection_unittest.isolate', ], }, - { - 'target_name': 'libjingle_unittest_run', - 'type': 'none', - 'dependencies': [ - 'libjingle_unittest', - ], - 'includes': [ - 'build/isolate.gypi', - 'libjingle_unittest.isolate', - ], - 'sources': [ - 'libjingle_unittest.isolate', - ], - }, ], }], ], diff --git a/libjingle_unittest.isolate b/libjingle_unittest.isolate index 0507f6a..9313beb 100644 --- a/libjingle_unittest.isolate +++ b/libjingle_unittest.isolate @@ -31,12 +31,9 @@ 'command': [ '<(PRODUCT_DIR)/libjingle_unittest<(EXECUTABLE_SUFFIX)', ], - 'isolate_dependency_tracked': [ + 'files': [ '<(PRODUCT_DIR)/libjingle_unittest<(EXECUTABLE_SUFFIX)', ], - 'isolate_dependency_untracked': [ - '<(DEPTH)/tools/swarming_client/', - ], }, }], ], diff --git a/media/base/fakemediaengine.h b/media/base/fakemediaengine.h index 87bc8a3..db5e2e4 100644 --- a/media/base/fakemediaengine.h +++ b/media/base/fakemediaengine.h @@ -38,7 +38,7 @@ #include "talk/media/base/mediaengine.h" #include "talk/media/base/rtputils.h" #include "talk/media/base/streamparams.h" -#include "talk/p2p/base/sessiondescription.h" +#include "webrtc/p2p/base/sessiondescription.h" #include "webrtc/base/buffer.h" #include "webrtc/base/stringutils.h" diff --git a/media/base/mediachannel.h b/media/base/mediachannel.h index b1164b5..5e03ef9 100644 --- a/media/base/mediachannel.h +++ b/media/base/mediachannel.h @@ -843,7 +843,6 @@ struct VideoSenderInfo : public MediaSenderInfo { capture_jitter_ms(0), avg_encode_ms(0), encode_usage_percent(0), - encode_rsd(0), capture_queue_delay_ms_per_s(0) { } @@ -865,7 +864,6 @@ struct VideoSenderInfo : public MediaSenderInfo { int capture_jitter_ms; int avg_encode_ms; int encode_usage_percent; - int encode_rsd; int capture_queue_delay_ms_per_s; VariableInfo<int> adapt_frame_drops; VariableInfo<int> effects_frame_drops; @@ -973,7 +971,7 @@ struct BandwidthEstimationInfo { // StatsOptions::include_received_propagation_stats is true. int total_received_propagation_delta_ms; std::vector<int> recent_received_propagation_delta_ms; - std::vector<int64> recent_received_packet_group_arrival_time_ms; + std::vector<int64_t> recent_received_packet_group_arrival_time_ms; }; struct VoiceMediaInfo { diff --git a/media/base/videoengine_unittest.h b/media/base/videoengine_unittest.h index 6ea5b8c..472d42a 100644 --- a/media/base/videoengine_unittest.h +++ b/media/base/videoengine_unittest.h @@ -783,10 +783,7 @@ class VideoMediaChannelTest : public testing::Test, void SetSendSetsTransportBufferSizes() { EXPECT_TRUE(SetOneCodec(DefaultCodec())); EXPECT_TRUE(SetSend(true)); - // TODO(sriniv): Remove or re-enable this. - // As part of b/8030474, send-buffer is size now controlled through - // portallocator flags. Its not set by channels. - // EXPECT_EQ(64 * 1024, network_interface_.sendbuf_size()); + EXPECT_EQ(64 * 1024, network_interface_.sendbuf_size()); EXPECT_EQ(64 * 1024, network_interface_.recvbuf_size()); } // Tests that we can send frames and the right payload type is used. @@ -1380,6 +1377,9 @@ class VideoMediaChannelTest : public testing::Test, EXPECT_EQ(0, renderer_.num_rendered_frames()); EXPECT_TRUE(SendFrame()); EXPECT_FRAME_WAIT(1, 640, 400, kTimeout); + // Wait for one frame so they don't get dropped because we send frames too + // tightly. + rtc::Thread::Current()->ProcessMessages(30); // Remove the capturer. EXPECT_TRUE(channel_->SetCapturer(kSsrc, NULL)); // Wait for one black frame for removing the capturer. diff --git a/media/webrtc/webrtcvideoengine.cc b/media/webrtc/webrtcvideoengine.cc index 4a4e29a..04092f3 100644 --- a/media/webrtc/webrtcvideoengine.cc +++ b/media/webrtc/webrtcvideoengine.cc @@ -66,6 +66,22 @@ namespace { +cricket::VideoFormat CreateVideoFormat(int width, int height, int framerate) { + return cricket::VideoFormat( + width, + height, + cricket::VideoFormat::FpsToInterval(framerate), + cricket::FOURCC_ANY); +} + +cricket::VideoFormat VideoFormatFromCodec(const cricket::VideoCodec& codec) { + return CreateVideoFormat(codec.width, codec.height, codec.framerate); +} + +cricket::VideoFormat VideoFormatFromVieCodec(const webrtc::VideoCodec& codec) { + return CreateVideoFormat(codec.width, codec.height, codec.maxFramerate); +} + template <class T> bool Changed(cricket::Settable<T> proposed, cricket::Settable<T> original) { @@ -172,10 +188,12 @@ static bool IsRembEnabled(const VideoCodec& codec) { } struct FlushBlackFrameData : public rtc::MessageData { - FlushBlackFrameData(uint32 s, int64 t) : ssrc(s), timestamp(t) { + FlushBlackFrameData(uint32 s, int64 t, int i) + : ssrc(s), timestamp(t), interval(i) { } uint32 ssrc; int64 timestamp; + int interval; }; class WebRtcRenderAdapter : public webrtc::ExternalRenderer { @@ -592,6 +610,15 @@ class WebRtcOveruseObserver : public webrtc::CpuOveruseObserver { class WebRtcVideoChannelSendInfo : public sigslot::has_slots<> { public: typedef std::map<int, webrtc::VideoEncoder*> EncoderMap; // key: payload type + + enum AdaptFormatType { + // This is how we make SetSendStreamFormat take precedence over + // SetSendCodecs. + kAdaptFormatTypeNone = 0, // Unset + kAdaptFormatTypeCodec = 1, // From SetSendCodec + kAdaptFormatTypeStream = 2, // From SetStreamFormat + }; + WebRtcVideoChannelSendInfo(int channel_id, int capture_id, webrtc::ViEExternalCapture* external_capture, rtc::CpuMonitor* cpu_monitor) @@ -602,15 +629,24 @@ class WebRtcVideoChannelSendInfo : public sigslot::has_slots<> { video_capturer_(NULL), encoder_observer_(channel_id), external_capture_(external_capture), - interval_(0), cpu_monitor_(cpu_monitor), - old_adaptation_changes_(0) { + old_adaptation_changes_(0), + adapt_format_type_(kAdaptFormatTypeNone) { } int channel_id() const { return channel_id_; } int capture_id() const { return capture_id_; } void set_sending(bool sending) { sending_ = sending; } bool sending() const { return sending_; } + void set_send_params(const VideoSendParams& send_params) { + send_params_ = send_params; + } + const VideoSendParams& send_params() const { + return send_params_; + } + const Settable<CapturedFrameInfo>& last_captured_frame_info() const { + return last_captured_frame_info_; + } void set_muted(bool on) { // TODO(asapersson): add support. // video_adapter_.SetBlackOutput(on); @@ -620,25 +656,51 @@ class WebRtcVideoChannelSendInfo : public sigslot::has_slots<> { WebRtcEncoderObserver* encoder_observer() { return &encoder_observer_; } webrtc::ViEExternalCapture* external_capture() { return external_capture_; } - const VideoFormat& video_format() const { - return video_format_; - } - void set_video_format(const VideoFormat& video_format) { - video_format_ = video_format; - if (video_format_ != cricket::VideoFormat()) { - interval_ = video_format_.interval; - } - CoordinatedVideoAdapter* adapter = video_adapter(); - if (adapter) { - adapter->OnOutputFormatRequest(video_format_); + const VideoFormat& adapt_format() const { return adapt_format_; } + AdaptFormatType adapt_format_type() const { return adapt_format_type_; } + bool adapt_format_set() const { + return adapt_format_type() != kAdaptFormatTypeNone; + } + + // Returns true if the last captured frame info changed. + void SetLastCapturedFrameInfo( + const VideoFrame* frame, bool screencast, bool* changed) { + CapturedFrameInfo last; + if (last_captured_frame_info_.Get(&last) && + frame->GetWidth() == last.width && + frame->GetHeight() == last.height && + screencast == last.screencast) { + *changed = false; + return; } + + last_captured_frame_info_.Set(CapturedFrameInfo( + frame->GetWidth(), frame->GetHeight(), screencast)); + *changed = true; } - void set_interval(int64 interval) { - if (video_format() == cricket::VideoFormat()) { - interval_ = interval; + + // Tells the video adapter to adapt down to a given format. The + // type indicates where the format came from, where different types + // have slightly different behavior and priority. + void SetAdaptFormat(const VideoFormat& format, AdaptFormatType type) { + if (type < adapt_format_type_) { + // Formats from SetSendStream format are higher priority than + // ones from SetSendCodecs wich is higher priority than not + // being set. If something lower-prioirty comes in, just ignore + // it. + return; + } + + // TODO(pthatcher): Use the adapter for all max size enforcement, + // both codec-based and SetSendStreamFormat-based. For now, we + // can't do that without fixing a lot of unit tests. + if (video_adapter() && type == kAdaptFormatTypeStream) { + video_adapter()->OnOutputFormatRequest(format); } + + adapt_format_ = format; + adapt_format_type_ = type; } - int64 interval() { return interval_; } int CurrentAdaptReason() const { if (!video_adapter()) { @@ -653,15 +715,20 @@ class WebRtcVideoChannelSendInfo : public sigslot::has_slots<> { return old_adaptation_changes_ + video_adapter()->adaptation_changes(); } - StreamParams* stream_params() { return stream_params_.get(); } void set_stream_params(const StreamParams& sp) { - stream_params_.reset(new StreamParams(sp)); + send_params_.stream = sp; + } + const StreamParams& stream_params() const { return send_params_.stream; } + // A default send channel can be non-active if a stream hasn't been + // added yet, or if all streams have been removed (at which point, + // Deactive is called). + bool IsActive() { + return stream_params().first_ssrc() != 0; } - void ClearStreamParams() { stream_params_.reset(); } - bool has_ssrc(uint32 local_ssrc) const { - return !stream_params_ ? false : - stream_params_->has_ssrc(local_ssrc); + void Deactivate() { + send_params_.stream = StreamParams(); } + WebRtcLocalStreamInfo* local_stream_info() { return &local_stream_info_; } @@ -696,6 +763,13 @@ class WebRtcVideoChannelSendInfo : public sigslot::has_slots<> { CoordinatedVideoAdapter* adapter = video_adapter(); ASSERT(adapter && "Video adapter should not be null here."); + // TODO(pthatcher): Use the adapter for all max size enforcement, + // both codec-based and SetSendStreamFormat-based. For now, we + // can't do that without fixing a lot of unit tests. + if (adapt_format_type_ == kAdaptFormatTypeStream) { + adapter->OnOutputFormatRequest(adapt_format_); + } + UpdateAdapterCpuOptions(); overuse_observer_.reset(new WebRtcOveruseObserver(adapter)); @@ -826,6 +900,9 @@ class WebRtcVideoChannelSendInfo : public sigslot::has_slots<> { private: int channel_id_; int capture_id_; + VideoSendParams send_params_; + // TODO(pthatcher): Merge CapturedFrameInfo and LocalStreamInfo. + Settable<CapturedFrameInfo> last_captured_frame_info_; bool sending_; bool muted_; VideoCapturer* video_capturer_; @@ -833,20 +910,17 @@ class WebRtcVideoChannelSendInfo : public sigslot::has_slots<> { webrtc::ViEExternalCapture* external_capture_; EncoderMap registered_encoders_; - VideoFormat video_format_; - - rtc::scoped_ptr<StreamParams> stream_params_; - WebRtcLocalStreamInfo local_stream_info_; - int64 interval_; - rtc::CpuMonitor* cpu_monitor_; rtc::scoped_ptr<WebRtcOveruseObserver> overuse_observer_; int old_adaptation_changes_; VideoOptions video_options_; + + VideoFormat adapt_format_; + AdaptFormatType adapt_format_type_; }; const WebRtcVideoEngine::VideoCodecPref @@ -861,17 +935,6 @@ const VideoFormatPod WebRtcVideoEngine::kDefaultMaxVideoFormat = {640, 400, FPS_TO_INTERVAL(30), FOURCC_ANY}; // TODO(ronghuawu): Change to 640x360. -static void UpdateVideoCodec(const cricket::VideoFormat& video_format, - webrtc::VideoCodec* target_codec) { - if ((!target_codec) || (video_format == cricket::VideoFormat())) { - return; - } - target_codec->width = video_format.width; - target_codec->height = video_format.height; - target_codec->maxFramerate = cricket::VideoFormat::IntervalToFps( - video_format.interval); -} - static bool GetCpuOveruseOptions(const VideoOptions& options, webrtc::CpuOveruseOptions* overuse_options) { int underuse_threshold = 0; @@ -1103,11 +1166,8 @@ bool WebRtcVideoEngine::SetDefaultCodec(const VideoCodec& codec) { } ASSERT(!video_codecs_.empty()); - default_codec_format_ = VideoFormat( - video_codecs_[0].width, - video_codecs_[0].height, - VideoFormat::FpsToInterval(video_codecs_[0].framerate), - FOURCC_ANY); + default_codec_format_ = VideoFormatFromCodec(video_codecs_[0]); + return true; } @@ -1774,15 +1834,15 @@ bool WebRtcVideoMediaChannel::MaybeRegisterExternalEncoder( const webrtc::VideoCodec& codec) { // Codec type not supported or encoder already registered, so // nothing to do. - if (!engine()->IsExternalEncoderCodecType(codec.codecType) - || send_channel->IsEncoderRegistered(codec.plType)) { + if (!engine()->IsExternalEncoderCodecType(codec.codecType) || + send_channel->IsEncoderRegistered(codec.plType)) { return true; } webrtc::VideoEncoder* encoder = engine()->CreateExternalEncoder(codec.codecType); if (!encoder) { - // No encoder factor, so nothing to do. + // No external encoder created, so nothing to do. return true; } @@ -1813,7 +1873,9 @@ bool WebRtcVideoMediaChannel::SetSendStreamFormat(uint32 ssrc, LOG(LS_ERROR) << "The specified ssrc " << ssrc << " is not in use."; return false; } - send_channel->set_video_format(format); + + send_channel->SetAdaptFormat( + format, WebRtcVideoChannelSendInfo::kAdaptFormatTypeStream); return true; } @@ -1897,7 +1959,7 @@ bool WebRtcVideoMediaChannel::AddSendStream(const StreamParams& sp) { // If the default channel is already used for sending create a new channel // otherwise use the default channel for sending. int channel_id = kChannelIdUnset; - if (!DefaultSendChannelInUse()) { + if (!DefaultSendChannelIsActive()) { channel_id = default_channel_id_; } else { if (!CreateChannel(ssrc_key, MD_SEND, &channel_id)) { @@ -1955,9 +2017,9 @@ bool WebRtcVideoMediaChannel::RemoveSendStream(uint32 ssrc) { } WebRtcVideoChannelSendInfo* send_channel = GetSendChannelBySsrcKey(ssrc_key); int channel_id = send_channel->channel_id(); - if (IsDefaultChannelId(channel_id) && !send_channel->stream_params()) { - // Default channel will still exist. However, if stream_params() is NULL - // there is no stream to remove. + if (IsDefaultChannelId(channel_id) && !send_channel->IsActive()) { + // Default channel will still exist. However, there is no stream + // to remove. return false; } if (sending_) { @@ -1979,7 +2041,7 @@ bool WebRtcVideoMediaChannel::RemoveSendStream(uint32 ssrc) { // The receive channels depend on the default channel, recycle it instead. if (IsDefaultChannelId(channel_id)) { SetCapturer(GetDefaultSendChannelSsrc(), NULL); - send_channel->ClearStreamParams(); + send_channel->Deactivate(); } else { return DeleteSendChannel(ssrc_key); } @@ -1995,7 +2057,7 @@ bool WebRtcVideoMediaChannel::AddRecvStream(const StreamParams& sp) { // TODO(zhurunz) Remove this once BWE works properly across different send // and receive channels. // Reuse default channel for recv stream in 1:1 call. - if (!InConferenceMode() && first_receive_ssrc_ == kSsrcUnset) { + if (!ConferenceModeIsEnabled() && first_receive_ssrc_ == kSsrcUnset) { LOG(LS_INFO) << "Recv stream " << sp.first_ssrc() << " reuse default channel #" << default_channel_id_; @@ -2211,11 +2273,11 @@ bool WebRtcVideoMediaChannel::SendIntraFrame() { bool WebRtcVideoMediaChannel::HasReadySendChannels() { return !send_channels_.empty() && - ((send_channels_.size() > 1) || DefaultSendChannelInUse()); + ((send_channels_.size() > 1) || DefaultSendChannelIsActive()); } -bool WebRtcVideoMediaChannel::DefaultSendChannelInUse() { - return GetDefaultSendChannel() && GetDefaultSendChannel()->stream_params(); +bool WebRtcVideoMediaChannel::DefaultSendChannelIsActive() { + return GetDefaultSendChannel() && GetDefaultSendChannel()->IsActive(); } bool WebRtcVideoMediaChannel::GetSendChannelSsrcKey(uint32 local_ssrc, @@ -2240,7 +2302,7 @@ bool WebRtcVideoMediaChannel::GetSendChannelSsrcKey(uint32 local_ssrc, for (SendChannelMap::iterator iter = send_channels_.begin(); iter != send_channels_.end(); ++iter) { WebRtcVideoChannelSendInfo* send_channel = iter->second; - if (send_channel->has_ssrc(local_ssrc)) { + if (send_channel->stream_params().has_ssrc(local_ssrc)) { *ssrc_key = iter->first; return true; } @@ -2284,7 +2346,7 @@ bool WebRtcVideoMediaChannel::CreateSendChannelSsrcKey(uint32 local_ssrc, // this point a duplicate SSRC has been detected. return false; } - if (!DefaultSendChannelInUse()) { + if (!DefaultSendChannelIsActive()) { // |ssrc_key| should be kDefaultChannelSsrcKey here as the default // channel should be re-used whenever it is not used. *ssrc_key = kDefaultChannelSsrcKey; @@ -2310,10 +2372,7 @@ int WebRtcVideoMediaChannel::GetSendChannelNum(VideoCapturer* capturer) { } uint32 WebRtcVideoMediaChannel::GetDefaultSendChannelSsrc() { - if (!DefaultSendChannelInUse()) { - return 0; - } - return GetDefaultSendChannel()->stream_params()->first_ssrc(); + return GetDefaultSendChannel()->stream_params().first_ssrc(); } bool WebRtcVideoMediaChannel::DeleteSendChannel(uint32 ssrc_key) { @@ -2377,7 +2436,8 @@ bool WebRtcVideoMediaChannel::RemoveCapturer(uint32 ssrc) { send_channel->set_video_capturer(NULL, engine()->vie()); const int64 timestamp = send_channel->local_stream_info()->time_stamp(); if (send_codec_) { - QueueBlackFrame(ssrc, timestamp, send_codec_->maxFramerate); + QueueBlackFrame(ssrc, timestamp, + VideoFormat::FpsToInterval(send_codec_->maxFramerate)); } return true; } @@ -2418,8 +2478,7 @@ bool WebRtcVideoMediaChannel::GetStats(const StatsOptions& options, WebRtcVideoChannelSendInfo* send_channel = iter->second; const int channel_id = send_channel->channel_id(); VideoSenderInfo sinfo; - const StreamParams* send_params = send_channel->stream_params(); - if (!send_params) { + if (!send_channel->IsActive()) { // This should only happen if the default vie channel is not in use. // This can happen if no streams have ever been added or the stream // corresponding to the default channel has been removed. Note that @@ -2439,8 +2498,9 @@ bool WebRtcVideoMediaChannel::GetStats(const StatsOptions& options, WebRtcLocalStreamInfo* channel_stream_info = send_channel->local_stream_info(); - for (size_t i = 0; i < send_params->ssrcs.size(); ++i) { - sinfo.add_ssrc(send_params->ssrcs[i]); + const StreamParams& sp = send_channel->stream_params(); + for (size_t i = 0; i < sp.ssrcs.size(); ++i) { + sinfo.add_ssrc(sp.ssrcs[i]); } sinfo.codec_name = send_codec_->plName; sinfo.bytes_sent = bytes_sent; @@ -2493,7 +2553,6 @@ bool WebRtcVideoMediaChannel::GetStats(const StatsOptions& options, sinfo.capture_jitter_ms = metrics.capture_jitter_ms; sinfo.avg_encode_ms = metrics.avg_encode_time_ms; sinfo.encode_usage_percent = metrics.encode_usage_percent; - sinfo.encode_rsd = metrics.encode_rsd; sinfo.capture_queue_delay_ms_per_s = metrics.capture_queue_delay_ms_per_s; #else sinfo.capture_jitter_ms = -1; @@ -2723,7 +2782,8 @@ bool WebRtcVideoMediaChannel::SetCapturer(uint32 ssrc, } const int64 timestamp = send_channel->local_stream_info()->time_stamp(); if (send_codec_) { - QueueBlackFrame(ssrc, timestamp, send_codec_->maxFramerate); + QueueBlackFrame(ssrc, timestamp, + VideoFormat::FpsToInterval(send_codec_->maxFramerate)); } return true; } @@ -2745,7 +2805,7 @@ void WebRtcVideoMediaChannel::OnPacketReceived( int processing_channel_id = GetRecvChannelId(ssrc); if (processing_channel_id == kChannelIdUnset) { // Allocate an unsignalled recv channel for processing in conference mode. - if (!InConferenceMode()) { + if (!ConferenceModeIsEnabled()) { // If we can't find or allocate one, use the default. processing_channel_id = default_channel_id_; } else if (!CreateUnsignalledRecvChannel(ssrc, &processing_channel_id)) { @@ -3048,12 +3108,13 @@ void WebRtcVideoMediaChannel::SetInterface(NetworkInterface* iface) { rtc::Socket::OPT_RCVBUF, kVideoRtpBufferSize); - // TODO(sriniv): Remove or re-enable this. - // As part of b/8030474, send-buffer is size now controlled through - // portallocator flags. - // network_interface_->SetOption(NetworkInterface::ST_RTP, - // rtc::Socket::OPT_SNDBUF, - // kVideoRtpBufferSize); + // Speculative change to increase the outbound socket buffer size. + // In b/15152257, we are seeing a significant number of packets discarded + // due to lack of socket buffer space, although it's not yet clear what the + // ideal value should be. + MediaChannel::SetOption(NetworkInterface::ST_RTP, + rtc::Socket::OPT_SNDBUF, + kVideoRtpBufferSize); } void WebRtcVideoMediaChannel::UpdateAspectRatio(int ratio_w, int ratio_h) { @@ -3140,23 +3201,32 @@ bool WebRtcVideoMediaChannel::SendFrame( // Send codec has not been set. No reason to process the frame any further. return false; } - const VideoFormat& video_format = send_channel->video_format(); + + // TODO(pthatcher): Move drop logic to adapter. // If the frame should be dropped. - const bool video_format_set = video_format != cricket::VideoFormat(); - if (video_format_set && - (video_format.width == 0 && video_format.height == 0)) { + if (send_channel->adapt_format_set() && + send_channel->adapt_format().width == 0 && + send_channel->adapt_format().height == 0) { return true; } - // Checks if we need to reset vie send codec. - if (!MaybeResetVieSendCodec(send_channel, - static_cast<int>(frame->GetWidth()), - static_cast<int>(frame->GetHeight()), - is_screencast, NULL)) { - LOG(LS_ERROR) << "MaybeResetVieSendCodec failed with " - << frame->GetWidth() << "x" << frame->GetHeight(); - return false; + bool changed; + send_channel->SetLastCapturedFrameInfo(frame, is_screencast, &changed); + if (changed) { + // If the last captured frame info changed, then calling + // SetSendParams will update to the latest resolution. + VideoSendParams send_params = send_channel->send_params(); + // Note: We must copy the send_params because otherwise the memory + // checker will complain. + if (!SetSendParams(send_channel, send_params)) { + LOG(LS_ERROR) << "SetSendParams from SendFrame failed with " + << frame->GetWidth() << "x" << frame->GetHeight() + << " screencast? " << is_screencast; + return false; + } + LogSendCodecChange("Captured frame size changed"); } + const VideoFrame* frame_out = frame; rtc::scoped_ptr<VideoFrame> processed_frame; // TODO(hellner): Remove the need for disabling mute when screencasting. @@ -3458,6 +3528,8 @@ bool WebRtcVideoMediaChannel::ConfigureSending(int channel_id, LOG_RTCERR2(ConnectCaptureDevice, vie_capture, channel_id); return false; } + + // Set up a new send channel. rtc::scoped_ptr<WebRtcVideoChannelSendInfo> send_channel( new WebRtcVideoChannelSendInfo(channel_id, vie_capture, external_capture, @@ -3549,7 +3621,7 @@ bool WebRtcVideoMediaChannel::SetNackFec(int channel_id, int fec_payload_type, bool nack_enabled) { bool enable = (red_payload_type != -1 && fec_payload_type != -1 && - !InConferenceMode()); + !ConferenceModeIsEnabled()); if (enable) { if (engine_->vie()->rtp()->SetHybridNACKFECStatus( channel_id, nack_enabled, red_payload_type, fec_payload_type) != 0) { @@ -3600,67 +3672,29 @@ bool WebRtcVideoMediaChannel::SetSendCodec( return false; } - const int channel_id = send_channel->channel_id(); - // Make a copy of the codec - webrtc::VideoCodec target_codec = codec; - - // Set the default number of temporal layers for VP8. - if (webrtc::kVideoCodecVP8 == codec.codecType) { - target_codec.codecSpecific.VP8.numberOfTemporalLayers = - kDefaultNumberOfTemporalLayers; + send_channel->SetAdaptFormat( + VideoFormatFromVieCodec(codec), + WebRtcVideoChannelSendInfo::kAdaptFormatTypeCodec); - // Turn off the VP8 error resilience - target_codec.codecSpecific.VP8.resilience = webrtc::kResilienceOff; + MaybeRegisterExternalEncoder(send_channel, codec); - bool enable_denoising = - options_.video_noise_reduction.GetWithDefaultIfUnset(true); - target_codec.codecSpecific.VP8.denoisingOn = enable_denoising; + VideoSendParams send_params = send_channel->send_params(); + send_params.codec = codec; + if (!SetSendParams(send_channel, send_params)) { + return false; } - MaybeRegisterExternalEncoder(send_channel, target_codec); - - // Resolution and framerate may vary for different send channels. - const VideoFormat& video_format = send_channel->video_format(); - UpdateVideoCodec(video_format, &target_codec); - - if (target_codec.width == 0 && target_codec.height == 0) { - const uint32 ssrc = send_channel->stream_params()->first_ssrc(); - LOG(LS_INFO) << "0x0 resolution selected. Captured frames will be dropped " - << "for ssrc: " << ssrc << "."; - } else { - StreamParams* send_params = send_channel->stream_params(); - SanitizeBitrates(channel_id, &target_codec); - webrtc::VideoCodec current_codec; - if (!engine()->vie()->codec()->GetSendCodec(channel_id, current_codec)) { - // Compare against existing configured send codec. - if (current_codec == target_codec) { - // Codec is already configured on channel. no need to apply. - return true; - } - } - - if (0 != engine()->vie()->codec()->SetSendCodec(channel_id, target_codec)) { - LOG_RTCERR2(SetSendCodec, channel_id, target_codec.plName); - return false; - } - - if (send_params) { - if (!SetSendSsrcs(channel_id, *send_params, target_codec)) { - return false; - } - } - // NOTE: SetRtxSendPayloadType must be called after all simulcast SSRCs - // are configured. Otherwise ssrc's configured after this point will use - // the primary PT for RTX. - if (send_rtx_type_ != -1 && - engine()->vie()->rtp()->SetRtxSendPayloadType(channel_id, - send_rtx_type_) != 0) { - LOG_RTCERR2(SetRtxSendPayloadType, channel_id, send_rtx_type_); - return false; - } + // NOTE: SetRtxSendPayloadType must be called after all simulcast SSRCs + // are configured. Otherwise ssrc's configured after this point will use + // the primary PT for RTX. + const int channel_id = send_channel->channel_id(); + if (send_rtx_type_ != -1 && + engine()->vie()->rtp()->SetRtxSendPayloadType(channel_id, + send_rtx_type_) != 0) { + LOG_RTCERR2(SetRtxSendPayloadType, channel_id, send_rtx_type_); + return false; } - send_channel->set_interval( - cricket::VideoFormat::FpsToInterval(target_codec.maxFramerate)); + return true; } @@ -3833,117 +3867,107 @@ int WebRtcVideoMediaChannel::GetRecvChannelId(uint32 ssrc) { return recv_channel_id; } -// If the new frame size is different from the send codec size we set on vie, -// we need to reset the send codec on vie. -// The new send codec size should not exceed send_codec_ which is controlled -// only by the 'jec' logic. -// TODO(pthatcher): Get rid of this function, so we only ever set up -// codecs in a single place. -bool WebRtcVideoMediaChannel::MaybeResetVieSendCodec( +bool WebRtcVideoMediaChannel::SetSendParams( WebRtcVideoChannelSendInfo* send_channel, - int new_width, - int new_height, - bool is_screencast, - bool* reset) { - if (reset) { - *reset = false; - } - ASSERT(send_codec_.get() != NULL); + const VideoSendParams& send_params) { + const int channel_id = send_channel->channel_id(); - webrtc::VideoCodec target_codec = *send_codec_; - const VideoFormat& video_format = send_channel->video_format(); - UpdateVideoCodec(video_format, &target_codec); - - // Vie send codec size should not exceed target_codec. - int target_width = new_width; - int target_height = new_height; - if (!is_screencast && - (new_width > target_codec.width || new_height > target_codec.height)) { - target_width = target_codec.width; - target_height = target_codec.height; - } + CapturedFrameInfo frame; + send_channel->last_captured_frame_info().Get(&frame); + + // TODO(pthatcher): This checking of the max height and width is + // only needed because some unit tests bypass the VideoAdapter, and + // others expect behavior from the adapter different than what it + // actually does. We should fix the tests and remove this block. + VideoFormat max = send_channel->adapt_format(); + size_t max_width = static_cast<size_t>(max.width); + size_t max_height = static_cast<size_t>(max.height); + if (!send_channel->last_captured_frame_info().IsSet() || + (!frame.screencast && + (frame.width > max_width || frame.height > max_height))) { + frame.width = max_width; + frame.height = max_height; + } + + webrtc::VideoCodec codec; + ConfigureVieCodecFromSendParams(channel_id, send_params, frame, &codec); + // TODO(pthatcher): Figure out a clean way to configure the max + // framerate and sanitize the bitrates inside of + // ConfigureVieCodecFromSendParams. + codec.maxFramerate = max.framerate(); + SanitizeBitrates(channel_id, &codec); // Get current vie codec. - webrtc::VideoCodec vie_codec; - const int channel_id = send_channel->channel_id(); - if (engine()->vie()->codec()->GetSendCodec(channel_id, vie_codec) != 0) { + webrtc::VideoCodec current; + if (engine()->vie()->codec()->GetSendCodec(channel_id, current) != 0) { LOG_RTCERR1(GetSendCodec, channel_id); return false; } - const int cur_width = vie_codec.width; - const int cur_height = vie_codec.height; - - // Only reset send codec when there is a size change. Additionally, - // automatic resize needs to be turned off when screencasting and on when - // not screencasting. - // Don't allow automatic resizing for screencasting. - bool automatic_resize = !is_screencast; - // Turn off VP8 frame dropping when screensharing as the current model does - // not work well at low fps. - bool vp8_frame_dropping = !is_screencast; - // TODO(pbos): Remove |video_noise_reduction| and enable it for all - // non-screencast. - bool enable_denoising = - options_.video_noise_reduction.GetWithDefaultIfUnset(true); - // Disable denoising for screencasting. - if (is_screencast) { - enable_denoising = false; - } - int screencast_min_bitrate = - options_.screencast_min_bitrate.GetWithDefaultIfUnset(0); - StreamParams* send_params = send_channel->stream_params(); - bool reset_send_codec = - target_width != cur_width || target_height != cur_height; - if (vie_codec.codecType == webrtc::kVideoCodecVP8) { - reset_send_codec = reset_send_codec || - automatic_resize != vie_codec.codecSpecific.VP8.automaticResizeOn || - enable_denoising != vie_codec.codecSpecific.VP8.denoisingOn || - vp8_frame_dropping != vie_codec.codecSpecific.VP8.frameDroppingOn; - } - - if (reset_send_codec) { - // Set the new codec on vie. - vie_codec.width = target_width; - vie_codec.height = target_height; - vie_codec.maxFramerate = target_codec.maxFramerate; - vie_codec.startBitrate = target_codec.startBitrate; - vie_codec.minBitrate = target_codec.minBitrate; - vie_codec.maxBitrate = target_codec.maxBitrate; - vie_codec.targetBitrate = 0; - if (vie_codec.codecType == webrtc::kVideoCodecVP8) { - vie_codec.codecSpecific.VP8.automaticResizeOn = automatic_resize; - vie_codec.codecSpecific.VP8.denoisingOn = enable_denoising; - vie_codec.codecSpecific.VP8.frameDroppingOn = vp8_frame_dropping; - } - SanitizeBitrates(channel_id, &vie_codec); - - if (engine()->vie()->codec()->SetSendCodec(channel_id, vie_codec) != 0) { + + if (current != codec) { + if (engine()->vie()->codec()->SetSendCodec(channel_id, codec) != 0) { LOG_RTCERR1(SetSendCodec, channel_id); return false; } + } - if (is_screencast) { - engine()->vie()->rtp()->SetMinTransmitBitrate(channel_id, - screencast_min_bitrate); - } else { - // In case of switching from screencast to regular capture, set - // min bitrate padding and pacer back to defaults. - engine()->vie()->rtp()->SetMinTransmitBitrate(channel_id, 0); - } - engine()->vie()->rtp()->SetTransmissionSmoothingStatus(channel_id, true); - // TODO(sriniv): SetSendCodec already sets ssrc's like below. - // Consider removing. - if (send_params) { - if (!SetSendSsrcs(channel_id, *send_params, target_codec)) { - return false; - } + if (frame.screencast) { + int screencast_min_bitrate = + options_.screencast_min_bitrate.GetWithDefaultIfUnset(0); + engine()->vie()->rtp()->SetMinTransmitBitrate(channel_id, + screencast_min_bitrate); + } else { + // In case of switching from screencast to regular capture, set + // min bitrate padding and pacer back to defaults. + engine()->vie()->rtp()->SetMinTransmitBitrate(channel_id, 0); + } + engine()->vie()->rtp()->SetTransmissionSmoothingStatus(channel_id, true); + + if (send_channel->IsActive()) { + if (!SetSendSsrcs(channel_id, send_params.stream, codec)) { + return false; } - if (reset) { - *reset = true; + } + + send_channel->set_send_params(send_params); + return true; +} + +bool WebRtcVideoMediaChannel::ConfigureVieCodecFromSendParams( + int channel_id, + const VideoSendParams& send_params, + const CapturedFrameInfo& last_captured_frame_info, + webrtc::VideoCodec* codec_out) { + webrtc::VideoCodec codec = send_params.codec; + + codec.width = static_cast<int>(last_captured_frame_info.width); + codec.height = static_cast<int>(last_captured_frame_info.height); + codec.targetBitrate = 0; + if (codec.codecType == webrtc::kVideoCodecVP8) { + codec.codecSpecific.VP8.numberOfTemporalLayers = + kDefaultNumberOfTemporalLayers; + codec.codecSpecific.VP8.resilience = webrtc::kResilienceOff; + } + + if (last_captured_frame_info.screencast) { + codec.mode = webrtc::kScreensharing; + if (codec.codecType == webrtc::kVideoCodecVP8) { + codec.codecSpecific.VP8.denoisingOn = false; + codec.codecSpecific.VP8.automaticResizeOn = false; + codec.codecSpecific.VP8.frameDroppingOn = false; + } + } else { + codec.mode = webrtc::kRealtimeVideo; + if (codec.codecType == webrtc::kVideoCodecVP8) { + // TODO(pthatcher): Pass in options in VideoSendParams. + codec.codecSpecific.VP8.denoisingOn = + options_.video_noise_reduction.GetWithDefaultIfUnset(true); + codec.codecSpecific.VP8.automaticResizeOn = true; + codec.codecSpecific.VP8.frameDroppingOn = true; } - LogSendCodecChange("Capture size changed"); } + *codec_out = codec; return true; } @@ -3987,10 +4011,9 @@ void WebRtcVideoMediaChannel::SanitizeBitrates( } void WebRtcVideoMediaChannel::OnMessage(rtc::Message* msg) { - FlushBlackFrameData* black_frame_data = - static_cast<FlushBlackFrameData*>(msg->pdata); - FlushBlackFrame(black_frame_data->ssrc, black_frame_data->timestamp); - delete black_frame_data; + FlushBlackFrameData* data = static_cast<FlushBlackFrameData*>(msg->pdata); + FlushBlackFrame(data->ssrc, data->timestamp, data->interval); + delete data; } int WebRtcVideoMediaChannel::SendPacket(int channel, const void* data, @@ -4007,19 +4030,18 @@ int WebRtcVideoMediaChannel::SendRTCPPacket(int channel, } void WebRtcVideoMediaChannel::QueueBlackFrame(uint32 ssrc, int64 timestamp, - int framerate) { + int interval) { if (timestamp) { FlushBlackFrameData* black_frame_data = new FlushBlackFrameData( - ssrc, - timestamp); + ssrc, timestamp, interval); const int delay_ms = static_cast<int>( - 2 * cricket::VideoFormat::FpsToInterval(framerate) * - rtc::kNumMillisecsPerSec / rtc::kNumNanosecsPerSec); + 2 * interval * rtc::kNumMillisecsPerSec / rtc::kNumNanosecsPerSec); worker_thread()->PostDelayed(delay_ms, this, 0, black_frame_data); } } -void WebRtcVideoMediaChannel::FlushBlackFrame(uint32 ssrc, int64 timestamp) { +void WebRtcVideoMediaChannel::FlushBlackFrame( + uint32 ssrc, int64 timestamp, int interval) { WebRtcVideoChannelSendInfo* send_channel = GetSendChannelBySsrc(ssrc); if (!send_channel) { return; @@ -4041,7 +4063,7 @@ void WebRtcVideoMediaChannel::FlushBlackFrame(uint32 ssrc, int64 timestamp) { WebRtcVideoFrame black_frame; // Black frame is not screencast. const bool screencasting = false; - const int64 timestamp_delta = send_channel->interval(); + const int64 timestamp_delta = interval; if (!black_frame.InitToBlack(send_codec_->width, send_codec_->height, 1, 1, last_frame_elapsed_time + timestamp_delta, last_frame_time_stamp + timestamp_delta) || @@ -4094,7 +4116,7 @@ bool WebRtcVideoMediaChannel::SetHeaderExtension(ExtensionSetterFunction setter, bool WebRtcVideoMediaChannel::SetPrimaryAndRtxSsrcs( int channel_id, int idx, uint32 primary_ssrc, - const StreamParams& send_params) { + const StreamParams& sp) { LOG(LS_INFO) << "Set primary ssrc " << primary_ssrc << " on channel " << channel_id << " idx " << idx; if (engine()->vie()->rtp()->SetLocalSSRC( @@ -4105,7 +4127,7 @@ bool WebRtcVideoMediaChannel::SetPrimaryAndRtxSsrcs( } uint32 rtx_ssrc = 0; - if (send_params.GetFidSsrc(primary_ssrc, &rtx_ssrc)) { + if (sp.GetFidSsrc(primary_ssrc, &rtx_ssrc)) { LOG(LS_INFO) << "Set rtx ssrc " << rtx_ssrc << " on channel " << channel_id << " idx " << idx; if (engine()->vie()->rtp()->SetLocalSSRC( diff --git a/media/webrtc/webrtcvideoengine.h b/media/webrtc/webrtcvideoengine.h index 72ba9df..cc81ee9 100644 --- a/media/webrtc/webrtcvideoengine.h +++ b/media/webrtc/webrtcvideoengine.h @@ -234,6 +234,22 @@ class WebRtcVideoEngine : public sigslot::has_slots<>, rtc::scoped_ptr<rtc::CpuMonitor> cpu_monitor_; }; +struct CapturedFrameInfo { + CapturedFrameInfo() : width(0), height(0), screencast(false) {} + CapturedFrameInfo(size_t width, size_t height, bool screencast) : + width(width), height(height), screencast(screencast) {} + + size_t width; + size_t height; + bool screencast; +}; + +// TODO(pthatcher): Add VideoOptions. +struct VideoSendParams { + webrtc::VideoCodec codec; + StreamParams stream; +}; + class WebRtcVideoMediaChannel : public rtc::MessageHandler, public VideoMediaChannel, public webrtc::Transport { @@ -316,6 +332,22 @@ class WebRtcVideoMediaChannel : public rtc::MessageHandler, virtual int SendPacket(int channel, const void* data, int len) OVERRIDE; virtual int SendRTCPPacket(int channel, const void* data, int len) OVERRIDE; + bool ConferenceModeIsEnabled() const { + return options_.conference_mode.GetWithDefaultIfUnset(false); + } + + // We take lots of things as input from applications (packaged in + // params), but ViE wants lots of those packed instead as a + // webrtc::VideoCodec. This is where we convert between the inputs + // we get from the applications and the input to give to ViE. We + // also configure the codec differently depending on the latest + // frame that we have received (in particular, depending on the + // resolution and whether the it was a screencast frame or not). + virtual bool ConfigureVieCodecFromSendParams( + int channel_id, + const VideoSendParams& send_params, + const CapturedFrameInfo& last_captured_frame_info, + webrtc::VideoCodec* codec); // Checks the current bitrate estimate and modifies the bitrates // accordingly, including converting kAutoBandwidth to the correct defaults. virtual void SanitizeBitrates( @@ -359,6 +391,9 @@ class WebRtcVideoMediaChannel : public rtc::MessageHandler, bool SetSendCodec(const webrtc::VideoCodec& codec); bool SetSendCodec(WebRtcVideoChannelSendInfo* send_channel, const webrtc::VideoCodec& codec); + bool SetSendParams(WebRtcVideoChannelSendInfo* send_channel, + const VideoSendParams& params); + // Prepares the channel with channel id |info->channel_id()| to receive all // codecs in |receive_codecs_| and start receive packets. bool SetReceiveCodecs(WebRtcVideoChannelRecvInfo* info); @@ -370,12 +405,6 @@ class WebRtcVideoMediaChannel : public rtc::MessageHandler, bool MaybeRegisterExternalEncoder( WebRtcVideoChannelSendInfo* send_channel, const webrtc::VideoCodec& codec); - // Given captured video frame size, checks if we need to reset vie send codec. - // |reset| is set to whether resetting has happened on vie or not. - // Returns false on error. - bool MaybeResetVieSendCodec(WebRtcVideoChannelSendInfo* send_channel, - int new_width, int new_height, bool is_screencast, - bool* reset); // Helper function for starting the sending of media on all channels or // |channel_id|. Note that these two function do not change |sending_|. bool StartSend(); @@ -387,7 +416,7 @@ class WebRtcVideoMediaChannel : public rtc::MessageHandler, bool SendIntraFrame(int channel_id); bool HasReadySendChannels(); - bool DefaultSendChannelInUse(); + bool DefaultSendChannelIsActive(); // Returns the ssrc key corresponding to the provided local SSRC in // |ssrc_key|. The return value is true upon success. If the local @@ -414,14 +443,11 @@ class WebRtcVideoMediaChannel : public rtc::MessageHandler, WebRtcVideoChannelRecvInfo* GetDefaultRecvChannel(); WebRtcVideoChannelRecvInfo* GetRecvChannelBySsrc(uint32 ssrc); - bool InConferenceMode() const { - return options_.conference_mode.GetWithDefaultIfUnset(false); - } bool RemoveCapturer(uint32 ssrc); rtc::MessageQueue* worker_thread() { return engine_->worker_thread(); } - void QueueBlackFrame(uint32 ssrc, int64 timestamp, int framerate); - void FlushBlackFrame(uint32 ssrc, int64 timestamp); + void QueueBlackFrame(uint32 ssrc, int64 timestamp, int interval); + void FlushBlackFrame(uint32 ssrc, int64 timestamp, int interval); void SetNetworkTransmissionState(bool is_transmitting); diff --git a/media/webrtc/webrtcvideoengine2.cc b/media/webrtc/webrtcvideoengine2.cc index 96df0f7..d79f71d 100644 --- a/media/webrtc/webrtcvideoengine2.cc +++ b/media/webrtc/webrtcvideoengine2.cc @@ -42,6 +42,7 @@ #include "webrtc/base/logging.h" #include "webrtc/base/stringutils.h" #include "webrtc/call.h" +#include "webrtc/video_decoder.h" #include "webrtc/video_encoder.h" #define UNIMPLEMENTED \ @@ -56,10 +57,18 @@ static bool CodecNameMatches(const std::string& name1, return _stricmp(name1.c_str(), name2.c_str()) == 0; } +const char* kInternallySupportedCodecs[] = { + kVp8CodecName, +}; + // True if codec is supported by a software implementation that's always // available. static bool CodecIsInternallySupported(const std::string& codec_name) { - return CodecNameMatches(codec_name, kVp8CodecName); + for (size_t i = 0; i < ARRAY_SIZE(kInternallySupportedCodecs); ++i) { + if (CodecNameMatches(codec_name, kInternallySupportedCodecs[i])) + return true; + } + return false; } static std::string CodecVectorToString(const std::vector<VideoCodec>& codecs) { @@ -117,6 +126,8 @@ static const int kDefaultQpMax = 56; static const int kDefaultRtcpReceiverReportSsrc = 1; +static const int kConferenceModeTemporalLayerBitrateBps = 100000; + // External video encoders are given payloads 120-127. This also means that we // only support up to 8 external payload types. static const int kExternalVideoPayloadTypeBase = 120; @@ -238,8 +249,9 @@ std::vector<webrtc::VideoStream> WebRtcVideoEncoderFactory2::CreateVideoStreams( const VideoOptions& options, size_t num_streams) { if (num_streams != 1) { - LOG(LS_ERROR) << "Unsupported number of streams: " << num_streams; - return std::vector<webrtc::VideoStream>(); + LOG(LS_WARNING) << "Unsupported number of streams (" << num_streams + << "), falling back to one."; + num_streams = 1; } webrtc::VideoStream stream; @@ -250,6 +262,12 @@ std::vector<webrtc::VideoStream> WebRtcVideoEncoderFactory2::CreateVideoStreams( int min_bitrate = kMinVideoBitrate; codec.GetParam(kCodecParamMinBitrate, &min_bitrate); + // Clamp the min video bitrate, this is set from JavaScript directly and needs + // to be sanitized. + if (min_bitrate < kMinVideoBitrate) { + min_bitrate = kMinVideoBitrate; + } + int max_bitrate = kMaxVideoBitrate; codec.GetParam(kCodecParamMaxBitrate, &max_bitrate); stream.min_bitrate_bps = min_bitrate * 1000; @@ -419,10 +437,12 @@ WebRtcVideoChannel2* WebRtcVideoEngine2::CreateChannel( assert(initialized_); LOG(LS_INFO) << "CreateChannel: " << (voice_channel != NULL ? "With" : "Without") - << " voice channel."; + << " voice channel. Options: " << options.ToString(); WebRtcVideoChannel2* channel = new WebRtcVideoChannel2(call_factory_, + voice_engine_, voice_channel, + options, external_encoder_factory_, external_decoder_factory_, GetVideoEncoderFactory()); @@ -430,7 +450,6 @@ WebRtcVideoChannel2* WebRtcVideoEngine2::CreateChannel( delete channel; return NULL; } - channel->SetOptions(options); channel->SetRecvCodecs(video_codecs_); return channel; } @@ -744,24 +763,35 @@ class WebRtcVideoRenderFrame : public VideoFrame { WebRtcVideoChannel2::WebRtcVideoChannel2( WebRtcCallFactory* call_factory, + WebRtcVoiceEngine* voice_engine, VoiceMediaChannel* voice_channel, + const VideoOptions& options, WebRtcVideoEncoderFactory* external_encoder_factory, WebRtcVideoDecoderFactory* external_decoder_factory, WebRtcVideoEncoderFactory2* encoder_factory) : unsignalled_ssrc_handler_(&default_unsignalled_ssrc_handler_), + voice_channel_(voice_channel), external_encoder_factory_(external_encoder_factory), external_decoder_factory_(external_decoder_factory), encoder_factory_(encoder_factory) { - // TODO(pbos): Connect the video and audio with |voice_channel|. + SetDefaultOptions(); + options_.SetAll(options); webrtc::Call::Config config(this); config.overuse_callback = this; + if (voice_engine != NULL) { + config.voice_engine = voice_engine->voe()->engine(); + } + + // Set start bitrate for the call. A default is provided by SetDefaultOptions. + int start_bitrate_kbps; + options_.video_start_bitrate.Get(&start_bitrate_kbps); + config.stream_start_bitrate_bps = start_bitrate_kbps * 1000; + call_.reset(call_factory->CreateCall(config)); rtcp_receiver_report_ssrc_ = kDefaultRtcpReceiverReportSsrc; sending_ = false; default_send_ssrc_ = 0; - - SetDefaultOptions(); } void WebRtcVideoChannel2::SetDefaultOptions() { @@ -769,6 +799,9 @@ void WebRtcVideoChannel2::SetDefaultOptions() { options_.suspend_below_min_bitrate.Set(false); options_.use_payload_padding.Set(false); options_.video_noise_reduction.Set(true); + options_.video_start_bitrate.Set( + webrtc::Call::Config::kDefaultStartBitrateBps / 1000); + options_.screencast_min_bitrate.Set(0); } WebRtcVideoChannel2::~WebRtcVideoChannel2() { @@ -789,6 +822,37 @@ WebRtcVideoChannel2::~WebRtcVideoChannel2() { bool WebRtcVideoChannel2::Init() { return true; } +bool WebRtcVideoChannel2::CodecIsExternallySupported( + const std::string& name) const { + if (external_encoder_factory_ == NULL) { + return false; + } + + const std::vector<WebRtcVideoEncoderFactory::VideoCodec> external_codecs = + external_encoder_factory_->codecs(); + for (size_t c = 0; c < external_codecs.size(); ++c) { + if (CodecNameMatches(name, external_codecs[c].name)) { + return true; + } + } + return false; +} + +std::vector<WebRtcVideoChannel2::VideoCodecSettings> +WebRtcVideoChannel2::FilterSupportedCodecs( + const std::vector<WebRtcVideoChannel2::VideoCodecSettings>& mapped_codecs) + const { + std::vector<VideoCodecSettings> supported_codecs; + for (size_t i = 0; i < mapped_codecs.size(); ++i) { + const VideoCodecSettings& codec = mapped_codecs[i]; + if (CodecIsInternallySupported(codec.codec.name) || + CodecIsExternallySupported(codec.codec.name)) { + supported_codecs.push_back(codec); + } + } + return supported_codecs; +} + bool WebRtcVideoChannel2::SetRecvCodecs(const std::vector<VideoCodec>& codecs) { LOG(LS_INFO) << "SetRecvCodecs: " << CodecVectorToString(codecs); if (!ValidateCodecFormats(codecs)) { @@ -797,21 +861,19 @@ bool WebRtcVideoChannel2::SetRecvCodecs(const std::vector<VideoCodec>& codecs) { const std::vector<VideoCodecSettings> mapped_codecs = MapCodecs(codecs); if (mapped_codecs.empty()) { - LOG(LS_ERROR) << "SetRecvCodecs called without video codec payloads."; + LOG(LS_ERROR) << "SetRecvCodecs called without any video codecs."; return false; } - // TODO(pbos): Add a decoder factory which controls supported codecs. - // Blocked on webrtc:2854. - for (size_t i = 0; i < mapped_codecs.size(); ++i) { - if (!CodecNameMatches(mapped_codecs[i].codec.name, kVp8CodecName)) { - LOG(LS_ERROR) << "SetRecvCodecs called with unsupported codec: '" - << mapped_codecs[i].codec.name << "'"; - return false; - } + const std::vector<VideoCodecSettings> supported_codecs = + FilterSupportedCodecs(mapped_codecs); + + if (mapped_codecs.size() != supported_codecs.size()) { + LOG(LS_ERROR) << "SetRecvCodecs called with unsupported video codecs."; + return false; } - recv_codecs_ = mapped_codecs; + recv_codecs_ = supported_codecs; rtc::CritScope stream_lock(&stream_crit_); for (std::map<uint32, WebRtcVideoReceiveStream*>::iterator it = @@ -998,8 +1060,20 @@ bool WebRtcVideoChannel2::AddRecvStream(const StreamParams& sp) { webrtc::VideoReceiveStream::Config config; ConfigureReceiverRtp(&config, sp); - receive_streams_[ssrc] = - new WebRtcVideoReceiveStream(call_.get(), config, recv_codecs_); + + // Set up A/V sync if there is a VoiceChannel. + // TODO(pbos): The A/V is synched by the receiving channel. So we need to know + // the SSRC of the remote audio channel in order to sync the correct webrtc + // VoiceEngine channel. For now sync the first channel in non-conference to + // match existing behavior in WebRtcVideoEngine. + if (voice_channel_ != NULL && receive_streams_.empty() && + !options_.conference_mode.GetWithDefaultIfUnset(false)) { + config.audio_channel_id = + static_cast<WebRtcVoiceMediaChannel*>(voice_channel_)->voe_channel(); + } + + receive_streams_[ssrc] = new WebRtcVideoReceiveStream( + call_.get(), external_decoder_factory_, config, recv_codecs_); return true; } @@ -1284,12 +1358,13 @@ void WebRtcVideoChannel2::SetInterface(NetworkInterface* iface) { rtc::Socket::OPT_RCVBUF, kVideoRtpBufferSize); - // TODO(sriniv): Remove or re-enable this. - // As part of b/8030474, send-buffer is size now controlled through - // portallocator flags. - // network_interface_->SetOption(NetworkInterface::ST_RTP, - // rtc::Socket::OPT_SNDBUF, - // kVideoRtpBufferSize); + // Speculative change to increase the outbound socket buffer size. + // In b/15152257, we are seeing a significant number of packets discarded + // due to lack of socket buffer space, although it's not yet clear what the + // ideal value should be. + MediaChannel::SetOption(NetworkInterface::ST_RTP, + rtc::Socket::OPT_SNDBUF, + kVideoRtpBufferSize); } void WebRtcVideoChannel2::UpdateAspectRatio(int ratio_w, int ratio_h) { @@ -1663,30 +1738,47 @@ void WebRtcVideoChannel2::WebRtcVideoSendStream::SetRtpExtensions( void WebRtcVideoChannel2::WebRtcVideoSendStream::SetDimensions( int width, int height, - bool override_max) { + bool is_screencast) { + if (last_dimensions_.width == width && last_dimensions_.height == height && + last_dimensions_.is_screencast == is_screencast) { + // Configured using the same parameters, do not reconfigure. + return; + } + + last_dimensions_.width = width; + last_dimensions_.height = height; + last_dimensions_.is_screencast = is_screencast; + assert(!parameters_.encoder_config.streams.empty()); LOG(LS_VERBOSE) << "SetDimensions: " << width << "x" << height; VideoCodecSettings codec_settings; parameters_.codec_settings.Get(&codec_settings); // Restrict dimensions according to codec max. - if (!override_max) { + if (!is_screencast) { if (codec_settings.codec.width < width) width = codec_settings.codec.width; if (codec_settings.codec.height < height) height = codec_settings.codec.height; } - if (parameters_.encoder_config.streams.back().width == width && - parameters_.encoder_config.streams.back().height == height) { - return; - } - webrtc::VideoEncoderConfig encoder_config = parameters_.encoder_config; encoder_config.encoder_specific_settings = encoder_factory_->CreateVideoEncoderSettings(codec_settings.codec, parameters_.options); + if (is_screencast) { + int screencast_min_bitrate_kbps; + parameters_.options.screencast_min_bitrate.Get( + &screencast_min_bitrate_kbps); + encoder_config.min_transmit_bitrate_bps = + screencast_min_bitrate_kbps * 1000; + encoder_config.content_type = webrtc::VideoEncoderConfig::kScreenshare; + } else { + encoder_config.min_transmit_bitrate_bps = 0; + encoder_config.content_type = webrtc::VideoEncoderConfig::kRealtimeVideo; + } + VideoCodec codec = codec_settings.codec; codec.width = width; codec.height = height; @@ -1694,6 +1786,14 @@ void WebRtcVideoChannel2::WebRtcVideoSendStream::SetDimensions( encoder_config.streams = encoder_factory_->CreateVideoStreams( codec, parameters_.options, parameters_.config.rtp.ssrcs.size()); + // Conference mode screencast uses 2 temporal layers split at 100kbit. + if (parameters_.options.conference_mode.GetWithDefaultIfUnset(false) && + is_screencast && encoder_config.streams.size() == 1) { + encoder_config.streams[0].temporal_layer_thresholds_bps.clear(); + encoder_config.streams[0].temporal_layer_thresholds_bps.push_back( + kConferenceModeTemporalLayerBitrateBps); + } + bool stream_reconfigured = stream_->ReconfigureVideoEncoder(encoder_config); encoder_factory_->DestroyVideoEncoderSettings( @@ -1826,11 +1926,13 @@ void WebRtcVideoChannel2::WebRtcVideoSendStream::RecreateWebRtcStream() { WebRtcVideoChannel2::WebRtcVideoReceiveStream::WebRtcVideoReceiveStream( webrtc::Call* call, + WebRtcVideoDecoderFactory* external_decoder_factory, const webrtc::VideoReceiveStream::Config& config, const std::vector<VideoCodecSettings>& recv_codecs) : call_(call), stream_(NULL), config_(config), + external_decoder_factory_(external_decoder_factory), renderer_(NULL), last_width_(-1), last_height_(-1) { @@ -1841,39 +1943,67 @@ WebRtcVideoChannel2::WebRtcVideoReceiveStream::WebRtcVideoReceiveStream( WebRtcVideoChannel2::WebRtcVideoReceiveStream::~WebRtcVideoReceiveStream() { call_->DestroyVideoReceiveStream(stream_); + ClearDecoders(&allocated_decoders_); +} + +WebRtcVideoChannel2::WebRtcVideoReceiveStream::AllocatedDecoder +WebRtcVideoChannel2::WebRtcVideoReceiveStream::CreateOrReuseVideoDecoder( + std::vector<AllocatedDecoder>* old_decoders, + const VideoCodec& codec) { + webrtc::VideoCodecType type = CodecTypeFromName(codec.name); + + for (size_t i = 0; i < old_decoders->size(); ++i) { + if ((*old_decoders)[i].type == type) { + AllocatedDecoder decoder = (*old_decoders)[i]; + (*old_decoders)[i] = old_decoders->back(); + old_decoders->pop_back(); + return decoder; + } + } + + if (external_decoder_factory_ != NULL) { + webrtc::VideoDecoder* decoder = + external_decoder_factory_->CreateVideoDecoder(type); + if (decoder != NULL) { + return AllocatedDecoder(decoder, type, true); + } + } + + if (type == webrtc::kVideoCodecVP8) { + return AllocatedDecoder( + webrtc::VideoDecoder::Create(webrtc::VideoDecoder::kVp8), type, false); + } + + // This shouldn't happen, we should not be trying to create something we don't + // support. + assert(false); + return AllocatedDecoder(NULL, webrtc::kVideoCodecUnknown, false); } void WebRtcVideoChannel2::WebRtcVideoReceiveStream::SetRecvCodecs( const std::vector<VideoCodecSettings>& recv_codecs) { - // TODO(pbos): Reconfigure RTX based on incoming recv_codecs. - // TODO(pbos): Base receive codecs off recv_codecs_ and set up using a - // DecoderFactory similar to send side. Pending webrtc:2854. - // Also set up default codecs if there's nothing in recv_codecs_. - webrtc::VideoCodec codec; - memset(&codec, 0, sizeof(codec)); - - codec.plType = kDefaultVideoCodecPref.payload_type; - strcpy(codec.plName, kDefaultVideoCodecPref.name); - codec.codecType = webrtc::kVideoCodecVP8; - codec.codecSpecific.VP8.resilience = webrtc::kResilientStream; - codec.codecSpecific.VP8.numberOfTemporalLayers = 1; - codec.codecSpecific.VP8.denoisingOn = true; - codec.codecSpecific.VP8.errorConcealmentOn = false; - codec.codecSpecific.VP8.automaticResizeOn = false; - codec.codecSpecific.VP8.frameDroppingOn = true; - codec.codecSpecific.VP8.keyFrameInterval = 3000; - // Bitrates don't matter and are ignored for the receiver. This is put in to - // have the current underlying implementation accept the VideoCodec. - codec.minBitrate = codec.startBitrate = codec.maxBitrate = 300; - config_.codecs.clear(); - config_.codecs.push_back(codec); + std::vector<AllocatedDecoder> old_decoders = allocated_decoders_; + allocated_decoders_.clear(); + config_.decoders.clear(); + for (size_t i = 0; i < recv_codecs.size(); ++i) { + AllocatedDecoder allocated_decoder = + CreateOrReuseVideoDecoder(&old_decoders, recv_codecs[i].codec); + allocated_decoders_.push_back(allocated_decoder); - config_.rtp.fec = recv_codecs.front().fec; + webrtc::VideoReceiveStream::Decoder decoder; + decoder.decoder = allocated_decoder.decoder; + decoder.payload_type = recv_codecs[i].codec.id; + decoder.payload_name = recv_codecs[i].codec.name; + config_.decoders.push_back(decoder); + } + // TODO(pbos): Reconfigure RTX based on incoming recv_codecs. + config_.rtp.fec = recv_codecs.front().fec; config_.rtp.nack.rtp_history_ms = IsNackEnabled(recv_codecs.begin()->codec) ? kNackHistoryMs : 0; config_.rtp.remb = IsRembEnabled(recv_codecs.begin()->codec); + ClearDecoders(&old_decoders); RecreateWebRtcStream(); } @@ -1891,6 +2021,19 @@ void WebRtcVideoChannel2::WebRtcVideoReceiveStream::RecreateWebRtcStream() { stream_->Start(); } +void WebRtcVideoChannel2::WebRtcVideoReceiveStream::ClearDecoders( + std::vector<AllocatedDecoder>* allocated_decoders) { + for (size_t i = 0; i < allocated_decoders->size(); ++i) { + if ((*allocated_decoders)[i].external) { + external_decoder_factory_->DestroyVideoDecoder( + (*allocated_decoders)[i].decoder); + } else { + delete (*allocated_decoders)[i].decoder; + } + } + allocated_decoders->clear(); +} + void WebRtcVideoChannel2::WebRtcVideoReceiveStream::RenderFrame( const webrtc::I420VideoFrame& frame, int time_to_render_ms) { @@ -2050,31 +2193,6 @@ WebRtcVideoChannel2::MapCodecs(const std::vector<VideoCodec>& codecs) { return video_codecs; } -std::vector<WebRtcVideoChannel2::VideoCodecSettings> -WebRtcVideoChannel2::FilterSupportedCodecs( - const std::vector<WebRtcVideoChannel2::VideoCodecSettings>& mapped_codecs) { - std::vector<VideoCodecSettings> supported_codecs; - for (size_t i = 0; i < mapped_codecs.size(); ++i) { - const VideoCodecSettings& codec = mapped_codecs[i]; - if (CodecIsInternallySupported(codec.codec.name)) { - supported_codecs.push_back(codec); - } - - if (external_encoder_factory_ == NULL) { - continue; - } - const std::vector<WebRtcVideoEncoderFactory::VideoCodec> external_codecs = - external_encoder_factory_->codecs(); - for (size_t c = 0; c < external_codecs.size(); ++c) { - if (CodecNameMatches(codec.codec.name, external_codecs[c].name)) { - supported_codecs.push_back(codec); - break; - } - } - } - return supported_codecs; -} - } // namespace cricket #endif // HAVE_WEBRTC_VIDEO diff --git a/media/webrtc/webrtcvideoengine2.h b/media/webrtc/webrtcvideoengine2.h index 8831a1b..0b812ef 100644 --- a/media/webrtc/webrtcvideoengine2.h +++ b/media/webrtc/webrtcvideoengine2.h @@ -211,7 +211,9 @@ class WebRtcVideoChannel2 : public rtc::MessageHandler, public webrtc::LoadObserver { public: WebRtcVideoChannel2(WebRtcCallFactory* call_factory, + WebRtcVoiceEngine* voice_engine, VoiceMediaChannel* voice_channel, + const VideoOptions& options, WebRtcVideoEncoderFactory* external_encoder_factory, WebRtcVideoDecoderFactory* external_decoder_factory, WebRtcVideoEncoderFactory2* encoder_factory); @@ -274,6 +276,7 @@ class WebRtcVideoChannel2 : public rtc::MessageHandler, private: void ConfigureReceiverRtp(webrtc::VideoReceiveStream::Config* config, const StreamParams& sp) const; + bool CodecIsExternallySupported(const std::string& name) const; struct VideoCodecSettings { VideoCodecSettings(); @@ -345,6 +348,13 @@ class WebRtcVideoChannel2 : public rtc::MessageHandler, bool external; }; + struct LastDimensions { + LastDimensions() : width(-1), height(-1), is_screencast(false) {} + int width; + int height; + bool is_screencast; + }; + AllocatedEncoder CreateVideoEncoder(const VideoCodec& codec) EXCLUSIVE_LOCKS_REQUIRED(lock_); void DestroyVideoEncoder(AllocatedEncoder* encoder); @@ -352,8 +362,7 @@ class WebRtcVideoChannel2 : public rtc::MessageHandler, const VideoOptions& options) EXCLUSIVE_LOCKS_REQUIRED(lock_); void RecreateWebRtcStream() EXCLUSIVE_LOCKS_REQUIRED(lock_); - // When |override_max| is false constrain width/height to codec dimensions. - void SetDimensions(int width, int height, bool override_max) + void SetDimensions(int width, int height, bool is_screencast) EXCLUSIVE_LOCKS_REQUIRED(lock_); webrtc::Call* const call_; @@ -364,6 +373,7 @@ class WebRtcVideoChannel2 : public rtc::MessageHandler, webrtc::VideoSendStream* stream_ GUARDED_BY(lock_); VideoSendStreamParameters parameters_ GUARDED_BY(lock_); AllocatedEncoder allocated_encoder_ GUARDED_BY(lock_); + LastDimensions last_dimensions_ GUARDED_BY(lock_); VideoCapturer* capturer_ GUARDED_BY(lock_); bool sending_ GUARDED_BY(lock_); @@ -381,6 +391,7 @@ class WebRtcVideoChannel2 : public rtc::MessageHandler, public: WebRtcVideoReceiveStream( webrtc::Call*, + WebRtcVideoDecoderFactory* external_decoder_factory, const webrtc::VideoReceiveStream::Config& config, const std::vector<VideoCodecSettings>& recv_codecs); ~WebRtcVideoReceiveStream(); @@ -397,14 +408,32 @@ class WebRtcVideoChannel2 : public rtc::MessageHandler, VideoReceiverInfo GetVideoReceiverInfo(); private: + struct AllocatedDecoder { + AllocatedDecoder(webrtc::VideoDecoder* decoder, + webrtc::VideoCodecType type, + bool external) + : decoder(decoder), type(type), external(external) {} + webrtc::VideoDecoder* decoder; + webrtc::VideoCodecType type; + bool external; + }; + void SetSize(int width, int height); void RecreateWebRtcStream(); + AllocatedDecoder CreateOrReuseVideoDecoder( + std::vector<AllocatedDecoder>* old_decoder, + const VideoCodec& codec); + void ClearDecoders(std::vector<AllocatedDecoder>* allocated_decoders); + webrtc::Call* const call_; webrtc::VideoReceiveStream* stream_; webrtc::VideoReceiveStream::Config config_; + WebRtcVideoDecoderFactory* const external_decoder_factory_; + std::vector<AllocatedDecoder> allocated_decoders_; + rtc::CriticalSection renderer_lock_; cricket::VideoRenderer* renderer_ GUARDED_BY(renderer_lock_); int last_width_ GUARDED_BY(renderer_lock_); @@ -423,7 +452,7 @@ class WebRtcVideoChannel2 : public rtc::MessageHandler, static std::vector<VideoCodecSettings> MapCodecs( const std::vector<VideoCodec>& codecs); std::vector<VideoCodecSettings> FilterSupportedCodecs( - const std::vector<VideoCodecSettings>& mapped_codecs); + const std::vector<VideoCodecSettings>& mapped_codecs) const; void FillSenderStats(VideoMediaInfo* info); void FillReceiverStats(VideoMediaInfo* info); @@ -449,6 +478,7 @@ class WebRtcVideoChannel2 : public rtc::MessageHandler, Settable<VideoCodecSettings> send_codec_; std::vector<webrtc::RtpExtension> send_rtp_extensions_; + VoiceMediaChannel* const voice_channel_; WebRtcVideoEncoderFactory* const external_encoder_factory_; WebRtcVideoDecoderFactory* const external_decoder_factory_; WebRtcVideoEncoderFactory2* const encoder_factory_; diff --git a/media/webrtc/webrtcvideoengine2_unittest.cc b/media/webrtc/webrtcvideoengine2_unittest.cc index 85b7b9e..0b85723 100644 --- a/media/webrtc/webrtcvideoengine2_unittest.cc +++ b/media/webrtc/webrtcvideoengine2_unittest.cc @@ -34,6 +34,7 @@ #include "talk/media/webrtc/webrtcvideochannelfactory.h" #include "talk/media/webrtc/webrtcvideoengine2.h" #include "talk/media/webrtc/webrtcvideoengine2_unittest.h" +#include "talk/media/webrtc/webrtcvoiceengine.h" #include "webrtc/base/gunit.h" #include "webrtc/base/stringutils.h" #include "webrtc/video_encoder.h" @@ -79,10 +80,14 @@ FakeVideoSendStream::FakeVideoSendStream( ReconfigureVideoEncoder(encoder_config); } -webrtc::VideoSendStream::Config FakeVideoSendStream::GetConfig() { +webrtc::VideoSendStream::Config FakeVideoSendStream::GetConfig() const { return config_; } +webrtc::VideoEncoderConfig FakeVideoSendStream::GetEncoderConfig() const { + return encoder_config_; +} + std::vector<webrtc::VideoStream> FakeVideoSendStream::GetVideoStreams() { return encoder_config_.streams; } @@ -170,9 +175,6 @@ void FakeVideoReceiveStream::Stop() { receiving_ = false; } -void FakeVideoReceiveStream::GetCurrentReceiveCodec(webrtc::VideoCodec* codec) { -} - FakeCall::FakeCall(const webrtc::Call::Config& config) : config_(config), network_state_(kNetworkUp) { SetVideoCodecs(GetDefaultVideoCodecs()); @@ -327,9 +329,32 @@ class WebRtcVideoEngine2Test : public ::testing::Test { } protected: + class FakeCallFactory : public WebRtcCallFactory { + public: + FakeCallFactory() : fake_call_(NULL) {} + FakeCall* GetCall() { return fake_call_; } + + private: + virtual webrtc::Call* CreateCall( + const webrtc::Call::Config& config) OVERRIDE { + assert(fake_call_ == NULL); + fake_call_ = new FakeCall(config); + return fake_call_; + } + + FakeCall* fake_call_; + }; + VideoMediaChannel* SetUpForExternalEncoderFactory( cricket::WebRtcVideoEncoderFactory* encoder_factory, const std::vector<VideoCodec>& codecs); + + VideoMediaChannel* SetUpForExternalDecoderFactory( + cricket::WebRtcVideoDecoderFactory* decoder_factory, + const std::vector<VideoCodec>& codecs); + + void TestStartBitrate(bool override_start_bitrate, int start_bitrate_bps); + WebRtcVideoEngine2 engine_; VideoCodec default_codec_; VideoCodec default_red_codec_; @@ -337,9 +362,42 @@ class WebRtcVideoEngine2Test : public ::testing::Test { VideoCodec default_rtx_codec_; }; -// TODO(pbos): Add test that verifies that sync is configured properly. -TEST_F(WebRtcVideoEngine2Test, DISABLED_CreateChannelWithVoiceEngine) { - FAIL() << "Not implemented."; // TODO(pbos): Implement. +TEST_F(WebRtcVideoEngine2Test, ConfiguresAvSyncForFirstReceiveChannel) { + FakeCallFactory call_factory; + engine_.SetCallFactory(&call_factory); + + WebRtcVoiceEngine voice_engine; + engine_.SetVoiceEngine(&voice_engine); + voice_engine.Init(rtc::Thread::Current()); + engine_.Init(rtc::Thread::Current()); + + rtc::scoped_ptr<VoiceMediaChannel> voice_channel( + voice_engine.CreateChannel()); + ASSERT_TRUE(voice_channel.get() != NULL); + WebRtcVoiceMediaChannel* webrtc_voice_channel = + static_cast<WebRtcVoiceMediaChannel*>(voice_channel.get()); + ASSERT_NE(webrtc_voice_channel->voe_channel(), -1); + rtc::scoped_ptr<VideoMediaChannel> channel( + engine_.CreateChannel(cricket::VideoOptions(), voice_channel.get())); + + FakeCall* fake_call = call_factory.GetCall(); + ASSERT_TRUE(fake_call != NULL); + + webrtc::Call::Config call_config = fake_call->GetConfig(); + + ASSERT_TRUE(voice_engine.voe()->engine() != NULL); + ASSERT_EQ(voice_engine.voe()->engine(), call_config.voice_engine); + + EXPECT_TRUE(channel->AddRecvStream(StreamParams::CreateLegacy(kSsrc))); + EXPECT_TRUE(channel->AddRecvStream(StreamParams::CreateLegacy(kSsrc + 1))); + std::vector<FakeVideoReceiveStream*> receive_streams = + fake_call->GetVideoReceiveStreams(); + + ASSERT_EQ(2u, receive_streams.size()); + EXPECT_EQ(webrtc_voice_channel->voe_channel(), + receive_streams[0]->GetConfig().audio_channel_id); + EXPECT_EQ(-1, receive_streams[1]->GetConfig().audio_channel_id) + << "AV sync should only be set up for the first receive channel."; } TEST_F(WebRtcVideoEngine2Test, FindCodec) { @@ -425,6 +483,39 @@ TEST_F(WebRtcVideoEngine2Test, SupportsAbsoluteSenderTimeHeaderExtension) { FAIL() << "Absolute Sender Time extension not in header-extension list."; } +void WebRtcVideoEngine2Test::TestStartBitrate(bool override_start_bitrate, + int start_bitrate_bps) { + FakeCallFactory call_factory; + engine_.SetCallFactory(&call_factory); + + engine_.Init(rtc::Thread::Current()); + + cricket::VideoOptions options; + if (override_start_bitrate) { + options.video_start_bitrate.Set(start_bitrate_bps / 1000); + } + + rtc::scoped_ptr<VideoMediaChannel> channel( + engine_.CreateChannel(options, NULL)); + + EXPECT_EQ(override_start_bitrate + ? start_bitrate_bps + : webrtc::Call::Config::kDefaultStartBitrateBps, + call_factory.GetCall()->GetConfig().stream_start_bitrate_bps); +} + +TEST_F(WebRtcVideoEngine2Test, UsesCorrectDefaultStartBitrate) { + TestStartBitrate(false, -1); +} + +TEST_F(WebRtcVideoEngine2Test, CreateChannelCanUseIncreasedStartBitrate) { + TestStartBitrate(true, 2 * webrtc::Call::Config::kDefaultStartBitrateBps); +} + +TEST_F(WebRtcVideoEngine2Test, CreateChannelCanUseDecreasedStartBitrate) { + TestStartBitrate(true, webrtc::Call::Config::kDefaultStartBitrateBps / 2); +} + TEST_F(WebRtcVideoEngine2Test, SetSendFailsBeforeSettingCodecs) { engine_.Init(rtc::Thread::Current()); rtc::scoped_ptr<VideoMediaChannel> channel( @@ -492,6 +583,19 @@ VideoMediaChannel* WebRtcVideoEngine2Test::SetUpForExternalEncoderFactory( return channel; } +VideoMediaChannel* WebRtcVideoEngine2Test::SetUpForExternalDecoderFactory( + cricket::WebRtcVideoDecoderFactory* decoder_factory, + const std::vector<VideoCodec>& codecs) { + engine_.SetExternalDecoderFactory(decoder_factory); + engine_.Init(rtc::Thread::Current()); + + VideoMediaChannel* channel = + engine_.CreateChannel(cricket::VideoOptions(), NULL); + EXPECT_TRUE(channel->SetRecvCodecs(codecs)); + + return channel; +} + TEST_F(WebRtcVideoEngine2Test, ChannelWithExternalH264CanChangeToInternalVp8) { cricket::FakeWebRtcVideoEncoderFactory encoder_factory; encoder_factory.AddSupportedVideoCodecType(webrtc::kVideoCodecH264, "H264"); @@ -545,6 +649,50 @@ TEST_F(WebRtcVideoEngine2Test, ReportSupportedExternalCodecs) { EXPECT_EQ("H264", external_codec.name); } +TEST_F(WebRtcVideoEngine2Test, RegisterExternalDecodersIfSupported) { + cricket::FakeWebRtcVideoDecoderFactory decoder_factory; + decoder_factory.AddSupportedVideoCodecType(webrtc::kVideoCodecVP8); + std::vector<cricket::VideoCodec> codecs; + codecs.push_back(kVp8Codec); + + rtc::scoped_ptr<VideoMediaChannel> channel( + SetUpForExternalDecoderFactory(&decoder_factory, codecs)); + + EXPECT_TRUE( + channel->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrc))); + ASSERT_EQ(1u, decoder_factory.decoders().size()); + + // Setting codecs of the same type should not reallocate the decoder. + EXPECT_TRUE(channel->SetRecvCodecs(codecs)); + EXPECT_EQ(1, decoder_factory.GetNumCreatedDecoders()); + + // Remove stream previously added to free the external decoder instance. + EXPECT_TRUE(channel->RemoveRecvStream(kSsrc)); + EXPECT_EQ(0u, decoder_factory.decoders().size()); +} + +// Verifies that we can set up decoders that are not internally supported. +TEST_F(WebRtcVideoEngine2Test, RegisterExternalH264DecoderIfSupported) { + // TODO(pbos): Do not assume that encoder/decoder support is symmetric. We + // can't even query the WebRtcVideoDecoderFactory for supported codecs. + // For now we add a FakeWebRtcVideoEncoderFactory to add H264 to supported + // codecs. + cricket::FakeWebRtcVideoEncoderFactory encoder_factory; + encoder_factory.AddSupportedVideoCodecType(webrtc::kVideoCodecH264, "H264"); + engine_.SetExternalEncoderFactory(&encoder_factory); + cricket::FakeWebRtcVideoDecoderFactory decoder_factory; + decoder_factory.AddSupportedVideoCodecType(webrtc::kVideoCodecH264); + std::vector<cricket::VideoCodec> codecs; + codecs.push_back(kH264Codec); + + rtc::scoped_ptr<VideoMediaChannel> channel( + SetUpForExternalDecoderFactory(&decoder_factory, codecs)); + + EXPECT_TRUE( + channel->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrc))); + ASSERT_EQ(1u, decoder_factory.decoders().size()); +} + class WebRtcVideoEngine2BaseTest : public VideoEngineTest<cricket::WebRtcVideoEngine2> { protected: @@ -670,7 +818,7 @@ TEST_F(WebRtcVideoChannel2BaseTest, DISABLED_SendVp8HdAndReceiveAdaptedVp8Vga) { EXPECT_TRUE(channel_->SetCapturer(kSsrc, video_capturer_.get())); // Capture format HD -> adapt (OnOutputFormatRequest VGA) -> VGA. - cricket::VideoCodec codec(100, "VP8", 1280, 720, 30, 0); + cricket::VideoCodec codec = kVp8Codec720p; EXPECT_TRUE(SetOneCodec(codec)); codec.width /= 2; codec.height /= 2; @@ -841,60 +989,6 @@ class WebRtcVideoChannel2Test : public WebRtcVideoEngine2Test, uint32 last_ssrc_; }; -TEST_F(WebRtcVideoChannel2Test, DISABLED_MaxBitrateResetsWithConferenceMode) { - FAIL() << "Not implemented."; // TODO(pbos): Implement. -} - -TEST_F(WebRtcVideoChannel2Test, DISABLED_StartSendBitrate) { - // TODO(pbos): Is this test testing vie_ ? this is confusing. No API to set - // start send bitrate from outside? Add defaults here that should be kept? - std::vector<cricket::VideoCodec> codec_list; - codec_list.push_back(kVp8Codec); - EXPECT_TRUE(channel_->SetSendCodecs(codec_list)); - const unsigned int kVideoMinSendBitrateKbps = 50; - const unsigned int kVideoTargetSendBitrateKbps = 300; - const unsigned int kVideoMaxSendBitrateKbps = 2000; - FakeVideoSendStream* stream = AddSendStream(); - std::vector<webrtc::VideoStream> video_streams = stream->GetVideoStreams(); - ASSERT_EQ(1u, video_streams.size()); - EXPECT_EQ(kVideoMinSendBitrateKbps, - video_streams.back().min_bitrate_bps / 1000); - EXPECT_EQ(kVideoTargetSendBitrateKbps, - video_streams.back().target_bitrate_bps / 1000); - EXPECT_EQ(kVideoMaxSendBitrateKbps, - video_streams.back().max_bitrate_bps / 1000); -#if 0 - // TODO(pbos): un-#if - VerifyVP8SendCodec(send_channel, kVP8Codec.width, kVP8Codec.height, 0, - kVideoMaxSendBitrateKbps, kVideoMinSendBitrateKbps, - kVideoDefaultStartSendBitrateKbps); - EXPECT_EQ(0, vie_.StartSend(send_channel)); - - // Increase the send bitrate and verify it is used as start bitrate. - const unsigned int kVideoSendBitrateBps = 768000; - vie_.SetSendBitrates(send_channel, kVideoSendBitrateBps, 0, 0); - EXPECT_TRUE(channel_->SetSendCodecs(codec_list)); - VerifyVP8SendCodec(send_channel, kVP8Codec.width, kVP8Codec.height, 0, - kVideoMaxSendBitrateKbps, kVideoMinSendBitrateKbps, - kVideoSendBitrateBps / 1000); - - // Never set a start bitrate higher than the max bitrate. - vie_.SetSendBitrates(send_channel, kVideoMaxSendBitrateKbps + 500, 0, 0); - EXPECT_TRUE(channel_->SetSendCodecs(codec_list)); - VerifyVP8SendCodec(send_channel, kVP8Codec.width, kVP8Codec.height, 0, - kVideoMaxSendBitrateKbps, kVideoMinSendBitrateKbps, - kVideoDefaultStartSendBitrateKbps); - - // Use the default start bitrate if the send bitrate is lower. - vie_.SetSendBitrates(send_channel, kVideoDefaultStartSendBitrateKbps - 50, 0, - 0); - EXPECT_TRUE(channel_->SetSendCodecs(codec_list)); - VerifyVP8SendCodec(send_channel, kVP8Codec.width, kVP8Codec.height, 0, - kVideoMaxSendBitrateKbps, kVideoMinSendBitrateKbps, - kVideoDefaultStartSendBitrateKbps); -#endif -} - TEST_F(WebRtcVideoChannel2Test, RecvStreamWithSimAndRtx) { EXPECT_TRUE(channel_->SetSendCodecs(engine_.codecs())); EXPECT_TRUE(channel_->SetSend(true)); @@ -1188,8 +1282,97 @@ TEST_F(WebRtcVideoChannel2Test, DISABLED_SetBandwidthInConference) { FAIL() << "Not implemented."; // TODO(pbos): Implement. } -TEST_F(WebRtcVideoChannel2Test, DISABLED_SetBandwidthScreencast) { - FAIL() << "Not implemented."; // TODO(pbos): Implement. +TEST_F(WebRtcVideoChannel2Test, UsesCorrectSettingsForScreencast) { + static const int kScreenshareMinBitrateKbps = 800; + cricket::VideoCodec codec = kVp8Codec360p; + std::vector<cricket::VideoCodec> codecs; + codecs.push_back(codec); + EXPECT_TRUE(channel_->SetSendCodecs(codecs)); + VideoOptions options; + options.screencast_min_bitrate.Set(kScreenshareMinBitrateKbps); + channel_->SetOptions(options); + + AddSendStream(); + + cricket::FakeVideoCapturer capturer; + capturer.SetScreencast(false); + EXPECT_TRUE(channel_->SetCapturer(last_ssrc_, &capturer)); + cricket::VideoFormat capture_format_hd = + capturer.GetSupportedFormats()->front(); + EXPECT_EQ(1280, capture_format_hd.width); + EXPECT_EQ(720, capture_format_hd.height); + EXPECT_EQ(cricket::CS_RUNNING, capturer.Start(capture_format_hd)); + + EXPECT_TRUE(channel_->SetSend(true)); + + EXPECT_TRUE(capturer.CaptureFrame()); + ASSERT_EQ(1u, fake_call_->GetVideoSendStreams().size()); + FakeVideoSendStream* send_stream = fake_call_->GetVideoSendStreams().front(); + + EXPECT_EQ(1, send_stream->GetNumberOfSwappedFrames()); + + // Verify non-screencast settings. + webrtc::VideoEncoderConfig encoder_config = send_stream->GetEncoderConfig(); + EXPECT_EQ(webrtc::VideoEncoderConfig::kRealtimeVideo, + encoder_config.content_type); + EXPECT_EQ(codec.width, encoder_config.streams.front().width); + EXPECT_EQ(codec.height, encoder_config.streams.front().height); + EXPECT_EQ(0, encoder_config.min_transmit_bitrate_bps) + << "Non-screenshare shouldn't use min-transmit bitrate."; + + capturer.SetScreencast(true); + EXPECT_TRUE(capturer.CaptureFrame()); + + EXPECT_EQ(2, send_stream->GetNumberOfSwappedFrames()); + + // Verify screencast settings. + encoder_config = send_stream->GetEncoderConfig(); + EXPECT_EQ(webrtc::VideoEncoderConfig::kScreenshare, + encoder_config.content_type); + EXPECT_EQ(kScreenshareMinBitrateKbps * 1000, + encoder_config.min_transmit_bitrate_bps); + + EXPECT_EQ(capture_format_hd.width, encoder_config.streams.front().width); + EXPECT_EQ(capture_format_hd.height, encoder_config.streams.front().height); + EXPECT_TRUE(encoder_config.streams[0].temporal_layer_thresholds_bps.empty()); + + EXPECT_TRUE(channel_->SetCapturer(last_ssrc_, NULL)); +} + +TEST_F(WebRtcVideoChannel2Test, + ConferenceModeScreencastConfiguresTemporalLayer) { + static const int kConferenceScreencastTemporalBitrateBps = 100000; + VideoOptions options; + options.conference_mode.Set(true); + channel_->SetOptions(options); + + AddSendStream(); + + cricket::FakeVideoCapturer capturer; + capturer.SetScreencast(true); + EXPECT_TRUE(channel_->SetCapturer(last_ssrc_, &capturer)); + cricket::VideoFormat capture_format_hd = + capturer.GetSupportedFormats()->front(); + EXPECT_EQ(cricket::CS_RUNNING, capturer.Start(capture_format_hd)); + + EXPECT_TRUE(channel_->SetSend(true)); + + EXPECT_TRUE(capturer.CaptureFrame()); + ASSERT_EQ(1u, fake_call_->GetVideoSendStreams().size()); + FakeVideoSendStream* send_stream = fake_call_->GetVideoSendStreams().front(); + + webrtc::VideoEncoderConfig encoder_config = send_stream->GetEncoderConfig(); + + // Verify screencast settings. + encoder_config = send_stream->GetEncoderConfig(); + EXPECT_EQ(webrtc::VideoEncoderConfig::kScreenshare, + encoder_config.content_type); + ASSERT_EQ(1u, encoder_config.streams.size()); + ASSERT_EQ(1u, encoder_config.streams[0].temporal_layer_thresholds_bps.size()); + EXPECT_EQ(kConferenceScreencastTemporalBitrateBps, + encoder_config.streams[0].temporal_layer_thresholds_bps[0]); + + EXPECT_TRUE(channel_->SetCapturer(last_ssrc_, NULL)); } TEST_F(WebRtcVideoChannel2Test, DISABLED_SetSendSsrcAndCname) { @@ -1289,7 +1472,7 @@ TEST_F(WebRtcVideoChannel2Test, DoesNotAdaptOnOveruseWhenDisabled) { } void WebRtcVideoChannel2Test::TestCpuAdaptation(bool enable_overuse) { - cricket::VideoCodec codec(100, "VP8", 1280, 720, 30, 0); + cricket::VideoCodec codec = kVp8Codec720p; std::vector<cricket::VideoCodec> codecs; codecs.push_back(codec); EXPECT_TRUE(channel_->SetSendCodecs(codecs)); @@ -1451,13 +1634,13 @@ TEST_F(WebRtcVideoChannel2Test, SetSendCodecsChangesExistingStreams) { } TEST_F(WebRtcVideoChannel2Test, SetSendCodecsWithMinMaxBitrate) { - SetSendCodecsShouldWorkForBitrates("10", "20"); + SetSendCodecsShouldWorkForBitrates("100", "200"); } TEST_F(WebRtcVideoChannel2Test, SetSendCodecsRejectsMaxLessThanMinBitrate) { std::vector<VideoCodec> video_codecs = engine_.codecs(); - video_codecs[0].params[kCodecParamMinBitrate] = "30"; - video_codecs[0].params[kCodecParamMaxBitrate] = "20"; + video_codecs[0].params[kCodecParamMinBitrate] = "300"; + video_codecs[0].params[kCodecParamMaxBitrate] = "200"; EXPECT_FALSE(channel_->SetSendCodecs(video_codecs)); } @@ -1565,8 +1748,8 @@ TEST_F(WebRtcVideoChannel2Test, SetRecvCodecsAcceptDefaultCodecs) { FakeVideoReceiveStream* stream = AddRecvStream(); webrtc::VideoReceiveStream::Config config = stream->GetConfig(); - EXPECT_STREQ(engine_.codecs()[0].name.c_str(), config.codecs[0].plName); - EXPECT_EQ(engine_.codecs()[0].id, config.codecs[0].plType); + EXPECT_EQ(engine_.codecs()[0].name, config.decoders[0].payload_name); + EXPECT_EQ(engine_.codecs()[0].id, config.decoders[0].payload_type); } TEST_F(WebRtcVideoChannel2Test, SetRecvCodecsRejectUnsupportedCodec) { @@ -1678,55 +1861,6 @@ TEST_F(WebRtcVideoChannel2Test, DISABLED_SetOptionsWithMaxBitrate) { FAIL() << "Not implemented."; // TODO(pbos): Implement. } -TEST_F(WebRtcVideoChannel2Test, DISABLED_SetOptionsWithLoweredBitrate) { - FAIL() << "Not implemented."; // TODO(pbos): Implement. -} - -TEST_F(WebRtcVideoChannel2Test, DISABLED_SetOptionsSucceedsWhenSending) { - FAIL() << "Not implemented."; // TODO(pbos): Implement. -} - -TEST_F(WebRtcVideoChannel2Test, DISABLED_ResetCodecOnScreencast) { - FAIL() << "Not implemented."; // TODO(pbos): Implement. -} - -TEST_F(WebRtcVideoChannel2Test, DISABLED_DontResetCodecOnSendFrame) { - FAIL() << "Not implemented."; // TODO(pbos): Implement. -} - -TEST_F(WebRtcVideoChannel2Test, - DISABLED_DontRegisterDecoderIfFactoryIsNotGiven) { - FAIL() << "Not implemented."; // TODO(pbos): Implement. -} - -TEST_F(WebRtcVideoChannel2Test, DISABLED_RegisterDecoderIfFactoryIsGiven) { - FAIL() << "Not implemented."; // TODO(pbos): Implement. -} - -TEST_F(WebRtcVideoChannel2Test, DISABLED_DontRegisterDecoderMultipleTimes) { - FAIL() << "Not implemented."; // TODO(pbos): Implement. -} - -TEST_F(WebRtcVideoChannel2Test, DISABLED_DontRegisterDecoderForNonVP8) { - FAIL() << "Not implemented."; // TODO(pbos): Implement. -} - -TEST_F(WebRtcVideoChannel2Test, DISABLED_FeedbackParamsForNonVP8) { - FAIL() << "Not implemented."; // TODO(pbos): Implement. -} - -TEST_F(WebRtcVideoChannel2Test, DISABLED_ExternalCodecAddedToTheEnd) { - FAIL() << "Not implemented."; // TODO(pbos): Implement. -} - -TEST_F(WebRtcVideoChannel2Test, DISABLED_ExternalCodecIgnored) { - FAIL() << "Not implemented."; // TODO(pbos): Implement. -} - -TEST_F(WebRtcVideoChannel2Test, DISABLED_UpdateEncoderCodecsAfterSetFactory) { - FAIL() << "Not implemented."; // TODO(pbos): Implement. -} - TEST_F(WebRtcVideoChannel2Test, OnReadyToSendSignalsNetworkState) { EXPECT_EQ(webrtc::Call::kNetworkUp, fake_call_->GetNetworkState()); @@ -1737,7 +1871,4 @@ TEST_F(WebRtcVideoChannel2Test, OnReadyToSendSignalsNetworkState) { EXPECT_EQ(webrtc::Call::kNetworkUp, fake_call_->GetNetworkState()); } -TEST_F(WebRtcVideoChannel2Test, DISABLED_CaptureFrameTimestampToNtpTimestamp) { - FAIL() << "Not implemented."; // TODO(pbos): Implement. -} } // namespace cricket diff --git a/media/webrtc/webrtcvideoengine2_unittest.h b/media/webrtc/webrtcvideoengine2_unittest.h index b3014ad..3b62289 100644 --- a/media/webrtc/webrtcvideoengine2_unittest.h +++ b/media/webrtc/webrtcvideoengine2_unittest.h @@ -41,7 +41,8 @@ class FakeVideoSendStream : public webrtc::VideoSendStream, public: FakeVideoSendStream(const webrtc::VideoSendStream::Config& config, const webrtc::VideoEncoderConfig& encoder_config); - webrtc::VideoSendStream::Config GetConfig(); + webrtc::VideoSendStream::Config GetConfig() const; + webrtc::VideoEncoderConfig GetEncoderConfig() const; std::vector<webrtc::VideoStream> GetVideoStreams(); bool IsSending() const; @@ -86,7 +87,6 @@ class FakeVideoReceiveStream : public webrtc::VideoReceiveStream { virtual void Start() OVERRIDE; virtual void Stop() OVERRIDE; - virtual void GetCurrentReceiveCodec(webrtc::VideoCodec* codec); webrtc::VideoReceiveStream::Config config_; bool receiving_; diff --git a/media/webrtc/webrtcvideoengine_unittest.cc b/media/webrtc/webrtcvideoengine_unittest.cc index 4060493..00d6e4b 100644 --- a/media/webrtc/webrtcvideoengine_unittest.cc +++ b/media/webrtc/webrtcvideoengine_unittest.cc @@ -1475,7 +1475,7 @@ TEST_F(WebRtcVideoEngineTestFake, SetStartBandwidthOption) { kMaxBandwidthKbps, kMinBandwidthKbps, kStartBandwidthKbps); // Set the start bitrate option. - int kBoostedStartBandwidthKbps = 1000; + unsigned int kBoostedStartBandwidthKbps = 1000; ASSERT_NE(kStartBandwidthKbps, kBoostedStartBandwidthKbps); cricket::VideoOptions options; options.video_start_bitrate.Set(kBoostedStartBandwidthKbps); diff --git a/media/webrtc/webrtcvoiceengine.cc b/media/webrtc/webrtcvoiceengine.cc index f9e7878..95e16e4 100644 --- a/media/webrtc/webrtcvoiceengine.cc +++ b/media/webrtc/webrtcvoiceengine.cc @@ -110,13 +110,26 @@ static const int kDefaultAudioDeviceId = 0; static const char kIsacCodecName[] = "ISAC"; static const char kL16CodecName[] = "L16"; -// Codec parameters for Opus. -static const int kOpusMonoBitrate = 32000; + // Parameter used for NACK. // This value is equivalent to 5 seconds of audio data at 20 ms per packet. static const int kNackMaxPackets = 250; -static const int kOpusStereoBitrate = 64000; + +// Codec parameters for Opus. // draft-spittka-payload-rtp-opus-03 + +// Recommended bitrates: +// 8-12 kb/s for NB speech, +// 16-20 kb/s for WB speech, +// 28-40 kb/s for FB speech, +// 48-64 kb/s for FB mono music, and +// 64-128 kb/s for FB stereo music. +// The current implementation applies the following values to mono signals, +// and multiplies them by 2 for stereo. +static const int kOpusBitrateNb = 12000; +static const int kOpusBitrateWb = 20000; +static const int kOpusBitrateFb = 32000; + // Opus bitrate should be in the range between 6000 and 510000. static const int kOpusMinBitrate = 6000; static const int kOpusMaxBitrate = 510000; @@ -405,22 +418,37 @@ static bool IsOpusStereoEnabled(const AudioCodec& codec) { return codec.GetParam(kCodecParamStereo, &value) && value == 1; } -// TODO(minyue): Clamp bitrate when invalid. -static bool IsValidOpusBitrate(int bitrate) { - return (bitrate >= kOpusMinBitrate && bitrate <= kOpusMaxBitrate); -} - -// Returns 0 if params[kCodecParamMaxAverageBitrate] is not defined or invalid. -// Returns the value of params[kCodecParamMaxAverageBitrate] otherwise. -static int GetOpusBitrateFromParams(const AudioCodec& codec) { +// Use params[kCodecParamMaxAverageBitrate] if it is defined, use codec.bitrate +// otherwise. If the value (either from params or codec.bitrate) <=0, use the +// default configuration. If the value is beyond feasible bit rate of Opus, +// clamp it. Returns the Opus bit rate for operation. +static int GetOpusBitrate(const AudioCodec& codec, int max_playback_rate) { int bitrate = 0; + bool use_param = true; if (!codec.GetParam(kCodecParamMaxAverageBitrate, &bitrate)) { - return 0; - } - if (!IsValidOpusBitrate(bitrate)) { - LOG(LS_WARNING) << "Codec parameter \"maxaveragebitrate\" has an " - << "invalid value: " << bitrate; - return 0; + bitrate = codec.bitrate; + use_param = false; + } + if (bitrate <= 0) { + if (max_playback_rate <= 8000) { + bitrate = kOpusBitrateNb; + } else if (max_playback_rate <= 16000) { + bitrate = kOpusBitrateWb; + } else { + bitrate = kOpusBitrateFb; + } + + if (IsOpusStereoEnabled(codec)) { + bitrate *= 2; + } + } else if (bitrate < kOpusMinBitrate || bitrate > kOpusMaxBitrate) { + bitrate = (bitrate < kOpusMinBitrate) ? kOpusMinBitrate : kOpusMaxBitrate; + std::string rate_source = + use_param ? "Codec parameter \"maxaveragebitrate\"" : + "Supplied Opus bitrate"; + LOG(LS_WARNING) << rate_source + << " is invalid and is replaced by: " + << bitrate; } return bitrate; } @@ -450,39 +478,11 @@ static void GetOpusConfig(const AudioCodec& codec, webrtc::CodecInst* voe_codec, // If OPUS, change what we send according to the "stereo" codec // parameter, and not the "channels" parameter. We set // voe_codec.channels to 2 if "stereo=1" and 1 otherwise. If - // the bitrate is not specified, i.e. is zero, we set it to the + // the bitrate is not specified, i.e. is <= zero, we set it to the // appropriate default value for mono or stereo Opus. - // TODO(minyue): The determination of bit rate might take the maximum playback - // rate into account. - - if (IsOpusStereoEnabled(codec)) { - voe_codec->channels = 2; - if (!IsValidOpusBitrate(codec.bitrate)) { - if (codec.bitrate != 0) { - LOG(LS_WARNING) << "Overrides the invalid supplied bitrate(" - << codec.bitrate - << ") with default opus stereo bitrate: " - << kOpusStereoBitrate; - } - voe_codec->rate = kOpusStereoBitrate; - } - } else { - voe_codec->channels = 1; - if (!IsValidOpusBitrate(codec.bitrate)) { - if (codec.bitrate != 0) { - LOG(LS_WARNING) << "Overrides the invalid supplied bitrate(" - << codec.bitrate - << ") with default opus mono bitrate: " - << kOpusMonoBitrate; - } - voe_codec->rate = kOpusMonoBitrate; - } - } - int bitrate_from_params = GetOpusBitrateFromParams(codec); - if (bitrate_from_params != 0) { - voe_codec->rate = bitrate_from_params; - } + voe_codec->channels = IsOpusStereoEnabled(codec) ? 2 : 1; + voe_codec->rate = GetOpusBitrate(codec, *max_playback_rate); } void WebRtcVoiceEngine::ConstructCodecs() { @@ -514,7 +514,7 @@ void WebRtcVoiceEngine::ConstructCodecs() { ARRAY_SIZE(kCodecPrefs) - (pref - kCodecPrefs)); LOG(LS_INFO) << ToString(codec); if (IsIsac(codec)) { - // Indicate auto-bandwidth in signaling. + // Indicate auto-bitrate in signaling. codec.bitrate = 0; } if (IsOpus(codec)) { @@ -1246,7 +1246,7 @@ bool WebRtcVoiceEngine::FindWebRtcCodec(const AudioCodec& in, // Apply codec-specific settings. if (IsIsac(codec)) { // If ISAC and an explicit bitrate is not specified, - // enable auto bandwidth adjustment. + // enable auto bitrate adjustment. voe_codec.rate = (in.bitrate > 0) ? in.bitrate : -1; } *out = voe_codec; @@ -1811,8 +1811,8 @@ WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(WebRtcVoiceEngine *engine) : WebRtcMediaChannel<VoiceMediaChannel, WebRtcVoiceEngine>( engine, engine->CreateMediaVoiceChannel()), - send_bw_setting_(false), - send_bw_bps_(0), + send_bitrate_setting_(false), + send_bitrate_bps_(0), options_(), dtmf_allowed_(false), desired_playout_(false), @@ -2047,9 +2047,7 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs( bool nack_enabled = nack_enabled_; bool enable_codec_fec = false; - // max_playback_rate <= 0 will not trigger setting of maximum encoding - // bandwidth. - int max_playback_rate = 0; + int opus_max_playback_rate = 0; // Set send codec (the first non-telephone-event/CN codec) for (std::vector<AudioCodec>::const_iterator it = codecs.begin(); @@ -2067,7 +2065,6 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs( continue; } - // We'll use the first codec in the list to actually send audio data. // Be sure to use the payload type requested by the remote side. // "red", for RED audio, is a special case where the actual codec to be @@ -2099,7 +2096,8 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs( // For Opus as the send codec, we are to enable inband FEC if requested // and set maximum playback rate. if (IsOpus(*it)) { - GetOpusConfig(*it, &send_codec, &enable_codec_fec, &max_playback_rate); + GetOpusConfig(*it, &send_codec, &enable_codec_fec, + &opus_max_playback_rate); } } found_send_codec = true; @@ -2135,15 +2133,16 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs( } // maxplaybackrate should be set after SetSendCodec. - if (max_playback_rate > 0) { + // If opus_max_playback_rate <= 0, the default maximum playback rate of 48 kHz + // will be used. + if (opus_max_playback_rate > 0) { LOG(LS_INFO) << "Attempt to set maximum playback rate to " - << max_playback_rate + << opus_max_playback_rate << " Hz on channel " << channel; #ifdef USE_WEBRTC_DEV_BRANCH - // (max_playback_rate + 1) >> 1 is to obtain ceil(max_playback_rate / 2.0). if (engine()->voe()->codec()->SetOpusMaxPlaybackRate( - channel, max_playback_rate) == -1) { + channel, opus_max_playback_rate) == -1) { LOG(LS_WARNING) << "Could not set maximum playback rate."; } #endif @@ -2152,8 +2151,8 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs( // Always update the |send_codec_| to the currently set send codec. send_codec_.reset(new webrtc::CodecInst(send_codec)); - if (send_bw_setting_) { - SetSendBandwidthInternal(send_bw_bps_); + if (send_bitrate_setting_) { + SetSendBitrateInternal(send_bitrate_bps_); } // Loop through the codecs list again to config the telephone-event/CN codec. @@ -3206,25 +3205,27 @@ bool WebRtcVoiceMediaChannel::MuteStream(uint32 ssrc, bool muted) { return true; } +// TODO(minyue): SetMaxSendBandwidth() is subject to be renamed to +// SetMaxSendBitrate() in future. bool WebRtcVoiceMediaChannel::SetMaxSendBandwidth(int bps) { - LOG(LS_INFO) << "WebRtcVoiceMediaChanne::SetSendBandwidth."; + LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetMaxSendBandwidth."; - return SetSendBandwidthInternal(bps); + return SetSendBitrateInternal(bps); } -bool WebRtcVoiceMediaChannel::SetSendBandwidthInternal(int bps) { - LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetSendBandwidthInternal."; +bool WebRtcVoiceMediaChannel::SetSendBitrateInternal(int bps) { + LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetSendBitrateInternal."; - send_bw_setting_ = true; - send_bw_bps_ = bps; + send_bitrate_setting_ = true; + send_bitrate_bps_ = bps; if (!send_codec_) { LOG(LS_INFO) << "The send codec has not been set up yet. " - << "The send bandwidth setting will be applied later."; + << "The send bitrate setting will be applied later."; return true; } - // Bandwidth is auto by default. + // Bitrate is auto by default. // TODO(bemasc): Fix this so that if SetMaxSendBandwidth(50) is followed by // SetMaxSendBandwith(0), the second call removes the previous limit. if (bps <= 0) diff --git a/media/webrtc/webrtcvoiceengine.h b/media/webrtc/webrtcvoiceengine.h index 8d762d4..f19059b 100644 --- a/media/webrtc/webrtcvoiceengine.h +++ b/media/webrtc/webrtcvoiceengine.h @@ -435,7 +435,7 @@ class WebRtcVoiceMediaChannel return channel_id == voe_channel(); } bool SetSendCodecs(int channel, const std::vector<AudioCodec>& codecs); - bool SetSendBandwidthInternal(int bps); + bool SetSendBitrateInternal(int bps); bool SetHeaderExtension(ExtensionSetterFunction setter, int channel_id, const RtpHeaderExtension* extension); @@ -453,8 +453,8 @@ class WebRtcVoiceMediaChannel std::vector<AudioCodec> recv_codecs_; std::vector<AudioCodec> send_codecs_; rtc::scoped_ptr<webrtc::CodecInst> send_codec_; - bool send_bw_setting_; - int send_bw_bps_; + bool send_bitrate_setting_; + int send_bitrate_bps_; AudioOptions options_; bool dtmf_allowed_; bool desired_playout_; diff --git a/media/webrtc/webrtcvoiceengine_unittest.cc b/media/webrtc/webrtcvoiceengine_unittest.cc index b044e92..5deabd2 100644 --- a/media/webrtc/webrtcvoiceengine_unittest.cc +++ b/media/webrtc/webrtcvoiceengine_unittest.cc @@ -40,7 +40,7 @@ #include "talk/media/webrtc/fakewebrtcvoiceengine.h" #include "talk/media/webrtc/webrtcvie.h" #include "talk/media/webrtc/webrtcvoiceengine.h" -#include "talk/p2p/base/fakesession.h" +#include "webrtc/p2p/base/fakesession.h" #include "talk/session/media/channel.h" // Tests for the WebRtcVoiceEngine/VoiceChannel code. @@ -876,21 +876,20 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodXBitrate0Stereo) { codecs[0].params["stereo"] = "0"; webrtc::CodecInst gcodec; - // bitrate that's out of the range between 6000 and 510000 will be considered - // as invalid and ignored. + // bitrate that's out of the range between 6000 and 510000 will be clamped. codecs[0].bitrate = 5999; EXPECT_TRUE(channel_->SetSendCodecs(codecs)); EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); EXPECT_STREQ("opus", gcodec.plname); EXPECT_EQ(1, gcodec.channels); - EXPECT_EQ(32000, gcodec.rate); + EXPECT_EQ(6000, gcodec.rate); codecs[0].bitrate = 510001; EXPECT_TRUE(channel_->SetSendCodecs(codecs)); EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); EXPECT_STREQ("opus", gcodec.plname); EXPECT_EQ(1, gcodec.channels); - EXPECT_EQ(32000, gcodec.rate); + EXPECT_EQ(510000, gcodec.rate); } // Test that with bitrate=0 and stereo=1, @@ -920,21 +919,20 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodXBitrate1Stereo) { codecs[0].params["stereo"] = "1"; webrtc::CodecInst gcodec; - // bitrate that's out of the range between 6000 and 510000 will be considered - // as invalid and ignored. + // bitrate that's out of the range between 6000 and 510000 will be clamped. codecs[0].bitrate = 5999; EXPECT_TRUE(channel_->SetSendCodecs(codecs)); EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); EXPECT_STREQ("opus", gcodec.plname); EXPECT_EQ(2, gcodec.channels); - EXPECT_EQ(64000, gcodec.rate); + EXPECT_EQ(6000, gcodec.rate); codecs[0].bitrate = 510001; EXPECT_TRUE(channel_->SetSendCodecs(codecs)); EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); EXPECT_STREQ("opus", gcodec.plname); EXPECT_EQ(2, gcodec.channels); - EXPECT_EQ(64000, gcodec.rate); + EXPECT_EQ(510000, gcodec.rate); } // Test that with bitrate=N and stereo unset, @@ -1020,13 +1018,13 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusMaxAverageBitrate) { codecs[0].params["maxaveragebitrate"] = "5999"; EXPECT_TRUE(channel_->SetSendCodecs(codecs)); EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); - EXPECT_EQ(30000, gcodec.rate); + EXPECT_EQ(6000, gcodec.rate); // Ignore if larger than 510000. codecs[0].params["maxaveragebitrate"] = "510001"; EXPECT_TRUE(channel_->SetSendCodecs(codecs)); EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); - EXPECT_EQ(30000, gcodec.rate); + EXPECT_EQ(510000, gcodec.rate); codecs[0].params["maxaveragebitrate"] = "200000"; EXPECT_TRUE(channel_->SetSendCodecs(codecs)); @@ -1256,9 +1254,12 @@ TEST_F(WebRtcVoiceEngineTestFake, SetOpusMaxPlaybackRateNb) { webrtc::CodecInst gcodec; EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); EXPECT_STREQ("opus", gcodec.plname); - // TODO(minyue): Default bit rate is not but can in future be affected by - // kCodecParamMaxPlaybackRate. - EXPECT_EQ(32000, gcodec.rate); + + EXPECT_EQ(12000, gcodec.rate); + codecs[0].SetParam(cricket::kCodecParamStereo, "1"); + EXPECT_TRUE(channel_->SetSendCodecs(codecs)); + EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); + EXPECT_EQ(24000, gcodec.rate); } // Test 8000 < maxplaybackrate <= 12000 triggers Opus medium band mode. @@ -1275,9 +1276,12 @@ TEST_F(WebRtcVoiceEngineTestFake, SetOpusMaxPlaybackRateMb) { webrtc::CodecInst gcodec; EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); EXPECT_STREQ("opus", gcodec.plname); - // TODO(minyue): Default bit rate is not but can in future be affected by - // kCodecParamMaxPlaybackRate. - EXPECT_EQ(32000, gcodec.rate); + + EXPECT_EQ(20000, gcodec.rate); + codecs[0].SetParam(cricket::kCodecParamStereo, "1"); + EXPECT_TRUE(channel_->SetSendCodecs(codecs)); + EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); + EXPECT_EQ(40000, gcodec.rate); } // Test 12000 < maxplaybackrate <= 16000 triggers Opus wide band mode. @@ -1294,9 +1298,12 @@ TEST_F(WebRtcVoiceEngineTestFake, SetOpusMaxPlaybackRateWb) { webrtc::CodecInst gcodec; EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); EXPECT_STREQ("opus", gcodec.plname); - // TODO(minyue): Default bit rate is not but can in future be affected by - // kCodecParamMaxPlaybackRate. - EXPECT_EQ(32000, gcodec.rate); + + EXPECT_EQ(20000, gcodec.rate); + codecs[0].SetParam(cricket::kCodecParamStereo, "1"); + EXPECT_TRUE(channel_->SetSendCodecs(codecs)); + EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); + EXPECT_EQ(40000, gcodec.rate); } // Test 16000 < maxplaybackrate <= 24000 triggers Opus super wide band mode. @@ -1313,9 +1320,12 @@ TEST_F(WebRtcVoiceEngineTestFake, SetOpusMaxPlaybackRateSwb) { webrtc::CodecInst gcodec; EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); EXPECT_STREQ("opus", gcodec.plname); - // TODO(minyue): Default bit rate is not but can in future be affected by - // kCodecParamMaxPlaybackRate. + EXPECT_EQ(32000, gcodec.rate); + codecs[0].SetParam(cricket::kCodecParamStereo, "1"); + EXPECT_TRUE(channel_->SetSendCodecs(codecs)); + EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); + EXPECT_EQ(64000, gcodec.rate); } // Test 24000 < maxplaybackrate triggers Opus full band mode. @@ -1332,9 +1342,12 @@ TEST_F(WebRtcVoiceEngineTestFake, SetOpusMaxPlaybackRateFb) { webrtc::CodecInst gcodec; EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); EXPECT_STREQ("opus", gcodec.plname); - // TODO(minyue): Default bit rate is not but can in future be affected by - // kCodecParamMaxPlaybackRate. + EXPECT_EQ(32000, gcodec.rate); + codecs[0].SetParam(cricket::kCodecParamStereo, "1"); + EXPECT_TRUE(channel_->SetSendCodecs(codecs)); + EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); + EXPECT_EQ(64000, gcodec.rate); } // Test Opus that without maxplaybackrate, default playback rate is used. diff --git a/p2p/base/asyncstuntcpsocket.cc b/p2p/base/asyncstuntcpsocket.cc index c1aeead..73bb152 100644 --- a/p2p/base/asyncstuntcpsocket.cc +++ b/p2p/base/asyncstuntcpsocket.cc @@ -25,11 +25,11 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/p2p/base/asyncstuntcpsocket.h" +#include "webrtc/p2p/base/asyncstuntcpsocket.h" #include <string.h> -#include "talk/p2p/base/stun.h" +#include "webrtc/p2p/base/stun.h" #include "webrtc/base/common.h" #include "webrtc/base/logging.h" diff --git a/p2p/base/asyncstuntcpsocket.h b/p2p/base/asyncstuntcpsocket.h index 136b4df..3675c9e 100644 --- a/p2p/base/asyncstuntcpsocket.h +++ b/p2p/base/asyncstuntcpsocket.h @@ -25,8 +25,8 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_BASE_ASYNCSTUNTCPSOCKET_H_ -#define TALK_P2P_BASE_ASYNCSTUNTCPSOCKET_H_ +#ifndef WEBRTC_P2P_BASE_ASYNCSTUNTCPSOCKET_H_ +#define WEBRTC_P2P_BASE_ASYNCSTUNTCPSOCKET_H_ #include "webrtc/base/asynctcpsocket.h" #include "webrtc/base/scoped_ptr.h" @@ -64,4 +64,4 @@ class AsyncStunTCPSocket : public rtc::AsyncTCPSocketBase { } // namespace cricket -#endif // TALK_P2P_BASE_ASYNCSTUNTCPSOCKET_H_ +#endif // WEBRTC_P2P_BASE_ASYNCSTUNTCPSOCKET_H_ diff --git a/p2p/base/asyncstuntcpsocket_unittest.cc b/p2p/base/asyncstuntcpsocket_unittest.cc index 46e5d63..ba407e7 100644 --- a/p2p/base/asyncstuntcpsocket_unittest.cc +++ b/p2p/base/asyncstuntcpsocket_unittest.cc @@ -25,7 +25,7 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/p2p/base/asyncstuntcpsocket.h" +#include "webrtc/p2p/base/asyncstuntcpsocket.h" #include "webrtc/base/asyncsocket.h" #include "webrtc/base/gunit.h" #include "webrtc/base/physicalsocketserver.h" diff --git a/p2p/base/basicpacketsocketfactory.cc b/p2p/base/basicpacketsocketfactory.cc index 2b3b06f..1e0ddf8 100644 --- a/p2p/base/basicpacketsocketfactory.cc +++ b/p2p/base/basicpacketsocketfactory.cc @@ -25,10 +25,10 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/p2p/base/basicpacketsocketfactory.h" +#include "webrtc/p2p/base/basicpacketsocketfactory.h" -#include "talk/p2p/base/asyncstuntcpsocket.h" -#include "talk/p2p/base/stun.h" +#include "webrtc/p2p/base/asyncstuntcpsocket.h" +#include "webrtc/p2p/base/stun.h" #include "webrtc/base/asynctcpsocket.h" #include "webrtc/base/asyncudpsocket.h" #include "webrtc/base/logging.h" diff --git a/p2p/base/basicpacketsocketfactory.h b/p2p/base/basicpacketsocketfactory.h index 77b1652..f895efa 100644 --- a/p2p/base/basicpacketsocketfactory.h +++ b/p2p/base/basicpacketsocketfactory.h @@ -25,10 +25,10 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_BASE_BASICPACKETSOCKETFACTORY_H_ -#define TALK_P2P_BASE_BASICPACKETSOCKETFACTORY_H_ +#ifndef WEBRTC_P2P_BASE_BASICPACKETSOCKETFACTORY_H_ +#define WEBRTC_P2P_BASE_BASICPACKETSOCKETFACTORY_H_ -#include "talk/p2p/base/packetsocketfactory.h" +#include "webrtc/p2p/base/packetsocketfactory.h" namespace rtc { @@ -65,4 +65,4 @@ class BasicPacketSocketFactory : public PacketSocketFactory { } // namespace rtc -#endif // TALK_P2P_BASE_BASICPACKETSOCKETFACTORY_H_ +#endif // WEBRTC_P2P_BASE_BASICPACKETSOCKETFACTORY_H_ diff --git a/p2p/base/candidate.h b/p2p/base/candidate.h index cb0b1bc..64872f3 100644 --- a/p2p/base/candidate.h +++ b/p2p/base/candidate.h @@ -25,8 +25,8 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_BASE_CANDIDATE_H_ -#define TALK_P2P_BASE_CANDIDATE_H_ +#ifndef WEBRTC_P2P_BASE_CANDIDATE_H_ +#define WEBRTC_P2P_BASE_CANDIDATE_H_ #include <limits.h> #include <math.h> @@ -35,7 +35,7 @@ #include <sstream> #include <string> -#include "talk/p2p/base/constants.h" +#include "webrtc/p2p/base/constants.h" #include "webrtc/base/basictypes.h" #include "webrtc/base/socketaddress.h" @@ -226,4 +226,4 @@ class Candidate { } // namespace cricket -#endif // TALK_P2P_BASE_CANDIDATE_H_ +#endif // WEBRTC_P2P_BASE_CANDIDATE_H_ diff --git a/p2p/base/common.h b/p2p/base/common.h index a33e9e0..0aeb77e 100644 --- a/p2p/base/common.h +++ b/p2p/base/common.h @@ -25,8 +25,8 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_BASE_COMMON_H_ -#define TALK_P2P_BASE_COMMON_H_ +#ifndef WEBRTC_P2P_BASE_COMMON_H_ +#define WEBRTC_P2P_BASE_COMMON_H_ #include "webrtc/base/logging.h" @@ -34,4 +34,4 @@ #define LOG_J(sev, obj) LOG(sev) << "Jingle:" << obj->ToString() << ": " #define LOG_JV(sev, obj) LOG_V(sev) << "Jingle:" << obj->ToString() << ": " -#endif // TALK_P2P_BASE_COMMON_H_ +#endif // WEBRTC_P2P_BASE_COMMON_H_ diff --git a/p2p/base/constants.cc b/p2p/base/constants.cc index 278a615..54cfb75 100644 --- a/p2p/base/constants.cc +++ b/p2p/base/constants.cc @@ -25,7 +25,7 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/p2p/base/constants.h" +#include "webrtc/p2p/base/constants.h" #include <string> diff --git a/p2p/base/constants.h b/p2p/base/constants.h index 4cd1166..9c53b7d 100644 --- a/p2p/base/constants.h +++ b/p2p/base/constants.h @@ -25,8 +25,8 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_BASE_CONSTANTS_H_ -#define TALK_P2P_BASE_CONSTANTS_H_ +#ifndef WEBRTC_P2P_BASE_CONSTANTS_H_ +#define WEBRTC_P2P_BASE_CONSTANTS_H_ #include <string> #include "webrtc/libjingle/xmllite/qname.h" @@ -273,4 +273,4 @@ extern const char CONNECTIONROLE_HOLDCONN_STR[]; } // namespace cricket -#endif // TALK_P2P_BASE_CONSTANTS_H_ +#endif // WEBRTC_P2P_BASE_CONSTANTS_H_ diff --git a/p2p/base/dtlstransport.h b/p2p/base/dtlstransport.h index 318c14a..66b08b4 100644 --- a/p2p/base/dtlstransport.h +++ b/p2p/base/dtlstransport.h @@ -25,11 +25,11 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_BASE_DTLSTRANSPORT_H_ -#define TALK_P2P_BASE_DTLSTRANSPORT_H_ +#ifndef WEBRTC_P2P_BASE_DTLSTRANSPORT_H_ +#define WEBRTC_P2P_BASE_DTLSTRANSPORT_H_ -#include "talk/p2p/base/dtlstransportchannel.h" -#include "talk/p2p/base/transport.h" +#include "webrtc/p2p/base/dtlstransportchannel.h" +#include "webrtc/p2p/base/transport.h" namespace rtc { class SSLIdentity; @@ -254,4 +254,4 @@ class DtlsTransport : public Base { } // namespace cricket -#endif // TALK_P2P_BASE_DTLSTRANSPORT_H_ +#endif // WEBRTC_P2P_BASE_DTLSTRANSPORT_H_ diff --git a/p2p/base/dtlstransportchannel.cc b/p2p/base/dtlstransportchannel.cc index e0f4141..78d4622 100644 --- a/p2p/base/dtlstransportchannel.cc +++ b/p2p/base/dtlstransportchannel.cc @@ -26,9 +26,9 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/p2p/base/dtlstransportchannel.h" +#include "webrtc/p2p/base/dtlstransportchannel.h" -#include "talk/p2p/base/common.h" +#include "webrtc/p2p/base/common.h" #include "webrtc/base/buffer.h" #include "webrtc/base/dscp.h" #include "webrtc/base/messagequeue.h" diff --git a/p2p/base/dtlstransportchannel.h b/p2p/base/dtlstransportchannel.h index 44769c9..6706013 100644 --- a/p2p/base/dtlstransportchannel.h +++ b/p2p/base/dtlstransportchannel.h @@ -26,13 +26,13 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_BASE_DTLSTRANSPORTCHANNEL_H_ -#define TALK_P2P_BASE_DTLSTRANSPORTCHANNEL_H_ +#ifndef WEBRTC_P2P_BASE_DTLSTRANSPORTCHANNEL_H_ +#define WEBRTC_P2P_BASE_DTLSTRANSPORTCHANNEL_H_ #include <string> #include <vector> -#include "talk/p2p/base/transportchannelimpl.h" +#include "webrtc/p2p/base/transportchannelimpl.h" #include "webrtc/base/buffer.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/base/sslstreamadapter.h" @@ -261,4 +261,4 @@ class DtlsTransportChannelWrapper : public TransportChannelImpl { } // namespace cricket -#endif // TALK_P2P_BASE_DTLSTRANSPORTCHANNEL_H_ +#endif // WEBRTC_P2P_BASE_DTLSTRANSPORTCHANNEL_H_ diff --git a/p2p/base/dtlstransportchannel_unittest.cc b/p2p/base/dtlstransportchannel_unittest.cc index 64825bc..44854c7 100644 --- a/p2p/base/dtlstransportchannel_unittest.cc +++ b/p2p/base/dtlstransportchannel_unittest.cc @@ -28,8 +28,8 @@ #include <set> -#include "talk/p2p/base/dtlstransport.h" -#include "talk/p2p/base/fakesession.h" +#include "webrtc/p2p/base/dtlstransport.h" +#include "webrtc/p2p/base/fakesession.h" #include "webrtc/base/common.h" #include "webrtc/base/dscp.h" #include "webrtc/base/gunit.h" diff --git a/p2p/base/fakesession.h b/p2p/base/fakesession.h index 7c99719..cb9111f 100644 --- a/p2p/base/fakesession.h +++ b/p2p/base/fakesession.h @@ -25,17 +25,17 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_BASE_FAKESESSION_H_ -#define TALK_P2P_BASE_FAKESESSION_H_ +#ifndef WEBRTC_P2P_BASE_FAKESESSION_H_ +#define WEBRTC_P2P_BASE_FAKESESSION_H_ #include <map> #include <string> #include <vector> -#include "talk/p2p/base/session.h" -#include "talk/p2p/base/transport.h" -#include "talk/p2p/base/transportchannel.h" -#include "talk/p2p/base/transportchannelimpl.h" +#include "webrtc/p2p/base/session.h" +#include "webrtc/p2p/base/transport.h" +#include "webrtc/p2p/base/transportchannel.h" +#include "webrtc/p2p/base/transportchannelimpl.h" #include "webrtc/base/buffer.h" #include "webrtc/base/fakesslidentity.h" #include "webrtc/base/messagequeue.h" @@ -506,4 +506,4 @@ class FakeSession : public BaseSession { } // namespace cricket -#endif // TALK_P2P_BASE_FAKESESSION_H_ +#endif // WEBRTC_P2P_BASE_FAKESESSION_H_ diff --git a/p2p/base/p2ptransport.cc b/p2p/base/p2ptransport.cc index 06941ac..bc316cf 100644 --- a/p2p/base/p2ptransport.cc +++ b/p2p/base/p2ptransport.cc @@ -25,19 +25,19 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/p2p/base/p2ptransport.h" +#include "webrtc/p2p/base/p2ptransport.h" #include <string> #include <vector> -#include "talk/p2p/base/constants.h" -#include "talk/p2p/base/p2ptransportchannel.h" -#include "talk/p2p/base/parsing.h" -#include "talk/p2p/base/sessionmanager.h" -#include "talk/p2p/base/sessionmessages.h" +#include "webrtc/p2p/base/constants.h" +#include "webrtc/p2p/base/p2ptransportchannel.h" +#include "webrtc/p2p/base/parsing.h" +#include "webrtc/p2p/base/sessionmanager.h" +#include "webrtc/p2p/base/sessionmessages.h" #include "webrtc/libjingle/xmllite/qname.h" #include "webrtc/libjingle/xmllite/xmlelement.h" -#include "talk/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/constants.h" #include "webrtc/base/base64.h" #include "webrtc/base/common.h" #include "webrtc/base/stringencode.h" diff --git a/p2p/base/p2ptransport.h b/p2p/base/p2ptransport.h index 500bb9b..fc65bab 100644 --- a/p2p/base/p2ptransport.h +++ b/p2p/base/p2ptransport.h @@ -25,12 +25,12 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_BASE_P2PTRANSPORT_H_ -#define TALK_P2P_BASE_P2PTRANSPORT_H_ +#ifndef WEBRTC_P2P_BASE_P2PTRANSPORT_H_ +#define WEBRTC_P2P_BASE_P2PTRANSPORT_H_ #include <string> #include <vector> -#include "talk/p2p/base/transport.h" +#include "webrtc/p2p/base/transport.h" namespace cricket { @@ -100,4 +100,4 @@ class P2PTransportParser : public TransportParser { } // namespace cricket -#endif // TALK_P2P_BASE_P2PTRANSPORT_H_ +#endif // WEBRTC_P2P_BASE_P2PTRANSPORT_H_ diff --git a/p2p/base/p2ptransportchannel.cc b/p2p/base/p2ptransportchannel.cc index b1dc223..daedbeb 100644 --- a/p2p/base/p2ptransportchannel.cc +++ b/p2p/base/p2ptransportchannel.cc @@ -25,12 +25,12 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/p2p/base/p2ptransportchannel.h" +#include "webrtc/p2p/base/p2ptransportchannel.h" #include <set> -#include "talk/p2p/base/common.h" -#include "talk/p2p/base/relayport.h" // For RELAY_PORT_TYPE. -#include "talk/p2p/base/stunport.h" // For STUN_PORT_TYPE. +#include "webrtc/p2p/base/common.h" +#include "webrtc/p2p/base/relayport.h" // For RELAY_PORT_TYPE. +#include "webrtc/p2p/base/stunport.h" // For STUN_PORT_TYPE. #include "webrtc/base/common.h" #include "webrtc/base/crc32.h" #include "webrtc/base/logging.h" diff --git a/p2p/base/p2ptransportchannel.h b/p2p/base/p2ptransportchannel.h index 229e512..6d45999 100644 --- a/p2p/base/p2ptransportchannel.h +++ b/p2p/base/p2ptransportchannel.h @@ -34,18 +34,18 @@ // When all of the available connections become invalid (non-writable), we // kick off a process of determining more candidates and more connections. // -#ifndef TALK_P2P_BASE_P2PTRANSPORTCHANNEL_H_ -#define TALK_P2P_BASE_P2PTRANSPORTCHANNEL_H_ +#ifndef WEBRTC_P2P_BASE_P2PTRANSPORTCHANNEL_H_ +#define WEBRTC_P2P_BASE_P2PTRANSPORTCHANNEL_H_ #include <map> #include <string> #include <vector> -#include "talk/p2p/base/candidate.h" -#include "talk/p2p/base/p2ptransport.h" -#include "talk/p2p/base/portallocator.h" -#include "talk/p2p/base/portinterface.h" -#include "talk/p2p/base/transport.h" -#include "talk/p2p/base/transportchannelimpl.h" +#include "webrtc/p2p/base/candidate.h" +#include "webrtc/p2p/base/p2ptransport.h" +#include "webrtc/p2p/base/portallocator.h" +#include "webrtc/p2p/base/portinterface.h" +#include "webrtc/p2p/base/transport.h" +#include "webrtc/p2p/base/transportchannelimpl.h" #include "webrtc/base/asyncpacketsocket.h" #include "webrtc/base/sigslot.h" @@ -256,4 +256,4 @@ class P2PTransportChannel : public TransportChannelImpl, } // namespace cricket -#endif // TALK_P2P_BASE_P2PTRANSPORTCHANNEL_H_ +#endif // WEBRTC_P2P_BASE_P2PTRANSPORTCHANNEL_H_ diff --git a/p2p/base/p2ptransportchannel_unittest.cc b/p2p/base/p2ptransportchannel_unittest.cc index 44c76ee..d7c0b9f 100644 --- a/p2p/base/p2ptransportchannel_unittest.cc +++ b/p2p/base/p2ptransportchannel_unittest.cc @@ -25,11 +25,11 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/p2p/base/p2ptransportchannel.h" -#include "talk/p2p/base/testrelayserver.h" -#include "talk/p2p/base/teststunserver.h" -#include "talk/p2p/base/testturnserver.h" -#include "talk/p2p/client/basicportallocator.h" +#include "webrtc/p2p/base/p2ptransportchannel.h" +#include "webrtc/p2p/base/testrelayserver.h" +#include "webrtc/p2p/base/teststunserver.h" +#include "webrtc/p2p/base/testturnserver.h" +#include "webrtc/p2p/client/basicportallocator.h" #include "webrtc/base/dscp.h" #include "webrtc/base/fakenetwork.h" #include "webrtc/base/firewallsocketserver.h" diff --git a/p2p/base/packetsocketfactory.h b/p2p/base/packetsocketfactory.h index 46767c2..0160cac 100644 --- a/p2p/base/packetsocketfactory.h +++ b/p2p/base/packetsocketfactory.h @@ -25,8 +25,8 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_BASE_PACKETSOCKETFACTORY_H_ -#define TALK_P2P_BASE_PACKETSOCKETFACTORY_H_ +#ifndef WEBRTC_P2P_BASE_PACKETSOCKETFACTORY_H_ +#define WEBRTC_P2P_BASE_PACKETSOCKETFACTORY_H_ #include "webrtc/base/proxyinfo.h" @@ -66,4 +66,4 @@ class PacketSocketFactory { } // namespace rtc -#endif // TALK_P2P_BASE_PACKETSOCKETFACTORY_H_ +#endif // WEBRTC_P2P_BASE_PACKETSOCKETFACTORY_H_ diff --git a/p2p/base/parsing.cc b/p2p/base/parsing.cc index 1465096..eace88b 100644 --- a/p2p/base/parsing.cc +++ b/p2p/base/parsing.cc @@ -25,7 +25,7 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/p2p/base/parsing.h" +#include "webrtc/p2p/base/parsing.h" #include <stdlib.h> #include <algorithm> diff --git a/p2p/base/parsing.h b/p2p/base/parsing.h index 2aab2f8..f150747 100644 --- a/p2p/base/parsing.h +++ b/p2p/base/parsing.h @@ -25,8 +25,8 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_BASE_PARSING_H_ -#define TALK_P2P_BASE_PARSING_H_ +#ifndef WEBRTC_P2P_BASE_PARSING_H_ +#define WEBRTC_P2P_BASE_PARSING_H_ #include <string> #include <vector> @@ -154,4 +154,4 @@ std::vector<buzz::XmlElement*> CopyOfXmlChildren(const buzz::XmlElement* elem); } // namespace cricket -#endif // TALK_P2P_BASE_PARSING_H_ +#endif // WEBRTC_P2P_BASE_PARSING_H_ diff --git a/p2p/base/port.cc b/p2p/base/port.cc index 0cf46eb..aa88872 100644 --- a/p2p/base/port.cc +++ b/p2p/base/port.cc @@ -25,12 +25,13 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/p2p/base/port.h" +#include "webrtc/p2p/base/port.h" #include <algorithm> #include <vector> -#include "talk/p2p/base/common.h" +#include "webrtc/p2p/base/common.h" +#include "webrtc/p2p/base/portallocator.h" #include "webrtc/base/base64.h" #include "webrtc/base/crc32.h" #include "webrtc/base/helpers.h" @@ -187,7 +188,8 @@ Port::Port(rtc::Thread* thread, rtc::PacketSocketFactory* factory, ice_protocol_(ICEPROTO_HYBRID), ice_role_(ICEROLE_UNKNOWN), tiebreaker_(0), - shared_socket_(true) { + shared_socket_(true), + candidate_filter_(CF_ALL) { Construct(); } @@ -213,7 +215,8 @@ Port::Port(rtc::Thread* thread, const std::string& type, ice_protocol_(ICEPROTO_HYBRID), ice_role_(ICEROLE_UNKNOWN), tiebreaker_(0), - shared_socket_(false) { + shared_socket_(false), + candidate_filter_(CF_ALL) { ASSERT(factory_ != NULL); Construct(); } @@ -1334,7 +1337,7 @@ void Connection::HandleRoleConflictFromPeer() { void Connection::OnMessage(rtc::Message *pmsg) { ASSERT(pmsg->message_id == MSG_DELETE); - LOG_J(LS_INFO, this) << "Connection deleted"; + LOG_J(LS_INFO, this) << "Connection deleted due to read or write timeout"; SignalDestroyed(this); delete this; } diff --git a/p2p/base/port.h b/p2p/base/port.h index 4893586..490aee5 100644 --- a/p2p/base/port.h +++ b/p2p/base/port.h @@ -25,20 +25,20 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_BASE_PORT_H_ -#define TALK_P2P_BASE_PORT_H_ +#ifndef WEBRTC_P2P_BASE_PORT_H_ +#define WEBRTC_P2P_BASE_PORT_H_ #include <map> #include <set> #include <string> #include <vector> -#include "talk/p2p/base/candidate.h" -#include "talk/p2p/base/packetsocketfactory.h" -#include "talk/p2p/base/portinterface.h" -#include "talk/p2p/base/stun.h" -#include "talk/p2p/base/stunrequest.h" -#include "talk/p2p/base/transport.h" +#include "webrtc/p2p/base/candidate.h" +#include "webrtc/p2p/base/packetsocketfactory.h" +#include "webrtc/p2p/base/portinterface.h" +#include "webrtc/p2p/base/stun.h" +#include "webrtc/p2p/base/stunrequest.h" +#include "webrtc/p2p/base/transport.h" #include "webrtc/base/asyncpacketsocket.h" #include "webrtc/base/network.h" #include "webrtc/base/proxyinfo.h" @@ -308,6 +308,10 @@ class Port : public PortInterface, public rtc::MessageHandler, // Returns if Hybrid ICE protocol is used. bool IsHybridIce() const; + void set_candidate_filter(uint32 candidate_filter) { + candidate_filter_ = candidate_filter; + } + protected: enum { MSG_CHECKTIMEOUT = 0, @@ -351,6 +355,8 @@ class Port : public PortInterface, public rtc::MessageHandler, return rtc::DSCP_NO_CHANGE; } + uint32 candidate_filter() { return candidate_filter_; } + private: void Construct(); // Called when one of our connections deletes itself. @@ -392,6 +398,12 @@ class Port : public PortInterface, public rtc::MessageHandler, std::string user_agent_; rtc::ProxyInfo proxy_; + // Candidate filter is pushed down to Port such that each Port could + // make its own decision on how to create candidates. For example, + // when IceTransportsType is set to relay, both RelayPort and + // TurnPort will hide raddr to avoid local address leakage. + uint32 candidate_filter_; + friend class Connection; }; @@ -604,4 +616,4 @@ class ProxyConnection : public Connection { } // namespace cricket -#endif // TALK_P2P_BASE_PORT_H_ +#endif // WEBRTC_P2P_BASE_PORT_H_ diff --git a/p2p/base/port_unittest.cc b/p2p/base/port_unittest.cc index efa3f65..bd4520f 100644 --- a/p2p/base/port_unittest.cc +++ b/p2p/base/port_unittest.cc @@ -25,16 +25,16 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/p2p/base/basicpacketsocketfactory.h" -#include "talk/p2p/base/portproxy.h" -#include "talk/p2p/base/relayport.h" -#include "talk/p2p/base/stunport.h" -#include "talk/p2p/base/tcpport.h" -#include "talk/p2p/base/testrelayserver.h" -#include "talk/p2p/base/teststunserver.h" -#include "talk/p2p/base/testturnserver.h" -#include "talk/p2p/base/transport.h" -#include "talk/p2p/base/turnport.h" +#include "webrtc/p2p/base/basicpacketsocketfactory.h" +#include "webrtc/p2p/base/portproxy.h" +#include "webrtc/p2p/base/relayport.h" +#include "webrtc/p2p/base/stunport.h" +#include "webrtc/p2p/base/tcpport.h" +#include "webrtc/p2p/base/testrelayserver.h" +#include "webrtc/p2p/base/teststunserver.h" +#include "webrtc/p2p/base/testturnserver.h" +#include "webrtc/p2p/base/transport.h" +#include "webrtc/p2p/base/turnport.h" #include "webrtc/base/crc32.h" #include "webrtc/base/gunit.h" #include "webrtc/base/helpers.h" diff --git a/p2p/base/portallocator.cc b/p2p/base/portallocator.cc index 369af86..84fb344 100644 --- a/p2p/base/portallocator.cc +++ b/p2p/base/portallocator.cc @@ -25,9 +25,9 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/p2p/base/portallocator.h" +#include "webrtc/p2p/base/portallocator.h" -#include "talk/p2p/base/portallocatorsessionproxy.h" +#include "webrtc/p2p/base/portallocatorsessionproxy.h" namespace cricket { diff --git a/p2p/base/portallocator.h b/p2p/base/portallocator.h index 5bc389e..29a1b0e 100644 --- a/p2p/base/portallocator.h +++ b/p2p/base/portallocator.h @@ -25,13 +25,13 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_BASE_PORTALLOCATOR_H_ -#define TALK_P2P_BASE_PORTALLOCATOR_H_ +#ifndef WEBRTC_P2P_BASE_PORTALLOCATOR_H_ +#define WEBRTC_P2P_BASE_PORTALLOCATOR_H_ #include <string> #include <vector> -#include "talk/p2p/base/portinterface.h" +#include "webrtc/p2p/base/portinterface.h" #include "webrtc/base/helpers.h" #include "webrtc/base/proxyinfo.h" #include "webrtc/base/sigslot.h" @@ -206,4 +206,4 @@ class PortAllocator : public sigslot::has_slots<> { } // namespace cricket -#endif // TALK_P2P_BASE_PORTALLOCATOR_H_ +#endif // WEBRTC_P2P_BASE_PORTALLOCATOR_H_ diff --git a/p2p/base/portallocatorsessionproxy.cc b/p2p/base/portallocatorsessionproxy.cc index a6c8065..2801f26 100644 --- a/p2p/base/portallocatorsessionproxy.cc +++ b/p2p/base/portallocatorsessionproxy.cc @@ -25,10 +25,10 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/p2p/base/portallocatorsessionproxy.h" +#include "webrtc/p2p/base/portallocatorsessionproxy.h" -#include "talk/p2p/base/portallocator.h" -#include "talk/p2p/base/portproxy.h" +#include "webrtc/p2p/base/portallocator.h" +#include "webrtc/p2p/base/portproxy.h" #include "webrtc/base/thread.h" namespace cricket { diff --git a/p2p/base/portallocatorsessionproxy.h b/p2p/base/portallocatorsessionproxy.h index 659c730..fdd5d02 100644 --- a/p2p/base/portallocatorsessionproxy.h +++ b/p2p/base/portallocatorsessionproxy.h @@ -25,13 +25,13 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_BASE_PORTALLOCATORSESSIONPROXY_H_ -#define TALK_P2P_BASE_PORTALLOCATORSESSIONPROXY_H_ +#ifndef WEBRTC_P2P_BASE_PORTALLOCATORSESSIONPROXY_H_ +#define WEBRTC_P2P_BASE_PORTALLOCATORSESSIONPROXY_H_ #include <string> -#include "talk/p2p/base/candidate.h" -#include "talk/p2p/base/portallocator.h" +#include "webrtc/p2p/base/candidate.h" +#include "webrtc/p2p/base/portallocator.h" namespace cricket { class PortAllocator; @@ -120,4 +120,4 @@ class PortAllocatorSessionProxy : public PortAllocatorSession { } // namespace cricket -#endif // TALK_P2P_BASE_PORTALLOCATORSESSIONPROXY_H_ +#endif // WEBRTC_P2P_BASE_PORTALLOCATORSESSIONPROXY_H_ diff --git a/p2p/base/portallocatorsessionproxy_unittest.cc b/p2p/base/portallocatorsessionproxy_unittest.cc index 3a69b46..30353ee 100644 --- a/p2p/base/portallocatorsessionproxy_unittest.cc +++ b/p2p/base/portallocatorsessionproxy_unittest.cc @@ -27,10 +27,10 @@ #include <vector> -#include "talk/p2p/base/basicpacketsocketfactory.h" -#include "talk/p2p/base/portallocatorsessionproxy.h" -#include "talk/p2p/client/basicportallocator.h" -#include "talk/p2p/client/fakeportallocator.h" +#include "webrtc/p2p/base/basicpacketsocketfactory.h" +#include "webrtc/p2p/base/portallocatorsessionproxy.h" +#include "webrtc/p2p/client/basicportallocator.h" +#include "webrtc/p2p/client/fakeportallocator.h" #include "webrtc/base/fakenetwork.h" #include "webrtc/base/gunit.h" #include "webrtc/base/thread.h" diff --git a/p2p/base/portinterface.h b/p2p/base/portinterface.h index 8cab5b4..d53dc2e 100644 --- a/p2p/base/portinterface.h +++ b/p2p/base/portinterface.h @@ -25,12 +25,12 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_BASE_PORTINTERFACE_H_ -#define TALK_P2P_BASE_PORTINTERFACE_H_ +#ifndef WEBRTC_P2P_BASE_PORTINTERFACE_H_ +#define WEBRTC_P2P_BASE_PORTINTERFACE_H_ #include <string> -#include "talk/p2p/base/transport.h" +#include "webrtc/p2p/base/transport.h" #include "webrtc/base/socketaddress.h" namespace rtc { @@ -140,4 +140,4 @@ class PortInterface { } // namespace cricket -#endif // TALK_P2P_BASE_PORTINTERFACE_H_ +#endif // WEBRTC_P2P_BASE_PORTINTERFACE_H_ diff --git a/p2p/base/portproxy.cc b/p2p/base/portproxy.cc index 841cd85..5a5d1ff 100644 --- a/p2p/base/portproxy.cc +++ b/p2p/base/portproxy.cc @@ -25,7 +25,7 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/p2p/base/portproxy.h" +#include "webrtc/p2p/base/portproxy.h" namespace cricket { diff --git a/p2p/base/portproxy.h b/p2p/base/portproxy.h index 8c28223..6f79318 100644 --- a/p2p/base/portproxy.h +++ b/p2p/base/portproxy.h @@ -25,10 +25,10 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_BASE_PORTPROXY_H_ -#define TALK_P2P_BASE_PORTPROXY_H_ +#ifndef WEBRTC_P2P_BASE_PORTPROXY_H_ +#define WEBRTC_P2P_BASE_PORTPROXY_H_ -#include "talk/p2p/base/portinterface.h" +#include "webrtc/p2p/base/portinterface.h" #include "webrtc/base/sigslot.h" namespace rtc { @@ -101,4 +101,4 @@ class PortProxy : public PortInterface, public sigslot::has_slots<> { } // namespace cricket -#endif // TALK_P2P_BASE_PORTPROXY_H_ +#endif // WEBRTC_P2P_BASE_PORTPROXY_H_ diff --git a/p2p/base/pseudotcp.cc b/p2p/base/pseudotcp.cc index 9a944f0..a82cd67 100644 --- a/p2p/base/pseudotcp.cc +++ b/p2p/base/pseudotcp.cc @@ -25,7 +25,7 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/p2p/base/pseudotcp.h" +#include "webrtc/p2p/base/pseudotcp.h" #include <stdio.h> #include <stdlib.h> diff --git a/p2p/base/pseudotcp.h b/p2p/base/pseudotcp.h index 46e9d3b..e3455de 100644 --- a/p2p/base/pseudotcp.h +++ b/p2p/base/pseudotcp.h @@ -25,8 +25,8 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_BASE_PSEUDOTCP_H_ -#define TALK_P2P_BASE_PSEUDOTCP_H_ +#ifndef WEBRTC_P2P_BASE_PSEUDOTCP_H_ +#define WEBRTC_P2P_BASE_PSEUDOTCP_H_ #include <list> @@ -255,4 +255,4 @@ class PseudoTcp { } // namespace cricket -#endif // TALK_P2P_BASE_PSEUDOTCP_H_ +#endif // WEBRTC_P2P_BASE_PSEUDOTCP_H_ diff --git a/p2p/base/pseudotcp_unittest.cc b/p2p/base/pseudotcp_unittest.cc index d9435cf..522eb18 100644 --- a/p2p/base/pseudotcp_unittest.cc +++ b/p2p/base/pseudotcp_unittest.cc @@ -27,7 +27,7 @@ #include <vector> -#include "talk/p2p/base/pseudotcp.h" +#include "webrtc/p2p/base/pseudotcp.h" #include "webrtc/base/gunit.h" #include "webrtc/base/helpers.h" #include "webrtc/base/messagehandler.h" diff --git a/p2p/base/rawtransport.cc b/p2p/base/rawtransport.cc index 2af1864..f363dd1 100644 --- a/p2p/base/rawtransport.cc +++ b/p2p/base/rawtransport.cc @@ -27,14 +27,14 @@ #include <string> #include <vector> -#include "talk/p2p/base/constants.h" -#include "talk/p2p/base/parsing.h" -#include "talk/p2p/base/rawtransport.h" -#include "talk/p2p/base/rawtransportchannel.h" -#include "talk/p2p/base/sessionmanager.h" +#include "webrtc/p2p/base/constants.h" +#include "webrtc/p2p/base/parsing.h" +#include "webrtc/p2p/base/rawtransport.h" +#include "webrtc/p2p/base/rawtransportchannel.h" +#include "webrtc/p2p/base/sessionmanager.h" #include "webrtc/libjingle/xmllite/qname.h" #include "webrtc/libjingle/xmllite/xmlelement.h" -#include "talk/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/constants.h" #include "webrtc/base/common.h" #if defined(FEATURE_ENABLE_PSTN) diff --git a/p2p/base/rawtransport.h b/p2p/base/rawtransport.h index 3a20ef5..7458191 100644 --- a/p2p/base/rawtransport.h +++ b/p2p/base/rawtransport.h @@ -25,11 +25,11 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_BASE_RAWTRANSPORT_H_ -#define TALK_P2P_BASE_RAWTRANSPORT_H_ +#ifndef WEBRTC_P2P_BASE_RAWTRANSPORT_H_ +#define WEBRTC_P2P_BASE_RAWTRANSPORT_H_ #include <string> -#include "talk/p2p/base/transport.h" +#include "webrtc/p2p/base/transport.h" #if defined(FEATURE_ENABLE_PSTN) namespace cricket { @@ -78,4 +78,4 @@ class RawTransport : public Transport, public TransportParser { #endif // defined(FEATURE_ENABLE_PSTN) -#endif // TALK_P2P_BASE_RAWTRANSPORT_H_ +#endif // WEBRTC_P2P_BASE_RAWTRANSPORT_H_ diff --git a/p2p/base/rawtransportchannel.cc b/p2p/base/rawtransportchannel.cc index ae268e3..ef2286b 100644 --- a/p2p/base/rawtransportchannel.cc +++ b/p2p/base/rawtransportchannel.cc @@ -25,20 +25,20 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/p2p/base/rawtransportchannel.h" +#include "webrtc/p2p/base/rawtransportchannel.h" #include <string> #include <vector> -#include "talk/p2p/base/constants.h" -#include "talk/p2p/base/portallocator.h" -#include "talk/p2p/base/portinterface.h" -#include "talk/p2p/base/rawtransport.h" -#include "talk/p2p/base/relayport.h" -#include "talk/p2p/base/sessionmanager.h" -#include "talk/p2p/base/stunport.h" +#include "webrtc/p2p/base/constants.h" +#include "webrtc/p2p/base/portallocator.h" +#include "webrtc/p2p/base/portinterface.h" +#include "webrtc/p2p/base/rawtransport.h" +#include "webrtc/p2p/base/relayport.h" +#include "webrtc/p2p/base/sessionmanager.h" +#include "webrtc/p2p/base/stunport.h" #include "webrtc/libjingle/xmllite/qname.h" #include "webrtc/libjingle/xmllite/xmlelement.h" -#include "talk/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/constants.h" #include "webrtc/base/common.h" #if defined(FEATURE_ENABLE_PSTN) diff --git a/p2p/base/rawtransportchannel.h b/p2p/base/rawtransportchannel.h index 8fbf073..25f6e4f 100644 --- a/p2p/base/rawtransportchannel.h +++ b/p2p/base/rawtransportchannel.h @@ -25,14 +25,14 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_BASE_RAWTRANSPORTCHANNEL_H_ -#define TALK_P2P_BASE_RAWTRANSPORTCHANNEL_H_ +#ifndef WEBRTC_P2P_BASE_RAWTRANSPORTCHANNEL_H_ +#define WEBRTC_P2P_BASE_RAWTRANSPORTCHANNEL_H_ #include <string> #include <vector> -#include "talk/p2p/base/candidate.h" -#include "talk/p2p/base/rawtransport.h" -#include "talk/p2p/base/transportchannelimpl.h" +#include "webrtc/p2p/base/candidate.h" +#include "webrtc/p2p/base/rawtransport.h" +#include "webrtc/p2p/base/transportchannelimpl.h" #include "webrtc/base/messagequeue.h" #if defined(FEATURE_ENABLE_PSTN) @@ -203,4 +203,4 @@ class RawTransportChannel : public TransportChannelImpl, } // namespace cricket #endif // defined(FEATURE_ENABLE_PSTN) -#endif // TALK_P2P_BASE_RAWTRANSPORTCHANNEL_H_ +#endif // WEBRTC_P2P_BASE_RAWTRANSPORTCHANNEL_H_ diff --git a/p2p/base/relayport.cc b/p2p/base/relayport.cc index 55052a5..2db42c9 100644 --- a/p2p/base/relayport.cc +++ b/p2p/base/relayport.cc @@ -25,7 +25,7 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/p2p/base/relayport.h" +#include "webrtc/p2p/base/relayport.h" #include "webrtc/base/asyncpacketsocket.h" #include "webrtc/base/helpers.h" #include "webrtc/base/logging.h" diff --git a/p2p/base/relayport.h b/p2p/base/relayport.h index f22d045..bc0c5d9 100644 --- a/p2p/base/relayport.h +++ b/p2p/base/relayport.h @@ -25,16 +25,16 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_BASE_RELAYPORT_H_ -#define TALK_P2P_BASE_RELAYPORT_H_ +#ifndef WEBRTC_P2P_BASE_RELAYPORT_H_ +#define WEBRTC_P2P_BASE_RELAYPORT_H_ #include <deque> #include <string> #include <utility> #include <vector> -#include "talk/p2p/base/port.h" -#include "talk/p2p/base/stunrequest.h" +#include "webrtc/p2p/base/port.h" +#include "webrtc/p2p/base/stunrequest.h" namespace cricket { @@ -115,4 +115,4 @@ class RelayPort : public Port { } // namespace cricket -#endif // TALK_P2P_BASE_RELAYPORT_H_ +#endif // WEBRTC_P2P_BASE_RELAYPORT_H_ diff --git a/p2p/base/relayport_unittest.cc b/p2p/base/relayport_unittest.cc index ebb16c5..2f4515e 100644 --- a/p2p/base/relayport_unittest.cc +++ b/p2p/base/relayport_unittest.cc @@ -25,9 +25,9 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/p2p/base/basicpacketsocketfactory.h" -#include "talk/p2p/base/relayport.h" -#include "talk/p2p/base/relayserver.h" +#include "webrtc/p2p/base/basicpacketsocketfactory.h" +#include "webrtc/p2p/base/relayport.h" +#include "webrtc/p2p/base/relayserver.h" #include "webrtc/base/gunit.h" #include "webrtc/base/helpers.h" #include "webrtc/base/logging.h" diff --git a/p2p/base/relayserver.cc b/p2p/base/relayserver.cc index b5d1ac6..ebf165b 100644 --- a/p2p/base/relayserver.cc +++ b/p2p/base/relayserver.cc @@ -25,7 +25,7 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/p2p/base/relayserver.h" +#include "webrtc/p2p/base/relayserver.h" #ifdef POSIX #include <errno.h> diff --git a/p2p/base/relayserver.h b/p2p/base/relayserver.h index 248d0e7..8065174 100644 --- a/p2p/base/relayserver.h +++ b/p2p/base/relayserver.h @@ -25,15 +25,15 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_BASE_RELAYSERVER_H_ -#define TALK_P2P_BASE_RELAYSERVER_H_ +#ifndef WEBRTC_P2P_BASE_RELAYSERVER_H_ +#define WEBRTC_P2P_BASE_RELAYSERVER_H_ #include <map> #include <string> #include <vector> -#include "talk/p2p/base/port.h" -#include "talk/p2p/base/stun.h" +#include "webrtc/p2p/base/port.h" +#include "webrtc/p2p/base/stun.h" #include "webrtc/base/asyncudpsocket.h" #include "webrtc/base/socketaddresspair.h" #include "webrtc/base/thread.h" @@ -249,4 +249,4 @@ class RelayServerBinding : public rtc::MessageHandler { } // namespace cricket -#endif // TALK_P2P_BASE_RELAYSERVER_H_ +#endif // WEBRTC_P2P_BASE_RELAYSERVER_H_ diff --git a/p2p/base/relayserver_unittest.cc b/p2p/base/relayserver_unittest.cc index 5d77ca6..5ea5e10 100644 --- a/p2p/base/relayserver_unittest.cc +++ b/p2p/base/relayserver_unittest.cc @@ -27,7 +27,7 @@ #include <string> -#include "talk/p2p/base/relayserver.h" +#include "webrtc/p2p/base/relayserver.h" #include "webrtc/base/gunit.h" #include "webrtc/base/helpers.h" #include "webrtc/base/logging.h" diff --git a/p2p/base/session.cc b/p2p/base/session.cc index a9bf815..c4db1cf 100644 --- a/p2p/base/session.cc +++ b/p2p/base/session.cc @@ -25,16 +25,16 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/p2p/base/session.h" - -#include "talk/p2p/base/dtlstransport.h" -#include "talk/p2p/base/p2ptransport.h" -#include "talk/p2p/base/sessionclient.h" -#include "talk/p2p/base/transport.h" -#include "talk/p2p/base/transportchannelproxy.h" -#include "talk/p2p/base/transportinfo.h" -#include "talk/xmpp/constants.h" -#include "talk/xmpp/jid.h" +#include "webrtc/p2p/base/session.h" + +#include "webrtc/p2p/base/dtlstransport.h" +#include "webrtc/p2p/base/p2ptransport.h" +#include "webrtc/p2p/base/sessionclient.h" +#include "webrtc/p2p/base/transport.h" +#include "webrtc/p2p/base/transportchannelproxy.h" +#include "webrtc/p2p/base/transportinfo.h" +#include "webrtc/libjingle/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/jid.h" #include "webrtc/base/bind.h" #include "webrtc/base/common.h" #include "webrtc/base/helpers.h" @@ -42,7 +42,7 @@ #include "webrtc/base/scoped_ptr.h" #include "webrtc/base/sslstreamadapter.h" -#include "talk/p2p/base/constants.h" +#include "webrtc/p2p/base/constants.h" namespace cricket { diff --git a/p2p/base/session.h b/p2p/base/session.h index e06cf00..cbe350f 100644 --- a/p2p/base/session.h +++ b/p2p/base/session.h @@ -25,22 +25,22 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_BASE_SESSION_H_ -#define TALK_P2P_BASE_SESSION_H_ +#ifndef WEBRTC_P2P_BASE_SESSION_H_ +#define WEBRTC_P2P_BASE_SESSION_H_ #include <list> #include <map> #include <string> #include <vector> -#include "talk/p2p/base/parsing.h" -#include "talk/p2p/base/port.h" -#include "talk/p2p/base/sessionclient.h" -#include "talk/p2p/base/sessionmanager.h" -#include "talk/p2p/base/sessionmessages.h" -#include "talk/p2p/base/transport.h" +#include "webrtc/p2p/base/parsing.h" +#include "webrtc/p2p/base/port.h" +#include "webrtc/p2p/base/sessionclient.h" +#include "webrtc/p2p/base/sessionmanager.h" +#include "webrtc/p2p/base/sessionmessages.h" +#include "webrtc/p2p/base/transport.h" #include "webrtc/libjingle/xmllite/xmlelement.h" -#include "talk/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/constants.h" #include "webrtc/base/refcount.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/base/scoped_ref_ptr.h" @@ -744,4 +744,4 @@ class Session : public BaseSession { } // namespace cricket -#endif // TALK_P2P_BASE_SESSION_H_ +#endif // WEBRTC_P2P_BASE_SESSION_H_ diff --git a/p2p/base/session_unittest.cc b/p2p/base/session_unittest.cc index 4674d2c..c7a3db9 100644 --- a/p2p/base/session_unittest.cc +++ b/p2p/base/session_unittest.cc @@ -31,22 +31,22 @@ #include <map> #include <sstream> -#include "talk/p2p/base/basicpacketsocketfactory.h" -#include "talk/p2p/base/constants.h" -#include "talk/p2p/base/p2ptransport.h" -#include "talk/p2p/base/parsing.h" -#include "talk/p2p/base/portallocator.h" -#include "talk/p2p/base/relayport.h" -#include "talk/p2p/base/relayserver.h" -#include "talk/p2p/base/session.h" -#include "talk/p2p/base/sessionclient.h" -#include "talk/p2p/base/sessionmanager.h" -#include "talk/p2p/base/stunport.h" -#include "talk/p2p/base/stunserver.h" -#include "talk/p2p/base/transportchannel.h" -#include "talk/p2p/base/transportchannelproxy.h" -#include "talk/p2p/base/udpport.h" -#include "talk/xmpp/constants.h" +#include "webrtc/p2p/base/basicpacketsocketfactory.h" +#include "webrtc/p2p/base/constants.h" +#include "webrtc/p2p/base/p2ptransport.h" +#include "webrtc/p2p/base/parsing.h" +#include "webrtc/p2p/base/portallocator.h" +#include "webrtc/p2p/base/relayport.h" +#include "webrtc/p2p/base/relayserver.h" +#include "webrtc/p2p/base/session.h" +#include "webrtc/p2p/base/sessionclient.h" +#include "webrtc/p2p/base/sessionmanager.h" +#include "webrtc/p2p/base/stunport.h" +#include "webrtc/p2p/base/stunserver.h" +#include "webrtc/p2p/base/transportchannel.h" +#include "webrtc/p2p/base/transportchannelproxy.h" +#include "webrtc/p2p/base/udpport.h" +#include "webrtc/libjingle/xmpp/constants.h" #include "webrtc/base/base64.h" #include "webrtc/base/common.h" #include "webrtc/base/gunit.h" diff --git a/p2p/base/sessionclient.h b/p2p/base/sessionclient.h index 10b0c92..f7952f0 100644 --- a/p2p/base/sessionclient.h +++ b/p2p/base/sessionclient.h @@ -25,10 +25,10 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_BASE_SESSIONCLIENT_H_ -#define TALK_P2P_BASE_SESSIONCLIENT_H_ +#ifndef WEBRTC_P2P_BASE_SESSIONCLIENT_H_ +#define WEBRTC_P2P_BASE_SESSIONCLIENT_H_ -#include "talk/p2p/base/constants.h" +#include "webrtc/p2p/base/constants.h" namespace buzz { class XmlElement; @@ -92,4 +92,4 @@ class SessionClient : public ContentParser { } // namespace cricket -#endif // TALK_P2P_BASE_SESSIONCLIENT_H_ +#endif // WEBRTC_P2P_BASE_SESSIONCLIENT_H_ diff --git a/p2p/base/sessiondescription.cc b/p2p/base/sessiondescription.cc index 7ad3d48..57f08c9 100644 --- a/p2p/base/sessiondescription.cc +++ b/p2p/base/sessiondescription.cc @@ -25,7 +25,7 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/p2p/base/sessiondescription.h" +#include "webrtc/p2p/base/sessiondescription.h" #include "webrtc/libjingle/xmllite/xmlelement.h" diff --git a/p2p/base/sessiondescription.h b/p2p/base/sessiondescription.h index 56dd412..bc76761 100644 --- a/p2p/base/sessiondescription.h +++ b/p2p/base/sessiondescription.h @@ -25,13 +25,13 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_BASE_SESSIONDESCRIPTION_H_ -#define TALK_P2P_BASE_SESSIONDESCRIPTION_H_ +#ifndef WEBRTC_P2P_BASE_SESSIONDESCRIPTION_H_ +#define WEBRTC_P2P_BASE_SESSIONDESCRIPTION_H_ #include <string> #include <vector> -#include "talk/p2p/base/transportinfo.h" +#include "webrtc/p2p/base/transportinfo.h" #include "webrtc/base/constructormagic.h" namespace cricket { @@ -199,4 +199,4 @@ enum ContentSource { } // namespace cricket -#endif // TALK_P2P_BASE_SESSIONDESCRIPTION_H_ +#endif // WEBRTC_P2P_BASE_SESSIONDESCRIPTION_H_ diff --git a/p2p/base/sessionid.h b/p2p/base/sessionid.h index 6942942..52ee990 100644 --- a/p2p/base/sessionid.h +++ b/p2p/base/sessionid.h @@ -25,8 +25,8 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_BASE_SESSIONID_H_ -#define TALK_P2P_BASE_SESSIONID_H_ +#ifndef WEBRTC_P2P_BASE_SESSIONID_H_ +#define WEBRTC_P2P_BASE_SESSIONID_H_ // TODO: Remove this file. @@ -34,4 +34,4 @@ namespace cricket { } // namespace cricket -#endif // TALK_P2P_BASE_SESSIONID_H_ +#endif // WEBRTC_P2P_BASE_SESSIONID_H_ diff --git a/p2p/base/sessionmanager.cc b/p2p/base/sessionmanager.cc index d173517..7f27f6c 100644 --- a/p2p/base/sessionmanager.cc +++ b/p2p/base/sessionmanager.cc @@ -25,13 +25,13 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/p2p/base/sessionmanager.h" +#include "webrtc/p2p/base/sessionmanager.h" -#include "talk/p2p/base/constants.h" -#include "talk/p2p/base/session.h" -#include "talk/p2p/base/sessionmessages.h" -#include "talk/xmpp/constants.h" -#include "talk/xmpp/jid.h" +#include "webrtc/p2p/base/constants.h" +#include "webrtc/p2p/base/session.h" +#include "webrtc/p2p/base/sessionmessages.h" +#include "webrtc/libjingle/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/jid.h" #include "webrtc/base/common.h" #include "webrtc/base/helpers.h" #include "webrtc/base/logging.h" diff --git a/p2p/base/sessionmanager.h b/p2p/base/sessionmanager.h index 92aeb3d..0e0e9a4 100644 --- a/p2p/base/sessionmanager.h +++ b/p2p/base/sessionmanager.h @@ -25,16 +25,16 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_BASE_SESSIONMANAGER_H_ -#define TALK_P2P_BASE_SESSIONMANAGER_H_ +#ifndef WEBRTC_P2P_BASE_SESSIONMANAGER_H_ +#define WEBRTC_P2P_BASE_SESSIONMANAGER_H_ #include <map> #include <string> #include <utility> #include <vector> -#include "talk/p2p/base/portallocator.h" -#include "talk/p2p/base/transportdescriptionfactory.h" +#include "webrtc/p2p/base/portallocator.h" +#include "webrtc/p2p/base/transportdescriptionfactory.h" #include "webrtc/base/sigslot.h" #include "webrtc/base/thread.h" @@ -208,4 +208,4 @@ class SessionManager : public sigslot::has_slots<> { } // namespace cricket -#endif // TALK_P2P_BASE_SESSIONMANAGER_H_ +#endif // WEBRTC_P2P_BASE_SESSIONMANAGER_H_ diff --git a/p2p/base/sessionmessages.cc b/p2p/base/sessionmessages.cc index b2fd9d6..0534271 100644 --- a/p2p/base/sessionmessages.cc +++ b/p2p/base/sessionmessages.cc @@ -25,19 +25,19 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/p2p/base/sessionmessages.h" +#include "webrtc/p2p/base/sessionmessages.h" #include <stdio.h> #include <string> -#include "talk/p2p/base/constants.h" -#include "talk/p2p/base/p2ptransport.h" -#include "talk/p2p/base/parsing.h" -#include "talk/p2p/base/sessionclient.h" -#include "talk/p2p/base/sessiondescription.h" -#include "talk/p2p/base/transport.h" +#include "webrtc/p2p/base/constants.h" +#include "webrtc/p2p/base/p2ptransport.h" +#include "webrtc/p2p/base/parsing.h" +#include "webrtc/p2p/base/sessionclient.h" +#include "webrtc/p2p/base/sessiondescription.h" +#include "webrtc/p2p/base/transport.h" #include "webrtc/libjingle/xmllite/xmlconstants.h" -#include "talk/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/constants.h" #include "webrtc/base/logging.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/base/stringutils.h" diff --git a/p2p/base/sessionmessages.h b/p2p/base/sessionmessages.h index 7e1f8ac..0ab56ce 100644 --- a/p2p/base/sessionmessages.h +++ b/p2p/base/sessionmessages.h @@ -25,17 +25,17 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_BASE_SESSIONMESSAGES_H_ -#define TALK_P2P_BASE_SESSIONMESSAGES_H_ +#ifndef WEBRTC_P2P_BASE_SESSIONMESSAGES_H_ +#define WEBRTC_P2P_BASE_SESSIONMESSAGES_H_ #include <map> #include <string> #include <vector> -#include "talk/p2p/base/constants.h" -#include "talk/p2p/base/parsing.h" -#include "talk/p2p/base/sessiondescription.h" // Needed to delete contents. -#include "talk/p2p/base/transportinfo.h" +#include "webrtc/p2p/base/constants.h" +#include "webrtc/p2p/base/parsing.h" +#include "webrtc/p2p/base/sessiondescription.h" // Needed to delete contents. +#include "webrtc/p2p/base/transportinfo.h" #include "webrtc/libjingle/xmllite/xmlelement.h" #include "webrtc/base/basictypes.h" @@ -240,4 +240,4 @@ bool FindSessionRedirect(const buzz::XmlElement* stanza, SessionRedirect* redirect); } // namespace cricket -#endif // TALK_P2P_BASE_SESSIONMESSAGES_H_ +#endif // WEBRTC_P2P_BASE_SESSIONMESSAGES_H_ diff --git a/p2p/base/stun.cc b/p2p/base/stun.cc index 061fd9a..6bdf77e 100644 --- a/p2p/base/stun.cc +++ b/p2p/base/stun.cc @@ -25,7 +25,7 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/p2p/base/stun.h" +#include "webrtc/p2p/base/stun.h" #include <string.h> diff --git a/p2p/base/stun.h b/p2p/base/stun.h index c4f522b..5a43a3c 100644 --- a/p2p/base/stun.h +++ b/p2p/base/stun.h @@ -25,8 +25,8 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_BASE_STUN_H_ -#define TALK_P2P_BASE_STUN_H_ +#ifndef WEBRTC_P2P_BASE_STUN_H_ +#define WEBRTC_P2P_BASE_STUN_H_ // This file contains classes for dealing with the STUN protocol, as specified // in RFC 5389, and its descendants. @@ -646,4 +646,4 @@ class IceMessage : public StunMessage { } // namespace cricket -#endif // TALK_P2P_BASE_STUN_H_ +#endif // WEBRTC_P2P_BASE_STUN_H_ diff --git a/p2p/base/stun_unittest.cc b/p2p/base/stun_unittest.cc index 00dffed..34a47cb 100644 --- a/p2p/base/stun_unittest.cc +++ b/p2p/base/stun_unittest.cc @@ -27,7 +27,7 @@ #include <string> -#include "talk/p2p/base/stun.h" +#include "webrtc/p2p/base/stun.h" #include "webrtc/base/bytebuffer.h" #include "webrtc/base/gunit.h" #include "webrtc/base/logging.h" diff --git a/p2p/base/stunport.cc b/p2p/base/stunport.cc index a185a35..0602c5a 100644 --- a/p2p/base/stunport.cc +++ b/p2p/base/stunport.cc @@ -25,10 +25,11 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/p2p/base/stunport.h" +#include "webrtc/p2p/base/stunport.h" -#include "talk/p2p/base/common.h" -#include "talk/p2p/base/stun.h" +#include "webrtc/p2p/base/common.h" +#include "webrtc/p2p/base/portallocator.h" +#include "webrtc/p2p/base/stun.h" #include "webrtc/base/common.h" #include "webrtc/base/helpers.h" #include "webrtc/base/logging.h" @@ -395,8 +396,17 @@ void UDPPort::OnStunBindingRequestSucceeded( // For STUN, related address is the local socket address. if ((!SharedSocket() || stun_reflected_addr != socket_->GetLocalAddress()) && !HasCandidateWithAddress(stun_reflected_addr)) { + + rtc::SocketAddress related_address = socket_->GetLocalAddress(); + if (!(candidate_filter() & CF_HOST)) { + // If candidate filter doesn't have CF_HOST specified, empty raddr to + // avoid local address leakage. + related_address = rtc::EmptySocketAddressWithFamily( + related_address.family()); + } + AddAddress(stun_reflected_addr, socket_->GetLocalAddress(), - socket_->GetLocalAddress(), UDP_PROTOCOL_NAME, "", + related_address, UDP_PROTOCOL_NAME, "", STUN_PORT_TYPE, ICE_TYPE_PREFERENCE_SRFLX, 0, false); } MaybeSetPortCompleteOrError(); diff --git a/p2p/base/stunport.h b/p2p/base/stunport.h index 1fc1f64..9767546 100644 --- a/p2p/base/stunport.h +++ b/p2p/base/stunport.h @@ -25,13 +25,13 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_BASE_STUNPORT_H_ -#define TALK_P2P_BASE_STUNPORT_H_ +#ifndef WEBRTC_P2P_BASE_STUNPORT_H_ +#define WEBRTC_P2P_BASE_STUNPORT_H_ #include <string> -#include "talk/p2p/base/port.h" -#include "talk/p2p/base/stunrequest.h" +#include "webrtc/p2p/base/port.h" +#include "webrtc/p2p/base/stunrequest.h" #include "webrtc/base/asyncpacketsocket.h" // TODO(mallinath) - Rename stunport.cc|h to udpport.cc|h. @@ -252,4 +252,4 @@ class StunPort : public UDPPort { } // namespace cricket -#endif // TALK_P2P_BASE_STUNPORT_H_ +#endif // WEBRTC_P2P_BASE_STUNPORT_H_ diff --git a/p2p/base/stunport_unittest.cc b/p2p/base/stunport_unittest.cc index 5c97ecc..7b289ea 100644 --- a/p2p/base/stunport_unittest.cc +++ b/p2p/base/stunport_unittest.cc @@ -25,9 +25,9 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/p2p/base/basicpacketsocketfactory.h" -#include "talk/p2p/base/stunport.h" -#include "talk/p2p/base/teststunserver.h" +#include "webrtc/p2p/base/basicpacketsocketfactory.h" +#include "webrtc/p2p/base/stunport.h" +#include "webrtc/p2p/base/teststunserver.h" #include "webrtc/base/gunit.h" #include "webrtc/base/helpers.h" #include "webrtc/base/physicalsocketserver.h" diff --git a/p2p/base/stunrequest.cc b/p2p/base/stunrequest.cc index 148718f..55d270e 100644 --- a/p2p/base/stunrequest.cc +++ b/p2p/base/stunrequest.cc @@ -25,7 +25,7 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/p2p/base/stunrequest.h" +#include "webrtc/p2p/base/stunrequest.h" #include "webrtc/base/common.h" #include "webrtc/base/helpers.h" diff --git a/p2p/base/stunrequest.h b/p2p/base/stunrequest.h index 374f7e7..be73445 100644 --- a/p2p/base/stunrequest.h +++ b/p2p/base/stunrequest.h @@ -25,12 +25,12 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_BASE_STUNREQUEST_H_ -#define TALK_P2P_BASE_STUNREQUEST_H_ +#ifndef WEBRTC_P2P_BASE_STUNREQUEST_H_ +#define WEBRTC_P2P_BASE_STUNREQUEST_H_ #include <map> #include <string> -#include "talk/p2p/base/stun.h" +#include "webrtc/p2p/base/stun.h" #include "webrtc/base/sigslot.h" #include "webrtc/base/thread.h" @@ -130,4 +130,4 @@ private: } // namespace cricket -#endif // TALK_P2P_BASE_STUNREQUEST_H_ +#endif // WEBRTC_P2P_BASE_STUNREQUEST_H_ diff --git a/p2p/base/stunrequest_unittest.cc b/p2p/base/stunrequest_unittest.cc index 2783aa3..5c7b97b 100644 --- a/p2p/base/stunrequest_unittest.cc +++ b/p2p/base/stunrequest_unittest.cc @@ -25,7 +25,7 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/p2p/base/stunrequest.h" +#include "webrtc/p2p/base/stunrequest.h" #include "webrtc/base/gunit.h" #include "webrtc/base/helpers.h" #include "webrtc/base/logging.h" diff --git a/p2p/base/stunserver.cc b/p2p/base/stunserver.cc index df428f9..a4beca4 100644 --- a/p2p/base/stunserver.cc +++ b/p2p/base/stunserver.cc @@ -25,7 +25,7 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/p2p/base/stunserver.h" +#include "webrtc/p2p/base/stunserver.h" #include "webrtc/base/bytebuffer.h" #include "webrtc/base/logging.h" diff --git a/p2p/base/stunserver.h b/p2p/base/stunserver.h index a141e5b..5bbe258 100644 --- a/p2p/base/stunserver.h +++ b/p2p/base/stunserver.h @@ -25,10 +25,10 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_BASE_STUNSERVER_H_ -#define TALK_P2P_BASE_STUNSERVER_H_ +#ifndef WEBRTC_P2P_BASE_STUNSERVER_H_ +#define WEBRTC_P2P_BASE_STUNSERVER_H_ -#include "talk/p2p/base/stun.h" +#include "webrtc/p2p/base/stun.h" #include "webrtc/base/asyncudpsocket.h" #include "webrtc/base/scoped_ptr.h" @@ -80,4 +80,4 @@ class StunServer : public sigslot::has_slots<> { } // namespace cricket -#endif // TALK_P2P_BASE_STUNSERVER_H_ +#endif // WEBRTC_P2P_BASE_STUNSERVER_H_ diff --git a/p2p/base/stunserver_unittest.cc b/p2p/base/stunserver_unittest.cc index 405d8ae..4cf3c33 100644 --- a/p2p/base/stunserver_unittest.cc +++ b/p2p/base/stunserver_unittest.cc @@ -27,7 +27,7 @@ #include <string> -#include "talk/p2p/base/stunserver.h" +#include "webrtc/p2p/base/stunserver.h" #include "webrtc/base/gunit.h" #include "webrtc/base/logging.h" #include "webrtc/base/physicalsocketserver.h" diff --git a/p2p/base/tcpport.cc b/p2p/base/tcpport.cc index cb5b726..aa08a92 100644 --- a/p2p/base/tcpport.cc +++ b/p2p/base/tcpport.cc @@ -25,9 +25,9 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/p2p/base/tcpport.h" +#include "webrtc/p2p/base/tcpport.h" -#include "talk/p2p/base/common.h" +#include "webrtc/p2p/base/common.h" #include "webrtc/base/common.h" #include "webrtc/base/logging.h" @@ -304,20 +304,23 @@ void TCPConnection::OnConnect(rtc::AsyncPacketSocket* socket) { // the one we asked for. This is seen in Chrome, where TCP sockets cannot be // given a binding address, and the platform is expected to pick the // correct local address. - if (socket->GetLocalAddress().ipaddr() == port()->ip()) { + const rtc::IPAddress& socket_ip = socket->GetLocalAddress().ipaddr(); + if (socket_ip == port()->ip()) { LOG_J(LS_VERBOSE, this) << "Connection established to " << socket->GetRemoteAddress().ToSensitiveString(); set_connected(true); } else { - LOG_J(LS_WARNING, this) << "Dropping connection as TCP socket bound to a " - << "different address from the local candidate."; + LOG_J(LS_WARNING, this) << "Dropping connection as TCP socket bound to IP " + << socket_ip.ToSensitiveString() + << ", different from the local candidate IP " + << port()->ip().ToSensitiveString(); socket_->Close(); } } void TCPConnection::OnClose(rtc::AsyncPacketSocket* socket, int error) { ASSERT(socket == socket_); - LOG_J(LS_VERBOSE, this) << "Connection closed with error " << error; + LOG_J(LS_INFO, this) << "Connection closed with error " << error; set_connected(false); set_write_state(STATE_WRITE_TIMEOUT); } diff --git a/p2p/base/tcpport.h b/p2p/base/tcpport.h index 97bd977..c331f6b 100644 --- a/p2p/base/tcpport.h +++ b/p2p/base/tcpport.h @@ -25,12 +25,12 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_BASE_TCPPORT_H_ -#define TALK_P2P_BASE_TCPPORT_H_ +#ifndef WEBRTC_P2P_BASE_TCPPORT_H_ +#define WEBRTC_P2P_BASE_TCPPORT_H_ #include <list> #include <string> -#include "talk/p2p/base/port.h" +#include "webrtc/p2p/base/port.h" #include "webrtc/base/asyncpacketsocket.h" namespace cricket { @@ -150,4 +150,4 @@ class TCPConnection : public Connection { } // namespace cricket -#endif // TALK_P2P_BASE_TCPPORT_H_ +#endif // WEBRTC_P2P_BASE_TCPPORT_H_ diff --git a/p2p/base/testrelayserver.h b/p2p/base/testrelayserver.h index eaeefc8..003b42c 100644 --- a/p2p/base/testrelayserver.h +++ b/p2p/base/testrelayserver.h @@ -25,10 +25,10 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_BASE_TESTRELAYSERVER_H_ -#define TALK_P2P_BASE_TESTRELAYSERVER_H_ +#ifndef WEBRTC_P2P_BASE_TESTRELAYSERVER_H_ +#define WEBRTC_P2P_BASE_TESTRELAYSERVER_H_ -#include "talk/p2p/base/relayserver.h" +#include "webrtc/p2p/base/relayserver.h" #include "webrtc/base/asynctcpsocket.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/base/sigslot.h" @@ -115,4 +115,4 @@ class TestRelayServer : public sigslot::has_slots<> { } // namespace cricket -#endif // TALK_P2P_BASE_TESTRELAYSERVER_H_ +#endif // WEBRTC_P2P_BASE_TESTRELAYSERVER_H_ diff --git a/p2p/base/teststunserver.h b/p2p/base/teststunserver.h index 840150b..eef15d7 100644 --- a/p2p/base/teststunserver.h +++ b/p2p/base/teststunserver.h @@ -25,10 +25,10 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_BASE_TESTSTUNSERVER_H_ -#define TALK_P2P_BASE_TESTSTUNSERVER_H_ +#ifndef WEBRTC_P2P_BASE_TESTSTUNSERVER_H_ +#define WEBRTC_P2P_BASE_TESTSTUNSERVER_H_ -#include "talk/p2p/base/stunserver.h" +#include "webrtc/p2p/base/stunserver.h" #include "webrtc/base/socketaddress.h" #include "webrtc/base/thread.h" @@ -72,4 +72,4 @@ class TestStunServer : StunServer { } // namespace cricket -#endif // TALK_P2P_BASE_TESTSTUNSERVER_H_ +#endif // WEBRTC_P2P_BASE_TESTSTUNSERVER_H_ diff --git a/p2p/base/testturnserver.h b/p2p/base/testturnserver.h index 6c30afe..64da877 100644 --- a/p2p/base/testturnserver.h +++ b/p2p/base/testturnserver.h @@ -25,15 +25,15 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_BASE_TESTTURNSERVER_H_ -#define TALK_P2P_BASE_TESTTURNSERVER_H_ +#ifndef WEBRTC_P2P_BASE_TESTTURNSERVER_H_ +#define WEBRTC_P2P_BASE_TESTTURNSERVER_H_ #include <string> #include <vector> -#include "talk/p2p/base/basicpacketsocketfactory.h" -#include "talk/p2p/base/stun.h" -#include "talk/p2p/base/turnserver.h" +#include "webrtc/p2p/base/basicpacketsocketfactory.h" +#include "webrtc/p2p/base/stun.h" +#include "webrtc/p2p/base/turnserver.h" #include "webrtc/base/asyncudpsocket.h" #include "webrtc/base/thread.h" @@ -117,4 +117,4 @@ class TestTurnServer : public TurnAuthInterface { } // namespace cricket -#endif // TALK_P2P_BASE_TESTTURNSERVER_H_ +#endif // WEBRTC_P2P_BASE_TESTTURNSERVER_H_ diff --git a/p2p/base/transport.cc b/p2p/base/transport.cc index d88f5e7..f33f942 100644 --- a/p2p/base/transport.cc +++ b/p2p/base/transport.cc @@ -25,16 +25,16 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/p2p/base/transport.h" - -#include "talk/p2p/base/candidate.h" -#include "talk/p2p/base/constants.h" -#include "talk/p2p/base/parsing.h" -#include "talk/p2p/base/port.h" -#include "talk/p2p/base/sessionmanager.h" -#include "talk/p2p/base/transportchannelimpl.h" +#include "webrtc/p2p/base/transport.h" + +#include "webrtc/p2p/base/candidate.h" +#include "webrtc/p2p/base/constants.h" +#include "webrtc/p2p/base/parsing.h" +#include "webrtc/p2p/base/port.h" +#include "webrtc/p2p/base/sessionmanager.h" +#include "webrtc/p2p/base/transportchannelimpl.h" #include "webrtc/libjingle/xmllite/xmlelement.h" -#include "talk/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/constants.h" #include "webrtc/base/bind.h" #include "webrtc/base/common.h" #include "webrtc/base/logging.h" diff --git a/p2p/base/transport.h b/p2p/base/transport.h index 21661f6..fbc7f6f 100644 --- a/p2p/base/transport.h +++ b/p2p/base/transport.h @@ -43,16 +43,16 @@ // It is not possible to do so here because the subclass constructor will // already have run. -#ifndef TALK_P2P_BASE_TRANSPORT_H_ -#define TALK_P2P_BASE_TRANSPORT_H_ +#ifndef WEBRTC_P2P_BASE_TRANSPORT_H_ +#define WEBRTC_P2P_BASE_TRANSPORT_H_ #include <map> #include <string> #include <vector> -#include "talk/p2p/base/candidate.h" -#include "talk/p2p/base/constants.h" -#include "talk/p2p/base/sessiondescription.h" -#include "talk/p2p/base/transportinfo.h" +#include "webrtc/p2p/base/candidate.h" +#include "webrtc/p2p/base/constants.h" +#include "webrtc/p2p/base/sessiondescription.h" +#include "webrtc/p2p/base/transportinfo.h" #include "webrtc/base/criticalsection.h" #include "webrtc/base/messagequeue.h" #include "webrtc/base/sigslot.h" @@ -527,4 +527,4 @@ TransportProtocol TransportProtocolFromDescription( } // namespace cricket -#endif // TALK_P2P_BASE_TRANSPORT_H_ +#endif // WEBRTC_P2P_BASE_TRANSPORT_H_ diff --git a/p2p/base/transport_unittest.cc b/p2p/base/transport_unittest.cc index 8f7dae2..9ef7d60 100644 --- a/p2p/base/transport_unittest.cc +++ b/p2p/base/transport_unittest.cc @@ -25,14 +25,14 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/p2p/base/constants.h" -#include "talk/p2p/base/fakesession.h" -#include "talk/p2p/base/p2ptransport.h" -#include "talk/p2p/base/parsing.h" -#include "talk/p2p/base/rawtransport.h" -#include "talk/p2p/base/sessionmessages.h" +#include "webrtc/p2p/base/constants.h" +#include "webrtc/p2p/base/fakesession.h" +#include "webrtc/p2p/base/p2ptransport.h" +#include "webrtc/p2p/base/parsing.h" +#include "webrtc/p2p/base/rawtransport.h" +#include "webrtc/p2p/base/sessionmessages.h" #include "webrtc/libjingle/xmllite/xmlelement.h" -#include "talk/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/constants.h" #include "webrtc/base/fakesslidentity.h" #include "webrtc/base/gunit.h" #include "webrtc/base/thread.h" diff --git a/p2p/base/transportchannel.cc b/p2p/base/transportchannel.cc index 50ebfb9..13048ed 100644 --- a/p2p/base/transportchannel.cc +++ b/p2p/base/transportchannel.cc @@ -26,7 +26,7 @@ */ #include <sstream> -#include "talk/p2p/base/transportchannel.h" +#include "webrtc/p2p/base/transportchannel.h" namespace cricket { diff --git a/p2p/base/transportchannel.h b/p2p/base/transportchannel.h index b0beb04..56d9467 100644 --- a/p2p/base/transportchannel.h +++ b/p2p/base/transportchannel.h @@ -25,15 +25,15 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_BASE_TRANSPORTCHANNEL_H_ -#define TALK_P2P_BASE_TRANSPORTCHANNEL_H_ +#ifndef WEBRTC_P2P_BASE_TRANSPORTCHANNEL_H_ +#define WEBRTC_P2P_BASE_TRANSPORTCHANNEL_H_ #include <string> #include <vector> -#include "talk/p2p/base/candidate.h" -#include "talk/p2p/base/transport.h" -#include "talk/p2p/base/transportdescription.h" +#include "webrtc/p2p/base/candidate.h" +#include "webrtc/p2p/base/transport.h" +#include "webrtc/p2p/base/transportdescription.h" #include "webrtc/base/asyncpacketsocket.h" #include "webrtc/base/basictypes.h" #include "webrtc/base/dscp.h" @@ -157,4 +157,4 @@ class TransportChannel : public sigslot::has_slots<> { } // namespace cricket -#endif // TALK_P2P_BASE_TRANSPORTCHANNEL_H_ +#endif // WEBRTC_P2P_BASE_TRANSPORTCHANNEL_H_ diff --git a/p2p/base/transportchannelimpl.h b/p2p/base/transportchannelimpl.h index fde980b..73afdee 100644 --- a/p2p/base/transportchannelimpl.h +++ b/p2p/base/transportchannelimpl.h @@ -25,12 +25,12 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_BASE_TRANSPORTCHANNELIMPL_H_ -#define TALK_P2P_BASE_TRANSPORTCHANNELIMPL_H_ +#ifndef WEBRTC_P2P_BASE_TRANSPORTCHANNELIMPL_H_ +#define WEBRTC_P2P_BASE_TRANSPORTCHANNELIMPL_H_ #include <string> -#include "talk/p2p/base/transport.h" -#include "talk/p2p/base/transportchannel.h" +#include "webrtc/p2p/base/transport.h" +#include "webrtc/p2p/base/transportchannel.h" namespace buzz { class XmlElement; } @@ -125,4 +125,4 @@ class TransportChannelImpl : public TransportChannel { } // namespace cricket -#endif // TALK_P2P_BASE_TRANSPORTCHANNELIMPL_H_ +#endif // WEBRTC_P2P_BASE_TRANSPORTCHANNELIMPL_H_ diff --git a/p2p/base/transportchannelproxy.cc b/p2p/base/transportchannelproxy.cc index 9fa0b6a..259b8dd 100644 --- a/p2p/base/transportchannelproxy.cc +++ b/p2p/base/transportchannelproxy.cc @@ -25,9 +25,9 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/p2p/base/transport.h" -#include "talk/p2p/base/transportchannelimpl.h" -#include "talk/p2p/base/transportchannelproxy.h" +#include "webrtc/p2p/base/transport.h" +#include "webrtc/p2p/base/transportchannelimpl.h" +#include "webrtc/p2p/base/transportchannelproxy.h" #include "webrtc/base/common.h" #include "webrtc/base/logging.h" #include "webrtc/base/thread.h" diff --git a/p2p/base/transportchannelproxy.h b/p2p/base/transportchannelproxy.h index 07b9090..dee4313 100644 --- a/p2p/base/transportchannelproxy.h +++ b/p2p/base/transportchannelproxy.h @@ -25,14 +25,14 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_BASE_TRANSPORTCHANNELPROXY_H_ -#define TALK_P2P_BASE_TRANSPORTCHANNELPROXY_H_ +#ifndef WEBRTC_P2P_BASE_TRANSPORTCHANNELPROXY_H_ +#define WEBRTC_P2P_BASE_TRANSPORTCHANNELPROXY_H_ #include <string> #include <utility> #include <vector> -#include "talk/p2p/base/transportchannel.h" +#include "webrtc/p2p/base/transportchannel.h" #include "webrtc/base/messagehandler.h" namespace rtc { @@ -109,4 +109,4 @@ class TransportChannelProxy : public TransportChannel, } // namespace cricket -#endif // TALK_P2P_BASE_TRANSPORTCHANNELPROXY_H_ +#endif // WEBRTC_P2P_BASE_TRANSPORTCHANNELPROXY_H_ diff --git a/p2p/base/transportdescription.cc b/p2p/base/transportdescription.cc index 08460d7..8bdcbb3 100644 --- a/p2p/base/transportdescription.cc +++ b/p2p/base/transportdescription.cc @@ -25,9 +25,9 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/p2p/base/transportdescription.h" +#include "webrtc/p2p/base/transportdescription.h" -#include "talk/p2p/base/constants.h" +#include "webrtc/p2p/base/constants.h" #include "webrtc/base/stringutils.h" namespace cricket { diff --git a/p2p/base/transportdescription.h b/p2p/base/transportdescription.h index efd04ff..5763d23 100644 --- a/p2p/base/transportdescription.h +++ b/p2p/base/transportdescription.h @@ -25,15 +25,15 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_BASE_TRANSPORTDESCRIPTION_H_ -#define TALK_P2P_BASE_TRANSPORTDESCRIPTION_H_ +#ifndef WEBRTC_P2P_BASE_TRANSPORTDESCRIPTION_H_ +#define WEBRTC_P2P_BASE_TRANSPORTDESCRIPTION_H_ #include <algorithm> #include <string> #include <vector> -#include "talk/p2p/base/candidate.h" -#include "talk/p2p/base/constants.h" +#include "webrtc/p2p/base/candidate.h" +#include "webrtc/p2p/base/constants.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/base/sslfingerprint.h" @@ -185,4 +185,4 @@ struct TransportDescription { } // namespace cricket -#endif // TALK_P2P_BASE_TRANSPORTDESCRIPTION_H_ +#endif // WEBRTC_P2P_BASE_TRANSPORTDESCRIPTION_H_ diff --git a/p2p/base/transportdescriptionfactory.cc b/p2p/base/transportdescriptionfactory.cc index 4eb603c..7260b9c 100644 --- a/p2p/base/transportdescriptionfactory.cc +++ b/p2p/base/transportdescriptionfactory.cc @@ -25,9 +25,9 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/p2p/base/transportdescriptionfactory.h" +#include "webrtc/p2p/base/transportdescriptionfactory.h" -#include "talk/p2p/base/transportdescription.h" +#include "webrtc/p2p/base/transportdescription.h" #include "webrtc/base/helpers.h" #include "webrtc/base/logging.h" #include "webrtc/base/messagedigest.h" diff --git a/p2p/base/transportdescriptionfactory.h b/p2p/base/transportdescriptionfactory.h index 84f25ac..bcd14c6 100644 --- a/p2p/base/transportdescriptionfactory.h +++ b/p2p/base/transportdescriptionfactory.h @@ -25,10 +25,10 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_BASE_TRANSPORTDESCRIPTIONFACTORY_H_ -#define TALK_P2P_BASE_TRANSPORTDESCRIPTIONFACTORY_H_ +#ifndef WEBRTC_P2P_BASE_TRANSPORTDESCRIPTIONFACTORY_H_ +#define WEBRTC_P2P_BASE_TRANSPORTDESCRIPTIONFACTORY_H_ -#include "talk/p2p/base/transportdescription.h" +#include "webrtc/p2p/base/transportdescription.h" namespace rtc { class SSLIdentity; @@ -80,4 +80,4 @@ class TransportDescriptionFactory { } // namespace cricket -#endif // TALK_P2P_BASE_TRANSPORTDESCRIPTIONFACTORY_H_ +#endif // WEBRTC_P2P_BASE_TRANSPORTDESCRIPTIONFACTORY_H_ diff --git a/p2p/base/transportdescriptionfactory_unittest.cc b/p2p/base/transportdescriptionfactory_unittest.cc index 5c17f96..a793853 100644 --- a/p2p/base/transportdescriptionfactory_unittest.cc +++ b/p2p/base/transportdescriptionfactory_unittest.cc @@ -28,9 +28,9 @@ #include <string> #include <vector> -#include "talk/p2p/base/constants.h" -#include "talk/p2p/base/transportdescription.h" -#include "talk/p2p/base/transportdescriptionfactory.h" +#include "webrtc/p2p/base/constants.h" +#include "webrtc/p2p/base/transportdescription.h" +#include "webrtc/p2p/base/transportdescriptionfactory.h" #include "webrtc/base/fakesslidentity.h" #include "webrtc/base/gunit.h" #include "webrtc/base/ssladapter.h" diff --git a/p2p/base/transportinfo.h b/p2p/base/transportinfo.h index 5ffb10a..414b6ca 100644 --- a/p2p/base/transportinfo.h +++ b/p2p/base/transportinfo.h @@ -25,15 +25,15 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_BASE_TRANSPORTINFO_H_ -#define TALK_P2P_BASE_TRANSPORTINFO_H_ +#ifndef WEBRTC_P2P_BASE_TRANSPORTINFO_H_ +#define WEBRTC_P2P_BASE_TRANSPORTINFO_H_ #include <string> #include <vector> -#include "talk/p2p/base/candidate.h" -#include "talk/p2p/base/constants.h" -#include "talk/p2p/base/transportdescription.h" +#include "webrtc/p2p/base/candidate.h" +#include "webrtc/p2p/base/constants.h" +#include "webrtc/p2p/base/transportdescription.h" #include "webrtc/base/helpers.h" namespace cricket { @@ -57,4 +57,4 @@ typedef std::vector<TransportInfo> TransportInfos; } // namespace cricket -#endif // TALK_P2P_BASE_TRANSPORTINFO_H_ +#endif // WEBRTC_P2P_BASE_TRANSPORTINFO_H_ diff --git a/p2p/base/turnport.cc b/p2p/base/turnport.cc index 2908d71..b38317b 100644 --- a/p2p/base/turnport.cc +++ b/p2p/base/turnport.cc @@ -25,12 +25,12 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/p2p/base/turnport.h" +#include "webrtc/p2p/base/turnport.h" #include <functional> -#include "talk/p2p/base/common.h" -#include "talk/p2p/base/stun.h" +#include "webrtc/p2p/base/common.h" +#include "webrtc/p2p/base/stun.h" #include "webrtc/base/asyncpacketsocket.h" #include "webrtc/base/byteorder.h" #include "webrtc/base/common.h" @@ -588,10 +588,18 @@ void TurnPort::OnStunAddress(const rtc::SocketAddress& address) { void TurnPort::OnAllocateSuccess(const rtc::SocketAddress& address, const rtc::SocketAddress& stun_address) { connected_ = true; + + rtc::SocketAddress related_address = stun_address; + if (!(candidate_filter() & CF_REFLEXIVE)) { + // If candidate filter only allows relay type of address, empty raddr to + // avoid local address leakage. + related_address = rtc::EmptySocketAddressWithFamily(stun_address.family()); + } + // For relayed candidate, Base is the candidate itself. - AddAddress(address, // Candidate address. - address, // Base address. - stun_address, // Related address. + AddAddress(address, // Candidate address. + address, // Base address. + related_address, // Related address. UDP_PROTOCOL_NAME, "", // TCP canddiate type, empty for turn candidates. RELAY_PORT_TYPE, diff --git a/p2p/base/turnport.h b/p2p/base/turnport.h index ab7d4e7..6345aea 100644 --- a/p2p/base/turnport.h +++ b/p2p/base/turnport.h @@ -25,16 +25,16 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_BASE_TURNPORT_H_ -#define TALK_P2P_BASE_TURNPORT_H_ +#ifndef WEBRTC_P2P_BASE_TURNPORT_H_ +#define WEBRTC_P2P_BASE_TURNPORT_H_ #include <stdio.h> #include <list> #include <set> #include <string> -#include "talk/p2p/base/port.h" -#include "talk/p2p/client/basicportallocator.h" +#include "webrtc/p2p/base/port.h" +#include "webrtc/p2p/client/basicportallocator.h" #include "webrtc/base/asyncpacketsocket.h" namespace rtc { @@ -251,4 +251,4 @@ class TurnPort : public Port { } // namespace cricket -#endif // TALK_P2P_BASE_TURNPORT_H_ +#endif // WEBRTC_P2P_BASE_TURNPORT_H_ diff --git a/p2p/base/turnport_unittest.cc b/p2p/base/turnport_unittest.cc index c5261e4..95615ff 100644 --- a/p2p/base/turnport_unittest.cc +++ b/p2p/base/turnport_unittest.cc @@ -28,12 +28,12 @@ #include <dirent.h> #endif -#include "talk/p2p/base/basicpacketsocketfactory.h" -#include "talk/p2p/base/constants.h" -#include "talk/p2p/base/tcpport.h" -#include "talk/p2p/base/testturnserver.h" -#include "talk/p2p/base/turnport.h" -#include "talk/p2p/base/udpport.h" +#include "webrtc/p2p/base/basicpacketsocketfactory.h" +#include "webrtc/p2p/base/constants.h" +#include "webrtc/p2p/base/tcpport.h" +#include "webrtc/p2p/base/testturnserver.h" +#include "webrtc/p2p/base/turnport.h" +#include "webrtc/p2p/base/udpport.h" #include "webrtc/base/asynctcpsocket.h" #include "webrtc/base/buffer.h" #include "webrtc/base/dscp.h" diff --git a/p2p/base/turnserver.cc b/p2p/base/turnserver.cc index dbcbcd4..28096ae 100644 --- a/p2p/base/turnserver.cc +++ b/p2p/base/turnserver.cc @@ -25,12 +25,12 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/p2p/base/turnserver.h" +#include "webrtc/p2p/base/turnserver.h" -#include "talk/p2p/base/asyncstuntcpsocket.h" -#include "talk/p2p/base/common.h" -#include "talk/p2p/base/packetsocketfactory.h" -#include "talk/p2p/base/stun.h" +#include "webrtc/p2p/base/asyncstuntcpsocket.h" +#include "webrtc/p2p/base/common.h" +#include "webrtc/p2p/base/packetsocketfactory.h" +#include "webrtc/p2p/base/stun.h" #include "webrtc/base/bytebuffer.h" #include "webrtc/base/helpers.h" #include "webrtc/base/logging.h" diff --git a/p2p/base/turnserver.h b/p2p/base/turnserver.h index 553d00c..44878fd 100644 --- a/p2p/base/turnserver.h +++ b/p2p/base/turnserver.h @@ -25,15 +25,15 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_BASE_TURNSERVER_H_ -#define TALK_P2P_BASE_TURNSERVER_H_ +#ifndef WEBRTC_P2P_BASE_TURNSERVER_H_ +#define WEBRTC_P2P_BASE_TURNSERVER_H_ #include <list> #include <map> #include <set> #include <string> -#include "talk/p2p/base/portinterface.h" +#include "webrtc/p2p/base/portinterface.h" #include "webrtc/base/asyncpacketsocket.h" #include "webrtc/base/messagequeue.h" #include "webrtc/base/sigslot.h" @@ -204,4 +204,4 @@ class TurnServer : public sigslot::has_slots<> { } // namespace cricket -#endif // TALK_P2P_BASE_TURNSERVER_H_ +#endif // WEBRTC_P2P_BASE_TURNSERVER_H_ diff --git a/p2p/base/udpport.h b/p2p/base/udpport.h index fc981fd..35be1a1 100644 --- a/p2p/base/udpport.h +++ b/p2p/base/udpport.h @@ -25,10 +25,10 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_BASE_UDPPORT_H_ -#define TALK_P2P_BASE_UDPPORT_H_ +#ifndef WEBRTC_P2P_BASE_UDPPORT_H_ +#define WEBRTC_P2P_BASE_UDPPORT_H_ // StunPort will be handling UDPPort functionality. -#include "talk/p2p/base/stunport.h" +#include "webrtc/p2p/base/stunport.h" -#endif // TALK_P2P_BASE_UDPPORT_H_ +#endif // WEBRTC_P2P_BASE_UDPPORT_H_ diff --git a/p2p/client/autoportallocator.h b/p2p/client/autoportallocator.h index ed87162..e7f2169 100644 --- a/p2p/client/autoportallocator.h +++ b/p2p/client/autoportallocator.h @@ -25,15 +25,15 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_CLIENT_AUTOPORTALLOCATOR_H_ -#define TALK_P2P_CLIENT_AUTOPORTALLOCATOR_H_ +#ifndef WEBRTC_P2P_CLIENT_AUTOPORTALLOCATOR_H_ +#define WEBRTC_P2P_CLIENT_AUTOPORTALLOCATOR_H_ #include <string> #include <vector> -#include "talk/p2p/client/httpportallocator.h" -#include "talk/xmpp/jingleinfotask.h" -#include "talk/xmpp/xmppclient.h" +#include "webrtc/p2p/client/httpportallocator.h" +#include "webrtc/libjingle/xmpp/jingleinfotask.h" +#include "webrtc/libjingle/xmpp/xmppclient.h" #include "webrtc/base/sigslot.h" // This class sets the relay and stun servers using XmppClient. @@ -66,4 +66,4 @@ class AutoPortAllocator : public cricket::HttpPortAllocator { } }; -#endif // TALK_P2P_CLIENT_AUTOPORTALLOCATOR_H_ +#endif // WEBRTC_P2P_CLIENT_AUTOPORTALLOCATOR_H_ diff --git a/p2p/client/basicportallocator.cc b/p2p/client/basicportallocator.cc index 0ec13e7..27d0db2 100644 --- a/p2p/client/basicportallocator.cc +++ b/p2p/client/basicportallocator.cc @@ -25,19 +25,19 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/p2p/client/basicportallocator.h" +#include "webrtc/p2p/client/basicportallocator.h" #include <string> #include <vector> -#include "talk/p2p/base/basicpacketsocketfactory.h" -#include "talk/p2p/base/common.h" -#include "talk/p2p/base/port.h" -#include "talk/p2p/base/relayport.h" -#include "talk/p2p/base/stunport.h" -#include "talk/p2p/base/tcpport.h" -#include "talk/p2p/base/turnport.h" -#include "talk/p2p/base/udpport.h" +#include "webrtc/p2p/base/basicpacketsocketfactory.h" +#include "webrtc/p2p/base/common.h" +#include "webrtc/p2p/base/port.h" +#include "webrtc/p2p/base/relayport.h" +#include "webrtc/p2p/base/stunport.h" +#include "webrtc/p2p/base/tcpport.h" +#include "webrtc/p2p/base/turnport.h" +#include "webrtc/p2p/base/udpport.h" #include "webrtc/base/common.h" #include "webrtc/base/helpers.h" #include "webrtc/base/logging.h" @@ -497,6 +497,9 @@ void BasicPortAllocatorSession::AddAllocatedPort(Port* port, port->set_send_retransmit_count_attribute((allocator_->flags() & PORTALLOCATOR_ENABLE_STUN_RETRANSMIT_ATTRIBUTE) != 0); + // Push down the candidate_filter to individual port. + port->set_candidate_filter(allocator_->candidate_filter()); + PortData data(port, seq); ports_.push_back(data); diff --git a/p2p/client/basicportallocator.h b/p2p/client/basicportallocator.h index d424772..e2389d3 100644 --- a/p2p/client/basicportallocator.h +++ b/p2p/client/basicportallocator.h @@ -25,14 +25,14 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_CLIENT_BASICPORTALLOCATOR_H_ -#define TALK_P2P_CLIENT_BASICPORTALLOCATOR_H_ +#ifndef WEBRTC_P2P_CLIENT_BASICPORTALLOCATOR_H_ +#define WEBRTC_P2P_CLIENT_BASICPORTALLOCATOR_H_ #include <string> #include <vector> -#include "talk/p2p/base/port.h" -#include "talk/p2p/base/portallocator.h" +#include "webrtc/p2p/base/port.h" +#include "webrtc/p2p/base/portallocator.h" #include "webrtc/base/messagequeue.h" #include "webrtc/base/network.h" #include "webrtc/base/scoped_ptr.h" @@ -255,4 +255,4 @@ struct PortConfiguration : public rtc::MessageData { } // namespace cricket -#endif // TALK_P2P_CLIENT_BASICPORTALLOCATOR_H_ +#endif // WEBRTC_P2P_CLIENT_BASICPORTALLOCATOR_H_ diff --git a/p2p/client/connectivitychecker.cc b/p2p/client/connectivitychecker.cc index 723c5a1..38a7dd2 100644 --- a/p2p/client/connectivitychecker.cc +++ b/p2p/client/connectivitychecker.cc @@ -27,14 +27,14 @@ #include <string> -#include "talk/p2p/client/connectivitychecker.h" - -#include "talk/p2p/base/candidate.h" -#include "talk/p2p/base/common.h" -#include "talk/p2p/base/constants.h" -#include "talk/p2p/base/port.h" -#include "talk/p2p/base/relayport.h" -#include "talk/p2p/base/stunport.h" +#include "webrtc/p2p/client/connectivitychecker.h" + +#include "webrtc/p2p/base/candidate.h" +#include "webrtc/p2p/base/common.h" +#include "webrtc/p2p/base/constants.h" +#include "webrtc/p2p/base/port.h" +#include "webrtc/p2p/base/relayport.h" +#include "webrtc/p2p/base/stunport.h" #include "webrtc/base/asynchttprequest.h" #include "webrtc/base/autodetectproxy.h" #include "webrtc/base/helpers.h" diff --git a/p2p/client/connectivitychecker.h b/p2p/client/connectivitychecker.h index d4cda1e..ba3d003 100644 --- a/p2p/client/connectivitychecker.h +++ b/p2p/client/connectivitychecker.h @@ -1,14 +1,14 @@ // Copyright 2011 Google Inc. All Rights Reserved. -#ifndef TALK_P2P_CLIENT_CONNECTIVITYCHECKER_H_ -#define TALK_P2P_CLIENT_CONNECTIVITYCHECKER_H_ +#ifndef WEBRTC_P2P_CLIENT_CONNECTIVITYCHECKER_H_ +#define WEBRTC_P2P_CLIENT_CONNECTIVITYCHECKER_H_ #include <map> #include <string> -#include "talk/p2p/base/basicpacketsocketfactory.h" -#include "talk/p2p/client/httpportallocator.h" +#include "webrtc/p2p/base/basicpacketsocketfactory.h" +#include "webrtc/p2p/client/httpportallocator.h" #include "webrtc/base/basictypes.h" #include "webrtc/base/messagehandler.h" #include "webrtc/base/network.h" @@ -271,4 +271,4 @@ class ConnectivityChecker } // namespace cricket -#endif // TALK_P2P_CLIENT_CONNECTIVITYCHECKER_H_ +#endif // WEBRTC_P2P_CLIENT_CONNECTIVITYCHECKER_H_ diff --git a/p2p/client/connectivitychecker_unittest.cc b/p2p/client/connectivitychecker_unittest.cc index 187505a..5a2a608 100644 --- a/p2p/client/connectivitychecker_unittest.cc +++ b/p2p/client/connectivitychecker_unittest.cc @@ -27,11 +27,11 @@ #include <string> -#include "talk/p2p/base/basicpacketsocketfactory.h" -#include "talk/p2p/base/relayport.h" -#include "talk/p2p/base/stunport.h" -#include "talk/p2p/client/connectivitychecker.h" -#include "talk/p2p/client/httpportallocator.h" +#include "webrtc/p2p/base/basicpacketsocketfactory.h" +#include "webrtc/p2p/base/relayport.h" +#include "webrtc/p2p/base/stunport.h" +#include "webrtc/p2p/client/connectivitychecker.h" +#include "webrtc/p2p/client/httpportallocator.h" #include "webrtc/base/asynchttprequest.h" #include "webrtc/base/fakenetwork.h" #include "webrtc/base/gunit.h" diff --git a/p2p/client/fakeportallocator.h b/p2p/client/fakeportallocator.h index e1a04dd..04c5f47 100644 --- a/p2p/client/fakeportallocator.h +++ b/p2p/client/fakeportallocator.h @@ -25,13 +25,13 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_CLIENT_FAKEPORTALLOCATOR_H_ -#define TALK_P2P_CLIENT_FAKEPORTALLOCATOR_H_ +#ifndef WEBRTC_P2P_CLIENT_FAKEPORTALLOCATOR_H_ +#define WEBRTC_P2P_CLIENT_FAKEPORTALLOCATOR_H_ #include <string> -#include "talk/p2p/base/basicpacketsocketfactory.h" -#include "talk/p2p/base/portallocator.h" -#include "talk/p2p/base/udpport.h" +#include "webrtc/p2p/base/basicpacketsocketfactory.h" +#include "webrtc/p2p/base/portallocator.h" +#include "webrtc/p2p/base/udpport.h" #include "webrtc/base/scoped_ptr.h" namespace rtc { @@ -135,4 +135,4 @@ class FakePortAllocator : public cricket::PortAllocator { } // namespace cricket -#endif // TALK_P2P_CLIENT_FAKEPORTALLOCATOR_H_ +#endif // WEBRTC_P2P_CLIENT_FAKEPORTALLOCATOR_H_ diff --git a/p2p/client/httpportallocator.cc b/p2p/client/httpportallocator.cc index 31c9b51..ce52593 100644 --- a/p2p/client/httpportallocator.cc +++ b/p2p/client/httpportallocator.cc @@ -25,7 +25,7 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/p2p/client/httpportallocator.h" +#include "webrtc/p2p/client/httpportallocator.h" #include <algorithm> #include <map> diff --git a/p2p/client/httpportallocator.h b/p2p/client/httpportallocator.h index 7ace943..cab8e7e 100644 --- a/p2p/client/httpportallocator.h +++ b/p2p/client/httpportallocator.h @@ -25,14 +25,14 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_CLIENT_HTTPPORTALLOCATOR_H_ -#define TALK_P2P_CLIENT_HTTPPORTALLOCATOR_H_ +#ifndef WEBRTC_P2P_CLIENT_HTTPPORTALLOCATOR_H_ +#define WEBRTC_P2P_CLIENT_HTTPPORTALLOCATOR_H_ #include <list> #include <string> #include <vector> -#include "talk/p2p/client/basicportallocator.h" +#include "webrtc/p2p/client/basicportallocator.h" class HttpPortAllocatorTest_TestSessionRequestUrl_Test; @@ -187,4 +187,4 @@ class HttpPortAllocatorSession : public HttpPortAllocatorSessionBase { } // namespace cricket -#endif // TALK_P2P_CLIENT_HTTPPORTALLOCATOR_H_ +#endif // WEBRTC_P2P_CLIENT_HTTPPORTALLOCATOR_H_ diff --git a/p2p/client/portallocator_unittest.cc b/p2p/client/portallocator_unittest.cc index 84b8d03..5b1a869 100644 --- a/p2p/client/portallocator_unittest.cc +++ b/p2p/client/portallocator_unittest.cc @@ -25,15 +25,15 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/p2p/base/basicpacketsocketfactory.h" -#include "talk/p2p/base/constants.h" -#include "talk/p2p/base/p2ptransportchannel.h" -#include "talk/p2p/base/portallocatorsessionproxy.h" -#include "talk/p2p/base/testrelayserver.h" -#include "talk/p2p/base/teststunserver.h" -#include "talk/p2p/base/testturnserver.h" -#include "talk/p2p/client/basicportallocator.h" -#include "talk/p2p/client/httpportallocator.h" +#include "webrtc/p2p/base/basicpacketsocketfactory.h" +#include "webrtc/p2p/base/constants.h" +#include "webrtc/p2p/base/p2ptransportchannel.h" +#include "webrtc/p2p/base/portallocatorsessionproxy.h" +#include "webrtc/p2p/base/testrelayserver.h" +#include "webrtc/p2p/base/teststunserver.h" +#include "webrtc/p2p/base/testturnserver.h" +#include "webrtc/p2p/client/basicportallocator.h" +#include "webrtc/p2p/client/httpportallocator.h" #include "webrtc/base/fakenetwork.h" #include "webrtc/base/firewallsocketserver.h" #include "webrtc/base/gunit.h" @@ -113,6 +113,8 @@ class PortAllocatorTest : public testing::Test, public sigslot::has_slots<> { candidate_allocation_done_(false) { cricket::ServerAddresses stun_servers; stun_servers.insert(kStunAddr); + // Passing the addresses of GTURN servers will enable GTURN in + // Basicportallocator. allocator_.reset(new cricket::BasicPortAllocator( &network_manager_, stun_servers, @@ -137,6 +139,14 @@ class PortAllocatorTest : public testing::Test, public sigslot::has_slots<> { allocator().set_step_delay(cricket::kMinimumStepDelay); } + // Create a BasicPortAllocator without GTURN and add the TURN servers. + void ResetWithTurnServers(const rtc::SocketAddress& udp_turn, + const rtc::SocketAddress& tcp_turn) { + allocator_.reset(new cricket::BasicPortAllocator(&network_manager_)); + allocator().set_step_delay(cricket::kMinimumStepDelay); + AddTurnServers(udp_turn, tcp_turn); + } + void AddTurnServers(const rtc::SocketAddress& udp_turn, const rtc::SocketAddress& tcp_turn) { cricket::RelayServerConfig relay_server(cricket::RELAY_TURN); @@ -565,17 +575,32 @@ TEST_F(PortAllocatorTest, TestGetAllPortsRestarts) { } // Test ICE candidate filter mechanism with options Relay/Host/Reflexive. +// This test also verifies that when the allocator is only allowed to use +// relay (i.e. IceTransportsType is relay), the raddr is an empty +// address with the correct family. This is to prevent any local +// reflective address leakage in the sdp line. TEST_F(PortAllocatorTest, TestCandidateFilterWithRelayOnly) { AddInterface(kClientAddr); + // GTURN is not configured here. + ResetWithTurnServers(kTurnUdpIntAddr, rtc::SocketAddress()); allocator().set_candidate_filter(cricket::CF_RELAY); EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); EXPECT_TRUE_WAIT(candidate_allocation_done_, kDefaultAllocationTimeout); - // Using GTURN, we will have 4 candidates. - EXPECT_EQ(4U, candidates_.size()); + EXPECT_PRED5(CheckCandidate, + candidates_[0], + cricket::ICE_CANDIDATE_COMPONENT_RTP, + "relay", + "udp", + rtc::SocketAddress(kTurnUdpExtAddr.ipaddr(), 0)); + + EXPECT_EQ(1U, candidates_.size()); EXPECT_EQ(1U, ports_.size()); // Only Relay port will be in ready state. for (size_t i = 0; i < candidates_.size(); ++i) { EXPECT_EQ(std::string(cricket::RELAY_PORT_TYPE), candidates_[i].type()); + EXPECT_EQ( + candidates_[0].related_address(), + rtc::EmptySocketAddressWithFamily(candidates_[0].address().family())); } } @@ -611,6 +636,9 @@ TEST_F(PortAllocatorTest, TestCandidateFilterWithReflexiveOnly) { EXPECT_EQ(1U, ports_.size()); // Only UDP port will be in ready state. for (size_t i = 0; i < candidates_.size(); ++i) { EXPECT_EQ(std::string(cricket::STUN_PORT_TYPE), candidates_[i].type()); + EXPECT_EQ( + candidates_[0].related_address(), + rtc::EmptySocketAddressWithFamily(candidates_[0].address().family())); } } diff --git a/p2p/client/sessionmanagertask.h b/p2p/client/sessionmanagertask.h index e16d9d6..753d938 100644 --- a/p2p/client/sessionmanagertask.h +++ b/p2p/client/sessionmanagertask.h @@ -25,13 +25,13 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_CLIENT_SESSIONMANAGERTASK_H_ -#define TALK_P2P_CLIENT_SESSIONMANAGERTASK_H_ +#ifndef WEBRTC_P2P_CLIENT_SESSIONMANAGERTASK_H_ +#define WEBRTC_P2P_CLIENT_SESSIONMANAGERTASK_H_ -#include "talk/p2p/base/sessionmanager.h" -#include "talk/p2p/client/sessionsendtask.h" -#include "talk/xmpp/xmppengine.h" -#include "talk/xmpp/xmpptask.h" +#include "webrtc/p2p/base/sessionmanager.h" +#include "webrtc/p2p/client/sessionsendtask.h" +#include "webrtc/libjingle/xmpp/xmppengine.h" +#include "webrtc/libjingle/xmpp/xmpptask.h" namespace cricket { @@ -90,4 +90,4 @@ class SessionManagerTask : public buzz::XmppTask { } // namespace cricket -#endif // TALK_P2P_CLIENT_SESSIONMANAGERTASK_H_ +#endif // WEBRTC_P2P_CLIENT_SESSIONMANAGERTASK_H_ diff --git a/p2p/client/sessionsendtask.h b/p2p/client/sessionsendtask.h index c8734d2..7dafc1e 100644 --- a/p2p/client/sessionsendtask.h +++ b/p2p/client/sessionsendtask.h @@ -25,14 +25,14 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_CLIENT_SESSIONSENDTASK_H_ -#define TALK_P2P_CLIENT_SESSIONSENDTASK_H_ - -#include "talk/p2p/base/sessionmanager.h" -#include "talk/xmpp/constants.h" -#include "talk/xmpp/xmppclient.h" -#include "talk/xmpp/xmppengine.h" -#include "talk/xmpp/xmpptask.h" +#ifndef WEBRTC_P2P_CLIENT_SESSIONSENDTASK_H_ +#define WEBRTC_P2P_CLIENT_SESSIONSENDTASK_H_ + +#include "webrtc/p2p/base/sessionmanager.h" +#include "webrtc/libjingle/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/xmppclient.h" +#include "webrtc/libjingle/xmpp/xmppengine.h" +#include "webrtc/libjingle/xmpp/xmpptask.h" #include "webrtc/base/common.h" namespace cricket { @@ -142,4 +142,4 @@ class SessionSendTask : public buzz::XmppTask { } -#endif // TALK_P2P_CLIENT_SESSIONSENDTASK_H_ +#endif // WEBRTC_P2P_CLIENT_SESSIONSENDTASK_H_ diff --git a/p2p/client/socketmonitor.cc b/p2p/client/socketmonitor.cc index 1924c70..a3a33e3 100644 --- a/p2p/client/socketmonitor.cc +++ b/p2p/client/socketmonitor.cc @@ -25,7 +25,7 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/p2p/client/socketmonitor.h" +#include "webrtc/p2p/client/socketmonitor.h" #include "webrtc/base/common.h" diff --git a/p2p/client/socketmonitor.h b/p2p/client/socketmonitor.h index 87b11cf..77241fe 100644 --- a/p2p/client/socketmonitor.h +++ b/p2p/client/socketmonitor.h @@ -25,12 +25,12 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_P2P_CLIENT_SOCKETMONITOR_H_ -#define TALK_P2P_CLIENT_SOCKETMONITOR_H_ +#ifndef WEBRTC_P2P_CLIENT_SOCKETMONITOR_H_ +#define WEBRTC_P2P_CLIENT_SOCKETMONITOR_H_ #include <vector> -#include "talk/p2p/base/transportchannel.h" +#include "webrtc/p2p/base/transportchannel.h" #include "webrtc/base/criticalsection.h" #include "webrtc/base/sigslot.h" #include "webrtc/base/thread.h" @@ -68,4 +68,4 @@ class SocketMonitor : public rtc::MessageHandler, } // namespace cricket -#endif // TALK_P2P_CLIENT_SOCKETMONITOR_H_ +#endif // WEBRTC_P2P_CLIENT_SOCKETMONITOR_H_ diff --git a/session/media/audiomonitor.h b/session/media/audiomonitor.h index e5585a4..454f0fe 100644 --- a/session/media/audiomonitor.h +++ b/session/media/audiomonitor.h @@ -29,7 +29,7 @@ #define TALK_SESSION_MEDIA_AUDIOMONITOR_H_ #include <vector> -#include "talk/p2p/base/port.h" +#include "webrtc/p2p/base/port.h" #include "webrtc/base/sigslot.h" #include "webrtc/base/thread.h" diff --git a/session/media/call.cc b/session/media/call.cc index ad28130..7e59ac6 100644 --- a/session/media/call.cc +++ b/session/media/call.cc @@ -28,7 +28,7 @@ #include <string> #include "talk/media/base/constants.h" #include "talk/media/base/screencastid.h" -#include "talk/p2p/base/parsing.h" +#include "webrtc/p2p/base/parsing.h" #include "talk/session/media/call.h" #include "talk/session/media/currentspeakermonitor.h" #include "talk/session/media/mediasessionclient.h" diff --git a/session/media/call.h b/session/media/call.h index be9397e..5a2d49b 100644 --- a/session/media/call.h +++ b/session/media/call.h @@ -37,13 +37,13 @@ #include "talk/media/base/screencastid.h" #include "talk/media/base/streamparams.h" #include "talk/media/base/videocommon.h" -#include "talk/p2p/base/session.h" -#include "talk/p2p/client/socketmonitor.h" +#include "webrtc/p2p/base/session.h" +#include "webrtc/p2p/client/socketmonitor.h" #include "talk/session/media/audiomonitor.h" #include "talk/session/media/currentspeakermonitor.h" #include "talk/session/media/mediamessages.h" #include "talk/session/media/mediasession.h" -#include "talk/xmpp/jid.h" +#include "webrtc/libjingle/xmpp/jid.h" #include "webrtc/base/messagequeue.h" namespace cricket { diff --git a/session/media/channel.cc b/session/media/channel.cc index 3615da8..414a131 100644 --- a/session/media/channel.cc +++ b/session/media/channel.cc @@ -29,7 +29,7 @@ #include "talk/media/base/constants.h" #include "talk/media/base/rtputils.h" -#include "talk/p2p/base/transportchannel.h" +#include "webrtc/p2p/base/transportchannel.h" #include "talk/session/media/channelmanager.h" #include "talk/session/media/mediamessages.h" #include "talk/session/media/typingmonitor.h" diff --git a/session/media/channel.h b/session/media/channel.h index 6da29a4..29980bb 100644 --- a/session/media/channel.h +++ b/session/media/channel.h @@ -35,8 +35,8 @@ #include "talk/media/base/mediaengine.h" #include "talk/media/base/streamparams.h" #include "talk/media/base/videocapturer.h" -#include "talk/p2p/base/session.h" -#include "talk/p2p/client/socketmonitor.h" +#include "webrtc/p2p/base/session.h" +#include "webrtc/p2p/client/socketmonitor.h" #include "talk/session/media/audiomonitor.h" #include "talk/session/media/bundlefilter.h" #include "talk/session/media/mediamonitor.h" diff --git a/session/media/channel_unittest.cc b/session/media/channel_unittest.cc index 698a34a..0fba14e 100644 --- a/session/media/channel_unittest.cc +++ b/session/media/channel_unittest.cc @@ -31,7 +31,7 @@ #include "talk/media/base/rtpdump.h" #include "talk/media/base/screencastid.h" #include "talk/media/base/testutils.h" -#include "talk/p2p/base/fakesession.h" +#include "webrtc/p2p/base/fakesession.h" #include "talk/session/media/channel.h" #include "talk/session/media/mediamessages.h" #include "talk/session/media/mediarecorder.h" diff --git a/session/media/channelmanager.h b/session/media/channelmanager.h index 98a3c96..764451d 100644 --- a/session/media/channelmanager.h +++ b/session/media/channelmanager.h @@ -33,7 +33,7 @@ #include "talk/media/base/capturemanager.h" #include "talk/media/base/mediaengine.h" -#include "talk/p2p/base/session.h" +#include "webrtc/p2p/base/session.h" #include "talk/session/media/voicechannel.h" #include "webrtc/base/criticalsection.h" #include "webrtc/base/fileutils.h" diff --git a/session/media/channelmanager_unittest.cc b/session/media/channelmanager_unittest.cc index 1363364..4c6f4ab 100644 --- a/session/media/channelmanager_unittest.cc +++ b/session/media/channelmanager_unittest.cc @@ -29,7 +29,7 @@ #include "talk/media/base/nullvideorenderer.h" #include "talk/media/base/testutils.h" #include "talk/media/devices/fakedevicemanager.h" -#include "talk/p2p/base/fakesession.h" +#include "webrtc/p2p/base/fakesession.h" #include "talk/session/media/channelmanager.h" #include "webrtc/base/gunit.h" #include "webrtc/base/logging.h" diff --git a/session/media/mediamessages.cc b/session/media/mediamessages.cc index 6c9f681..011fd7d 100644 --- a/session/media/mediamessages.cc +++ b/session/media/mediamessages.cc @@ -31,8 +31,8 @@ #include "talk/session/media/mediamessages.h" -#include "talk/p2p/base/constants.h" -#include "talk/p2p/base/parsing.h" +#include "webrtc/p2p/base/constants.h" +#include "webrtc/p2p/base/parsing.h" #include "talk/session/media/mediasessionclient.h" #include "webrtc/libjingle/xmllite/xmlelement.h" #include "webrtc/base/logging.h" diff --git a/session/media/mediamessages.h b/session/media/mediamessages.h index 4a56e4b..d918e3e 100644 --- a/session/media/mediamessages.h +++ b/session/media/mediamessages.h @@ -41,8 +41,8 @@ #include "talk/media/base/mediachannel.h" // For RtpHeaderExtension #include "talk/media/base/streamparams.h" -#include "talk/p2p/base/parsing.h" -#include "talk/p2p/base/sessiondescription.h" +#include "webrtc/p2p/base/parsing.h" +#include "webrtc/p2p/base/sessiondescription.h" #include "webrtc/base/basictypes.h" namespace cricket { diff --git a/session/media/mediamessages_unittest.cc b/session/media/mediamessages_unittest.cc index 9ebd38c..a9f00e4 100644 --- a/session/media/mediamessages_unittest.cc +++ b/session/media/mediamessages_unittest.cc @@ -30,7 +30,7 @@ #include <string> #include <vector> -#include "talk/p2p/base/constants.h" +#include "webrtc/p2p/base/constants.h" #include "talk/session/media/mediasessionclient.h" #include "webrtc/libjingle/xmllite/xmlelement.h" #include "webrtc/base/gunit.h" diff --git a/session/media/mediarecorder_unittest.cc b/session/media/mediarecorder_unittest.cc index 9feb22a..b71a984 100644 --- a/session/media/mediarecorder_unittest.cc +++ b/session/media/mediarecorder_unittest.cc @@ -28,7 +28,7 @@ #include "talk/media/base/fakemediaengine.h" #include "talk/media/base/rtpdump.h" #include "talk/media/base/testutils.h" -#include "talk/p2p/base/fakesession.h" +#include "webrtc/p2p/base/fakesession.h" #include "talk/session/media/channel.h" #include "talk/session/media/mediarecorder.h" #include "webrtc/base/bytebuffer.h" diff --git a/session/media/mediasession.cc b/session/media/mediasession.cc index 54eeffa..96e45a6 100644 --- a/session/media/mediasession.cc +++ b/session/media/mediasession.cc @@ -34,10 +34,10 @@ #include "talk/media/base/constants.h" #include "talk/media/base/cryptoparams.h" -#include "talk/p2p/base/constants.h" +#include "webrtc/p2p/base/constants.h" #include "talk/session/media/channelmanager.h" #include "talk/session/media/srtpfilter.h" -#include "talk/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/constants.h" #include "webrtc/base/helpers.h" #include "webrtc/base/logging.h" #include "webrtc/base/scoped_ptr.h" @@ -1191,8 +1191,14 @@ SessionDescription* MediaSessionDescriptionFactory::CreateOffer( } video_added = true; } else if (IsMediaContentOfType(&*it, MEDIA_TYPE_DATA)) { - if (!AddDataContentForOffer(options, current_description, &data_codecs, - ¤t_streams, offer.get())) { + MediaSessionOptions options_copy(options); + if (IsSctp(static_cast<const MediaContentDescription*>( + it->description))) { + options_copy.data_channel_type = DCT_SCTP; + } + if (!AddDataContentForOffer(options_copy, current_description, + &data_codecs, ¤t_streams, + offer.get())) { return NULL; } data_added = true; diff --git a/session/media/mediasession.h b/session/media/mediasession.h index 992439c..462ddd2 100644 --- a/session/media/mediasession.h +++ b/session/media/mediasession.h @@ -40,9 +40,9 @@ #include "talk/media/base/mediachannel.h" #include "talk/media/base/mediaengine.h" // For DataChannelType #include "talk/media/base/streamparams.h" -#include "talk/p2p/base/sessiondescription.h" -#include "talk/p2p/base/transport.h" -#include "talk/p2p/base/transportdescriptionfactory.h" +#include "webrtc/p2p/base/sessiondescription.h" +#include "webrtc/p2p/base/transport.h" +#include "webrtc/p2p/base/transportdescriptionfactory.h" #include "webrtc/base/scoped_ptr.h" namespace cricket { diff --git a/session/media/mediasession_unittest.cc b/session/media/mediasession_unittest.cc index ef155f0..9ee6065 100644 --- a/session/media/mediasession_unittest.cc +++ b/session/media/mediasession_unittest.cc @@ -30,9 +30,9 @@ #include "talk/media/base/codec.h" #include "talk/media/base/testutils.h" -#include "talk/p2p/base/constants.h" -#include "talk/p2p/base/transportdescription.h" -#include "talk/p2p/base/transportinfo.h" +#include "webrtc/p2p/base/constants.h" +#include "webrtc/p2p/base/transportdescription.h" +#include "webrtc/p2p/base/transportinfo.h" #include "talk/session/media/mediasession.h" #include "talk/session/media/srtpfilter.h" #include "webrtc/base/fakesslidentity.h" @@ -628,6 +628,33 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateSctpDataOffer) { EXPECT_TRUE(offer->GetContentByName("data") != NULL); } +// Test creating an sctp data channel from an already generated offer. +TEST_F(MediaSessionDescriptionFactoryTest, TestCreateImplicitSctpDataOffer) { + MediaSessionOptions opts; + opts.recv_audio = false; + opts.bundle_enabled = true; + opts.data_channel_type = cricket::DCT_SCTP; + f1_.set_secure(SEC_ENABLED); + rtc::scoped_ptr<SessionDescription> offer1(f1_.CreateOffer(opts, NULL)); + ASSERT_TRUE(offer1.get() != NULL); + const ContentInfo* data = offer1->GetContentByName("data"); + ASSERT_TRUE(data != NULL); + const MediaContentDescription* mdesc = + static_cast<const MediaContentDescription*>(data->description); + ASSERT_EQ(cricket::kMediaProtocolSctp, mdesc->protocol()); + + // Now set data_channel_type to 'none' (default) and make sure that the + // datachannel type that gets generated from the previous offer, is of the + // same type. + opts.data_channel_type = cricket::DCT_NONE; + rtc::scoped_ptr<SessionDescription> offer2( + f1_.CreateOffer(opts, offer1.get())); + data = offer2->GetContentByName("data"); + ASSERT_TRUE(data != NULL); + mdesc = static_cast<const MediaContentDescription*>(data->description); + EXPECT_EQ(cricket::kMediaProtocolSctp, mdesc->protocol()); +} + // Create an audio, video offer without legacy StreamParams. TEST_F(MediaSessionDescriptionFactoryTest, TestCreateOfferWithoutLegacyStreams) { diff --git a/session/media/mediasessionclient.cc b/session/media/mediasessionclient.cc index 826909a..5cb7917 100644 --- a/session/media/mediasessionclient.cc +++ b/session/media/mediasessionclient.cc @@ -32,13 +32,13 @@ #include "talk/media/base/capturemanager.h" #include "talk/media/base/cryptoparams.h" #include "talk/media/sctp/sctpdataengine.h" -#include "talk/p2p/base/constants.h" -#include "talk/p2p/base/parsing.h" +#include "webrtc/p2p/base/constants.h" +#include "webrtc/p2p/base/parsing.h" #include "talk/session/media/mediamessages.h" #include "talk/session/media/srtpfilter.h" #include "webrtc/libjingle/xmllite/qname.h" #include "webrtc/libjingle/xmllite/xmlconstants.h" -#include "talk/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/constants.h" #include "webrtc/base/helpers.h" #include "webrtc/base/logging.h" #include "webrtc/base/stringencode.h" diff --git a/session/media/mediasessionclient.h b/session/media/mediasessionclient.h index 61d1f1d..c66c221 100644 --- a/session/media/mediasessionclient.h +++ b/session/media/mediasessionclient.h @@ -33,10 +33,10 @@ #include <string> #include <vector> #include "talk/media/base/cryptoparams.h" -#include "talk/p2p/base/session.h" -#include "talk/p2p/base/sessionclient.h" -#include "talk/p2p/base/sessiondescription.h" -#include "talk/p2p/base/sessionmanager.h" +#include "webrtc/p2p/base/session.h" +#include "webrtc/p2p/base/sessionclient.h" +#include "webrtc/p2p/base/sessiondescription.h" +#include "webrtc/p2p/base/sessionmanager.h" #include "talk/session/media/call.h" #include "talk/session/media/channelmanager.h" #include "talk/session/media/mediasession.h" diff --git a/session/media/mediasessionclient_unittest.cc b/session/media/mediasessionclient_unittest.cc index bb3043d..3e8a90f 100644 --- a/session/media/mediasessionclient_unittest.cc +++ b/session/media/mediasessionclient_unittest.cc @@ -31,13 +31,13 @@ #include "talk/media/base/fakemediaengine.h" #include "talk/media/base/testutils.h" #include "talk/media/devices/fakedevicemanager.h" -#include "talk/p2p/base/constants.h" -#include "talk/p2p/client/basicportallocator.h" +#include "webrtc/p2p/base/constants.h" +#include "webrtc/p2p/client/basicportallocator.h" #include "talk/session/media/mediasessionclient.h" #include "webrtc/libjingle/xmllite/xmlbuilder.h" #include "webrtc/libjingle/xmllite/xmlelement.h" #include "webrtc/libjingle/xmllite/xmlprinter.h" -#include "talk/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/constants.h" #include "webrtc/base/gunit.h" #include "webrtc/base/logging.h" #include "webrtc/base/scoped_ptr.h" diff --git a/session/media/rtcpmuxfilter.h b/session/media/rtcpmuxfilter.h index 40951a2..948b3c3 100644 --- a/session/media/rtcpmuxfilter.h +++ b/session/media/rtcpmuxfilter.h @@ -28,7 +28,7 @@ #ifndef TALK_SESSION_MEDIA_RTCPMUXFILTER_H_ #define TALK_SESSION_MEDIA_RTCPMUXFILTER_H_ -#include "talk/p2p/base/sessiondescription.h" +#include "webrtc/p2p/base/sessiondescription.h" #include "webrtc/base/basictypes.h" namespace cricket { diff --git a/session/media/srtpfilter.cc b/session/media/srtpfilter.cc index c79bf13..5a774db 100644 --- a/session/media/srtpfilter.cc +++ b/session/media/srtpfilter.cc @@ -150,7 +150,7 @@ bool SrtpFilter::SetRtpParams(const std::string& send_cs, const uint8* send_key, int send_key_len, const std::string& recv_cs, const uint8* recv_key, int recv_key_len) { - if (state_ == ST_ACTIVE) { + if (IsActive()) { LOG(LS_ERROR) << "Tried to set SRTP Params when filter already active"; return false; } @@ -212,6 +212,7 @@ bool SrtpFilter::ProtectRtp(void* p, int in_len, int max_len, int* out_len) { LOG(LS_WARNING) << "Failed to ProtectRtp: SRTP not active"; return false; } + ASSERT(send_session_ != NULL); return send_session_->ProtectRtp(p, in_len, max_len, out_len); } @@ -221,7 +222,7 @@ bool SrtpFilter::ProtectRtp(void* p, int in_len, int max_len, int* out_len, LOG(LS_WARNING) << "Failed to ProtectRtp: SRTP not active"; return false; } - + ASSERT(send_session_ != NULL); return send_session_->ProtectRtp(p, in_len, max_len, out_len, index); } @@ -233,6 +234,7 @@ bool SrtpFilter::ProtectRtcp(void* p, int in_len, int max_len, int* out_len) { if (send_rtcp_session_) { return send_rtcp_session_->ProtectRtcp(p, in_len, max_len, out_len); } else { + ASSERT(send_session_ != NULL); return send_session_->ProtectRtcp(p, in_len, max_len, out_len); } } @@ -242,6 +244,7 @@ bool SrtpFilter::UnprotectRtp(void* p, int in_len, int* out_len) { LOG(LS_WARNING) << "Failed to UnprotectRtp: SRTP not active"; return false; } + ASSERT(recv_session_ != NULL); return recv_session_->UnprotectRtp(p, in_len, out_len); } @@ -253,6 +256,7 @@ bool SrtpFilter::UnprotectRtcp(void* p, int in_len, int* out_len) { if (recv_rtcp_session_) { return recv_rtcp_session_->UnprotectRtcp(p, in_len, out_len); } else { + ASSERT(recv_session_ != NULL); return recv_session_->UnprotectRtcp(p, in_len, out_len); } } @@ -263,13 +267,16 @@ bool SrtpFilter::GetRtpAuthParams(uint8** key, int* key_len, int* tag_len) { return false; } + ASSERT(send_session_ != NULL); return send_session_->GetRtpAuthParams(key, key_len, tag_len); } void SrtpFilter::set_signal_silent_time(uint32 signal_silent_time_in_ms) { signal_silent_time_in_ms_ = signal_silent_time_in_ms; - if (state_ == ST_ACTIVE) { + if (IsActive()) { + ASSERT(send_session_ != NULL); send_session_->set_signal_silent_time(signal_silent_time_in_ms); + ASSERT(recv_session_ != NULL); recv_session_->set_signal_silent_time(signal_silent_time_in_ms); if (send_rtcp_session_) send_rtcp_session_->set_signal_silent_time(signal_silent_time_in_ms); @@ -292,7 +299,7 @@ bool SrtpFilter::StoreParams(const std::vector<CryptoParams>& params, offer_params_ = params; if (state_ == ST_INIT) { state_ = (source == CS_LOCAL) ? ST_SENTOFFER : ST_RECEIVEDOFFER; - } else { // state >= ST_ACTIVE + } else if (state_ == ST_ACTIVE) { state_ = (source == CS_LOCAL) ? ST_SENTUPDATEDOFFER : ST_RECEIVEDUPDATEDOFFER; } diff --git a/session/media/srtpfilter.h b/session/media/srtpfilter.h index ce3368f..3134418 100644 --- a/session/media/srtpfilter.h +++ b/session/media/srtpfilter.h @@ -34,7 +34,7 @@ #include <vector> #include "talk/media/base/cryptoparams.h" -#include "talk/p2p/base/sessiondescription.h" +#include "webrtc/p2p/base/sessiondescription.h" #include "webrtc/base/basictypes.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/base/sigslotrepeater.h" diff --git a/session/media/srtpfilter_unittest.cc b/session/media/srtpfilter_unittest.cc index 19efa2a..51d701d 100644 --- a/session/media/srtpfilter_unittest.cc +++ b/session/media/srtpfilter_unittest.cc @@ -27,7 +27,7 @@ #include "talk/media/base/cryptoparams.h" #include "talk/media/base/fakertp.h" -#include "talk/p2p/base/sessiondescription.h" +#include "webrtc/p2p/base/sessiondescription.h" #include "talk/session/media/srtpfilter.h" #include "webrtc/base/byteorder.h" #include "webrtc/base/gunit.h" @@ -82,9 +82,12 @@ class SrtpFilterTest : public testing::Test { const std::vector<CryptoParams>& params2) { EXPECT_TRUE(f1_.SetOffer(params1, CS_LOCAL)); EXPECT_TRUE(f2_.SetOffer(params1, CS_REMOTE)); + EXPECT_FALSE(f1_.IsActive()); + EXPECT_FALSE(f2_.IsActive()); EXPECT_TRUE(f2_.SetAnswer(params2, CS_LOCAL)); EXPECT_TRUE(f1_.SetAnswer(params2, CS_REMOTE)); EXPECT_TRUE(f1_.IsActive()); + EXPECT_TRUE(f2_.IsActive()); } void TestProtectUnprotect(const std::string& cs1, const std::string& cs2) { char rtp_packet[sizeof(kPcmuFrame) + 10]; @@ -139,6 +142,7 @@ class SrtpFilterTest : public testing::Test { // Test that we can set up the session and keys properly. TEST_F(SrtpFilterTest, TestGoodSetupOneCipherSuite) { EXPECT_TRUE(f1_.SetOffer(MakeVector(kTestCryptoParams1), CS_LOCAL)); + EXPECT_FALSE(f1_.IsActive()); EXPECT_TRUE(f1_.SetAnswer(MakeVector(kTestCryptoParams2), CS_REMOTE)); EXPECT_TRUE(f1_.IsActive()); } @@ -153,6 +157,7 @@ TEST_F(SrtpFilterTest, TestGoodSetupMultipleCipherSuites) { answer[0].tag = 2; answer[0].cipher_suite = CS_AES_CM_128_HMAC_SHA1_32; EXPECT_TRUE(f1_.SetOffer(offer, CS_LOCAL)); + EXPECT_FALSE(f1_.IsActive()); EXPECT_TRUE(f1_.SetAnswer(answer, CS_REMOTE)); EXPECT_TRUE(f1_.IsActive()); } @@ -188,6 +193,7 @@ TEST_F(SrtpFilterTest, TestBadSetup) { TEST_F(SrtpFilterTest, TestGoodSetupMultipleOffers) { EXPECT_TRUE(f1_.SetOffer(MakeVector(kTestCryptoParams1), CS_LOCAL)); EXPECT_TRUE(f1_.SetOffer(MakeVector(kTestCryptoParams2), CS_LOCAL)); + EXPECT_FALSE(f1_.IsActive()); EXPECT_TRUE(f1_.SetAnswer(MakeVector(kTestCryptoParams2), CS_REMOTE)); EXPECT_TRUE(f1_.IsActive()); EXPECT_TRUE(f1_.SetOffer(MakeVector(kTestCryptoParams1), CS_LOCAL)); @@ -196,6 +202,7 @@ TEST_F(SrtpFilterTest, TestGoodSetupMultipleOffers) { EXPECT_TRUE(f2_.SetOffer(MakeVector(kTestCryptoParams1), CS_REMOTE)); EXPECT_TRUE(f2_.SetOffer(MakeVector(kTestCryptoParams2), CS_REMOTE)); + EXPECT_FALSE(f2_.IsActive()); EXPECT_TRUE(f2_.SetAnswer(MakeVector(kTestCryptoParams2), CS_LOCAL)); EXPECT_TRUE(f2_.IsActive()); EXPECT_TRUE(f2_.SetOffer(MakeVector(kTestCryptoParams1), CS_REMOTE)); @@ -206,6 +213,7 @@ TEST_F(SrtpFilterTest, TestGoodSetupMultipleOffers) { TEST_F(SrtpFilterTest, TestBadSetupMultipleOffers) { EXPECT_TRUE(f1_.SetOffer(MakeVector(kTestCryptoParams1), CS_LOCAL)); EXPECT_FALSE(f1_.SetOffer(MakeVector(kTestCryptoParams2), CS_REMOTE)); + EXPECT_FALSE(f1_.IsActive()); EXPECT_TRUE(f1_.SetAnswer(MakeVector(kTestCryptoParams1), CS_REMOTE)); EXPECT_TRUE(f1_.IsActive()); EXPECT_TRUE(f1_.SetOffer(MakeVector(kTestCryptoParams2), CS_LOCAL)); @@ -214,6 +222,7 @@ TEST_F(SrtpFilterTest, TestBadSetupMultipleOffers) { EXPECT_TRUE(f2_.SetOffer(MakeVector(kTestCryptoParams2), CS_REMOTE)); EXPECT_FALSE(f2_.SetOffer(MakeVector(kTestCryptoParams1), CS_LOCAL)); + EXPECT_FALSE(f2_.IsActive()); EXPECT_TRUE(f2_.SetAnswer(MakeVector(kTestCryptoParams2), CS_LOCAL)); EXPECT_TRUE(f2_.IsActive()); EXPECT_TRUE(f2_.SetOffer(MakeVector(kTestCryptoParams2), CS_REMOTE)); @@ -402,6 +411,8 @@ TEST_F(SrtpFilterTest, TestProvisionalAnswer) { EXPECT_TRUE(f1_.SetOffer(offer, CS_LOCAL)); EXPECT_TRUE(f2_.SetOffer(offer, CS_REMOTE)); + EXPECT_FALSE(f1_.IsActive()); + EXPECT_FALSE(f2_.IsActive()); EXPECT_TRUE(f2_.SetProvisionalAnswer(answer, CS_LOCAL)); EXPECT_TRUE(f1_.SetProvisionalAnswer(answer, CS_REMOTE)); EXPECT_TRUE(f1_.IsActive()); @@ -425,6 +436,8 @@ TEST_F(SrtpFilterTest, TestProvisionalAnswerWithoutCrypto) { EXPECT_TRUE(f1_.SetOffer(offer, CS_LOCAL)); EXPECT_TRUE(f2_.SetOffer(offer, CS_REMOTE)); + EXPECT_FALSE(f1_.IsActive()); + EXPECT_FALSE(f2_.IsActive()); EXPECT_TRUE(f2_.SetProvisionalAnswer(answer, CS_LOCAL)); EXPECT_TRUE(f1_.SetProvisionalAnswer(answer, CS_REMOTE)); EXPECT_FALSE(f1_.IsActive()); @@ -438,6 +451,33 @@ TEST_F(SrtpFilterTest, TestProvisionalAnswerWithoutCrypto) { TestProtectUnprotect(CS_AES_CM_128_HMAC_SHA1_80, CS_AES_CM_128_HMAC_SHA1_80); } +// Test that if we get a new local offer after a provisional answer +// with no crypto, that we are in an inactive state. +TEST_F(SrtpFilterTest, TestLocalOfferAfterProvisionalAnswerWithoutCrypto) { + std::vector<CryptoParams> offer(MakeVector(kTestCryptoParams1)); + std::vector<CryptoParams> answer; + + EXPECT_TRUE(f1_.SetOffer(offer, CS_LOCAL)); + EXPECT_TRUE(f2_.SetOffer(offer, CS_REMOTE)); + EXPECT_TRUE(f1_.SetProvisionalAnswer(answer, CS_REMOTE)); + EXPECT_TRUE(f2_.SetProvisionalAnswer(answer, CS_LOCAL)); + EXPECT_FALSE(f1_.IsActive()); + EXPECT_FALSE(f2_.IsActive()); + // The calls to set an offer after a provisional answer fail, so the + // state doesn't change. + EXPECT_FALSE(f1_.SetOffer(offer, CS_LOCAL)); + EXPECT_FALSE(f2_.SetOffer(offer, CS_REMOTE)); + EXPECT_FALSE(f1_.IsActive()); + EXPECT_FALSE(f2_.IsActive()); + + answer.push_back(kTestCryptoParams2); + EXPECT_TRUE(f2_.SetAnswer(answer, CS_LOCAL)); + EXPECT_TRUE(f1_.SetAnswer(answer, CS_REMOTE)); + EXPECT_TRUE(f1_.IsActive()); + EXPECT_TRUE(f2_.IsActive()); + TestProtectUnprotect(CS_AES_CM_128_HMAC_SHA1_80, CS_AES_CM_128_HMAC_SHA1_80); +} + // Test that we can disable encryption. TEST_F(SrtpFilterTest, TestDisableEncryption) { std::vector<CryptoParams> offer(MakeVector(kTestCryptoParams1)); diff --git a/session/media/typingmonitor_unittest.cc b/session/media/typingmonitor_unittest.cc index 95fa7a4..66e2401 100644 --- a/session/media/typingmonitor_unittest.cc +++ b/session/media/typingmonitor_unittest.cc @@ -26,7 +26,7 @@ */ #include "talk/media/base/fakemediaengine.h" -#include "talk/p2p/base/fakesession.h" +#include "webrtc/p2p/base/fakesession.h" #include "talk/session/media/channel.h" #include "talk/session/media/currentspeakermonitor.h" #include "talk/session/media/typingmonitor.h" diff --git a/session/tunnel/pseudotcpchannel.cc b/session/tunnel/pseudotcpchannel.cc index f277614..861e178 100644 --- a/session/tunnel/pseudotcpchannel.cc +++ b/session/tunnel/pseudotcpchannel.cc @@ -27,8 +27,8 @@ #include <string> #include "pseudotcpchannel.h" -#include "talk/p2p/base/candidate.h" -#include "talk/p2p/base/transportchannel.h" +#include "webrtc/p2p/base/candidate.h" +#include "webrtc/p2p/base/transportchannel.h" #include "webrtc/base/basictypes.h" #include "webrtc/base/common.h" #include "webrtc/base/logging.h" diff --git a/session/tunnel/pseudotcpchannel.h b/session/tunnel/pseudotcpchannel.h index 5ccfa4c..ad6d907 100644 --- a/session/tunnel/pseudotcpchannel.h +++ b/session/tunnel/pseudotcpchannel.h @@ -28,8 +28,8 @@ #ifndef TALK_SESSION_TUNNEL_PSEUDOTCPCHANNEL_H_ #define TALK_SESSION_TUNNEL_PSEUDOTCPCHANNEL_H_ -#include "talk/p2p/base/pseudotcp.h" -#include "talk/p2p/base/session.h" +#include "webrtc/p2p/base/pseudotcp.h" +#include "webrtc/p2p/base/session.h" #include "webrtc/base/criticalsection.h" #include "webrtc/base/messagequeue.h" #include "webrtc/base/stream.h" diff --git a/session/tunnel/securetunnelsessionclient.cc b/session/tunnel/securetunnelsessionclient.cc index cb41c3b..4ed8185 100644 --- a/session/tunnel/securetunnelsessionclient.cc +++ b/session/tunnel/securetunnelsessionclient.cc @@ -27,7 +27,7 @@ // SecureTunnelSessionClient and SecureTunnelSession implementation. -#include "talk/p2p/base/transportchannel.h" +#include "webrtc/p2p/base/transportchannel.h" #include "talk/session/tunnel/pseudotcpchannel.h" #include "talk/session/tunnel/securetunnelsessionclient.h" #include "webrtc/libjingle/xmllite/xmlelement.h" diff --git a/session/tunnel/tunnelsessionclient.cc b/session/tunnel/tunnelsessionclient.cc index 7221db4..7d2a7d1 100644 --- a/session/tunnel/tunnelsessionclient.cc +++ b/session/tunnel/tunnelsessionclient.cc @@ -26,8 +26,8 @@ */ #include "pseudotcpchannel.h" -#include "talk/p2p/base/constants.h" -#include "talk/p2p/base/transportchannel.h" +#include "webrtc/p2p/base/constants.h" +#include "webrtc/p2p/base/transportchannel.h" #include "webrtc/libjingle/xmllite/xmlelement.h" #include "tunnelsessionclient.h" #include "webrtc/base/basicdefs.h" diff --git a/session/tunnel/tunnelsessionclient.h b/session/tunnel/tunnelsessionclient.h index d3fa64a..87beaed 100644 --- a/session/tunnel/tunnelsessionclient.h +++ b/session/tunnel/tunnelsessionclient.h @@ -30,14 +30,14 @@ #include <vector> -#include "talk/p2p/base/constants.h" -#include "talk/p2p/base/pseudotcp.h" -#include "talk/p2p/base/session.h" -#include "talk/p2p/base/sessionclient.h" -#include "talk/p2p/base/sessiondescription.h" -#include "talk/p2p/base/sessionmanager.h" +#include "webrtc/p2p/base/constants.h" +#include "webrtc/p2p/base/pseudotcp.h" +#include "webrtc/p2p/base/session.h" +#include "webrtc/p2p/base/sessionclient.h" +#include "webrtc/p2p/base/sessiondescription.h" +#include "webrtc/p2p/base/sessionmanager.h" #include "webrtc/libjingle/xmllite/qname.h" -#include "talk/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/constants.h" #include "webrtc/base/criticalsection.h" #include "webrtc/base/stream.h" diff --git a/session/tunnel/tunnelsessionclient_unittest.cc b/session/tunnel/tunnelsessionclient_unittest.cc index 6d46918..6bd746a 100644 --- a/session/tunnel/tunnelsessionclient_unittest.cc +++ b/session/tunnel/tunnelsessionclient_unittest.cc @@ -26,9 +26,9 @@ */ #include <string> -#include "talk/p2p/base/sessionmanager.h" -#include "talk/p2p/base/transport.h" -#include "talk/p2p/client/fakeportallocator.h" +#include "webrtc/p2p/base/sessionmanager.h" +#include "webrtc/p2p/base/transport.h" +#include "webrtc/p2p/client/fakeportallocator.h" #include "talk/session/tunnel/tunnelsessionclient.h" #include "webrtc/base/gunit.h" #include "webrtc/base/messagehandler.h" diff --git a/xmpp/chatroommodule.h b/xmpp/chatroommodule.h index 8358fc1..145ab89 100644 --- a/xmpp/chatroommodule.h +++ b/xmpp/chatroommodule.h @@ -28,8 +28,8 @@ #ifndef TALK_XMPP_CHATROOMMODULE_H_ #define TALK_XMPP_CHATROOMMODULE_H_ -#include "talk/xmpp/module.h" -#include "talk/xmpp/rostermodule.h" +#include "webrtc/libjingle/xmpp/module.h" +#include "webrtc/libjingle/xmpp/rostermodule.h" namespace buzz { diff --git a/xmpp/chatroommoduleimpl.cc b/xmpp/chatroommoduleimpl.cc index 45db014..33c752e 100644 --- a/xmpp/chatroommoduleimpl.cc +++ b/xmpp/chatroommoduleimpl.cc @@ -31,9 +31,9 @@ #include <sstream> #include <string> #include <vector> -#include "talk/xmpp/chatroommodule.h" -#include "talk/xmpp/constants.h" -#include "talk/xmpp/moduleimpl.h" +#include "webrtc/libjingle/xmpp/chatroommodule.h" +#include "webrtc/libjingle/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/moduleimpl.h" #include "webrtc/base/common.h" namespace buzz { diff --git a/xmpp/constants.cc b/xmpp/constants.cc index 297eafd..fce9c75 100644 --- a/xmpp/constants.cc +++ b/xmpp/constants.cc @@ -25,14 +25,14 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/constants.h" #include <string> #include "webrtc/libjingle/xmllite/qname.h" #include "webrtc/libjingle/xmllite/xmlconstants.h" #include "webrtc/libjingle/xmllite/xmlelement.h" -#include "talk/xmpp/jid.h" +#include "webrtc/libjingle/xmpp/jid.h" #include "webrtc/base/basicdefs.h" namespace buzz { diff --git a/xmpp/constants.h b/xmpp/constants.h index 6aa1a54..52088ca 100644 --- a/xmpp/constants.h +++ b/xmpp/constants.h @@ -30,7 +30,7 @@ #include <string> #include "webrtc/libjingle/xmllite/qname.h" -#include "talk/xmpp/jid.h" +#include "webrtc/libjingle/xmpp/jid.h" namespace buzz { diff --git a/xmpp/discoitemsquerytask.cc b/xmpp/discoitemsquerytask.cc index d900b85..b739ba3 100644 --- a/xmpp/discoitemsquerytask.cc +++ b/xmpp/discoitemsquerytask.cc @@ -25,9 +25,9 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/xmpp/constants.h" -#include "talk/xmpp/discoitemsquerytask.h" -#include "talk/xmpp/xmpptask.h" +#include "webrtc/libjingle/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/discoitemsquerytask.h" +#include "webrtc/libjingle/xmpp/xmpptask.h" #include "webrtc/base/scoped_ptr.h" namespace buzz { diff --git a/xmpp/discoitemsquerytask.h b/xmpp/discoitemsquerytask.h index 409fc90..c31d261 100644 --- a/xmpp/discoitemsquerytask.h +++ b/xmpp/discoitemsquerytask.h @@ -54,7 +54,7 @@ #include <string> #include <vector> -#include "talk/xmpp/iqtask.h" +#include "webrtc/libjingle/xmpp/iqtask.h" namespace buzz { diff --git a/xmpp/fakexmppclient.h b/xmpp/fakexmppclient.h index 3522ba9..2e37dd1 100644 --- a/xmpp/fakexmppclient.h +++ b/xmpp/fakexmppclient.h @@ -33,7 +33,7 @@ #include <string> #include <vector> -#include "talk/xmpp/xmpptask.h" +#include "webrtc/libjingle/xmpp/xmpptask.h" namespace buzz { diff --git a/xmpp/hangoutpubsubclient.cc b/xmpp/hangoutpubsubclient.cc index 63f5bcf..dacccf1 100644 --- a/xmpp/hangoutpubsubclient.cc +++ b/xmpp/hangoutpubsubclient.cc @@ -25,12 +25,12 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/xmpp/hangoutpubsubclient.h" +#include "webrtc/libjingle/xmpp/hangoutpubsubclient.h" #include "webrtc/libjingle/xmllite/qname.h" #include "webrtc/libjingle/xmllite/xmlelement.h" -#include "talk/xmpp/constants.h" -#include "talk/xmpp/jid.h" +#include "webrtc/libjingle/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/jid.h" #include "webrtc/base/logging.h" diff --git a/xmpp/hangoutpubsubclient.h b/xmpp/hangoutpubsubclient.h index 5692fc6..03c96f2 100644 --- a/xmpp/hangoutpubsubclient.h +++ b/xmpp/hangoutpubsubclient.h @@ -32,9 +32,9 @@ #include <string> #include <vector> -#include "talk/xmpp/jid.h" -#include "talk/xmpp/pubsubclient.h" -#include "talk/xmpp/pubsubstateclient.h" +#include "webrtc/libjingle/xmpp/jid.h" +#include "webrtc/libjingle/xmpp/pubsubclient.h" +#include "webrtc/libjingle/xmpp/pubsubstateclient.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/base/sigslot.h" #include "webrtc/base/sigslotrepeater.h" diff --git a/xmpp/hangoutpubsubclient_unittest.cc b/xmpp/hangoutpubsubclient_unittest.cc index 555ee5c..574e21a 100644 --- a/xmpp/hangoutpubsubclient_unittest.cc +++ b/xmpp/hangoutpubsubclient_unittest.cc @@ -5,10 +5,10 @@ #include "webrtc/libjingle/xmllite/qname.h" #include "webrtc/libjingle/xmllite/xmlelement.h" -#include "talk/xmpp/constants.h" -#include "talk/xmpp/fakexmppclient.h" -#include "talk/xmpp/hangoutpubsubclient.h" -#include "talk/xmpp/jid.h" +#include "webrtc/libjingle/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/fakexmppclient.h" +#include "webrtc/libjingle/xmpp/hangoutpubsubclient.h" +#include "webrtc/libjingle/xmpp/jid.h" #include "webrtc/base/faketaskrunner.h" #include "webrtc/base/gunit.h" #include "webrtc/base/sigslot.h" diff --git a/xmpp/iqtask.cc b/xmpp/iqtask.cc index f6a21d4..d4cf310 100644 --- a/xmpp/iqtask.cc +++ b/xmpp/iqtask.cc @@ -25,10 +25,10 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/xmpp/iqtask.h" +#include "webrtc/libjingle/xmpp/iqtask.h" -#include "talk/xmpp/constants.h" -#include "talk/xmpp/xmppclient.h" +#include "webrtc/libjingle/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/xmppclient.h" namespace buzz { diff --git a/xmpp/iqtask.h b/xmpp/iqtask.h index 312c4af..8b0c04b 100644 --- a/xmpp/iqtask.h +++ b/xmpp/iqtask.h @@ -30,8 +30,8 @@ #include <string> -#include "talk/xmpp/xmppengine.h" -#include "talk/xmpp/xmpptask.h" +#include "webrtc/libjingle/xmpp/xmppengine.h" +#include "webrtc/libjingle/xmpp/xmpptask.h" namespace buzz { diff --git a/xmpp/jid.cc b/xmpp/jid.cc index 702e477..0410458 100644 --- a/xmpp/jid.cc +++ b/xmpp/jid.cc @@ -25,14 +25,14 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/xmpp/jid.h" +#include "webrtc/libjingle/xmpp/jid.h" #include <ctype.h> #include <algorithm> #include <string> -#include "talk/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/constants.h" #include "webrtc/base/common.h" #include "webrtc/base/logging.h" diff --git a/xmpp/jid_unittest.cc b/xmpp/jid_unittest.cc index 05835cf..9cae2f8 100644 --- a/xmpp/jid_unittest.cc +++ b/xmpp/jid_unittest.cc @@ -1,7 +1,7 @@ // Copyright 2004 Google Inc. All Rights Reserved -#include "talk/xmpp/jid.h" +#include "webrtc/libjingle/xmpp/jid.h" #include "webrtc/base/gunit.h" using buzz::Jid; diff --git a/xmpp/jingleinfotask.cc b/xmpp/jingleinfotask.cc index 3ceae07..3b8fdde 100644 --- a/xmpp/jingleinfotask.cc +++ b/xmpp/jingleinfotask.cc @@ -25,11 +25,11 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/xmpp/jingleinfotask.h" +#include "webrtc/libjingle/xmpp/jingleinfotask.h" -#include "talk/xmpp/constants.h" -#include "talk/xmpp/xmppclient.h" -#include "talk/xmpp/xmpptask.h" +#include "webrtc/libjingle/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/xmppclient.h" +#include "webrtc/libjingle/xmpp/xmpptask.h" #include "webrtc/base/socketaddress.h" namespace buzz { diff --git a/xmpp/jingleinfotask.h b/xmpp/jingleinfotask.h index 18cb5a9..ac03ecf 100644 --- a/xmpp/jingleinfotask.h +++ b/xmpp/jingleinfotask.h @@ -30,9 +30,9 @@ #include <vector> -#include "talk/p2p/client/httpportallocator.h" -#include "talk/xmpp/xmppengine.h" -#include "talk/xmpp/xmpptask.h" +#include "webrtc/p2p/client/httpportallocator.h" +#include "webrtc/libjingle/xmpp/xmppengine.h" +#include "webrtc/libjingle/xmpp/xmpptask.h" #include "webrtc/base/sigslot.h" namespace buzz { diff --git a/xmpp/module.h b/xmpp/module.h index a5d0687..f1cff11 100644 --- a/xmpp/module.h +++ b/xmpp/module.h @@ -28,7 +28,7 @@ #ifndef TALK_XMPP_MODULE_H_ #define TALK_XMPP_MODULE_H_ -#include "talk/xmpp/xmppengine.h" +#include "webrtc/libjingle/xmpp/xmppengine.h" namespace buzz { diff --git a/xmpp/moduleimpl.cc b/xmpp/moduleimpl.cc index 4635bee..8ac755d 100644 --- a/xmpp/moduleimpl.cc +++ b/xmpp/moduleimpl.cc @@ -25,7 +25,7 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/xmpp/moduleimpl.h" +#include "webrtc/libjingle/xmpp/moduleimpl.h" #include "webrtc/base/common.h" namespace buzz { diff --git a/xmpp/moduleimpl.h b/xmpp/moduleimpl.h index 897bfce..f1986f3 100644 --- a/xmpp/moduleimpl.h +++ b/xmpp/moduleimpl.h @@ -28,8 +28,8 @@ #ifndef TALK_XMPP_MODULEIMPL_H_ #define TALK_XMPP_MODULEIMPL_H_ -#include "talk/xmpp/module.h" -#include "talk/xmpp/xmppengine.h" +#include "webrtc/libjingle/xmpp/module.h" +#include "webrtc/libjingle/xmpp/xmppengine.h" namespace buzz { diff --git a/xmpp/mucroomconfigtask.cc b/xmpp/mucroomconfigtask.cc index 536eb9b..86ba5f6 100644 --- a/xmpp/mucroomconfigtask.cc +++ b/xmpp/mucroomconfigtask.cc @@ -28,9 +28,9 @@ #include <string> #include <vector> -#include "talk/xmpp/mucroomconfigtask.h" +#include "webrtc/libjingle/xmpp/mucroomconfigtask.h" -#include "talk/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/constants.h" #include "webrtc/base/scoped_ptr.h" namespace buzz { diff --git a/xmpp/mucroomconfigtask.h b/xmpp/mucroomconfigtask.h index ba0dbaa..5626d3e 100644 --- a/xmpp/mucroomconfigtask.h +++ b/xmpp/mucroomconfigtask.h @@ -29,7 +29,7 @@ #define TALK_XMPP_MUCROOMCONFIGTASK_H_ #include <string> -#include "talk/xmpp/iqtask.h" +#include "webrtc/libjingle/xmpp/iqtask.h" namespace buzz { diff --git a/xmpp/mucroomconfigtask_unittest.cc b/xmpp/mucroomconfigtask_unittest.cc index bf5e7b4..e9e1281 100644 --- a/xmpp/mucroomconfigtask_unittest.cc +++ b/xmpp/mucroomconfigtask_unittest.cc @@ -29,9 +29,9 @@ #include <vector> #include "webrtc/libjingle/xmllite/xmlelement.h" -#include "talk/xmpp/constants.h" -#include "talk/xmpp/fakexmppclient.h" -#include "talk/xmpp/mucroomconfigtask.h" +#include "webrtc/libjingle/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/fakexmppclient.h" +#include "webrtc/libjingle/xmpp/mucroomconfigtask.h" #include "webrtc/base/faketaskrunner.h" #include "webrtc/base/gunit.h" #include "webrtc/base/sigslot.h" diff --git a/xmpp/mucroomdiscoverytask.cc b/xmpp/mucroomdiscoverytask.cc index c7477ae..0ec4fbe 100644 --- a/xmpp/mucroomdiscoverytask.cc +++ b/xmpp/mucroomdiscoverytask.cc @@ -25,9 +25,9 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/xmpp/mucroomdiscoverytask.h" +#include "webrtc/libjingle/xmpp/mucroomdiscoverytask.h" -#include "talk/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/constants.h" namespace buzz { diff --git a/xmpp/mucroomdiscoverytask.h b/xmpp/mucroomdiscoverytask.h index 4097cc6..dc5918d 100644 --- a/xmpp/mucroomdiscoverytask.h +++ b/xmpp/mucroomdiscoverytask.h @@ -30,7 +30,7 @@ #include <map> #include <string> -#include "talk/xmpp/iqtask.h" +#include "webrtc/libjingle/xmpp/iqtask.h" namespace buzz { diff --git a/xmpp/mucroomdiscoverytask_unittest.cc b/xmpp/mucroomdiscoverytask_unittest.cc index e1a633e..77fdf86 100644 --- a/xmpp/mucroomdiscoverytask_unittest.cc +++ b/xmpp/mucroomdiscoverytask_unittest.cc @@ -29,9 +29,9 @@ #include <vector> #include "webrtc/libjingle/xmllite/xmlelement.h" -#include "talk/xmpp/constants.h" -#include "talk/xmpp/fakexmppclient.h" -#include "talk/xmpp/mucroomdiscoverytask.h" +#include "webrtc/libjingle/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/fakexmppclient.h" +#include "webrtc/libjingle/xmpp/mucroomdiscoverytask.h" #include "webrtc/base/faketaskrunner.h" #include "webrtc/base/gunit.h" #include "webrtc/base/sigslot.h" diff --git a/xmpp/mucroomlookuptask.cc b/xmpp/mucroomlookuptask.cc index 8e1895c..6832ade 100644 --- a/xmpp/mucroomlookuptask.cc +++ b/xmpp/mucroomlookuptask.cc @@ -25,9 +25,9 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/xmpp/mucroomlookuptask.h" +#include "webrtc/libjingle/xmpp/mucroomlookuptask.h" -#include "talk/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/constants.h" #include "webrtc/base/logging.h" #include "webrtc/base/scoped_ptr.h" diff --git a/xmpp/mucroomlookuptask.h b/xmpp/mucroomlookuptask.h index 48f2484..9bcca2c 100644 --- a/xmpp/mucroomlookuptask.h +++ b/xmpp/mucroomlookuptask.h @@ -29,7 +29,7 @@ #define TALK_XMPP_MUCROOMLOOKUPTASK_H_ #include <string> -#include "talk/xmpp/iqtask.h" +#include "webrtc/libjingle/xmpp/iqtask.h" namespace buzz { diff --git a/xmpp/mucroomlookuptask_unittest.cc b/xmpp/mucroomlookuptask_unittest.cc index 03be292..961e664 100644 --- a/xmpp/mucroomlookuptask_unittest.cc +++ b/xmpp/mucroomlookuptask_unittest.cc @@ -29,9 +29,9 @@ #include <vector> #include "webrtc/libjingle/xmllite/xmlelement.h" -#include "talk/xmpp/constants.h" -#include "talk/xmpp/fakexmppclient.h" -#include "talk/xmpp/mucroomlookuptask.h" +#include "webrtc/libjingle/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/fakexmppclient.h" +#include "webrtc/libjingle/xmpp/mucroomlookuptask.h" #include "webrtc/base/faketaskrunner.h" #include "webrtc/base/gunit.h" #include "webrtc/base/sigslot.h" diff --git a/xmpp/mucroomuniquehangoutidtask.cc b/xmpp/mucroomuniquehangoutidtask.cc index 78a8edf..db94741 100644 --- a/xmpp/mucroomuniquehangoutidtask.cc +++ b/xmpp/mucroomuniquehangoutidtask.cc @@ -1,9 +1,9 @@ // Copyright 2012 Google Inc. All Rights Reserved. -#include "talk/xmpp/mucroomuniquehangoutidtask.h" +#include "webrtc/libjingle/xmpp/mucroomuniquehangoutidtask.h" -#include "talk/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/constants.h" namespace buzz { diff --git a/xmpp/mucroomuniquehangoutidtask.h b/xmpp/mucroomuniquehangoutidtask.h index d222bac..83de434 100644 --- a/xmpp/mucroomuniquehangoutidtask.h +++ b/xmpp/mucroomuniquehangoutidtask.h @@ -4,7 +4,7 @@ #ifndef TALK_XMPP_MUCROOMUNIQUEHANGOUTIDTASK_H_ #define TALK_XMPP_MUCROOMUNIQUEHANGOUTIDTASK_H_ -#include "talk/xmpp/iqtask.h" +#include "webrtc/libjingle/xmpp/iqtask.h" namespace buzz { diff --git a/xmpp/mucroomuniquehangoutidtask_unittest.cc b/xmpp/mucroomuniquehangoutidtask_unittest.cc index 42bed13..d8f7c20 100644 --- a/xmpp/mucroomuniquehangoutidtask_unittest.cc +++ b/xmpp/mucroomuniquehangoutidtask_unittest.cc @@ -29,9 +29,9 @@ #include <vector> #include "webrtc/libjingle/xmllite/xmlelement.h" -#include "talk/xmpp/constants.h" -#include "talk/xmpp/fakexmppclient.h" -#include "talk/xmpp/mucroomuniquehangoutidtask.h" +#include "webrtc/libjingle/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/fakexmppclient.h" +#include "webrtc/libjingle/xmpp/mucroomuniquehangoutidtask.h" #include "webrtc/base/faketaskrunner.h" #include "webrtc/base/gunit.h" #include "webrtc/base/sigslot.h" diff --git a/xmpp/pingtask.cc b/xmpp/pingtask.cc index 7922ea0..619c3a8 100644 --- a/xmpp/pingtask.cc +++ b/xmpp/pingtask.cc @@ -1,9 +1,9 @@ // Copyright 2011 Google Inc. All Rights Reserved. -#include "talk/xmpp/pingtask.h" +#include "webrtc/libjingle/xmpp/pingtask.h" -#include "talk/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/constants.h" #include "webrtc/base/logging.h" #include "webrtc/base/scoped_ptr.h" diff --git a/xmpp/pingtask.h b/xmpp/pingtask.h index 2a869a3..87d74b4 100644 --- a/xmpp/pingtask.h +++ b/xmpp/pingtask.h @@ -28,7 +28,7 @@ #ifndef TALK_XMPP_PINGTASK_H_ #define TALK_XMPP_PINGTASK_H_ -#include "talk/xmpp/xmpptask.h" +#include "webrtc/libjingle/xmpp/xmpptask.h" #include "webrtc/base/messagehandler.h" #include "webrtc/base/messagequeue.h" diff --git a/xmpp/pingtask_unittest.cc b/xmpp/pingtask_unittest.cc index fe88a5c..a0b6618 100644 --- a/xmpp/pingtask_unittest.cc +++ b/xmpp/pingtask_unittest.cc @@ -29,9 +29,9 @@ #include <vector> #include "webrtc/libjingle/xmllite/xmlelement.h" -#include "talk/xmpp/constants.h" -#include "talk/xmpp/fakexmppclient.h" -#include "talk/xmpp/pingtask.h" +#include "webrtc/libjingle/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/fakexmppclient.h" +#include "webrtc/libjingle/xmpp/pingtask.h" #include "webrtc/base/faketaskrunner.h" #include "webrtc/base/gunit.h" #include "webrtc/base/sigslot.h" diff --git a/xmpp/plainsaslhandler.h b/xmpp/plainsaslhandler.h index 31032e4..f23782d 100644 --- a/xmpp/plainsaslhandler.h +++ b/xmpp/plainsaslhandler.h @@ -29,8 +29,8 @@ #define TALK_XMPP_PLAINSASLHANDLER_H_ #include <algorithm> -#include "talk/xmpp/saslhandler.h" -#include "talk/xmpp/saslplainmechanism.h" +#include "webrtc/libjingle/xmpp/saslhandler.h" +#include "webrtc/libjingle/xmpp/saslplainmechanism.h" #include "webrtc/base/cryptstring.h" namespace buzz { diff --git a/xmpp/presenceouttask.cc b/xmpp/presenceouttask.cc index a77d9d6..39ef646 100644 --- a/xmpp/presenceouttask.cc +++ b/xmpp/presenceouttask.cc @@ -27,9 +27,9 @@ #include <time.h> #include <sstream> -#include "talk/xmpp/constants.h" -#include "talk/xmpp/presenceouttask.h" -#include "talk/xmpp/xmppclient.h" +#include "webrtc/libjingle/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/presenceouttask.h" +#include "webrtc/libjingle/xmpp/xmppclient.h" #include "webrtc/base/stringencode.h" namespace buzz { diff --git a/xmpp/presenceouttask.h b/xmpp/presenceouttask.h index 53bbae5..f6b04af 100644 --- a/xmpp/presenceouttask.h +++ b/xmpp/presenceouttask.h @@ -28,9 +28,9 @@ #ifndef TALK_XMPP_PRESENCEOUTTASK_H_ #define TALK_XMPP_PRESENCEOUTTASK_H_ -#include "talk/xmpp/presencestatus.h" -#include "talk/xmpp/xmppengine.h" -#include "talk/xmpp/xmpptask.h" +#include "webrtc/libjingle/xmpp/presencestatus.h" +#include "webrtc/libjingle/xmpp/xmppengine.h" +#include "webrtc/libjingle/xmpp/xmpptask.h" namespace buzz { diff --git a/xmpp/presencereceivetask.cc b/xmpp/presencereceivetask.cc index 940c53b..0a159ac 100644 --- a/xmpp/presencereceivetask.cc +++ b/xmpp/presencereceivetask.cc @@ -25,9 +25,9 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/xmpp/presencereceivetask.h" +#include "webrtc/libjingle/xmpp/presencereceivetask.h" -#include "talk/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/constants.h" #include "webrtc/base/stringencode.h" namespace buzz { diff --git a/xmpp/presencereceivetask.h b/xmpp/presencereceivetask.h index 6a090f3..80e11d8 100644 --- a/xmpp/presencereceivetask.h +++ b/xmpp/presencereceivetask.h @@ -30,8 +30,8 @@ #include "webrtc/base/sigslot.h" -#include "talk/xmpp/presencestatus.h" -#include "talk/xmpp/xmpptask.h" +#include "webrtc/libjingle/xmpp/presencestatus.h" +#include "webrtc/libjingle/xmpp/xmpptask.h" namespace buzz { diff --git a/xmpp/presencestatus.cc b/xmpp/presencestatus.cc index c75b705..2ab89b2 100644 --- a/xmpp/presencestatus.cc +++ b/xmpp/presencestatus.cc @@ -25,7 +25,7 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/xmpp/presencestatus.h" +#include "webrtc/libjingle/xmpp/presencestatus.h" namespace buzz { PresenceStatus::PresenceStatus() diff --git a/xmpp/presencestatus.h b/xmpp/presencestatus.h index 45c5471..c133e40 100644 --- a/xmpp/presencestatus.h +++ b/xmpp/presencestatus.h @@ -28,8 +28,8 @@ #ifndef THIRD_PARTY_LIBJINGLE_FILES_TALK_XMPP_PRESENCESTATUS_H_ #define THIRD_PARTY_LIBJINGLE_FILES_TALK_XMPP_PRESENCESTATUS_H_ -#include "talk/xmpp/constants.h" -#include "talk/xmpp/jid.h" +#include "webrtc/libjingle/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/jid.h" namespace buzz { diff --git a/xmpp/prexmppauth.h b/xmpp/prexmppauth.h index 71fcae4..7ddc85b 100644 --- a/xmpp/prexmppauth.h +++ b/xmpp/prexmppauth.h @@ -28,7 +28,7 @@ #ifndef TALK_XMPP_PREXMPPAUTH_H_ #define TALK_XMPP_PREXMPPAUTH_H_ -#include "talk/xmpp/saslhandler.h" +#include "webrtc/libjingle/xmpp/saslhandler.h" #include "webrtc/base/cryptstring.h" #include "webrtc/base/sigslot.h" diff --git a/xmpp/pubsub_task.cc b/xmpp/pubsub_task.cc index 3552764..98446fd 100644 --- a/xmpp/pubsub_task.cc +++ b/xmpp/pubsub_task.cc @@ -25,13 +25,13 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/xmpp/pubsub_task.h" +#include "webrtc/libjingle/xmpp/pubsub_task.h" #include <map> #include <string> -#include "talk/xmpp/constants.h" -#include "talk/xmpp/xmppengine.h" +#include "webrtc/libjingle/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/xmppengine.h" #include "webrtc/base/common.h" namespace buzz { diff --git a/xmpp/pubsub_task.h b/xmpp/pubsub_task.h index 2787cbc..a219b60 100644 --- a/xmpp/pubsub_task.h +++ b/xmpp/pubsub_task.h @@ -31,8 +31,8 @@ #include <map> #include <string> #include "webrtc/libjingle/xmllite/xmlelement.h" -#include "talk/xmpp/jid.h" -#include "talk/xmpp/xmpptask.h" +#include "webrtc/libjingle/xmpp/jid.h" +#include "webrtc/libjingle/xmpp/xmpptask.h" namespace buzz { diff --git a/xmpp/pubsubclient.cc b/xmpp/pubsubclient.cc index b627587..a5793ef 100644 --- a/xmpp/pubsubclient.cc +++ b/xmpp/pubsubclient.cc @@ -25,14 +25,14 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/xmpp/pubsubclient.h" +#include "webrtc/libjingle/xmpp/pubsubclient.h" #include <string> #include <vector> -#include "talk/xmpp/constants.h" -#include "talk/xmpp/jid.h" -#include "talk/xmpp/pubsubtasks.h" +#include "webrtc/libjingle/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/jid.h" +#include "webrtc/libjingle/xmpp/pubsubtasks.h" namespace buzz { diff --git a/xmpp/pubsubclient.h b/xmpp/pubsubclient.h index d7b5066..6465957 100644 --- a/xmpp/pubsubclient.h +++ b/xmpp/pubsubclient.h @@ -31,8 +31,8 @@ #include <string> #include <vector> -#include "talk/xmpp/jid.h" -#include "talk/xmpp/pubsubtasks.h" +#include "webrtc/libjingle/xmpp/jid.h" +#include "webrtc/libjingle/xmpp/pubsubtasks.h" #include "webrtc/base/sigslot.h" #include "webrtc/base/sigslotrepeater.h" #include "webrtc/base/task.h" diff --git a/xmpp/pubsubclient_unittest.cc b/xmpp/pubsubclient_unittest.cc index f191a18..b3049d8 100644 --- a/xmpp/pubsubclient_unittest.cc +++ b/xmpp/pubsubclient_unittest.cc @@ -5,10 +5,10 @@ #include "webrtc/libjingle/xmllite/qname.h" #include "webrtc/libjingle/xmllite/xmlelement.h" -#include "talk/xmpp/constants.h" -#include "talk/xmpp/fakexmppclient.h" -#include "talk/xmpp/jid.h" -#include "talk/xmpp/pubsubclient.h" +#include "webrtc/libjingle/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/fakexmppclient.h" +#include "webrtc/libjingle/xmpp/jid.h" +#include "webrtc/libjingle/xmpp/pubsubclient.h" #include "webrtc/base/faketaskrunner.h" #include "webrtc/base/gunit.h" #include "webrtc/base/sigslot.h" diff --git a/xmpp/pubsubstateclient.cc b/xmpp/pubsubstateclient.cc index 5cd7b1a..2ed7b20 100644 --- a/xmpp/pubsubstateclient.cc +++ b/xmpp/pubsubstateclient.cc @@ -25,7 +25,7 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/xmpp/pubsubstateclient.h" +#include "webrtc/libjingle/xmpp/pubsubstateclient.h" namespace buzz { diff --git a/xmpp/pubsubstateclient.h b/xmpp/pubsubstateclient.h index 09ef0f4..79b1329 100644 --- a/xmpp/pubsubstateclient.h +++ b/xmpp/pubsubstateclient.h @@ -34,9 +34,9 @@ #include "webrtc/libjingle/xmllite/qname.h" #include "webrtc/libjingle/xmllite/xmlelement.h" -#include "talk/xmpp/constants.h" -#include "talk/xmpp/jid.h" -#include "talk/xmpp/pubsubclient.h" +#include "webrtc/libjingle/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/jid.h" +#include "webrtc/libjingle/xmpp/pubsubclient.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/base/sigslot.h" #include "webrtc/base/sigslotrepeater.h" diff --git a/xmpp/pubsubtasks.cc b/xmpp/pubsubtasks.cc index 015708e..091a7ad 100644 --- a/xmpp/pubsubtasks.cc +++ b/xmpp/pubsubtasks.cc @@ -25,13 +25,13 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/xmpp/pubsubtasks.h" +#include "webrtc/libjingle/xmpp/pubsubtasks.h" #include <string> #include <vector> -#include "talk/xmpp/constants.h" -#include "talk/xmpp/receivetask.h" +#include "webrtc/libjingle/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/receivetask.h" // An implementation of the tasks for XEP-0060 // (http://xmpp.org/extensions/xep-0060.html). diff --git a/xmpp/pubsubtasks.h b/xmpp/pubsubtasks.h index 94be276..280c613 100644 --- a/xmpp/pubsubtasks.h +++ b/xmpp/pubsubtasks.h @@ -30,8 +30,8 @@ #include <vector> -#include "talk/xmpp/iqtask.h" -#include "talk/xmpp/receivetask.h" +#include "webrtc/libjingle/xmpp/iqtask.h" +#include "webrtc/libjingle/xmpp/receivetask.h" #include "webrtc/base/sigslot.h" namespace buzz { diff --git a/xmpp/pubsubtasks_unittest.cc b/xmpp/pubsubtasks_unittest.cc index 48cd04a..7ea21e7 100644 --- a/xmpp/pubsubtasks_unittest.cc +++ b/xmpp/pubsubtasks_unittest.cc @@ -5,11 +5,11 @@ #include "webrtc/libjingle/xmllite/qname.h" #include "webrtc/libjingle/xmllite/xmlelement.h" -#include "talk/xmpp/constants.h" -#include "talk/xmpp/fakexmppclient.h" -#include "talk/xmpp/iqtask.h" -#include "talk/xmpp/jid.h" -#include "talk/xmpp/pubsubtasks.h" +#include "webrtc/libjingle/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/fakexmppclient.h" +#include "webrtc/libjingle/xmpp/iqtask.h" +#include "webrtc/libjingle/xmpp/jid.h" +#include "webrtc/libjingle/xmpp/pubsubtasks.h" #include "webrtc/base/faketaskrunner.h" #include "webrtc/base/gunit.h" #include "webrtc/base/sigslot.h" diff --git a/xmpp/receivetask.cc b/xmpp/receivetask.cc index c2fb244..4c4fae5 100644 --- a/xmpp/receivetask.cc +++ b/xmpp/receivetask.cc @@ -25,8 +25,8 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/xmpp/constants.h" -#include "talk/xmpp/receivetask.h" +#include "webrtc/libjingle/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/receivetask.h" namespace buzz { diff --git a/xmpp/receivetask.h b/xmpp/receivetask.h index b18e0f0..7055339 100644 --- a/xmpp/receivetask.h +++ b/xmpp/receivetask.h @@ -28,7 +28,7 @@ #ifndef TALK_XMPP_RECEIVETASK_H_ #define TALK_XMPP_RECEIVETASK_H_ -#include "talk/xmpp/xmpptask.h" +#include "webrtc/libjingle/xmpp/xmpptask.h" namespace buzz { diff --git a/xmpp/rostermodule.h b/xmpp/rostermodule.h index dfb647d..9ae7f70 100644 --- a/xmpp/rostermodule.h +++ b/xmpp/rostermodule.h @@ -28,7 +28,7 @@ #ifndef TALK_XMPP_ROSTERMODULE_H_ #define TALK_XMPP_ROSTERMODULE_H_ -#include "talk/xmpp/module.h" +#include "webrtc/libjingle/xmpp/module.h" namespace buzz { diff --git a/xmpp/rostermodule_unittest.cc b/xmpp/rostermodule_unittest.cc index cb7f773..fb82546 100644 --- a/xmpp/rostermodule_unittest.cc +++ b/xmpp/rostermodule_unittest.cc @@ -30,10 +30,10 @@ #include <string> #include "webrtc/libjingle/xmllite/xmlelement.h" -#include "talk/xmpp/constants.h" -#include "talk/xmpp/rostermodule.h" -#include "talk/xmpp/util_unittest.h" -#include "talk/xmpp/xmppengine.h" +#include "webrtc/libjingle/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/rostermodule.h" +#include "webrtc/libjingle/xmpp/util_unittest.h" +#include "webrtc/libjingle/xmpp/xmppengine.h" #include "webrtc/base/gunit.h" #include "webrtc/base/scoped_ptr.h" diff --git a/xmpp/rostermoduleimpl.cc b/xmpp/rostermoduleimpl.cc index e52e78e..8841db6 100644 --- a/xmpp/rostermoduleimpl.cc +++ b/xmpp/rostermoduleimpl.cc @@ -31,8 +31,8 @@ #include <sstream> #include <string> #include <vector> -#include "talk/xmpp/constants.h" -#include "talk/xmpp/rostermoduleimpl.h" +#include "webrtc/libjingle/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/rostermoduleimpl.h" #include "webrtc/base/common.h" #include "webrtc/base/stringencode.h" diff --git a/xmpp/rostermoduleimpl.h b/xmpp/rostermoduleimpl.h index 37d1117..efef5ed 100644 --- a/xmpp/rostermoduleimpl.h +++ b/xmpp/rostermoduleimpl.h @@ -28,8 +28,8 @@ #ifndef TALK_XMPP_XMPPTHREAD_H_ #define TALK_XMPP_XMPPTHREAD_H_ -#include "talk/xmpp/moduleimpl.h" -#include "talk/xmpp/rostermodule.h" +#include "webrtc/libjingle/xmpp/moduleimpl.h" +#include "webrtc/libjingle/xmpp/rostermodule.h" namespace buzz { diff --git a/xmpp/saslcookiemechanism.h b/xmpp/saslcookiemechanism.h index eda142c..0f193f2 100644 --- a/xmpp/saslcookiemechanism.h +++ b/xmpp/saslcookiemechanism.h @@ -30,8 +30,8 @@ #include "webrtc/libjingle/xmllite/qname.h" #include "webrtc/libjingle/xmllite/xmlelement.h" -#include "talk/xmpp/constants.h" -#include "talk/xmpp/saslmechanism.h" +#include "webrtc/libjingle/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/saslmechanism.h" namespace buzz { diff --git a/xmpp/saslmechanism.cc b/xmpp/saslmechanism.cc index 8877084..9fb01bc 100644 --- a/xmpp/saslmechanism.cc +++ b/xmpp/saslmechanism.cc @@ -26,8 +26,8 @@ */ #include "webrtc/libjingle/xmllite/xmlelement.h" -#include "talk/xmpp/constants.h" -#include "talk/xmpp/saslmechanism.h" +#include "webrtc/libjingle/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/saslmechanism.h" #include "webrtc/base/base64.h" using rtc::Base64; diff --git a/xmpp/saslplainmechanism.h b/xmpp/saslplainmechanism.h index 36b9bb9..c9d428e 100644 --- a/xmpp/saslplainmechanism.h +++ b/xmpp/saslplainmechanism.h @@ -28,7 +28,7 @@ #ifndef TALK_XMPP_SASLPLAINMECHANISM_H_ #define TALK_XMPP_SASLPLAINMECHANISM_H_ -#include "talk/xmpp/saslmechanism.h" +#include "webrtc/libjingle/xmpp/saslmechanism.h" #include "webrtc/base/cryptstring.h" namespace buzz { diff --git a/xmpp/util_unittest.cc b/xmpp/util_unittest.cc index 3e47d3f..d245efe 100644 --- a/xmpp/util_unittest.cc +++ b/xmpp/util_unittest.cc @@ -5,8 +5,8 @@ #include <sstream> #include <string> #include "webrtc/libjingle/xmllite/xmlelement.h" -#include "talk/xmpp/util_unittest.h" -#include "talk/xmpp/xmppengine.h" +#include "webrtc/libjingle/xmpp/util_unittest.h" +#include "webrtc/libjingle/xmpp/xmppengine.h" #include "webrtc/base/gunit.h" namespace buzz { diff --git a/xmpp/util_unittest.h b/xmpp/util_unittest.h index c9377ff..806b505 100644 --- a/xmpp/util_unittest.h +++ b/xmpp/util_unittest.h @@ -30,7 +30,7 @@ #include <sstream> #include <string> -#include "talk/xmpp/xmppengine.h" +#include "webrtc/libjingle/xmpp/xmppengine.h" namespace buzz { diff --git a/xmpp/xmppauth.cc b/xmpp/xmppauth.cc index d828475..255f3a7 100644 --- a/xmpp/xmppauth.cc +++ b/xmpp/xmppauth.cc @@ -25,13 +25,13 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/xmpp/xmppauth.h" +#include "webrtc/libjingle/xmpp/xmppauth.h" #include <algorithm> -#include "talk/xmpp/constants.h" -#include "talk/xmpp/saslcookiemechanism.h" -#include "talk/xmpp/saslplainmechanism.h" +#include "webrtc/libjingle/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/saslcookiemechanism.h" +#include "webrtc/libjingle/xmpp/saslplainmechanism.h" XmppAuth::XmppAuth() : done_(false) { } diff --git a/xmpp/xmppauth.h b/xmpp/xmppauth.h index fec73e6..9fabd5e 100644 --- a/xmpp/xmppauth.h +++ b/xmpp/xmppauth.h @@ -30,9 +30,9 @@ #include <vector> -#include "talk/xmpp/jid.h" -#include "talk/xmpp/prexmppauth.h" -#include "talk/xmpp/saslhandler.h" +#include "webrtc/libjingle/xmpp/jid.h" +#include "webrtc/libjingle/xmpp/prexmppauth.h" +#include "webrtc/libjingle/xmpp/saslhandler.h" #include "webrtc/base/cryptstring.h" #include "webrtc/base/sigslot.h" diff --git a/xmpp/xmppclient.cc b/xmpp/xmppclient.cc index 27aa5e4..66d1e97 100644 --- a/xmpp/xmppclient.cc +++ b/xmpp/xmppclient.cc @@ -25,12 +25,12 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/xmpp/xmppclient.h" +#include "webrtc/libjingle/xmpp/xmppclient.h" -#include "talk/xmpp/constants.h" -#include "talk/xmpp/plainsaslhandler.h" -#include "talk/xmpp/prexmppauth.h" -#include "talk/xmpp/saslplainmechanism.h" +#include "webrtc/libjingle/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/plainsaslhandler.h" +#include "webrtc/libjingle/xmpp/prexmppauth.h" +#include "webrtc/libjingle/xmpp/saslplainmechanism.h" #include "webrtc/base/logging.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/base/sigslot.h" diff --git a/xmpp/xmppclient.h b/xmpp/xmppclient.h index 53b2d24..74e4030 100644 --- a/xmpp/xmppclient.h +++ b/xmpp/xmppclient.h @@ -29,10 +29,10 @@ #define TALK_XMPP_XMPPCLIENT_H_ #include <string> -#include "talk/xmpp/asyncsocket.h" -#include "talk/xmpp/xmppclientsettings.h" -#include "talk/xmpp/xmppengine.h" -#include "talk/xmpp/xmpptask.h" +#include "webrtc/libjingle/xmpp/asyncsocket.h" +#include "webrtc/libjingle/xmpp/xmppclientsettings.h" +#include "webrtc/libjingle/xmpp/xmppengine.h" +#include "webrtc/libjingle/xmpp/xmpptask.h" #include "webrtc/base/basicdefs.h" #include "webrtc/base/sigslot.h" #include "webrtc/base/task.h" diff --git a/xmpp/xmppclientsettings.h b/xmpp/xmppclientsettings.h index 50028b7..04acf36 100644 --- a/xmpp/xmppclientsettings.h +++ b/xmpp/xmppclientsettings.h @@ -28,8 +28,8 @@ #ifndef TALK_XMPP_XMPPCLIENTSETTINGS_H_ #define TALK_XMPP_XMPPCLIENTSETTINGS_H_ -#include "talk/p2p/base/port.h" -#include "talk/xmpp/xmppengine.h" +#include "webrtc/p2p/base/port.h" +#include "webrtc/libjingle/xmpp/xmppengine.h" #include "webrtc/base/cryptstring.h" namespace buzz { diff --git a/xmpp/xmppengine.h b/xmpp/xmppengine.h index 461e90f..1806ba9 100644 --- a/xmpp/xmppengine.h +++ b/xmpp/xmppengine.h @@ -31,7 +31,7 @@ // also part of the API #include "webrtc/libjingle/xmllite/qname.h" #include "webrtc/libjingle/xmllite/xmlelement.h" -#include "talk/xmpp/jid.h" +#include "webrtc/libjingle/xmpp/jid.h" namespace buzz { diff --git a/xmpp/xmppengine_unittest.cc b/xmpp/xmppengine_unittest.cc index b519a65..6b0dc96 100644 --- a/xmpp/xmppengine_unittest.cc +++ b/xmpp/xmppengine_unittest.cc @@ -5,11 +5,11 @@ #include <sstream> #include <string> #include "webrtc/libjingle/xmllite/xmlelement.h" -#include "talk/xmpp/constants.h" -#include "talk/xmpp/plainsaslhandler.h" -#include "talk/xmpp/saslplainmechanism.h" -#include "talk/xmpp/util_unittest.h" -#include "talk/xmpp/xmppengine.h" +#include "webrtc/libjingle/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/plainsaslhandler.h" +#include "webrtc/libjingle/xmpp/saslplainmechanism.h" +#include "webrtc/libjingle/xmpp/util_unittest.h" +#include "webrtc/libjingle/xmpp/xmppengine.h" #include "webrtc/base/common.h" #include "webrtc/base/gunit.h" diff --git a/xmpp/xmppengineimpl.cc b/xmpp/xmppengineimpl.cc index 5de9de7..98d89e2 100644 --- a/xmpp/xmppengineimpl.cc +++ b/xmpp/xmppengineimpl.cc @@ -25,7 +25,7 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/xmpp/xmppengineimpl.h" +#include "webrtc/libjingle/xmpp/xmppengineimpl.h" #include <algorithm> #include <sstream> @@ -33,9 +33,9 @@ #include "webrtc/libjingle/xmllite/xmlelement.h" #include "webrtc/libjingle/xmllite/xmlprinter.h" -#include "talk/xmpp/constants.h" -#include "talk/xmpp/saslhandler.h" -#include "talk/xmpp/xmpplogintask.h" +#include "webrtc/libjingle/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/saslhandler.h" +#include "webrtc/libjingle/xmpp/xmpplogintask.h" #include "webrtc/base/common.h" namespace buzz { diff --git a/xmpp/xmppengineimpl.h b/xmpp/xmppengineimpl.h index 4eacf2f..a3da795 100644 --- a/xmpp/xmppengineimpl.h +++ b/xmpp/xmppengineimpl.h @@ -30,8 +30,8 @@ #include <sstream> #include <vector> -#include "talk/xmpp/xmppengine.h" -#include "talk/xmpp/xmppstanzaparser.h" +#include "webrtc/libjingle/xmpp/xmppengine.h" +#include "webrtc/libjingle/xmpp/xmppstanzaparser.h" namespace buzz { diff --git a/xmpp/xmppengineimpl_iq.cc b/xmpp/xmppengineimpl_iq.cc index 208e164..d48f021 100644 --- a/xmpp/xmppengineimpl_iq.cc +++ b/xmpp/xmppengineimpl_iq.cc @@ -27,8 +27,8 @@ #include <algorithm> #include <vector> -#include "talk/xmpp/constants.h" -#include "talk/xmpp/xmppengineimpl.h" +#include "webrtc/libjingle/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/xmppengineimpl.h" #include "webrtc/base/common.h" namespace buzz { diff --git a/xmpp/xmpplogintask.cc b/xmpp/xmpplogintask.cc index a48a94c..7788c0d 100644 --- a/xmpp/xmpplogintask.cc +++ b/xmpp/xmpplogintask.cc @@ -25,16 +25,16 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/xmpp/xmpplogintask.h" +#include "webrtc/libjingle/xmpp/xmpplogintask.h" #include <string> #include <vector> #include "webrtc/libjingle/xmllite/xmlelement.h" -#include "talk/xmpp/constants.h" -#include "talk/xmpp/jid.h" -#include "talk/xmpp/saslmechanism.h" -#include "talk/xmpp/xmppengineimpl.h" +#include "webrtc/libjingle/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/jid.h" +#include "webrtc/libjingle/xmpp/saslmechanism.h" +#include "webrtc/libjingle/xmpp/xmppengineimpl.h" #include "webrtc/base/base64.h" #include "webrtc/base/common.h" diff --git a/xmpp/xmpplogintask.h b/xmpp/xmpplogintask.h index 8685658..631ebdf 100644 --- a/xmpp/xmpplogintask.h +++ b/xmpp/xmpplogintask.h @@ -31,8 +31,8 @@ #include <string> #include <vector> -#include "talk/xmpp/jid.h" -#include "talk/xmpp/xmppengine.h" +#include "webrtc/libjingle/xmpp/jid.h" +#include "webrtc/libjingle/xmpp/xmppengine.h" #include "webrtc/base/logging.h" #include "webrtc/base/scoped_ptr.h" diff --git a/xmpp/xmpplogintask_unittest.cc b/xmpp/xmpplogintask_unittest.cc index ae9a554..82973c4 100644 --- a/xmpp/xmpplogintask_unittest.cc +++ b/xmpp/xmpplogintask_unittest.cc @@ -5,11 +5,11 @@ #include <sstream> #include <string> #include "webrtc/libjingle/xmllite/xmlelement.h" -#include "talk/xmpp/constants.h" -#include "talk/xmpp/plainsaslhandler.h" -#include "talk/xmpp/saslplainmechanism.h" -#include "talk/xmpp/util_unittest.h" -#include "talk/xmpp/xmppengine.h" +#include "webrtc/libjingle/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/plainsaslhandler.h" +#include "webrtc/libjingle/xmpp/saslplainmechanism.h" +#include "webrtc/libjingle/xmpp/util_unittest.h" +#include "webrtc/libjingle/xmpp/xmppengine.h" #include "webrtc/base/common.h" #include "webrtc/base/cryptstring.h" #include "webrtc/base/gunit.h" diff --git a/xmpp/xmpppump.cc b/xmpp/xmpppump.cc index cf7aa7b..e9e7823 100644 --- a/xmpp/xmpppump.cc +++ b/xmpp/xmpppump.cc @@ -25,9 +25,9 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/xmpp/xmpppump.h" +#include "webrtc/libjingle/xmpp/xmpppump.h" -#include "talk/xmpp/xmppauth.h" +#include "webrtc/libjingle/xmpp/xmppauth.h" namespace buzz { diff --git a/xmpp/xmpppump.h b/xmpp/xmpppump.h index ff72f54..082afb9 100644 --- a/xmpp/xmpppump.h +++ b/xmpp/xmpppump.h @@ -28,9 +28,9 @@ #ifndef TALK_XMPP_XMPPPUMP_H_ #define TALK_XMPP_XMPPPUMP_H_ -#include "talk/xmpp/xmppclient.h" -#include "talk/xmpp/xmppengine.h" -#include "talk/xmpp/xmpptask.h" +#include "webrtc/libjingle/xmpp/xmppclient.h" +#include "webrtc/libjingle/xmpp/xmppengine.h" +#include "webrtc/libjingle/xmpp/xmpptask.h" #include "webrtc/base/messagequeue.h" #include "webrtc/base/taskrunner.h" #include "webrtc/base/thread.h" diff --git a/xmpp/xmppsocket.h b/xmpp/xmppsocket.h index 3d6df52..240bbe9 100644 --- a/xmpp/xmppsocket.h +++ b/xmpp/xmppsocket.h @@ -28,8 +28,8 @@ #ifndef TALK_XMPP_XMPPSOCKET_H_ #define TALK_XMPP_XMPPSOCKET_H_ -#include "talk/xmpp/asyncsocket.h" -#include "talk/xmpp/xmppengine.h" +#include "webrtc/libjingle/xmpp/asyncsocket.h" +#include "webrtc/libjingle/xmpp/xmppengine.h" #include "webrtc/base/asyncsocket.h" #include "webrtc/base/bytebuffer.h" #include "webrtc/base/sigslot.h" diff --git a/xmpp/xmppstanzaparser.cc b/xmpp/xmppstanzaparser.cc index 4795839..a4ca2ed 100644 --- a/xmpp/xmppstanzaparser.cc +++ b/xmpp/xmppstanzaparser.cc @@ -25,10 +25,10 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/xmpp/xmppstanzaparser.h" +#include "webrtc/libjingle/xmpp/xmppstanzaparser.h" #include "webrtc/libjingle/xmllite/xmlelement.h" -#include "talk/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/constants.h" #include "webrtc/base/common.h" #ifdef EXPAT_RELATIVE_PATH #include "expat.h" diff --git a/xmpp/xmppstanzaparser_unittest.cc b/xmpp/xmppstanzaparser_unittest.cc index 0b114c0..433de5e 100644 --- a/xmpp/xmppstanzaparser_unittest.cc +++ b/xmpp/xmppstanzaparser_unittest.cc @@ -5,7 +5,7 @@ #include <sstream> #include <string> #include "webrtc/libjingle/xmllite/xmlelement.h" -#include "talk/xmpp/xmppstanzaparser.h" +#include "webrtc/libjingle/xmpp/xmppstanzaparser.h" #include "webrtc/base/common.h" #include "webrtc/base/gunit.h" diff --git a/xmpp/xmpptask.cc b/xmpp/xmpptask.cc index c5e86fa..31aba62 100644 --- a/xmpp/xmpptask.cc +++ b/xmpp/xmpptask.cc @@ -25,10 +25,10 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/xmpp/constants.h" -#include "talk/xmpp/xmppclient.h" -#include "talk/xmpp/xmppengine.h" -#include "talk/xmpp/xmpptask.h" +#include "webrtc/libjingle/xmpp/constants.h" +#include "webrtc/libjingle/xmpp/xmppclient.h" +#include "webrtc/libjingle/xmpp/xmppengine.h" +#include "webrtc/libjingle/xmpp/xmpptask.h" namespace buzz { diff --git a/xmpp/xmpptask.h b/xmpp/xmpptask.h index f6dd156..a8d1124 100644 --- a/xmpp/xmpptask.h +++ b/xmpp/xmpptask.h @@ -30,7 +30,7 @@ #include <deque> #include <string> -#include "talk/xmpp/xmppengine.h" +#include "webrtc/libjingle/xmpp/xmppengine.h" #include "webrtc/base/sigslot.h" #include "webrtc/base/task.h" #include "webrtc/base/taskparent.h" diff --git a/xmpp/xmppthread.cc b/xmpp/xmppthread.cc index e67bffe..ad9246b 100644 --- a/xmpp/xmppthread.cc +++ b/xmpp/xmppthread.cc @@ -25,10 +25,10 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/xmpp/xmppthread.h" +#include "webrtc/libjingle/xmpp/xmppthread.h" -#include "talk/xmpp/xmppauth.h" -#include "talk/xmpp/xmppclientsettings.h" +#include "webrtc/libjingle/xmpp/xmppauth.h" +#include "webrtc/libjingle/xmpp/xmppclientsettings.h" namespace buzz { namespace { diff --git a/xmpp/xmppthread.h b/xmpp/xmppthread.h index bfb74d6..5a77f00 100644 --- a/xmpp/xmppthread.h +++ b/xmpp/xmppthread.h @@ -28,10 +28,10 @@ #ifndef TALK_XMPP_XMPPTHREAD_H_ #define TALK_XMPP_XMPPTHREAD_H_ -#include "talk/xmpp/xmppclientsettings.h" -#include "talk/xmpp/xmppengine.h" -#include "talk/xmpp/xmpppump.h" -#include "talk/xmpp/xmppsocket.h" +#include "webrtc/libjingle/xmpp/xmppclientsettings.h" +#include "webrtc/libjingle/xmpp/xmppengine.h" +#include "webrtc/libjingle/xmpp/xmpppump.h" +#include "webrtc/libjingle/xmpp/xmppsocket.h" #include "webrtc/base/thread.h" namespace buzz { |
