* Update libjingle to 50389769.
* Together with "Add texture support for i420 video frame." from
wuchengli@chromium.org.
https://webrtc-codereview.appspot.com/1413004
RISK=P1
TESTED=try bots
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1967004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4489 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/talk/app/webrtc/peerconnection.cc b/talk/app/webrtc/peerconnection.cc
index 6d3417a..c91b7c8 100644
--- a/talk/app/webrtc/peerconnection.cc
+++ b/talk/app/webrtc/peerconnection.cc
@@ -235,17 +235,6 @@
return false;
}
- bool audio_track_exist = false;
- for (size_t j = 0; j < current_streams->count(); ++j) {
- if (!audio_track_exist) {
- audio_track_exist = current_streams->at(j)->GetAudioTracks().size() > 0;
- }
- }
- if (audio_track_exist && (new_stream->GetAudioTracks().size() > 0)) {
- LOG(LS_ERROR) << "AddStream - Currently only one audio track is supported"
- << "per PeerConnection.";
- return false;
- }
return true;
}
diff --git a/talk/app/webrtc/peerconnectioninterface_unittest.cc b/talk/app/webrtc/peerconnectioninterface_unittest.cc
index 698de3e..d743684 100644
--- a/talk/app/webrtc/peerconnectioninterface_unittest.cc
+++ b/talk/app/webrtc/peerconnectioninterface_unittest.cc
@@ -551,20 +551,27 @@
AddVoiceStream(kStreamLabel2);
ASSERT_EQ(2u, pc_->local_streams()->count());
- // Fail to add another stream with audio since we already have an audio track.
+ // Test we can add multiple local streams to one peerconnection.
scoped_refptr<MediaStreamInterface> stream(
pc_factory_->CreateLocalMediaStream(kStreamLabel3));
scoped_refptr<AudioTrackInterface> audio_track(
pc_factory_->CreateAudioTrack(
kStreamLabel3, static_cast<AudioSourceInterface*>(NULL)));
stream->AddTrack(audio_track.get());
- EXPECT_FALSE(pc_->AddStream(stream, NULL));
-
- // Remove the stream with the audio track.
- pc_->RemoveStream(pc_->local_streams()->at(1));
-
- // Test that we now can add the stream with the audio track.
EXPECT_TRUE(pc_->AddStream(stream, NULL));
+ EXPECT_EQ(3u, pc_->local_streams()->count());
+
+ // Remove the third stream.
+ pc_->RemoveStream(pc_->local_streams()->at(2));
+ EXPECT_EQ(2u, pc_->local_streams()->count());
+
+ // Remove the second stream.
+ pc_->RemoveStream(pc_->local_streams()->at(1));
+ EXPECT_EQ(1u, pc_->local_streams()->count());
+
+ // Remove the first stream.
+ pc_->RemoveStream(pc_->local_streams()->at(0));
+ EXPECT_EQ(0u, pc_->local_streams()->count());
}
TEST_F(PeerConnectionInterfaceTest, RemoveStream) {
diff --git a/talk/libjingle.gyp b/talk/libjingle.gyp
index 8317150..f74ddc1 100755
--- a/talk/libjingle.gyp
+++ b/talk/libjingle.gyp
@@ -835,6 +835,8 @@
'media/webrtc/webrtcmediaengine.h',
'media/webrtc/webrtcpassthroughrender.cc',
'media/webrtc/webrtcpassthroughrender.h',
+ 'media/webrtc/webrtctexturevideoframe.cc',
+ 'media/webrtc/webrtctexturevideoframe.h',
'media/webrtc/webrtcvideocapturer.cc',
'media/webrtc/webrtcvideocapturer.h',
'media/webrtc/webrtcvideodecoderfactory.h',
diff --git a/talk/libjingle_tests.gyp b/talk/libjingle_tests.gyp
index 9ec8ca3..9d57503 100755
--- a/talk/libjingle_tests.gyp
+++ b/talk/libjingle_tests.gyp
@@ -402,7 +402,7 @@
}, # target libjingle_peerconnection_unittest
],
'conditions': [
- ['OS=="linux" or OS=="android"', {
+ ['OS=="linux"', {
'targets': [
{
'target_name': 'libjingle_peerconnection_test_jar',
diff --git a/talk/media/base/nullvideoframe.h b/talk/media/base/nullvideoframe.h
index ff29129..26741ad 100644
--- a/talk/media/base/nullvideoframe.h
+++ b/talk/media/base/nullvideoframe.h
@@ -57,6 +57,7 @@
virtual int32 GetYPitch() const { return 0; }
virtual int32 GetUPitch() const { return 0; }
virtual int32 GetVPitch() const { return 0; }
+ virtual void* GetNativeHandle() const { return NULL; }
virtual size_t GetPixelWidth() const { return 1; }
virtual size_t GetPixelHeight() const { return 1; }
diff --git a/talk/media/base/videoframe.h b/talk/media/base/videoframe.h
index 2f641ff..fe5ff01 100644
--- a/talk/media/base/videoframe.h
+++ b/talk/media/base/videoframe.h
@@ -66,16 +66,23 @@
size_t GetChromaWidth() const { return (GetWidth() + 1) / 2; }
size_t GetChromaHeight() const { return (GetHeight() + 1) / 2; }
size_t GetChromaSize() const { return GetUPitch() * GetChromaHeight(); }
+ // These can return NULL if the object is not backed by a buffer.
virtual const uint8 *GetYPlane() const = 0;
virtual const uint8 *GetUPlane() const = 0;
virtual const uint8 *GetVPlane() const = 0;
virtual uint8 *GetYPlane() = 0;
virtual uint8 *GetUPlane() = 0;
virtual uint8 *GetVPlane() = 0;
+
virtual int32 GetYPitch() const = 0;
virtual int32 GetUPitch() const = 0;
virtual int32 GetVPitch() const = 0;
+ // Returns the handle of the underlying video frame. This is used when the
+ // frame is backed by a texture. The object should be destroyed when it is no
+ // longer in use, so the underlying resource can be freed.
+ virtual void* GetNativeHandle() const = 0;
+
// For retrieving the aspect ratio of each pixel. Usually this is 1x1, but
// the aspect_ratio_idc parameter of H.264 can specify non-square pixels.
virtual size_t GetPixelWidth() const = 0;
@@ -165,7 +172,7 @@
bool crop) const;
// Sets the video frame to black.
- bool SetToBlack();
+ virtual bool SetToBlack();
// Tests if sample is valid. Returns true if valid.
static bool Validate(uint32 fourcc, int w, int h, const uint8 *sample,
diff --git a/talk/media/webrtc/fakewebrtcvideoengine.h b/talk/media/webrtc/fakewebrtcvideoengine.h
index 1c406ec..df53904 100644
--- a/talk/media/webrtc/fakewebrtcvideoengine.h
+++ b/talk/media/webrtc/fakewebrtcvideoengine.h
@@ -597,10 +597,8 @@
channels_.erase(channel);
return 0;
}
-#ifdef USE_WEBRTC_DEV_BRANCH
WEBRTC_STUB(RegisterCpuOveruseObserver,
(int channel, webrtc::CpuOveruseObserver* observer));
-#endif
WEBRTC_STUB(ConnectAudioChannel, (const int, const int));
WEBRTC_STUB(DisconnectAudioChannel, (const int));
WEBRTC_FUNC(StartSend, (const int channel)) {
diff --git a/talk/media/webrtc/fakewebrtcvoiceengine.h b/talk/media/webrtc/fakewebrtcvoiceengine.h
index 7202e15..3a8cfb3 100644
--- a/talk/media/webrtc/fakewebrtcvoiceengine.h
+++ b/talk/media/webrtc/fakewebrtcvoiceengine.h
@@ -161,6 +161,14 @@
}
bool IsInited() const { return inited_; }
int GetLastChannel() const { return last_channel_; }
+ int GetChannelFromLocalSsrc(uint32 local_ssrc) const {
+ for (std::map<int, Channel*>::const_iterator iter = channels_.begin();
+ iter != channels_.end(); ++iter) {
+ if (local_ssrc == iter->second->send_ssrc)
+ return iter->first;
+ }
+ return -1;
+ }
int GetNumChannels() const { return channels_.size(); }
bool GetPlayout(int channel) {
return channels_[channel]->playout;
diff --git a/talk/media/webrtc/webrtctexturevideoframe.cc b/talk/media/webrtc/webrtctexturevideoframe.cc
new file mode 100644
index 0000000..08f63a5
--- /dev/null
+++ b/talk/media/webrtc/webrtctexturevideoframe.cc
@@ -0,0 +1,183 @@
+/*
+ * libjingle
+ * Copyright 2013 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.
+ */
+
+#include "talk/media/webrtc/webrtctexturevideoframe.h"
+
+#include "talk/base/common.h"
+#include "talk/base/logging.h"
+#include "talk/base/stream.h"
+
+#define UNIMPLEMENTED \
+ LOG(LS_ERROR) << "Call to unimplemented function "<< __FUNCTION__; \
+ ASSERT(false)
+
+namespace cricket {
+
+WebRtcTextureVideoFrame::WebRtcTextureVideoFrame(
+ webrtc::NativeHandle* handle, int width, int height, int64 elapsed_time,
+ int64 time_stamp)
+ : handle_(handle), width_(width), height_(height),
+ elapsed_time_(elapsed_time), time_stamp_(time_stamp) {}
+
+WebRtcTextureVideoFrame::~WebRtcTextureVideoFrame() {}
+
+bool WebRtcTextureVideoFrame::InitToBlack(
+ int w, int h, size_t pixel_width, size_t pixel_height, int64 elapsed_time,
+ int64 time_stamp) {
+ UNIMPLEMENTED;
+ return false;
+}
+
+bool WebRtcTextureVideoFrame::Reset(
+ uint32 fourcc, int w, int h, int dw, int dh, uint8* sample,
+ size_t sample_size, size_t pixel_width, size_t pixel_height,
+ int64 elapsed_time, int64 time_stamp, int rotation) {
+ UNIMPLEMENTED;
+ return false;
+}
+
+const uint8* WebRtcTextureVideoFrame::GetYPlane() const {
+ UNIMPLEMENTED;
+ return NULL;
+}
+
+const uint8* WebRtcTextureVideoFrame::GetUPlane() const {
+ UNIMPLEMENTED;
+ return NULL;
+}
+
+const uint8* WebRtcTextureVideoFrame::GetVPlane() const {
+ UNIMPLEMENTED;
+ return NULL;
+}
+
+uint8* WebRtcTextureVideoFrame::GetYPlane() {
+ UNIMPLEMENTED;
+ return NULL;
+}
+
+uint8* WebRtcTextureVideoFrame::GetUPlane() {
+ UNIMPLEMENTED;
+ return NULL;
+}
+
+uint8* WebRtcTextureVideoFrame::GetVPlane() {
+ UNIMPLEMENTED;
+ return NULL;
+}
+
+int32 WebRtcTextureVideoFrame::GetYPitch() const {
+ UNIMPLEMENTED;
+ return width_;
+}
+
+int32 WebRtcTextureVideoFrame::GetUPitch() const {
+ UNIMPLEMENTED;
+ return (width_ + 1) / 2;
+}
+
+int32 WebRtcTextureVideoFrame::GetVPitch() const {
+ UNIMPLEMENTED;
+ return (width_ + 1) / 2;
+}
+
+VideoFrame* WebRtcTextureVideoFrame::Copy() const {
+ return new WebRtcTextureVideoFrame(
+ handle_, width_, height_, elapsed_time_, time_stamp_);
+}
+
+bool WebRtcTextureVideoFrame::MakeExclusive() {
+ UNIMPLEMENTED;
+ return false;
+}
+
+size_t WebRtcTextureVideoFrame::CopyToBuffer(uint8* buffer, size_t size) const {
+ UNIMPLEMENTED;
+ return 0;
+}
+
+size_t WebRtcTextureVideoFrame::ConvertToRgbBuffer(
+ uint32 to_fourcc, uint8* buffer, size_t size, int stride_rgb) const {
+ UNIMPLEMENTED;
+ return 0;
+}
+
+bool WebRtcTextureVideoFrame::CopyToPlanes(
+ uint8* dst_y, uint8* dst_u, uint8* dst_v, int32 dst_pitch_y,
+ int32 dst_pitch_u, int32 dst_pitch_v) const {
+ UNIMPLEMENTED;
+ return false;
+}
+
+void WebRtcTextureVideoFrame::CopyToFrame(VideoFrame* dst) const {
+ UNIMPLEMENTED;
+}
+
+talk_base::StreamResult WebRtcTextureVideoFrame::Write(
+ talk_base::StreamInterface* stream, int* error) {
+ UNIMPLEMENTED;
+ return talk_base::SR_ERROR;
+}
+void WebRtcTextureVideoFrame::StretchToPlanes(
+ uint8* dst_y, uint8* dst_u, uint8* dst_v, int32 dst_pitch_y,
+ int32 dst_pitch_u, int32 dst_pitch_v, size_t width, size_t height,
+ bool interpolate, bool vert_crop) const {
+ UNIMPLEMENTED;
+}
+
+size_t WebRtcTextureVideoFrame::StretchToBuffer(
+ size_t dst_width, size_t dst_height, uint8* dst_buffer, size_t size,
+ bool interpolate, bool vert_crop) const {
+ UNIMPLEMENTED;
+ return 0;
+}
+
+void WebRtcTextureVideoFrame::StretchToFrame(
+ VideoFrame* dst, bool interpolate, bool vert_crop) const {
+ UNIMPLEMENTED;
+}
+
+VideoFrame* WebRtcTextureVideoFrame::Stretch(
+ size_t dst_width, size_t dst_height, bool interpolate,
+ bool vert_crop) const {
+ UNIMPLEMENTED;
+ return NULL;
+}
+
+bool WebRtcTextureVideoFrame::SetToBlack() {
+ UNIMPLEMENTED;
+ return false;
+}
+
+VideoFrame* WebRtcTextureVideoFrame::CreateEmptyFrame(
+ int w, int h, size_t pixel_width, size_t pixel_height, int64 elapsed_time,
+ int64 time_stamp) const {
+ UNIMPLEMENTED;
+ return NULL;
+}
+
+} // namespace cricket
diff --git a/talk/media/webrtc/webrtctexturevideoframe.h b/talk/media/webrtc/webrtctexturevideoframe.h
new file mode 100644
index 0000000..05b50f7
--- /dev/null
+++ b/talk/media/webrtc/webrtctexturevideoframe.h
@@ -0,0 +1,120 @@
+/*
+ * libjingle
+ * Copyright 2013 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.
+ */
+
+#ifndef TALK_MEDIA_WEBRTC_WEBRTCTEXTUREVIDEOFRAME_H_
+#define TALK_MEDIA_WEBRTC_WEBRTCTEXTUREVIDEOFRAME_H_
+
+#include "talk/base/refcount.h"
+#include "talk/base/scoped_ref_ptr.h"
+#include "talk/media/base/videoframe.h"
+#ifdef USE_WEBRTC_DEV_BRANCH
+#include "webrtc/common_video/interface/native_handle.h"
+#else
+#include "webrtc/common_video/interface/i420_video_frame.h"
+// Define NativeHandle to an existing type so we don't need to add lots of
+// USE_WEBRTC_DEV_BRANCH.
+#define NativeHandle I420VideoFrame
+#endif
+
+namespace cricket {
+
+// A video frame backed by the texture via a native handle.
+class WebRtcTextureVideoFrame : public VideoFrame {
+ public:
+ WebRtcTextureVideoFrame(webrtc::NativeHandle* handle, int width, int height,
+ int64 elapsed_time, int64 time_stamp);
+ virtual ~WebRtcTextureVideoFrame();
+
+ // From base class VideoFrame.
+ virtual bool InitToBlack(int w, int h, size_t pixel_width,
+ size_t pixel_height, int64 elapsed_time,
+ int64 time_stamp);
+ virtual bool Reset(uint32 fourcc, int w, int h, int dw, int dh, uint8* sample,
+ size_t sample_size, size_t pixel_width,
+ size_t pixel_height, int64 elapsed_time, int64 time_stamp,
+ int rotation);
+ virtual size_t GetWidth() const { return width_; }
+ virtual size_t GetHeight() const { return height_; }
+ virtual const uint8* GetYPlane() const;
+ virtual const uint8* GetUPlane() const;
+ virtual const uint8* GetVPlane() const;
+ virtual uint8* GetYPlane();
+ virtual uint8* GetUPlane();
+ virtual uint8* GetVPlane();
+ virtual int32 GetYPitch() const;
+ virtual int32 GetUPitch() const;
+ virtual int32 GetVPitch() const;
+ virtual size_t GetPixelWidth() const { return 1; }
+ virtual size_t GetPixelHeight() const { return 1; }
+ virtual int64 GetElapsedTime() const { return elapsed_time_; }
+ virtual int64 GetTimeStamp() const { return time_stamp_; }
+ virtual void SetElapsedTime(int64 elapsed_time) {
+ elapsed_time_ = elapsed_time;
+ }
+ virtual void SetTimeStamp(int64 time_stamp) { time_stamp_ = time_stamp; }
+ virtual int GetRotation() const { return 0; }
+ virtual VideoFrame* Copy() const;
+ virtual bool MakeExclusive();
+ virtual size_t CopyToBuffer(uint8* buffer, size_t size) const;
+ virtual size_t ConvertToRgbBuffer(uint32 to_fourcc, uint8* buffer,
+ size_t size, int stride_rgb) const;
+ virtual void* GetNativeHandle() const { return handle_.get(); }
+
+ virtual bool CopyToPlanes(
+ uint8* dst_y, uint8* dst_u, uint8* dst_v,
+ int32 dst_pitch_y, int32 dst_pitch_u, int32 dst_pitch_v) const;
+ virtual void CopyToFrame(VideoFrame* target) const;
+ virtual talk_base::StreamResult Write(talk_base::StreamInterface* stream,
+ int* error);
+ virtual void StretchToPlanes(
+ uint8* y, uint8* u, uint8* v, int32 pitchY, int32 pitchU, int32 pitchV,
+ size_t width, size_t height, bool interpolate, bool crop) const;
+ virtual size_t StretchToBuffer(size_t w, size_t h, uint8* buffer, size_t size,
+ bool interpolate, bool crop) const;
+ virtual void StretchToFrame(VideoFrame* target, bool interpolate,
+ bool crop) const;
+ virtual VideoFrame* Stretch(size_t w, size_t h, bool interpolate,
+ bool crop) const;
+ virtual bool SetToBlack();
+
+ protected:
+ virtual VideoFrame* CreateEmptyFrame(int w, int h, size_t pixel_width,
+ size_t pixel_height, int64 elapsed_time,
+ int64 time_stamp) const;
+
+ private:
+ // The handle of the underlying video frame.
+ talk_base::scoped_refptr<webrtc::NativeHandle> handle_;
+ int width_;
+ int height_;
+ int64 elapsed_time_;
+ int64 time_stamp_;
+};
+
+} // namespace cricket
+
+#endif // TALK_MEDIA_WEBRTC_WEBRTCTEXTUREVIDEOFRAME_H_
diff --git a/talk/media/webrtc/webrtctexturevideoframe_unittest.cc b/talk/media/webrtc/webrtctexturevideoframe_unittest.cc
new file mode 100644
index 0000000..9ac16da
--- /dev/null
+++ b/talk/media/webrtc/webrtctexturevideoframe_unittest.cc
@@ -0,0 +1,84 @@
+/*
+ * libjingle
+ * Copyright 2013 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.
+ */
+
+#include "talk/media/webrtc/webrtctexturevideoframe.h"
+
+#include "talk/base/gunit.h"
+#include "talk/media/base/videocommon.h"
+
+class NativeHandleImpl : public webrtc::NativeHandle {
+ public:
+ NativeHandleImpl() : ref_count_(0) {}
+ virtual ~NativeHandleImpl() {}
+ virtual int32_t AddRef() { return ++ref_count_; }
+ virtual int32_t Release() { return --ref_count_; }
+ virtual void* GetHandle() { return NULL; }
+
+ int32_t ref_count() { return ref_count_; }
+ private:
+ int32_t ref_count_;
+};
+
+TEST(WebRtcTextureVideoFrameTest, InitialValues) {
+ NativeHandleImpl handle;
+ cricket::WebRtcTextureVideoFrame frame(&handle, 640, 480, 100, 200);
+ EXPECT_EQ(&handle, frame.GetNativeHandle());
+ EXPECT_EQ(640u, frame.GetWidth());
+ EXPECT_EQ(480u, frame.GetHeight());
+ EXPECT_EQ(100, frame.GetElapsedTime());
+ EXPECT_EQ(200, frame.GetTimeStamp());
+ frame.SetElapsedTime(300);
+ EXPECT_EQ(300, frame.GetElapsedTime());
+ frame.SetTimeStamp(400);
+ EXPECT_EQ(400, frame.GetTimeStamp());
+}
+
+TEST(WebRtcTextureVideoFrameTest, CopyFrame) {
+ NativeHandleImpl handle;
+ cricket::WebRtcTextureVideoFrame frame1(&handle, 640, 480, 100, 200);
+ cricket::VideoFrame* frame2 = frame1.Copy();
+ EXPECT_EQ(frame1.GetNativeHandle(), frame2->GetNativeHandle());
+ EXPECT_EQ(frame1.GetWidth(), frame2->GetWidth());
+ EXPECT_EQ(frame1.GetHeight(), frame2->GetHeight());
+ EXPECT_EQ(frame1.GetElapsedTime(), frame2->GetElapsedTime());
+ EXPECT_EQ(frame1.GetTimeStamp(), frame2->GetTimeStamp());
+ delete frame2;
+}
+
+TEST(WebRtcTextureVideoFrameTest, RefCount) {
+ NativeHandleImpl handle;
+ EXPECT_EQ(0, handle.ref_count());
+ cricket::WebRtcTextureVideoFrame* frame1 =
+ new cricket::WebRtcTextureVideoFrame(&handle, 640, 480, 100, 200);
+ EXPECT_EQ(1, handle.ref_count());
+ cricket::VideoFrame* frame2 = frame1->Copy();
+ EXPECT_EQ(2, handle.ref_count());
+ delete frame2;
+ EXPECT_EQ(1, handle.ref_count());
+ delete frame1;
+ EXPECT_EQ(0, handle.ref_count());
+}
diff --git a/talk/media/webrtc/webrtcvideoengine.cc b/talk/media/webrtc/webrtcvideoengine.cc
index 10cdd8e..873b249 100644
--- a/talk/media/webrtc/webrtcvideoengine.cc
+++ b/talk/media/webrtc/webrtcvideoengine.cc
@@ -51,10 +51,11 @@
#include "talk/media/base/videocapturer.h"
#include "talk/media/base/videorenderer.h"
#include "talk/media/devices/filevideocapturer.h"
+#include "talk/media/webrtc/webrtcpassthroughrender.h"
+#include "talk/media/webrtc/webrtctexturevideoframe.h"
+#include "talk/media/webrtc/webrtcvideocapturer.h"
#include "talk/media/webrtc/webrtcvideodecoderfactory.h"
#include "talk/media/webrtc/webrtcvideoencoderfactory.h"
-#include "talk/media/webrtc/webrtcpassthroughrender.h"
-#include "talk/media/webrtc/webrtcvideocapturer.h"
#include "talk/media/webrtc/webrtcvideoframe.h"
#include "talk/media/webrtc/webrtcvie.h"
#include "talk/media/webrtc/webrtcvoe.h"
@@ -176,12 +177,15 @@
explicit WebRtcRenderAdapter(VideoRenderer* renderer)
: renderer_(renderer), width_(0), height_(0), watermark_enabled_(false) {
}
+
virtual ~WebRtcRenderAdapter() {
}
+
void set_watermark_enabled(bool enable) {
talk_base::CritScope cs(&crit_);
watermark_enabled_ = enable;
}
+
void SetRenderer(VideoRenderer* renderer) {
talk_base::CritScope cs(&crit_);
renderer_ = renderer;
@@ -198,6 +202,7 @@
}
}
}
+
// Implementation of webrtc::ExternalRenderer.
virtual int FrameSizeChange(unsigned int width, unsigned int height,
unsigned int /*number_of_streams*/) {
@@ -213,14 +218,18 @@
}
return renderer_->SetSize(width_, height_, 0) ? 0 : -1;
}
+
virtual int DeliverFrame(unsigned char* buffer, int buffer_size,
- uint32_t time_stamp, int64_t render_time) {
+ uint32_t time_stamp, int64_t render_time
+#ifdef USE_WEBRTC_DEV_BRANCH
+ , void* handle
+#endif
+ ) {
talk_base::CritScope cs(&crit_);
frame_rate_tracker_.Update(1);
if (renderer_ == NULL) {
return 0;
}
- WebRtcVideoFrame video_frame;
// Convert 90K rtp timestamp to ns timestamp.
int64 rtp_time_stamp_in_ns = (time_stamp / 90) *
talk_base::kNumNanosecsPerMillisec;
@@ -229,9 +238,26 @@
talk_base::kNumNanosecsPerMillisec;
// Send the rtp timestamp to renderer as the VideoFrame timestamp.
// and the render timestamp as the VideoFrame elapsed_time.
+#ifdef USE_WEBRTC_DEV_BRANCH
+ if (handle == NULL) {
+#endif
+ return DeliverBufferFrame(buffer, buffer_size, render_time_stamp_in_ns,
+ rtp_time_stamp_in_ns);
+#ifdef USE_WEBRTC_DEV_BRANCH
+ } else {
+ return DeliverTextureFrame(handle, render_time_stamp_in_ns,
+ rtp_time_stamp_in_ns);
+ }
+#endif
+ }
+
+ virtual bool IsTextureSupported() { return true; }
+
+ int DeliverBufferFrame(unsigned char* buffer, int buffer_size,
+ int64 elapsed_time, int64 time_stamp) {
+ WebRtcVideoFrame video_frame;
video_frame.Attach(buffer, buffer_size, width_, height_,
- 1, 1, render_time_stamp_in_ns,
- rtp_time_stamp_in_ns, 0);
+ 1, 1, elapsed_time, time_stamp, 0);
// Sanity check on decoded frame size.
@@ -247,18 +273,28 @@
return ret;
}
+ int DeliverTextureFrame(void* handle, int64 elapsed_time, int64 time_stamp) {
+ WebRtcTextureVideoFrame video_frame(
+ static_cast<webrtc::NativeHandle*>(handle), width_, height_,
+ elapsed_time, time_stamp);
+ return renderer_->RenderFrame(&video_frame);
+ }
+
unsigned int width() {
talk_base::CritScope cs(&crit_);
return width_;
}
+
unsigned int height() {
talk_base::CritScope cs(&crit_);
return height_;
}
+
int framerate() {
talk_base::CritScope cs(&crit_);
return static_cast<int>(frame_rate_tracker_.units_second());
}
+
VideoRenderer* renderer() {
talk_base::CritScope cs(&crit_);
return renderer_;
diff --git a/talk/media/webrtc/webrtcvideoframe.h b/talk/media/webrtc/webrtcvideoframe.h
index 03a3196..18475a6 100644
--- a/talk/media/webrtc/webrtcvideoframe.h
+++ b/talk/media/webrtc/webrtcvideoframe.h
@@ -106,6 +106,7 @@
virtual int32 GetYPitch() const { return frame()->Width(); }
virtual int32 GetUPitch() const { return (frame()->Width() + 1) / 2; }
virtual int32 GetVPitch() const { return (frame()->Width() + 1) / 2; }
+ virtual void* GetNativeHandle() const { return NULL; }
virtual size_t GetPixelWidth() const { return pixel_width_; }
virtual size_t GetPixelHeight() const { return pixel_height_; }
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index a974d0e..855a9e4 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -548,6 +548,12 @@
}
#endif
+ // Disable the DTMF playout when a tone is sent.
+ // PlayDtmfTone will be used if local playout is needed.
+ if (voe_wrapper_->dtmf()->SetDtmfFeedbackStatus(false) == -1) {
+ LOG_RTCERR1(SetDtmfFeedbackStatus, false);
+ }
+
initialized_ = true;
return true;
}
@@ -675,6 +681,7 @@
options.experimental_aec.Set(false);
#endif
+
LOG(LS_INFO) << "Applying audio options: " << options.ToString();
webrtc::VoEAudioProcessing* voep = voe_wrapper_->processing();
@@ -1490,41 +1497,22 @@
playout_(false),
desired_send_(SEND_NOTHING),
send_(SEND_NOTHING),
- send_ssrc_(0),
- local_renderer_(NULL),
default_receive_ssrc_(0) {
engine->RegisterChannel(this);
LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel "
<< voe_channel();
- // Register external transport
- if (engine->voe()->network()->RegisterExternalTransport(
- voe_channel(), *static_cast<Transport*>(this)) == -1) {
- LOG_RTCERR2(RegisterExternalTransport, voe_channel(), this);
- }
-
- // Enable RTCP (for quality stats and feedback messages)
- EnableRtcp(voe_channel());
-
- // Reset all recv codecs; they will be enabled via SetRecvCodecs.
- ResetRecvCodecs(voe_channel());
-
- // Disable the DTMF playout when a tone is sent.
- // PlayDtmfTone will be used if local playout is needed.
- if (engine->voe()->dtmf()->SetDtmfFeedbackStatus(false) == -1) {
- LOG_RTCERR1(SetDtmfFeedbackStatus, false);
- }
+ ConfigureSendChannel(voe_channel());
}
WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel() {
LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel "
<< voe_channel();
- // DeRegister external transport
- if (engine()->voe()->network()->DeRegisterExternalTransport(
- voe_channel()) == -1) {
- LOG_RTCERR1(DeRegisterExternalTransport, voe_channel());
- }
+ // Remove any remaining send streams, the default channel will be deleted
+ // later.
+ while (!send_channels_.empty())
+ RemoveSendStream(send_channels_.begin()->first);
// Unregister ourselves from the engine.
engine()->UnregisterChannel(this);
@@ -1533,16 +1521,17 @@
RemoveRecvStream(receive_channels_.begin()->first);
}
- // Delete the primary channel.
- if (engine()->voe()->base()->DeleteChannel(voe_channel()) == -1) {
- LOG_RTCERR1(DeleteChannel, voe_channel());
- }
+ // Delete the default channel.
+ DeleteChannel(voe_channel());
}
bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) {
LOG(LS_INFO) << "Setting voice channel options: "
<< options.ToString();
+ // TODO(xians): Add support to set different options for different send
+ // streams after we support multiple APMs.
+
// We retain all of the existing options, and apply the given ones
// on top. This means there is no way to "clear" options such that
// they go back to the engine default.
@@ -1644,11 +1633,17 @@
bool WebRtcVoiceMediaChannel::SetSendCodecs(
const std::vector<AudioCodec>& codecs) {
+ // TODO(xians): Break down this function into SetSendCodecs(channel, codecs)
+ // to support per-channel codecs.
+
// Disable DTMF, VAD, and FEC unless we know the other side wants them.
dtmf_allowed_ = false;
- engine()->voe()->codec()->SetVADStatus(voe_channel(), false);
- engine()->voe()->rtp()->SetNACKStatus(voe_channel(), false, 0);
- engine()->voe()->rtp()->SetFECStatus(voe_channel(), false);
+ for (ChannelMap::iterator iter = send_channels_.begin();
+ iter != send_channels_.end(); ++iter) {
+ engine()->voe()->codec()->SetVADStatus(iter->second.channel, false);
+ engine()->voe()->rtp()->SetNACKStatus(iter->second.channel, false, 0);
+ engine()->voe()->rtp()->SetFECStatus(iter->second.channel, false);
+ }
// Scan through the list to figure out the codec to use for sending, along
// with the proper configuration for VAD and DTMF.
@@ -1701,13 +1696,18 @@
}
}
- // Find the DTMF telephone event "codec" and tell VoiceEngine about it.
+ // Find the DTMF telephone event "codec" and tell VoiceEngine channels
+ // about it.
if (_stricmp(it->name.c_str(), "telephone-event") == 0 ||
_stricmp(it->name.c_str(), "audio/telephone-event") == 0) {
- if (engine()->voe()->dtmf()->SetSendTelephoneEventPayloadType(
- voe_channel(), it->id) == -1) {
- LOG_RTCERR2(SetSendTelephoneEventPayloadType, voe_channel(), it->id);
- return false;
+ for (ChannelMap::iterator iter = send_channels_.begin();
+ iter != send_channels_.end(); ++iter) {
+ if (engine()->voe()->dtmf()->SetSendTelephoneEventPayloadType(
+ iter->second.channel, it->id) == -1) {
+ LOG_RTCERR2(SetSendTelephoneEventPayloadType,
+ iter->second.channel, it->id);
+ return false;
+ }
}
dtmf_allowed_ = true;
}
@@ -1732,28 +1732,35 @@
<< " not supported.";
continue;
}
- // The CN payload type for 8000 Hz clockrate is fixed at 13.
- if (cn_freq != webrtc::kFreq8000Hz) {
- if (engine()->voe()->codec()->SetSendCNPayloadType(voe_channel(),
- it->id, cn_freq) == -1) {
- LOG_RTCERR3(SetSendCNPayloadType, voe_channel(), it->id, cn_freq);
- // TODO(ajm): This failure condition will be removed from VoE.
- // Restore the return here when we update to a new enough webrtc.
- //
- // Not returning false because the SetSendCNPayloadType will fail if
- // the channel is already sending.
- // This can happen if the remote description is applied twice, for
- // example in the case of ROAP on top of JSEP, where both side will
- // send the offer.
+ // Loop through the existing send channels and set the CN payloadtype
+ // and the VAD status.
+ for (ChannelMap::iterator iter = send_channels_.begin();
+ iter != send_channels_.end(); ++iter) {
+ int channel = iter->second.channel;
+ // The CN payload type for 8000 Hz clockrate is fixed at 13.
+ if (cn_freq != webrtc::kFreq8000Hz) {
+ if (engine()->voe()->codec()->SetSendCNPayloadType(
+ channel, it->id, cn_freq) == -1) {
+ LOG_RTCERR3(SetSendCNPayloadType, channel, it->id, cn_freq);
+ // TODO(ajm): This failure condition will be removed from VoE.
+ // Restore the return here when we update to a new enough webrtc.
+ //
+ // Not returning false because the SetSendCNPayloadType will fail if
+ // the channel is already sending.
+ // This can happen if the remote description is applied twice, for
+ // example in the case of ROAP on top of JSEP, where both side will
+ // send the offer.
+ }
}
- }
- // Only turn on VAD if we have a CN payload type that matches the
- // clockrate for the codec we are going to use.
- if (it->clockrate == send_codec.plfreq) {
- LOG(LS_INFO) << "Enabling VAD";
- if (engine()->voe()->codec()->SetVADStatus(voe_channel(), true) == -1) {
- LOG_RTCERR2(SetVADStatus, voe_channel(), true);
- return false;
+
+ // Only turn on VAD if we have a CN payload type that matches the
+ // clockrate for the codec we are going to use.
+ if (it->clockrate == send_codec.plfreq) {
+ LOG(LS_INFO) << "Enabling VAD";
+ if (engine()->voe()->codec()->SetVADStatus(channel, true) == -1) {
+ LOG_RTCERR2(SetVADStatus, channel, true);
+ return false;
+ }
}
}
}
@@ -1773,15 +1780,18 @@
// Enable redundant encoding of the specified codec. Treat any
// failure as a fatal internal error.
LOG(LS_INFO) << "Enabling FEC";
- if (engine()->voe()->rtp()->SetFECStatus(voe_channel(),
- true, it->id) == -1) {
- LOG_RTCERR3(SetFECStatus, voe_channel(), true, it->id);
- return false;
+ for (ChannelMap::iterator iter = send_channels_.begin();
+ iter != send_channels_.end(); ++iter) {
+ if (engine()->voe()->rtp()->SetFECStatus(iter->second.channel,
+ true, it->id) == -1) {
+ LOG_RTCERR3(SetFECStatus, iter->second.channel, true, it->id);
+ return false;
+ }
}
} else {
send_codec = voe_codec;
nack_enabled_ = IsNackEnabled(*it);
- SetNack(send_ssrc_, voe_channel(), nack_enabled_);
+ SetNack(send_channels_, nack_enabled_);
}
first = false;
// Set the codec immediately, since SetVADStatus() depends on whether
@@ -1790,10 +1800,7 @@
return false;
}
}
- for (ChannelMap::iterator it = receive_channels_.begin();
- it != receive_channels_.end(); ++it) {
- SetNack(it->first, it->second.channel, nack_enabled_);
- }
+ SetNack(receive_channels_, nack_enabled_);
// If we're being asked to set an empty list of codecs, due to a buggy client,
@@ -1808,6 +1815,15 @@
return true;
}
+
+void WebRtcVoiceMediaChannel::SetNack(const ChannelMap& channels,
+ bool nack_enabled) {
+ for (ChannelMap::const_iterator it = channels.begin();
+ it != channels.end(); ++it) {
+ SetNack(it->first, it->second.channel, nack_enabled_);
+ }
+}
+
void WebRtcVoiceMediaChannel::SetNack(uint32 ssrc, int channel,
bool nack_enabled) {
if (nack_enabled) {
@@ -1819,17 +1835,32 @@
}
}
-
bool WebRtcVoiceMediaChannel::SetSendCodec(
const webrtc::CodecInst& send_codec) {
LOG(LS_INFO) << "Selected voice codec " << ToString(send_codec)
<< ", bitrate=" << send_codec.rate;
- if (engine()->voe()->codec()->SetSendCodec(voe_channel(),
- send_codec) == -1) {
- LOG_RTCERR2(SetSendCodec, voe_channel(), ToString(send_codec));
+ for (ChannelMap::iterator iter = send_channels_.begin();
+ iter != send_channels_.end(); ++iter) {
+ if (!SetSendCodec(iter->second.channel, send_codec))
+ return false;
+ }
+
+ // All SetSendCodec calls were successful. Update the global state
+ // accordingly.
+ send_codec_.reset(new webrtc::CodecInst(send_codec));
+
+ return true;
+}
+
+bool WebRtcVoiceMediaChannel::SetSendCodec(
+ int channel, const webrtc::CodecInst& send_codec) {
+ LOG(LS_INFO) << "Send channel " << channel << " selected voice codec "
+ << ToString(send_codec) << ", bitrate=" << send_codec.rate;
+
+ if (engine()->voe()->codec()->SetSendCodec(channel, send_codec) == -1) {
+ LOG_RTCERR2(SetSendCodec, channel, ToString(send_codec));
return false;
}
- send_codec_.reset(new webrtc::CodecInst(send_codec));
return true;
}
@@ -1862,10 +1893,14 @@
}
LOG(LS_INFO) << "Enabling audio level header extension with ID " << id;
- if (engine()->voe()->rtp()->SetRTPAudioLevelIndicationStatus(
- voe_channel(), enable, id) == -1) {
- LOG_RTCERR3(SetRTPAudioLevelIndicationStatus, voe_channel(), enable, id);
- return false;
+ for (ChannelMap::const_iterator iter = send_channels_.begin();
+ iter != send_channels_.end(); ++iter) {
+ if (engine()->voe()->rtp()->SetRTPAudioLevelIndicationStatus(
+ iter->second.channel, enable, id) == -1) {
+ LOG_RTCERR3(SetRTPAudioLevelIndicationStatus,
+ iter->second.channel, enable, id);
+ return false;
+ }
}
return true;
@@ -1912,7 +1947,7 @@
bool WebRtcVoiceMediaChannel::SetSend(SendFlags send) {
desired_send_ = send;
- if (send_ssrc_ != 0)
+ if (!send_channels_.empty())
return ChangeSend(desired_send_);
return true;
}
@@ -1930,131 +1965,177 @@
return true;
}
- if (send == SEND_MICROPHONE) {
+ // Change the settings on each send channel.
+ if (send == SEND_MICROPHONE)
engine()->SetOptionOverrides(options_);
- // VoiceEngine resets sequence number when StopSend is called. This
- // sometimes causes libSRTP to complain about packets being
- // replayed. To get around this we store the last sent sequence
- // number and initializes the channel with the next to continue on
- // the same sequence.
- if (sequence_number() != -1) {
- LOG(LS_INFO) << "WebRtcVoiceMediaChannel restores seqnum="
- << sequence_number() + 1;
- if (engine()->voe()->sync()->SetInitSequenceNumber(
- voe_channel(), sequence_number() + 1) == -1) {
- LOG_RTCERR2(SetInitSequenceNumber, voe_channel(),
- sequence_number() + 1);
- }
- }
- if (engine()->voe()->base()->StartSend(voe_channel()) == -1) {
- LOG_RTCERR1(StartSend, voe_channel());
+ // Change the settings on each send channel.
+ for (ChannelMap::iterator iter = send_channels_.begin();
+ iter != send_channels_.end(); ++iter) {
+ if (!ChangeSend(iter->second.channel, send))
return false;
- }
- // It's OK not to have file() here, since we don't need to call Stop if
- // no file is playing.
- if (engine()->voe()->file() &&
- engine()->voe()->file()->StopPlayingFileAsMicrophone(
- voe_channel()) == -1) {
- LOG_RTCERR1(StopPlayingFileAsMicrophone, voe_channel());
- return false;
- }
- } else if (send == SEND_RINGBACKTONE) {
- ASSERT(ringback_tone_);
- if (!ringback_tone_) {
- return false;
- }
- if (engine()->voe()->file() &&
- engine()->voe()->file()->StartPlayingFileAsMicrophone(
- voe_channel(), ringback_tone_.get(), false) != -1) {
- LOG(LS_INFO) << "File StartPlayingFileAsMicrophone Succeeded. channel:"
- << voe_channel();
- } else {
- LOG_RTCERR3(StartPlayingFileAsMicrophone, voe_channel(),
- ringback_tone_.get(), false);
- return false;
- }
- // VoiceEngine resets sequence number when StopSend is called. This
- // sometimes causes libSRTP to complain about packets being
- // replayed. To get around this we store the last sent sequence
- // number and initializes the channel with the next to continue on
- // the same sequence.
- if (sequence_number() != -1) {
- LOG(LS_INFO) << "WebRtcVoiceMediaChannel restores seqnum="
- << sequence_number() + 1;
- if (engine()->voe()->sync()->SetInitSequenceNumber(
- voe_channel(), sequence_number() + 1) == -1) {
- LOG_RTCERR2(SetInitSequenceNumber, voe_channel(),
- sequence_number() + 1);
- }
- }
- if (engine()->voe()->base()->StartSend(voe_channel()) == -1) {
- LOG_RTCERR1(StartSend, voe_channel());
- return false;
- }
- } else { // SEND_NOTHING
- if (engine()->voe()->base()->StopSend(voe_channel()) == -1) {
- LOG_RTCERR1(StopSend, voe_channel());
- }
-
- engine()->ClearOptionOverrides();
}
+
+ // Clear up the options after stopping sending.
+ if (send == SEND_NOTHING)
+ engine()->ClearOptionOverrides();
+
send_ = send;
return true;
}
-bool WebRtcVoiceMediaChannel::AddSendStream(const StreamParams& sp) {
- if (send_ssrc_ != 0) {
- LOG(LS_ERROR) << "WebRtcVoiceMediaChannel supports one sending channel.";
- return false;
- }
-
- if (engine()->voe()->rtp()->SetLocalSSRC(voe_channel(), sp.first_ssrc())
- == -1) {
- LOG_RTCERR2(SetSendSSRC, voe_channel(), sp.first_ssrc());
- return false;
- }
- // Set the SSRC on the receive channels.
- // Receive channels have to have the same SSRC in order to send receiver
- // reports with this SSRC.
- for (ChannelMap::const_iterator it = receive_channels_.begin();
- it != receive_channels_.end(); ++it) {
- int channel_id = it->second.channel;
- if (channel_id != voe_channel()) {
- if (engine()->voe()->rtp()->SetLocalSSRC(channel_id,
- sp.first_ssrc()) != 0) {
- LOG_RTCERR1(SetLocalSSRC, it->first);
- return false;
- }
+bool WebRtcVoiceMediaChannel::ChangeSend(int channel, SendFlags send) {
+ if (send == SEND_MICROPHONE) {
+ if (engine()->voe()->base()->StartSend(channel) == -1) {
+ LOG_RTCERR1(StartSend, channel);
+ return false;
+ }
+ if (engine()->voe()->file() &&
+ engine()->voe()->file()->StopPlayingFileAsMicrophone(channel) == -1) {
+ LOG_RTCERR1(StopPlayingFileAsMicrophone, channel);
+ return false;
+ }
+ } else { // SEND_NOTHING
+ ASSERT(send == SEND_NOTHING);
+ if (engine()->voe()->base()->StopSend(channel) == -1) {
+ LOG_RTCERR1(StopSend, channel);
+ return false;
}
}
- if (engine()->voe()->rtp()->SetRTCP_CNAME(voe_channel(),
- sp.cname.c_str()) == -1) {
- LOG_RTCERR2(SetRTCP_CNAME, voe_channel(), sp.cname);
- return false;
- }
-
- send_ssrc_ = sp.first_ssrc();
- if (desired_send_ != send_)
- return ChangeSend(desired_send_);
-
- if (local_renderer_)
- local_renderer_->AddChannel(voe_channel());
-
return true;
}
-bool WebRtcVoiceMediaChannel::RemoveSendStream(uint32 ssrc) {
- if (ssrc != send_ssrc_) {
+void WebRtcVoiceMediaChannel::ConfigureSendChannel(int channel) {
+ if (engine()->voe()->network()->RegisterExternalTransport(
+ channel, *this) == -1) {
+ LOG_RTCERR2(RegisterExternalTransport, channel, this);
+ }
+
+ // Enable RTCP (for quality stats and feedback messages)
+ EnableRtcp(channel);
+
+ // Reset all recv codecs; they will be enabled via SetRecvCodecs.
+ ResetRecvCodecs(channel);
+}
+
+bool WebRtcVoiceMediaChannel::DeleteChannel(int channel) {
+ if (engine()->voe()->network()->DeRegisterExternalTransport(channel) == -1) {
+ LOG_RTCERR1(DeRegisterExternalTransport, channel);
+ }
+
+ if (engine()->voe()->base()->DeleteChannel(channel) == -1) {
+ LOG_RTCERR1(DeleteChannel, channel);
return false;
}
- if (local_renderer_)
- local_renderer_->RemoveChannel(voe_channel());
+ return true;
+}
- send_ssrc_ = 0;
- ChangeSend(SEND_NOTHING);
+bool WebRtcVoiceMediaChannel::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 = GetSendChannelNum(sp.first_ssrc());
+ if (channel != -1) {
+ LOG(LS_ERROR) << "Stream already exists with ssrc " << sp.first_ssrc();
+ return false;
+ }
+
+ bool default_channel_is_available = true;
+ for (ChannelMap::const_iterator iter = send_channels_.begin();
+ iter != send_channels_.end(); ++iter) {
+ if (IsDefaultChannel(iter->second.channel)) {
+ default_channel_is_available = false;
+ break;
+ }
+ }
+ if (default_channel_is_available) {
+ channel = voe_channel();
+ } else {
+ // Create a new channel for sending audio data.
+ channel = engine()->voe()->base()->CreateChannel();
+ if (channel == -1) {
+ LOG_RTCERR0(CreateChannel);
+ return false;
+ }
+
+ ConfigureSendChannel(channel);
+ }
+
+ // Save the channel to send_channels_, so that RemoveSendStream() can still
+ // delete the channel in case failure happens below.
+ send_channels_[sp.first_ssrc()] = WebRtcVoiceChannelInfo(channel, NULL);
+
+ // Set the send (local) SSRC.
+ // If there are multiple send SSRCs, we can only set the first one here, and
+ // the rest of the SSRC(s) need to be set after SetSendCodec has been called
+ // (with a codec requires multiple SSRC(s)).
+ if (engine()->voe()->rtp()->SetLocalSSRC(channel, sp.first_ssrc()) == -1) {
+ LOG_RTCERR2(SetSendSSRC, channel, sp.first_ssrc());
+ return false;
+ }
+
+ // At this point the channel's local SSRC has been updated. If the channel is
+ // the default channel make sure that all the receive channels are updated as
+ // well. Receive channels have to have the same SSRC as the default channel in
+ // order to send receiver reports with this SSRC.
+ if (IsDefaultChannel(channel)) {
+ for (ChannelMap::const_iterator it = receive_channels_.begin();
+ it != receive_channels_.end(); ++it) {
+ // Only update the SSRC for non-default channels.
+ if (!IsDefaultChannel(it->second.channel)) {
+ if (engine()->voe()->rtp()->SetLocalSSRC(it->second.channel,
+ sp.first_ssrc()) != 0) {
+ LOG_RTCERR2(SetLocalSSRC, it->second.channel, sp.first_ssrc());
+ return false;
+ }
+ }
+ }
+ }
+
+ if (engine()->voe()->rtp()->SetRTCP_CNAME(channel, sp.cname.c_str()) == -1) {
+ LOG_RTCERR2(SetRTCP_CNAME, channel, sp.cname);
+ return false;
+ }
+
+ // Set the current codec to be used for the new channel.
+ if (send_codec_ && !SetSendCodec(channel, *send_codec_))
+ return false;
+
+ return ChangeSend(channel, desired_send_);
+}
+
+bool WebRtcVoiceMediaChannel::RemoveSendStream(uint32 ssrc) {
+ ChannelMap::iterator it = send_channels_.find(ssrc);
+ if (it == send_channels_.end()) {
+ LOG(LS_WARNING) << "Try to remove stream with ssrc " << ssrc
+ << " which doesn't exist.";
+ return false;
+ }
+
+ int channel = it->second.channel;
+ ChangeSend(channel, SEND_NOTHING);
+
+ // Notify the audio renderer that the send channel is going away.
+ if (it->second.renderer)
+ it->second.renderer->RemoveChannel(channel);
+
+ if (IsDefaultChannel(channel)) {
+ // Do not delete the default channel since the receive channels depend on
+ // the default channel, recycle it instead.
+ ChangeSend(channel, SEND_NOTHING);
+ } else {
+ // Clean up and delete the send channel.
+ LOG(LS_INFO) << "Removing audio send stream " << ssrc
+ << " with VoiceEngine channel #" << channel << ".";
+ if (!DeleteChannel(channel))
+ return false;
+ }
+
+ send_channels_.erase(it);
+ if (send_channels_.empty())
+ ChangeSend(SEND_NOTHING);
+
return true;
}
@@ -2157,11 +2238,14 @@
bool WebRtcVoiceMediaChannel::RemoveRecvStream(uint32 ssrc) {
talk_base::CritScope lock(&receive_channels_cs_);
ChannelMap::iterator it = receive_channels_.find(ssrc);
- if (it == receive_channels_.end())
+ if (it == receive_channels_.end()) {
+ LOG(LS_WARNING) << "Try to remove stream with ssrc " << ssrc
+ << " which doesn't exist.";
return false;
+ }
if (ssrc == default_receive_ssrc_) {
- ASSERT(voe_channel() == it->second.channel);
+ ASSERT(IsDefaultChannel(it->second.channel));
// Recycle the default channel is for recv stream.
if (playout_)
SetPlayout(voe_channel(), false);
@@ -2179,16 +2263,9 @@
if (it->second.renderer)
it->second.renderer->RemoveChannel(it->second.channel);
- if (engine()->voe()->network()->DeRegisterExternalTransport(
- it->second.channel) == -1) {
- LOG_RTCERR1(DeRegisterExternalTransport, it->second.channel);
- }
-
LOG(LS_INFO) << "Removing audio stream " << ssrc
- << " with VoiceEngine channel #"
- << it->second.channel << ".";
- if (engine()->voe()->base()->DeleteChannel(it->second.channel) == -1) {
- LOG_RTCERR1(DeleteChannel, voe_channel());
+ << " with VoiceEngine channel #" << it->second.channel << ".";
+ if (!DeleteChannel(it->second.channel)) {
// Erase the entry anyhow.
receive_channels_.erase(it);
return false;
@@ -2224,7 +2301,7 @@
if (it == receive_channels_.end()) {
if (renderer) {
// Return an error if trying to set a valid renderer with an invalid ssrc.
- LOG_RTCERR1(SetRemoteRenderer, ssrc);
+ LOG(LS_ERROR) << "SetRemoteRenderer failed with ssrc "<< ssrc;
return false;
}
@@ -2232,43 +2309,47 @@
return true;
}
+ AudioRenderer* remote_renderer = it->second.renderer;
if (renderer) {
- ASSERT(it->second.renderer == NULL || it->second.renderer == renderer);
- if (!it->second.renderer) {
+ ASSERT(remote_renderer == NULL || remote_renderer == renderer);
+ if (!remote_renderer) {
renderer->AddChannel(it->second.channel);
}
- } else if (it->second.renderer) {
+ } else if (remote_renderer) {
// |renderer| == NULL, remove the channel from the renderer.
- it->second.renderer->RemoveChannel(it->second.channel);
+ remote_renderer->RemoveChannel(it->second.channel);
}
+ // Assign the new value to the struct.
it->second.renderer = renderer;
return true;
}
bool WebRtcVoiceMediaChannel::SetLocalRenderer(uint32 ssrc,
AudioRenderer* renderer) {
- if (!renderer && !local_renderer_)
+ ChannelMap::iterator it = send_channels_.find(ssrc);
+ if (it == send_channels_.end()) {
+ if (renderer) {
+ // Return an error if trying to set a valid renderer with an invalid ssrc.
+ LOG(LS_ERROR) << "SetLocalRenderer failed with ssrc "<< ssrc;
+ return false;
+ }
+
+ // The channel likely has gone away, do nothing.
return true;
-
- int channel = GetSendChannelNum(ssrc);
- if (channel == -1) {
- // Invalidate the |local_renderer_| before quitting.
- if (!renderer)
- local_renderer_ = NULL;
-
- return false;
}
+ AudioRenderer* local_renderer = it->second.renderer;
if (renderer) {
- ASSERT(local_renderer_ == NULL || local_renderer_ == renderer);
- if (!local_renderer_)
- renderer->AddChannel(channel);
- } else {
- local_renderer_->RemoveChannel(channel);
+ ASSERT(local_renderer == NULL || local_renderer == renderer);
+ if (!local_renderer)
+ renderer->AddChannel(it->second.channel);
+ } else if (local_renderer) {
+ local_renderer->RemoveChannel(it->second.channel);
}
- local_renderer_ = renderer;
+ // Assign the new value to the struct.
+ it->second.renderer = renderer;
return true;
}
@@ -2466,15 +2547,16 @@
// Send the event.
if (flags & cricket::DF_SEND) {
- if (send_ssrc_ != ssrc && ssrc != 0) {
+ int channel = (ssrc == 0) ? voe_channel() : GetSendChannelNum(ssrc);
+ if (channel == -1) {
LOG(LS_WARNING) << "InsertDtmf - The specified ssrc "
<< ssrc << " is not in use.";
return false;
}
// Send DTMF using out-of-band DTMF. ("true", as 3rd arg)
- if (engine()->voe()->dtmf()->SendTelephoneEvent(voe_channel(),
- event, true, duration) == -1) {
- LOG_RTCERR4(SendTelephoneEvent, voe_channel(), event, true, duration);
+ if (engine()->voe()->dtmf()->SendTelephoneEvent(
+ channel, event, true, duration) == -1) {
+ LOG_RTCERR4(SendTelephoneEvent, channel, event, true, duration);
return false;
}
}
@@ -2525,27 +2607,56 @@
}
void WebRtcVoiceMediaChannel::OnRtcpReceived(talk_base::Buffer* packet) {
- // See above.
- int which_channel = GetReceiveChannelNum(
- ParseSsrc(packet->data(), packet->length(), true));
- if (which_channel == -1) {
- which_channel = voe_channel();
+ // Sending channels need all RTCP packets with feedback information.
+ // Even sender reports can contain attached report blocks.
+ // Receiving channels need sender reports in order to create
+ // correct receiver reports.
+ int type = 0;
+ if (!GetRtcpType(packet->data(), packet->length(), &type)) {
+ LOG(LS_WARNING) << "Failed to parse type from received RTCP packet";
+ return;
}
- engine()->voe()->network()->ReceivedRTCPPacket(
- which_channel,
- packet->data(),
- static_cast<unsigned int>(packet->length()));
+ // If it is a sender report, find the channel that is listening.
+ bool has_sent_to_default_channel = false;
+ if (type == kRtcpTypeSR) {
+ int which_channel = GetReceiveChannelNum(
+ ParseSsrc(packet->data(), packet->length(), true));
+ if (which_channel != -1) {
+ engine()->voe()->network()->ReceivedRTCPPacket(
+ which_channel,
+ packet->data(),
+ static_cast<unsigned int>(packet->length()));
+
+ if (IsDefaultChannel(which_channel))
+ has_sent_to_default_channel = true;
+ }
+ }
+
+ // SR may continue RR and any RR entry may correspond to any one of the send
+ // channels. So all RTCP packets must be forwarded all send channels. VoE
+ // will filter out RR internally.
+ for (ChannelMap::iterator iter = send_channels_.begin();
+ iter != send_channels_.end(); ++iter) {
+ // Make sure not sending the same packet to default channel more than once.
+ if (IsDefaultChannel(iter->second.channel) && has_sent_to_default_channel)
+ continue;
+
+ engine()->voe()->network()->ReceivedRTCPPacket(
+ iter->second.channel,
+ packet->data(),
+ static_cast<unsigned int>(packet->length()));
+ }
}
bool WebRtcVoiceMediaChannel::MuteStream(uint32 ssrc, bool muted) {
- if (send_ssrc_ != ssrc && ssrc != 0) {
+ int channel = (ssrc == 0) ? voe_channel() : GetSendChannelNum(ssrc);
+ if (channel == -1) {
LOG(LS_WARNING) << "The specified ssrc " << ssrc << " is not in use.";
return false;
}
- if (engine()->voe()->volume()->SetInputMute(voe_channel(),
- muted) == -1) {
- LOG_RTCERR2(SetInputMute, voe_channel(), muted);
+ if (engine()->voe()->volume()->SetInputMute(channel, muted) == -1) {
+ LOG_RTCERR2(SetInputMute, channel, muted);
return false;
}
return true;
@@ -2590,96 +2701,106 @@
}
bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
- // In VoiceEngine 3.5, GetRTCPStatistics will return 0 even when it fails,
- // causing the stats to contain garbage information. To prevent this, we
- // zero the stats structure before calling this API.
- // TODO(juberti): Remove this workaround.
+ bool echo_metrics_on = false;
+ // These can take on valid negative values, so use the lowest possible level
+ // as default rather than -1.
+ int echo_return_loss = -100;
+ int echo_return_loss_enhancement = -100;
+ // These can also be negative, but in practice -1 is only used to signal
+ // insufficient data, since the resolution is limited to multiples of 4 ms.
+ int echo_delay_median_ms = -1;
+ int echo_delay_std_ms = -1;
+ if (engine()->voe()->processing()->GetEcMetricsStatus(
+ echo_metrics_on) != -1 && echo_metrics_on) {
+ // TODO(ajm): we may want to use VoECallReport::GetEchoMetricsSummary
+ // here, but it appears to be unsuitable currently. Revisit after this is
+ // investigated: http://b/issue?id=5666755
+ int erl, erle, rerl, anlp;
+ if (engine()->voe()->processing()->GetEchoMetrics(
+ erl, erle, rerl, anlp) != -1) {
+ echo_return_loss = erl;
+ echo_return_loss_enhancement = erle;
+ }
+
+ int median, std;
+ if (engine()->voe()->processing()->GetEcDelayMetrics(median, std) != -1) {
+ echo_delay_median_ms = median;
+ echo_delay_std_ms = std;
+ }
+ }
+
+
webrtc::CallStatistics cs;
unsigned int ssrc;
webrtc::CodecInst codec;
unsigned int level;
- // Fill in the sender info, based on what we know, and what the
- // remote side told us it got from its RTCP report.
- VoiceSenderInfo sinfo;
+ for (ChannelMap::const_iterator channel_iter = send_channels_.begin();
+ channel_iter != send_channels_.end(); ++channel_iter) {
+ const int channel = channel_iter->second.channel;
- // Data we obtain locally.
- memset(&cs, 0, sizeof(cs));
- if (engine()->voe()->rtp()->GetRTCPStatistics(voe_channel(), cs) == -1 ||
- engine()->voe()->rtp()->GetLocalSSRC(voe_channel(), ssrc) == -1) {
- return false;
- }
+ // Fill in the sender info, based on what we know, and what the
+ // remote side told us it got from its RTCP report.
+ VoiceSenderInfo sinfo;
- sinfo.ssrc = ssrc;
- sinfo.codec_name = send_codec_.get() ? send_codec_->plname : "";
- sinfo.bytes_sent = cs.bytesSent;
- sinfo.packets_sent = cs.packetsSent;
- // RTT isn't known until a RTCP report is received. Until then, VoiceEngine
- // returns 0 to indicate an error value.
- sinfo.rtt_ms = (cs.rttMs > 0) ? cs.rttMs : -1;
+ if (engine()->voe()->rtp()->GetRTCPStatistics(channel, cs) == -1 ||
+ engine()->voe()->rtp()->GetLocalSSRC(channel, ssrc) == -1) {
+ continue;
+ }
- // Get data from the last remote RTCP report. Use default values if no data
- // available.
- sinfo.fraction_lost = -1.0;
- sinfo.jitter_ms = -1;
- sinfo.packets_lost = -1;
- sinfo.ext_seqnum = -1;
- std::vector<webrtc::ReportBlock> receive_blocks;
- if (engine()->voe()->rtp()->GetRemoteRTCPReportBlocks(
- voe_channel(), &receive_blocks) != -1 &&
- engine()->voe()->codec()->GetSendCodec(voe_channel(),
- codec) != -1) {
- std::vector<webrtc::ReportBlock>::iterator iter;
- for (iter = receive_blocks.begin(); iter != receive_blocks.end(); ++iter) {
- // Lookup report for send ssrc only.
- if (iter->source_SSRC == sinfo.ssrc) {
- // Convert Q8 to floating point.
- sinfo.fraction_lost = static_cast<float>(iter->fraction_lost) / 256;
- // Convert samples to milliseconds.
- if (codec.plfreq / 1000 > 0) {
- sinfo.jitter_ms = iter->interarrival_jitter / (codec.plfreq / 1000);
+ sinfo.ssrc = ssrc;
+ sinfo.codec_name = send_codec_.get() ? send_codec_->plname : "";
+ sinfo.bytes_sent = cs.bytesSent;
+ sinfo.packets_sent = cs.packetsSent;
+ // RTT isn't known until a RTCP report is received. Until then, VoiceEngine
+ // returns 0 to indicate an error value.
+ sinfo.rtt_ms = (cs.rttMs > 0) ? cs.rttMs : -1;
+
+ // Get data from the last remote RTCP report. Use default values if no data
+ // available.
+ sinfo.fraction_lost = -1.0;
+ sinfo.jitter_ms = -1;
+ sinfo.packets_lost = -1;
+ sinfo.ext_seqnum = -1;
+ std::vector<webrtc::ReportBlock> receive_blocks;
+ if (engine()->voe()->rtp()->GetRemoteRTCPReportBlocks(
+ channel, &receive_blocks) != -1 &&
+ engine()->voe()->codec()->GetSendCodec(channel, codec) != -1) {
+ std::vector<webrtc::ReportBlock>::iterator iter;
+ for (iter = receive_blocks.begin(); iter != receive_blocks.end();
+ ++iter) {
+ // Lookup report for send ssrc only.
+ if (iter->source_SSRC == sinfo.ssrc) {
+ // Convert Q8 to floating point.
+ sinfo.fraction_lost = static_cast<float>(iter->fraction_lost) / 256;
+ // Convert samples to milliseconds.
+ if (codec.plfreq / 1000 > 0) {
+ sinfo.jitter_ms = iter->interarrival_jitter / (codec.plfreq / 1000);
+ }
+ sinfo.packets_lost = iter->cumulative_num_packets_lost;
+ sinfo.ext_seqnum = iter->extended_highest_sequence_number;
+ break;
}
- sinfo.packets_lost = iter->cumulative_num_packets_lost;
- sinfo.ext_seqnum = iter->extended_highest_sequence_number;
- break;
}
}
+
+ // Local speech level.
+ sinfo.audio_level = (engine()->voe()->volume()->
+ GetSpeechInputLevelFullRange(level) != -1) ? level : -1;
+
+ // TODO(xians): We are injecting the same APM logging to all the send
+ // channels here because there is no good way to know which send channel
+ // is using the APM. The correct fix is to allow the send channels to have
+ // their own APM so that we can feed the correct APM logging to different
+ // send channels. See issue crbug/264611 .
+ sinfo.echo_return_loss = echo_return_loss;
+ sinfo.echo_return_loss_enhancement = echo_return_loss_enhancement;
+ sinfo.echo_delay_median_ms = echo_delay_median_ms;
+ sinfo.echo_delay_std_ms = echo_delay_std_ms;
+
+ info->senders.push_back(sinfo);
}
- // Local speech level.
- sinfo.audio_level = (engine()->voe()->volume()->
- GetSpeechInputLevelFullRange(level) != -1) ? level : -1;
-
- bool echo_metrics_on = false;
- // These can take on valid negative values, so use the lowest possible level
- // as default rather than -1.
- sinfo.echo_return_loss = -100;
- sinfo.echo_return_loss_enhancement = -100;
- // These can also be negative, but in practice -1 is only used to signal
- // insufficient data, since the resolution is limited to multiples of 4 ms.
- sinfo.echo_delay_median_ms = -1;
- sinfo.echo_delay_std_ms = -1;
- if (engine()->voe()->processing()->GetEcMetricsStatus(echo_metrics_on) !=
- -1 && echo_metrics_on) {
- // TODO(ajm): we may want to use VoECallReport::GetEchoMetricsSummary
- // here, but it appears to be unsuitable currently. Revisit after this is
- // investigated: http://b/issue?id=5666755
- int erl, erle, rerl, anlp;
- if (engine()->voe()->processing()->GetEchoMetrics(erl, erle, rerl, anlp) !=
- -1) {
- sinfo.echo_return_loss = erl;
- sinfo.echo_return_loss_enhancement = erle;
- }
-
- int median, std;
- if (engine()->voe()->processing()->GetEcDelayMetrics(median, std) != -1) {
- sinfo.echo_delay_median_ms = median;
- sinfo.echo_delay_std_ms = std;
- }
- }
-
- info->senders.push_back(sinfo);
-
// Build the list of receivers, one for each receiving channel, or 1 in
// a 1:1 call.
std::vector<int> channels;
@@ -2749,15 +2870,7 @@
bool WebRtcVoiceMediaChannel::FindSsrc(int channel_num, uint32* ssrc) {
talk_base::CritScope lock(&receive_channels_cs_);
ASSERT(ssrc != NULL);
- if (channel_num == voe_channel()) {
- unsigned local_ssrc = 0;
- // This is a sending channel.
- if (engine()->voe()->rtp()->GetLocalSSRC(
- channel_num, local_ssrc) != -1) {
- *ssrc = local_ssrc;
- }
- return true;
- } else if (channel_num == -1 && send_ != SEND_NOTHING) {
+ if (channel_num == -1 && send_ != SEND_NOTHING) {
// Sometimes the VoiceEngine core will throw error with channel_num = -1.
// This means the error is not limited to a specific channel. Signal the
// message using ssrc=0. If the current channel is sending, use this
@@ -2765,6 +2878,20 @@
*ssrc = 0;
return true;
} else {
+ // Check whether this is a sending channel.
+ for (ChannelMap::const_iterator it = send_channels_.begin();
+ it != send_channels_.end(); ++it) {
+ if (it->second.channel == channel_num) {
+ // This is a sending channel.
+ uint32 local_ssrc = 0;
+ if (engine()->voe()->rtp()->GetLocalSSRC(
+ channel_num, local_ssrc) != -1) {
+ *ssrc = local_ssrc;
+ }
+ return true;
+ }
+ }
+
// Check whether this is a receiving channel.
for (ChannelMap::const_iterator it = receive_channels_.begin();
it != receive_channels_.end(); ++it) {
@@ -2796,7 +2923,11 @@
}
int WebRtcVoiceMediaChannel::GetSendChannelNum(uint32 ssrc) {
- return (ssrc == send_ssrc_) ? voe_channel() : -1;
+ ChannelMap::iterator it = send_channels_.find(ssrc);
+ if (it != send_channels_.end())
+ return it->second.channel;
+
+ return -1;
}
bool WebRtcVoiceMediaChannel::GetRedSendCodec(const AudioCodec& red_codec,
@@ -2848,7 +2979,7 @@
bool WebRtcVoiceMediaChannel::EnableRtcp(int channel) {
if (engine()->voe()->rtp()->SetRTCPStatus(channel, true) == -1) {
- LOG_RTCERR2(SetRTCPStatus, voe_channel(), 1);
+ LOG_RTCERR2(SetRTCPStatus, channel, 1);
return false;
}
// TODO(juberti): Enable VQMon and RTCP XR reports, once we know what
diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h
index 036117f..0c2b613 100644
--- a/talk/media/webrtc/webrtcvoiceengine.h
+++ b/talk/media/webrtc/webrtcvoiceengine.h
@@ -275,7 +275,7 @@
class WebRtcMediaChannel : public T, public webrtc::Transport {
public:
WebRtcMediaChannel(E *engine, int channel)
- : engine_(engine), voe_channel_(channel), sequence_number_(-1) {}
+ : engine_(engine), voe_channel_(channel) {}
E *engine() { return engine_; }
int voe_channel() const { return voe_channel_; }
bool valid() const { return voe_channel_ != -1; }
@@ -283,23 +283,10 @@
protected:
// implements Transport interface
virtual int SendPacket(int channel, const void *data, int len) {
- // We need to store the sequence number to be able to pick up
- // the same sequence when the device is restarted.
- // TODO(oja): Remove when WebRtc has fixed the problem.
- int seq_num;
- if (!GetRtpSeqNum(data, len, &seq_num)) {
- return -1;
- }
- if (sequence_number() == -1) {
- LOG(INFO) << "WebRtcVoiceMediaChannel sends first packet seqnum="
- << seq_num;
- }
-
talk_base::Buffer packet(data, len, kMaxRtpPacketLen);
if (!T::SendPacket(&packet)) {
return -1;
}
- sequence_number_ = seq_num;
return len;
}
@@ -308,14 +295,9 @@
return T::SendRtcp(&packet) ? len : -1;
}
- int sequence_number() const {
- return sequence_number_;
- }
-
private:
E *engine_;
int voe_channel_;
- int sequence_number_;
};
// WebRtcVoiceMediaChannel is an implementation of VoiceMediaChannel that uses
@@ -393,16 +375,24 @@
private:
struct WebRtcVoiceChannelInfo;
+ typedef std::map<uint32, WebRtcVoiceChannelInfo> ChannelMap;
void SetNack(uint32 ssrc, int channel, bool nack_enabled);
+ void SetNack(const ChannelMap& channels, bool nack_enabled);
bool SetSendCodec(const webrtc::CodecInst& send_codec);
+ bool SetSendCodec(int channel, const webrtc::CodecInst& send_codec);
bool ChangePlayout(bool playout);
bool ChangeSend(SendFlags send);
+ bool ChangeSend(int channel, SendFlags send);
+ void ConfigureSendChannel(int channel);
+ bool DeleteChannel(int channel);
bool InConferenceMode() const {
return options_.conference_mode.GetWithDefaultIfUnset(false);
}
+ bool IsDefaultChannel(int channel_id) const {
+ return channel_id == voe_channel();
+ }
- typedef std::map<uint32, WebRtcVoiceChannelInfo> ChannelMap;
talk_base::scoped_ptr<WebRtcSoundclipStream> ringback_tone_;
std::set<int> ringback_channels_; // channels playing ringback
std::vector<AudioCodec> recv_codecs_;
@@ -415,17 +405,14 @@
SendFlags desired_send_;
SendFlags send_;
- // TODO(xians): Add support for multiple send channels.
- uint32 send_ssrc_;
- // Weak pointer to the renderer of the local audio track. It is owned by the
- // track and will set to NULL when the track is going away or channel gets
- // deleted. Used to notify the audio track that the media channel is added
- // or removed.
- AudioRenderer* local_renderer_;
+ // send_channels_ contains the channels which are being used for sending.
+ // When the default channel (voe_channel) is used for sending, it is
+ // contained in send_channels_, otherwise not.
+ ChannelMap send_channels_;
uint32 default_receive_ssrc_;
// Note the default channel (voe_channel()) can reside in both
- // receive_channels_ and send channel in non-conference mode and in that case
- // it will only be there if a non-zero default_receive_ssrc_ is set.
+ // receive_channels_ and send_channels_ in non-conference mode and in that
+ // case it will only be there if a non-zero default_receive_ssrc_ is set.
ChannelMap receive_channels_; // for multiple sources
// receive_channels_ can be read from WebRtc callback thread. Access from
// the WebRtc thread must be synchronized with edits on the worker thread.
diff --git a/talk/media/webrtc/webrtcvoiceengine_unittest.cc b/talk/media/webrtc/webrtcvoiceengine_unittest.cc
index b476b4d..31596cd 100644
--- a/talk/media/webrtc/webrtcvoiceengine_unittest.cc
+++ b/talk/media/webrtc/webrtcvoiceengine_unittest.cc
@@ -121,6 +121,18 @@
}
return result;
}
+ void SetupForMultiSendStream() {
+ EXPECT_TRUE(SetupEngine());
+ // Remove stream added in Setup, which is corresponding to default channel.
+ int default_channel_num = voe_.GetLastChannel();
+ uint32 default_send_ssrc;
+ EXPECT_EQ(0, voe_.GetLocalSSRC(default_channel_num, default_send_ssrc));
+ EXPECT_EQ(kSsrc1, default_send_ssrc);
+ EXPECT_TRUE(channel_->RemoveSendStream(default_send_ssrc));
+
+ // Verify the default channel still exists.
+ EXPECT_EQ(0, voe_.GetLocalSSRC(default_channel_num, default_send_ssrc));
+ }
void DeliverPacket(const void* data, int len) {
talk_base::Buffer packet(data, len);
channel_->OnPacketReceived(&packet);
@@ -205,6 +217,47 @@
}
+ void TestSetSendRtpHeaderExtensions(int channel_id) {
+ std::vector<cricket::RtpHeaderExtension> extensions;
+ bool enable = false;
+ unsigned char id = 0;
+
+ // Ensure audio levels are off by default.
+ EXPECT_EQ(0, voe_.GetRTPAudioLevelIndicationStatus(
+ channel_id, enable, id));
+ EXPECT_FALSE(enable);
+
+ // Ensure unknown extensions won't cause an error.
+ extensions.push_back(cricket::RtpHeaderExtension(
+ "urn:ietf:params:unknowextention", 1));
+ EXPECT_TRUE(channel_->SetSendRtpHeaderExtensions(extensions));
+ EXPECT_EQ(0, voe_.GetRTPAudioLevelIndicationStatus(
+ channel_id, enable, id));
+ EXPECT_FALSE(enable);
+
+ // Ensure audio levels stay off with an empty list of headers.
+ EXPECT_TRUE(channel_->SetSendRtpHeaderExtensions(extensions));
+ EXPECT_EQ(0, voe_.GetRTPAudioLevelIndicationStatus(
+ channel_id, enable, id));
+ EXPECT_FALSE(enable);
+
+ // Ensure audio levels are enabled if the audio-level header is specified.
+ extensions.push_back(cricket::RtpHeaderExtension(
+ "urn:ietf:params:rtp-hdrext:ssrc-audio-level", 8));
+ EXPECT_TRUE(channel_->SetSendRtpHeaderExtensions(extensions));
+ EXPECT_EQ(0, voe_.GetRTPAudioLevelIndicationStatus(
+ channel_id, enable, id));
+ EXPECT_TRUE(enable);
+ EXPECT_EQ(8, id);
+
+ // Ensure audio levels go back off with an empty list.
+ extensions.clear();
+ EXPECT_TRUE(channel_->SetSendRtpHeaderExtensions(extensions));
+ EXPECT_EQ(0, voe_.GetRTPAudioLevelIndicationStatus(
+ channel_id, enable, id));
+ EXPECT_FALSE(enable);
+ }
+
protected:
cricket::FakeWebRtcVoiceEngine voe_;
cricket::FakeWebRtcVoiceEngine voe_sc_;
@@ -1336,43 +1389,7 @@
EXPECT_TRUE(SetupEngine());
std::vector<cricket::RtpHeaderExtension> extensions;
int channel_num = voe_.GetLastChannel();
- bool enable = false;
- unsigned char id = 0;
-
- // Ensure audio levels are off by default.
- EXPECT_EQ(0, voe_.GetRTPAudioLevelIndicationStatus(
- channel_num, enable, id));
- EXPECT_FALSE(enable);
-
- // Ensure unknown extentions won't cause an error.
- extensions.push_back(cricket::RtpHeaderExtension(
- "urn:ietf:params:unknowextention", 1));
- EXPECT_TRUE(channel_->SetSendRtpHeaderExtensions(extensions));
- EXPECT_EQ(0, voe_.GetRTPAudioLevelIndicationStatus(
- channel_num, enable, id));
- EXPECT_FALSE(enable);
-
- // Ensure audio levels stay off with an empty list of headers.
- EXPECT_TRUE(channel_->SetSendRtpHeaderExtensions(extensions));
- EXPECT_EQ(0, voe_.GetRTPAudioLevelIndicationStatus(
- channel_num, enable, id));
- EXPECT_FALSE(enable);
-
- // Ensure audio levels are enabled if the audio-level header is specified.
- extensions.push_back(cricket::RtpHeaderExtension(
- "urn:ietf:params:rtp-hdrext:ssrc-audio-level", 8));
- EXPECT_TRUE(channel_->SetSendRtpHeaderExtensions(extensions));
- EXPECT_EQ(0, voe_.GetRTPAudioLevelIndicationStatus(
- channel_num, enable, id));
- EXPECT_TRUE(enable);
- EXPECT_EQ(8, id);
-
- // Ensure audio levels go back off with an empty list.
- extensions.clear();
- EXPECT_TRUE(channel_->SetSendRtpHeaderExtensions(extensions));
- EXPECT_EQ(0, voe_.GetRTPAudioLevelIndicationStatus(
- channel_num, enable, id));
- EXPECT_FALSE(enable);
+ TestSetSendRtpHeaderExtensions(channel_num);
}
// Test that we can create a channel and start sending/playing out on it.
@@ -1392,9 +1409,169 @@
EXPECT_FALSE(voe_.GetPlayout(channel_num));
}
-// Test that we can add and remove streams, and do proper send/playout.
-// We can receive on multiple streams, but will only send on one.
-TEST_F(WebRtcVoiceEngineTestFake, SendAndPlayoutWithMultipleStreams) {
+// Test that we can add and remove send streams.
+TEST_F(WebRtcVoiceEngineTestFake, CreateAndDeleteMultipleSendStreams) {
+ SetupForMultiSendStream();
+
+ static const uint32 kSsrcs4[] = {1, 2, 3, 4};
+
+ // Set the global state for sending.
+ EXPECT_TRUE(channel_->SetSend(cricket::SEND_MICROPHONE));
+
+ for (unsigned int i = 0; i < ARRAY_SIZE(kSsrcs4); ++i) {
+ EXPECT_TRUE(channel_->AddSendStream(
+ cricket::StreamParams::CreateLegacy(kSsrcs4[i])));
+
+ // Verify that we are in a sending state for all the created streams.
+ int channel_num = voe_.GetChannelFromLocalSsrc(kSsrcs4[i]);
+ EXPECT_TRUE(voe_.GetSend(channel_num));
+ }
+
+ // Remove the first send channel, which is the default channel. It will only
+ // recycle the default channel but not delete it.
+ EXPECT_TRUE(channel_->RemoveSendStream(kSsrcs4[0]));
+ // Stream should already be Removed from the send stream list.
+ EXPECT_FALSE(channel_->RemoveSendStream(kSsrcs4[0]));
+ // But the default still exists.
+ EXPECT_EQ(0, voe_.GetChannelFromLocalSsrc(kSsrcs4[0]));
+
+ // Delete the rest of send channel streams.
+ for (unsigned int i = 1; i < ARRAY_SIZE(kSsrcs4); ++i) {
+ EXPECT_TRUE(channel_->RemoveSendStream(kSsrcs4[i]));
+ // Stream should already be deleted.
+ EXPECT_FALSE(channel_->RemoveSendStream(kSsrcs4[i]));
+ EXPECT_EQ(-1, voe_.GetChannelFromLocalSsrc(kSsrcs4[i]));
+ }
+}
+
+// Test SetSendCodecs correctly configure the codecs in all send streams.
+TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsWithMultipleSendStreams) {
+ SetupForMultiSendStream();
+
+ static const uint32 kSsrcs4[] = {1, 2, 3, 4};
+ // Create send streams.
+ for (unsigned int i = 0; i < ARRAY_SIZE(kSsrcs4); ++i) {
+ EXPECT_TRUE(channel_->AddSendStream(
+ cricket::StreamParams::CreateLegacy(kSsrcs4[i])));
+ }
+
+ std::vector<cricket::AudioCodec> codecs;
+ // Set ISAC(16K) and CN(16K). VAD should be activated.
+ codecs.push_back(kIsacCodec);
+ codecs.push_back(kCn16000Codec);
+ codecs[1].id = 97;
+ EXPECT_TRUE(channel_->SetSendCodecs(codecs));
+
+ // Verify ISAC and VAD are corrected configured on all send channels.
+ webrtc::CodecInst gcodec;
+ for (unsigned int i = 0; i < ARRAY_SIZE(kSsrcs4); ++i) {
+ int channel_num = voe_.GetChannelFromLocalSsrc(kSsrcs4[i]);
+ EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
+ EXPECT_STREQ("ISAC", gcodec.plname);
+ EXPECT_TRUE(voe_.GetVAD(channel_num));
+ EXPECT_EQ(97, voe_.GetSendCNPayloadType(channel_num, true));
+ }
+
+ // Change to PCMU(8K) and CN(16K). VAD should not be activated.
+ codecs[0] = kPcmuCodec;
+ EXPECT_TRUE(channel_->SetSendCodecs(codecs));
+ for (unsigned int i = 0; i < ARRAY_SIZE(kSsrcs4); ++i) {
+ int channel_num = voe_.GetChannelFromLocalSsrc(kSsrcs4[i]);
+ EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
+ EXPECT_STREQ("PCMU", gcodec.plname);
+ EXPECT_FALSE(voe_.GetVAD(channel_num));
+ }
+}
+
+// Test we can SetSend on all send streams correctly.
+TEST_F(WebRtcVoiceEngineTestFake, SetSendWithMultipleSendStreams) {
+ SetupForMultiSendStream();
+
+ static const uint32 kSsrcs4[] = {1, 2, 3, 4};
+ // Create the send channels and they should be a SEND_NOTHING date.
+ for (unsigned int i = 0; i < ARRAY_SIZE(kSsrcs4); ++i) {
+ EXPECT_TRUE(channel_->AddSendStream(
+ cricket::StreamParams::CreateLegacy(kSsrcs4[i])));
+ int channel_num = voe_.GetLastChannel();
+ EXPECT_FALSE(voe_.GetSend(channel_num));
+ }
+
+ // Set the global state for starting sending.
+ EXPECT_TRUE(channel_->SetSend(cricket::SEND_MICROPHONE));
+ for (unsigned int i = 0; i < ARRAY_SIZE(kSsrcs4); ++i) {
+ // Verify that we are in a sending state for all the send streams.
+ int channel_num = voe_.GetChannelFromLocalSsrc(kSsrcs4[i]);
+ EXPECT_TRUE(voe_.GetSend(channel_num));
+ }
+
+ // Set the global state for stopping sending.
+ EXPECT_TRUE(channel_->SetSend(cricket::SEND_NOTHING));
+ for (unsigned int i = 1; i < ARRAY_SIZE(kSsrcs4); ++i) {
+ // Verify that we are in a stop state for all the send streams.
+ int channel_num = voe_.GetChannelFromLocalSsrc(kSsrcs4[i]);
+ EXPECT_FALSE(voe_.GetSend(channel_num));
+ }
+}
+
+// Test we can set the correct statistics on all send streams.
+TEST_F(WebRtcVoiceEngineTestFake, GetStatsWithMultipleSendStreams) {
+ SetupForMultiSendStream();
+
+ static const uint32 kSsrcs4[] = {1, 2, 3, 4};
+ // Create send streams.
+ for (unsigned int i = 0; i < ARRAY_SIZE(kSsrcs4); ++i) {
+ EXPECT_TRUE(channel_->AddSendStream(
+ cricket::StreamParams::CreateLegacy(kSsrcs4[i])));
+ }
+
+ // We need send codec to be set to get all stats.
+ std::vector<cricket::AudioCodec> codecs;
+ codecs.push_back(kPcmuCodec);
+ EXPECT_TRUE(channel_->SetSendCodecs(codecs));
+
+ cricket::VoiceMediaInfo info;
+ EXPECT_EQ(true, channel_->GetStats(&info));
+ EXPECT_EQ(static_cast<size_t>(ARRAY_SIZE(kSsrcs4)), info.senders.size());
+
+ // Verify the statistic information is correct.
+ for (unsigned int i = 0; i < ARRAY_SIZE(kSsrcs4); ++i) {
+ EXPECT_EQ(kSsrcs4[i], info.senders[i].ssrc);
+ EXPECT_EQ(kPcmuCodec.name, info.senders[i].codec_name);
+ EXPECT_EQ(cricket::kIntStatValue, info.senders[i].bytes_sent);
+ EXPECT_EQ(cricket::kIntStatValue, info.senders[i].packets_sent);
+ EXPECT_EQ(cricket::kIntStatValue, info.senders[i].packets_lost);
+ EXPECT_EQ(cricket::kFractionLostStatValue, info.senders[i].fraction_lost);
+ EXPECT_EQ(cricket::kIntStatValue, info.senders[i].ext_seqnum);
+ EXPECT_EQ(cricket::kIntStatValue, info.senders[i].rtt_ms);
+ EXPECT_EQ(cricket::kIntStatValue, info.senders[i].jitter_ms);
+ }
+
+ EXPECT_EQ(1u, info.receivers.size());
+}
+
+// Test that we support setting certain send header extensions on multiple
+// send streams.
+TEST_F(WebRtcVoiceEngineTestFake,
+ SetSendRtpHeaderExtensionsWithMultpleSendStreams) {
+ SetupForMultiSendStream();
+
+ static const uint32 kSsrcs4[] = {1, 2, 3, 4};
+ // Create send streams.
+ for (unsigned int i = 0; i < ARRAY_SIZE(kSsrcs4); ++i) {
+ EXPECT_TRUE(channel_->AddSendStream(
+ cricket::StreamParams::CreateLegacy(kSsrcs4[i])));
+ }
+
+ // Test SendRtpHeaderExtensions on each send channel.
+ for (unsigned int i = 0; i < ARRAY_SIZE(kSsrcs4); ++i) {
+ int channel_num = voe_.GetChannelFromLocalSsrc(kSsrcs4[i]);
+ TestSetSendRtpHeaderExtensions(channel_num);
+ }
+}
+
+// Test that we can add and remove receive streams, and do proper send/playout.
+// We can receive on multiple streams while sending one stream.
+TEST_F(WebRtcVoiceEngineTestFake, PlayoutWithMultipleStreams) {
EXPECT_TRUE(SetupEngine());
int channel_num1 = voe_.GetLastChannel();
diff --git a/talk/p2p/base/sessionmanager.cc b/talk/p2p/base/sessionmanager.cc
index 6b3d60a..8f4ddf2 100644
--- a/talk/p2p/base/sessionmanager.cc
+++ b/talk/p2p/base/sessionmanager.cc
@@ -79,9 +79,16 @@
Session* SessionManager::CreateSession(const std::string& local_name,
const std::string& content_type) {
- return CreateSession(local_name, local_name,
- talk_base::ToString(talk_base::CreateRandomId64()),
- content_type, false);
+ std::string id;
+ return CreateSession(id, local_name, content_type);
+}
+
+Session* SessionManager::CreateSession(const std::string& id,
+ const std::string& local_name,
+ const std::string& content_type) {
+ std::string sid =
+ id.empty() ? talk_base::ToString(talk_base::CreateRandomId64()) : id;
+ return CreateSession(local_name, local_name, sid, content_type, false);
}
Session* SessionManager::CreateSession(
diff --git a/talk/p2p/base/sessionmanager.h b/talk/p2p/base/sessionmanager.h
index dcdf1ed..d88e050 100644
--- a/talk/p2p/base/sessionmanager.h
+++ b/talk/p2p/base/sessionmanager.h
@@ -92,6 +92,10 @@
Session *CreateSession(const std::string& local_name,
const std::string& content_type);
+ Session *CreateSession(const std::string& id,
+ const std::string& local_name,
+ const std::string& content_type);
+
// Destroys the given session.
void DestroySession(Session *session);
diff --git a/talk/session/media/call.cc b/talk/session/media/call.cc
index 7ba6b10..c963c36 100644
--- a/talk/session/media/call.cc
+++ b/talk/session/media/call.cc
@@ -59,6 +59,19 @@
return (it != map.end()) ? it->second : NULL;
}
+
+bool ContentContainsCrypto(const cricket::ContentInfo* content) {
+ if (content != NULL) {
+ const cricket::MediaContentDescription* desc =
+ static_cast<const cricket::MediaContentDescription*>(
+ content->description);
+ if (!desc || desc->cryptos().empty()) {
+ return false;
+ }
+ }
+ return true;
+}
+
}
Call::Call(MediaSessionClient* session_client)
@@ -85,22 +98,16 @@
Session* Call::InitiateSession(const buzz::Jid& to,
const buzz::Jid& initiator,
const CallOptions& options) {
- const SessionDescription* offer = session_client_->CreateOffer(options);
+ std::string id;
+ std::string initiator_name = initiator.Str();
+ return InternalInitiateSession(id, to, initiator_name, options);
+}
- Session* session = session_client_->CreateSession(this);
- session->set_initiator_name(initiator.Str());
-
- AddSession(session, offer);
- session->Initiate(to.Str(), offer);
-
- // After this timeout, terminate the call because the callee isn't
- // answering
- session_client_->session_manager()->signaling_thread()->Clear(this,
- MSG_TERMINATECALL);
- session_client_->session_manager()->signaling_thread()->PostDelayed(
- send_to_voicemail_ ? kSendToVoicemailTimeout : kNoVoicemailTimeout,
- this, MSG_TERMINATECALL);
- return session;
+Session *Call::InitiateSession(const std::string& id,
+ const buzz::Jid& to,
+ const CallOptions& options) {
+ std::string initiator_name;
+ return InternalInitiateSession(id, to, initiator_name, options);
}
void Call::IncomingSession(Session* session, const SessionDescription* offer) {
@@ -1025,4 +1032,66 @@
SignalReceivedTerminateReason(this, session, reason);
}
+// TODO(mdodd): Get ride of this method since all Hangouts are using a secure
+// connection.
+bool Call::secure() const {
+ if (session_client_->secure() == SEC_DISABLED) {
+ return false;
+ }
+
+ bool ret = true;
+ int i = 0;
+
+ MediaSessionMap::const_iterator it;
+ for (it = media_session_map_.begin(); it != media_session_map_.end(); ++it) {
+ LOG_F(LS_VERBOSE) << "session[" << i
+ << "], check local and remote descriptions";
+ i++;
+
+ if (!SessionDescriptionContainsCrypto(
+ it->second.session->local_description()) ||
+ !SessionDescriptionContainsCrypto(
+ it->second.session->remote_description())) {
+ ret = false;
+ break;
+ }
+ }
+
+ LOG_F(LS_VERBOSE) << "secure=" << ret;
+ return ret;
+}
+
+bool Call::SessionDescriptionContainsCrypto(
+ const SessionDescription* sdesc) const {
+ if (sdesc == NULL) {
+ LOG_F(LS_VERBOSE) << "sessionDescription is NULL";
+ return false;
+ }
+
+ return ContentContainsCrypto(sdesc->GetContentByName(CN_AUDIO)) &&
+ ContentContainsCrypto(sdesc->GetContentByName(CN_VIDEO));
+}
+
+Session* Call::InternalInitiateSession(const std::string& id,
+ const buzz::Jid& to,
+ const std::string& initiator_name,
+ const CallOptions& options) {
+ const SessionDescription* offer = session_client_->CreateOffer(options);
+
+ Session* session = session_client_->CreateSession(id, this);
+ session->set_initiator_name(initiator_name);
+
+ AddSession(session, offer);
+ session->Initiate(to.Str(), offer);
+
+ // After this timeout, terminate the call because the callee isn't
+ // answering
+ session_client_->session_manager()->signaling_thread()->Clear(this,
+ MSG_TERMINATECALL);
+ session_client_->session_manager()->signaling_thread()->PostDelayed(
+ send_to_voicemail_ ? kSendToVoicemailTimeout : kNoVoicemailTimeout,
+ this, MSG_TERMINATECALL);
+ return session;
+}
+
} // namespace cricket
diff --git a/talk/session/media/call.h b/talk/session/media/call.h
index a604a2b..9b0a6c9 100644
--- a/talk/session/media/call.h
+++ b/talk/session/media/call.h
@@ -66,6 +66,8 @@
// |initiator| can be empty.
Session* InitiateSession(const buzz::Jid& to, const buzz::Jid& initiator,
const CallOptions& options);
+ Session* InitiateSession(const std::string& id, const buzz::Jid& to,
+ const CallOptions& options);
void AcceptSession(Session* session, const CallOptions& options);
void RejectSession(Session* session);
void TerminateSession(Session* session);
@@ -100,6 +102,8 @@
bool has_video() const { return has_video_; }
bool has_data() const { return has_data_; }
bool muted() const { return muted_; }
+ bool video() const { return has_video_; }
+ bool secure() const;
bool video_muted() const { return video_muted_; }
const std::vector<StreamParams>* GetDataRecvStreams(Session* session) const {
MediaStreams* recv_streams = GetMediaStreams(session);
@@ -222,6 +226,11 @@
void ContinuePlayDTMF();
bool StopScreencastWithoutSendingUpdate(Session* session, uint32 ssrc);
bool StopAllScreencastsWithoutSendingUpdate(Session* session);
+ bool SessionDescriptionContainsCrypto(const SessionDescription* sdesc) const;
+ Session* InternalInitiateSession(const std::string& id,
+ const buzz::Jid& to,
+ const std::string& initiator_name,
+ const CallOptions& options);
uint32 id_;
MediaSessionClient* session_client_;
diff --git a/talk/session/media/channelmanager.cc b/talk/session/media/channelmanager.cc
index 88e7f79..c16b066 100644
--- a/talk/session/media/channelmanager.cc
+++ b/talk/session/media/channelmanager.cc
@@ -48,6 +48,7 @@
#include "talk/media/sctp/sctpdataengine.h"
#endif
#include "talk/session/media/soundclip.h"
+#include "talk/session/media/srtpfilter.h"
namespace cricket {
@@ -138,8 +139,15 @@
}
ChannelManager::~ChannelManager() {
- if (initialized_)
+ if (initialized_) {
Terminate();
+ // If srtp is initialized (done by the Channel) then we must call
+ // srtp_shutdown to free all crypto kernel lists. But we need to make sure
+ // shutdown always called at the end, after channels are destroyed.
+ // ChannelManager d'tor is always called last, it's safe place to call
+ // shutdown.
+ ShutdownSrtp();
+ }
}
bool ChannelManager::SetVideoRtxEnabled(bool enable) {
diff --git a/talk/session/media/mediasessionclient.cc b/talk/session/media/mediasessionclient.cc
index 584c342..265906f 100644
--- a/talk/session/media/mediasessionclient.cc
+++ b/talk/session/media/mediasessionclient.cc
@@ -220,8 +220,13 @@
}
Session *MediaSessionClient::CreateSession(Call *call) {
+ std::string id;
+ return CreateSession(id, call);
+}
+
+Session *MediaSessionClient::CreateSession(const std::string& id, Call* call) {
const std::string& type = NS_JINGLE_RTP;
- Session *session = session_manager_->CreateSession(jid().Str(), type);
+ Session *session = session_manager_->CreateSession(id, jid().Str(), type);
session_map_[session->id()] = call;
return session;
}
diff --git a/talk/session/media/mediasessionclient.h b/talk/session/media/mediasessionclient.h
index c76f0e9..1ade753 100644
--- a/talk/session/media/mediasessionclient.h
+++ b/talk/session/media/mediasessionclient.h
@@ -152,6 +152,7 @@
void OnSessionState(BaseSession *session, BaseSession::State state);
void OnSessionDestroy(Session *session);
Session *CreateSession(Call *call);
+ Session *CreateSession(const std::string& id, Call* call);
Call *FindCallByRemoteName(const std::string &remote_name);
buzz::Jid jid_;
diff --git a/talk/session/media/srtpfilter.cc b/talk/session/media/srtpfilter.cc
index e5104db..8e1c2c1 100644
--- a/talk/session/media/srtpfilter.cc
+++ b/talk/session/media/srtpfilter.cc
@@ -97,6 +97,15 @@
#endif // HAVE_SRTP
}
+// NOTE: This is called from ChannelManager D'tor.
+void ShutdownSrtp() {
+#ifdef HAVE_SRTP
+ // If srtp_dealloc is not executed then this will clear all existing sessions.
+ // This should be called when application is shutting down.
+ SrtpSession::Terminate();
+#endif
+}
+
SrtpFilter::SrtpFilter()
: state_(ST_INIT),
signal_silent_time_in_ms_(0) {
@@ -621,6 +630,17 @@
return true;
}
+void SrtpSession::Terminate() {
+ if (inited_) {
+ int err = srtp_shutdown();
+ if (err) {
+ LOG(LS_ERROR) << "srtp_shutdown failed. err=" << err;
+ return;
+ }
+ inited_ = false;
+ }
+}
+
void SrtpSession::HandleEvent(const srtp_event_data_t* ev) {
switch (ev->event) {
case event_ssrc_collision:
diff --git a/talk/session/media/srtpfilter.h b/talk/session/media/srtpfilter.h
index 7d97eff..9b48dcd 100644
--- a/talk/session/media/srtpfilter.h
+++ b/talk/session/media/srtpfilter.h
@@ -65,6 +65,7 @@
class SrtpStat;
void EnableSrtpDebugging();
+void ShutdownSrtp();
// Class to transform SRTP to/from RTP.
// Initialize by calling SetSend with the local security params, then call
@@ -208,6 +209,9 @@
// Update the silent threshold (in ms) for signaling errors.
void set_signal_silent_time(uint32 signal_silent_time_in_ms);
+ // Calls srtp_shutdown if it's initialized.
+ static void Terminate();
+
sigslot::repeater3<uint32, SrtpFilter::Mode, SrtpFilter::Error>
SignalSrtpError;
diff --git a/talk/xmpp/constants.cc b/talk/xmpp/constants.cc
index 193ae2b..196a1ec 100644
--- a/talk/xmpp/constants.cc
+++ b/talk/xmpp/constants.cc
@@ -119,6 +119,27 @@
const char STR_MUC_ROOM_FEATURE_BROADCAST[] = "broadcast";
const char STR_MUC_ROOM_FEATURE_MULTI_USER_VC[] = "muc_muvc";
+const char STR_ID_TYPE_CONVERSATION[] = "conversation";
+const char NS_GOOGLE_MUC_HANGOUT[] = "google:muc#hangout";
+const StaticQName QN_GOOGLE_MUC_HANGOUT_INVITE =
+ { NS_GOOGLE_MUC_HANGOUT, "invite" };
+const StaticQName QN_GOOGLE_MUC_HANGOUT_INVITE_TYPE =
+ { NS_GOOGLE_MUC_HANGOUT, "invite-type" };
+const StaticQName QN_ATTR_CREATE_ACTIVITY =
+ { STR_EMPTY, "create-activity" };
+const StaticQName QN_GOOGLE_MUC_HANGOUT_PUBLIC =
+ { NS_GOOGLE_MUC_HANGOUT, "public" };
+const StaticQName QN_GOOGLE_MUC_HANGOUT_INVITEE =
+ { NS_GOOGLE_MUC_HANGOUT, "invitee" };
+const StaticQName QN_GOOGLE_MUC_HANGOUT_NOTIFICATION_STATUS =
+ { NS_GOOGLE_MUC_HANGOUT, "notification-status" };
+const StaticQName QN_GOOGLE_MUC_HANGOUT_NOTIFICATION_TYPE = {
+ NS_GOOGLE_MUC_HANGOUT, "notification-type" };
+const StaticQName QN_GOOGLE_MUC_HANGOUT_HANGOUT_START_CONTEXT = {
+ NS_GOOGLE_MUC_HANGOUT, "hangout-start-context" };
+const StaticQName QN_GOOGLE_MUC_HANGOUT_CONVERSATION_ID = {
+ NS_GOOGLE_MUC_HANGOUT, "conversation-id" };
+
const StaticQName QN_STREAM_STREAM = { NS_STREAM, STR_STREAM };
const StaticQName QN_STREAM_FEATURES = { NS_STREAM, "features" };
const StaticQName QN_STREAM_ERROR = { NS_STREAM, "error" };
diff --git a/talk/xmpp/constants.h b/talk/xmpp/constants.h
index e01a798..cd6d2b7 100644
--- a/talk/xmpp/constants.h
+++ b/talk/xmpp/constants.h
@@ -112,6 +112,17 @@
extern const char STR_MUC_ROOM_FEATURE_BROADCAST[];
extern const char STR_MUC_ROOM_FEATURE_MULTI_USER_VC[];
+extern const char STR_ID_TYPE_CONVERSATION[];
+extern const char NS_GOOGLE_MUC_HANGOUT[];
+extern const StaticQName QN_GOOGLE_MUC_HANGOUT_INVITE;
+extern const StaticQName QN_GOOGLE_MUC_HANGOUT_INVITE_TYPE;
+extern const StaticQName QN_ATTR_CREATE_ACTIVITY;
+extern const StaticQName QN_GOOGLE_MUC_HANGOUT_PUBLIC;
+extern const StaticQName QN_GOOGLE_MUC_HANGOUT_INVITEE;
+extern const StaticQName QN_GOOGLE_MUC_HANGOUT_NOTIFICATION_TYPE;
+extern const StaticQName QN_GOOGLE_MUC_HANGOUT_HANGOUT_START_CONTEXT;
+extern const StaticQName QN_GOOGLE_MUC_HANGOUT_CONVERSATION_ID;
+
extern const StaticQName QN_STREAM_STREAM;
extern const StaticQName QN_STREAM_FEATURES;
extern const StaticQName QN_STREAM_ERROR;
diff --git a/talk/xmpp/mucroomdiscoverytask.cc b/talk/xmpp/mucroomdiscoverytask.cc
index e0770fd..a5055d2 100644
--- a/talk/xmpp/mucroomdiscoverytask.cc
+++ b/talk/xmpp/mucroomdiscoverytask.cc
@@ -49,12 +49,20 @@
std::map<std::string, std::string> extended_info;
const XmlElement* identity = query->FirstNamed(QN_DISCO_IDENTITY);
if (identity == NULL || !identity->HasAttr(QN_NAME)) {
- SignalResult(this, false, "", features, extended_info);
+ SignalResult(this, false, "", "", features, extended_info);
return;
}
const std::string name(identity->Attr(QN_NAME));
+ // Get the conversation id
+ const XmlElement* convIdElement =
+ identity->FirstNamed(QN_GOOGLE_MUC_HANGOUT_CONVERSATION_ID);
+ std::string conversation_id;
+ if (convIdElement != NULL) {
+ conversation_id = convIdElement->BodyText();
+ }
+
for (const XmlElement* feature = query->FirstNamed(QN_DISCO_FEATURE);
feature != NULL; feature = feature->NextNamed(QN_DISCO_FEATURE)) {
features.insert(feature->Attr(QN_VAR));
@@ -69,7 +77,7 @@
}
}
- SignalResult(this, true, name, features, extended_info);
+ SignalResult(this, true, name, conversation_id, features, extended_info);
}
} // namespace buzz
diff --git a/talk/xmpp/mucroomdiscoverytask.h b/talk/xmpp/mucroomdiscoverytask.h
index 6e3a21a..4097cc6 100644
--- a/talk/xmpp/mucroomdiscoverytask.h
+++ b/talk/xmpp/mucroomdiscoverytask.h
@@ -41,10 +41,11 @@
MucRoomDiscoveryTask(XmppTaskParentInterface* parent,
const Jid& room_jid);
- // Signal (exists, name, features, extended_info)
- sigslot::signal5<MucRoomDiscoveryTask*,
+ // Signal (exists, name, conversationId, features, extended_info)
+ sigslot::signal6<MucRoomDiscoveryTask*,
bool,
const std::string&,
+ const std::string&,
const std::set<std::string>&,
const std::map<std::string, std::string>& > SignalResult;
diff --git a/talk/xmpp/mucroomdiscoverytask_unittest.cc b/talk/xmpp/mucroomdiscoverytask_unittest.cc
index b88b6f2..354503f 100644
--- a/talk/xmpp/mucroomdiscoverytask_unittest.cc
+++ b/talk/xmpp/mucroomdiscoverytask_unittest.cc
@@ -43,10 +43,12 @@
void OnResult(buzz::MucRoomDiscoveryTask* task,
bool exists,
const std::string& name,
+ const std::string& conversation_id,
const std::set<std::string>& features,
const std::map<std::string, std::string>& extended_info) {
last_exists = exists;
last_name = name;
+ last_conversation_id = conversation_id;
last_features = features;
last_extended_info = extended_info;
}
@@ -58,6 +60,7 @@
bool last_exists;
std::string last_name;
+ std::string last_conversation_id;
std::set<std::string> last_features;
std::map<std::string, std::string> last_extended_info;
int error_count;
@@ -67,7 +70,8 @@
public:
MucRoomDiscoveryTaskTest() :
room_jid("muc-jid-ponies@domain.com"),
- room_name("ponies") {
+ room_name("ponies"),
+ conversation_id("test_conversation_id") {
}
virtual void SetUp() {
@@ -87,6 +91,7 @@
MucRoomDiscoveryListener* listener;
buzz::Jid room_jid;
std::string room_name;
+ std::string conversation_id;
};
TEST_F(MucRoomDiscoveryTaskTest, TestDiscovery) {
@@ -107,12 +112,16 @@
EXPECT_EQ(expected_iq, xmpp_client->sent_stanzas()[0]->Str());
EXPECT_EQ("", listener->last_name);
+ EXPECT_EQ("", listener->last_conversation_id);
std::string response_iq =
"<iq xmlns='jabber:client'"
" from='muc-jid-ponies@domain.com' id='0' type='result'>"
" <info:query xmlns:info='http://jabber.org/protocol/disco#info'>"
- " <info:identity name='ponies'/>"
+ " <info:identity name='ponies'>"
+ " <han:conversation-id xmlns:han='google:muc#hangout'>"
+ "test_conversation_id</han:conversation-id>"
+ " </info:identity>"
" <info:feature var='feature1'/>"
" <info:feature var='feature2'/>"
" <data:x xmlns:data='jabber:x:data'>"
@@ -126,6 +135,7 @@
EXPECT_EQ(true, listener->last_exists);
EXPECT_EQ(room_name, listener->last_name);
+ EXPECT_EQ(conversation_id, listener->last_conversation_id);
EXPECT_EQ(2U, listener->last_features.size());
EXPECT_EQ(1U, listener->last_features.count("feature1"));
EXPECT_EQ(2U, listener->last_extended_info.size());
diff --git a/talk/xmpp/mucroomlookuptask.h b/talk/xmpp/mucroomlookuptask.h
index 60001ff..48f2484 100644
--- a/talk/xmpp/mucroomlookuptask.h
+++ b/talk/xmpp/mucroomlookuptask.h
@@ -46,6 +46,11 @@
class MucRoomLookupTask : public IqTask {
public:
+ enum IdType {
+ ID_TYPE_CONVERSATION,
+ ID_TYPE_HANGOUT
+ };
+
static MucRoomLookupTask*
CreateLookupTaskForRoomName(XmppTaskParentInterface* parent,
const Jid& lookup_server_jid,
diff --git a/webrtc/common_video/common_video.gyp b/webrtc/common_video/common_video.gyp
index 8be60cd..f8b7124 100644
--- a/webrtc/common_video/common_video.gyp
+++ b/webrtc/common_video/common_video.gyp
@@ -54,6 +54,7 @@
],
'sources': [
'interface/i420_video_frame.h',
+ 'interface/texture_video_frame.h',
'i420_video_frame.cc',
'jpeg/include/jpeg.h',
'jpeg/data_manager.cc',
@@ -65,6 +66,7 @@
'libyuv/scaler.cc',
'plane.h',
'plane.cc',
+ 'texture_video_frame.cc'
],
# Silence jpeg struct padding warnings.
'msvs_disabled_warnings': [ 4324, ],
@@ -88,6 +90,7 @@
'libyuv/libyuv_unittest.cc',
'libyuv/scaler_unittest.cc',
'plane_unittest.cc',
+ 'texture_video_frame_unittest.cc'
],
# Disable warnings to enable Win64 build, issue 1323.
'msvs_disabled_warnings': [
diff --git a/webrtc/common_video/i420_video_frame.cc b/webrtc/common_video/i420_video_frame.cc
index 77b7648..e369ffe 100644
--- a/webrtc/common_video/i420_video_frame.cc
+++ b/webrtc/common_video/i420_video_frame.cc
@@ -142,6 +142,8 @@
v_plane_.ResetSize();
}
+void* I420VideoFrame::native_handle() const { return NULL; }
+
int I420VideoFrame::CheckDimensions(int width, int height,
int stride_y, int stride_u, int stride_v) {
int half_width = (width + 1) / 2;
@@ -179,5 +181,4 @@
return NULL;
}
-
} // namespace webrtc
diff --git a/webrtc/common_video/interface/i420_video_frame.h b/webrtc/common_video/interface/i420_video_frame.h
index 5aaf8c0..45f2ec3 100644
--- a/webrtc/common_video/interface/i420_video_frame.h
+++ b/webrtc/common_video/interface/i420_video_frame.h
@@ -16,6 +16,7 @@
// Storing and handling of YUV (I420) video frames.
#include "webrtc/common_video/plane.h"
+#include "webrtc/system_wrappers/interface/scoped_refptr.h"
#include "webrtc/typedefs.h"
/*
@@ -49,74 +50,81 @@
// If required size is bigger than the allocated one, new buffers of adequate
// size will be allocated.
// Return value: 0 on success ,-1 on error.
- int CreateEmptyFrame(int width, int height,
- int stride_y, int stride_u, int stride_v);
+ virtual int CreateEmptyFrame(int width, int height,
+ int stride_y, int stride_u, int stride_v);
// CreateFrame: Sets the frame's members and buffers. If required size is
// bigger than allocated one, new buffers of adequate size will be allocated.
// Return value: 0 on success ,-1 on error.
- int CreateFrame(int size_y, const uint8_t* buffer_y,
- int size_u, const uint8_t* buffer_u,
- int size_v, const uint8_t* buffer_v,
- int width, int height,
- int stride_y, int stride_u, int stride_v);
+ virtual int CreateFrame(int size_y, const uint8_t* buffer_y,
+ int size_u, const uint8_t* buffer_u,
+ int size_v, const uint8_t* buffer_v,
+ int width, int height,
+ int stride_y, int stride_u, int stride_v);
// Copy frame: If required size is bigger than allocated one, new buffers of
// adequate size will be allocated.
// Return value: 0 on success ,-1 on error.
- int CopyFrame(const I420VideoFrame& videoFrame);
+ virtual int CopyFrame(const I420VideoFrame& videoFrame);
// Swap Frame.
- void SwapFrame(I420VideoFrame* videoFrame);
+ virtual void SwapFrame(I420VideoFrame* videoFrame);
// Get pointer to buffer per plane.
- uint8_t* buffer(PlaneType type);
+ virtual uint8_t* buffer(PlaneType type);
// Overloading with const.
- const uint8_t* buffer(PlaneType type) const;
+ virtual const uint8_t* buffer(PlaneType type) const;
// Get allocated size per plane.
- int allocated_size(PlaneType type) const;
+ virtual int allocated_size(PlaneType type) const;
// Get allocated stride per plane.
- int stride(PlaneType type) const;
+ virtual int stride(PlaneType type) const;
// Set frame width.
- int set_width(int width);
+ virtual int set_width(int width);
// Set frame height.
- int set_height(int height);
+ virtual int set_height(int height);
// Get frame width.
- int width() const {return width_;}
+ virtual int width() const {return width_;}
// Get frame height.
- int height() const {return height_;}
+ virtual int height() const {return height_;}
// Set frame timestamp (90kHz).
- void set_timestamp(uint32_t timestamp) {timestamp_ = timestamp;}
+ virtual void set_timestamp(uint32_t timestamp) {timestamp_ = timestamp;}
// Get frame timestamp (90kHz).
- uint32_t timestamp() const {return timestamp_;}
+ virtual uint32_t timestamp() const {return timestamp_;}
// Set render time in miliseconds.
- void set_render_time_ms(int64_t render_time_ms) {render_time_ms_ =
+ virtual void set_render_time_ms(int64_t render_time_ms) {render_time_ms_ =
render_time_ms;}
// Get render time in miliseconds.
- int64_t render_time_ms() const {return render_time_ms_;}
+ virtual int64_t render_time_ms() const {return render_time_ms_;}
// Return true if underlying plane buffers are of zero size, false if not.
- bool IsZeroSize() const;
+ virtual bool IsZeroSize() const;
// Reset underlying plane buffers sizes to 0. This function doesn't
// clear memory.
- void ResetSize();
+ virtual void ResetSize();
+
+ // Return the handle of the underlying video frame. This is used when the
+ // frame is backed by a texture. The object should be destroyed when it is no
+ // longer in use, so the underlying resource can be freed.
+ virtual void* native_handle() const;
+
+ protected:
+ // Verifies legality of parameters.
+ // Return value: 0 on success, -1 on error.
+ virtual int CheckDimensions(int width, int height,
+ int stride_y, int stride_u, int stride_v);
private:
- // Verifies legality of parameters.
- // Return value: 0 on success ,-1 on error.
- int CheckDimensions(int width, int height,
- int stride_y, int stride_u, int stride_v);
// Get the pointer to a specific plane.
const Plane* GetPlane(PlaneType type) const;
// Overloading with non-const.
diff --git a/webrtc/common_video/interface/native_handle.h b/webrtc/common_video/interface/native_handle.h
new file mode 100644
index 0000000..d078d4c
--- /dev/null
+++ b/webrtc/common_video/interface/native_handle.h
@@ -0,0 +1,36 @@
+/*
+ * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef COMMON_VIDEO_INTERFACE_NATIVEHANDLE_H_
+#define COMMON_VIDEO_INTERFACE_NATIVEHANDLE_H_
+
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+// A class to store an opaque handle of the underlying video frame. This is used
+// when the frame is backed by a texture. WebRTC carries the handle in
+// TextureVideoFrame. This object keeps a reference to the handle. The reference
+// is cleared when the object is destroyed. It is important to destroy the
+// object as soon as possible so the texture can be recycled.
+class NativeHandle {
+ public:
+ virtual ~NativeHandle() {}
+ // For scoped_refptr
+ virtual int32_t AddRef() = 0;
+ virtual int32_t Release() = 0;
+
+ // Gets the handle.
+ virtual void* GetHandle() = 0;
+};
+
+} // namespace webrtc
+
+#endif // COMMON_VIDEO_INTERFACE_NATIVEHANDLE_H_
diff --git a/webrtc/common_video/interface/texture_video_frame.h b/webrtc/common_video/interface/texture_video_frame.h
new file mode 100644
index 0000000..e905ea7
--- /dev/null
+++ b/webrtc/common_video/interface/texture_video_frame.h
@@ -0,0 +1,72 @@
+/*
+ * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef COMMON_VIDEO_INTERFACE_TEXTURE_VIDEO_FRAME_H
+#define COMMON_VIDEO_INTERFACE_TEXTURE_VIDEO_FRAME_H
+
+// TextureVideoFrame class
+//
+// Storing and handling of video frames backed by textures.
+
+#include "webrtc/common_video/interface/i420_video_frame.h"
+#include "webrtc/common_video/interface/native_handle.h"
+#include "webrtc/system_wrappers/interface/scoped_refptr.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class TextureVideoFrame : public I420VideoFrame {
+ public:
+ TextureVideoFrame(NativeHandle* handle,
+ int width,
+ int height,
+ uint32_t timestamp,
+ int64_t render_time_ms);
+ virtual ~TextureVideoFrame();
+
+ // I420VideoFrame implementation
+ virtual int CreateEmptyFrame(int width,
+ int height,
+ int stride_y,
+ int stride_u,
+ int stride_v) OVERRIDE;
+ virtual int CreateFrame(int size_y,
+ const uint8_t* buffer_y,
+ int size_u,
+ const uint8_t* buffer_u,
+ int size_v,
+ const uint8_t* buffer_v,
+ int width,
+ int height,
+ int stride_y,
+ int stride_u,
+ int stride_v) OVERRIDE;
+ virtual int CopyFrame(const I420VideoFrame& videoFrame) OVERRIDE;
+ virtual void SwapFrame(I420VideoFrame* videoFrame) OVERRIDE;
+ virtual uint8_t* buffer(PlaneType type) OVERRIDE;
+ virtual const uint8_t* buffer(PlaneType type) const OVERRIDE;
+ virtual int allocated_size(PlaneType type) const OVERRIDE;
+ virtual int stride(PlaneType type) const OVERRIDE;
+ virtual bool IsZeroSize() const OVERRIDE;
+ virtual void ResetSize() OVERRIDE;
+ virtual void* native_handle() const OVERRIDE;
+
+ protected:
+ virtual int CheckDimensions(
+ int width, int height, int stride_y, int stride_u, int stride_v) OVERRIDE;
+
+ private:
+ // An opaque handle that stores the underlying video frame.
+ scoped_refptr<NativeHandle> handle_;
+};
+
+} // namespace webrtc
+
+#endif // COMMON_VIDEO_INTERFACE_TEXTURE_VIDEO_FRAME_H
diff --git a/webrtc/common_video/texture_video_frame.cc b/webrtc/common_video/texture_video_frame.cc
new file mode 100644
index 0000000..ea53dc2
--- /dev/null
+++ b/webrtc/common_video/texture_video_frame.cc
@@ -0,0 +1,108 @@
+/*
+ * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/common_video/interface/texture_video_frame.h"
+
+#include <assert.h>
+
+#include "webrtc/system_wrappers/interface/trace.h"
+
+#define NOTREACHED() \
+ do { \
+ WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, -1, "Not reached"); \
+ assert(false); \
+ } while (0)
+
+namespace webrtc {
+
+TextureVideoFrame::TextureVideoFrame(NativeHandle* handle,
+ int width,
+ int height,
+ uint32_t timestamp,
+ int64_t render_time_ms)
+ : handle_(handle) {
+ set_width(width);
+ set_height(height);
+ set_timestamp(timestamp);
+ set_render_time_ms(render_time_ms);
+}
+
+TextureVideoFrame::~TextureVideoFrame() {}
+
+int TextureVideoFrame::CreateEmptyFrame(int width,
+ int height,
+ int stride_y,
+ int stride_u,
+ int stride_v) {
+ NOTREACHED();
+ return -1;
+}
+
+int TextureVideoFrame::CreateFrame(int size_y,
+ const uint8_t* buffer_y,
+ int size_u,
+ const uint8_t* buffer_u,
+ int size_v,
+ const uint8_t* buffer_v,
+ int width,
+ int height,
+ int stride_y,
+ int stride_u,
+ int stride_v) {
+ NOTREACHED();
+ return -1;
+}
+
+int TextureVideoFrame::CopyFrame(const I420VideoFrame& videoFrame) {
+ NOTREACHED();
+ return -1;
+}
+
+void TextureVideoFrame::SwapFrame(I420VideoFrame* videoFrame) {
+ NOTREACHED();
+}
+
+uint8_t* TextureVideoFrame::buffer(PlaneType type) {
+ NOTREACHED();
+ return NULL;
+}
+
+const uint8_t* TextureVideoFrame::buffer(PlaneType type) const {
+ NOTREACHED();
+ return NULL;
+}
+
+int TextureVideoFrame::allocated_size(PlaneType type) const {
+ NOTREACHED();
+ return -1;
+}
+
+int TextureVideoFrame::stride(PlaneType type) const {
+ NOTREACHED();
+ return -1;
+}
+
+bool TextureVideoFrame::IsZeroSize() const {
+ NOTREACHED();
+ return true;
+}
+
+void TextureVideoFrame::ResetSize() {
+ NOTREACHED();
+}
+
+void* TextureVideoFrame::native_handle() const { return handle_.get(); }
+
+int TextureVideoFrame::CheckDimensions(
+ int width, int height, int stride_y, int stride_u, int stride_v) {
+ return 0;
+}
+
+} // namespace webrtc
diff --git a/webrtc/common_video/texture_video_frame_unittest.cc b/webrtc/common_video/texture_video_frame_unittest.cc
new file mode 100644
index 0000000..04e09a6
--- /dev/null
+++ b/webrtc/common_video/texture_video_frame_unittest.cc
@@ -0,0 +1,58 @@
+/*
+ * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/common_video/interface/native_handle.h"
+#include "webrtc/common_video/interface/texture_video_frame.h"
+
+namespace webrtc {
+
+class NativeHandleImpl : public NativeHandle {
+ public:
+ NativeHandleImpl() : ref_count_(0) {}
+ virtual ~NativeHandleImpl() {}
+ virtual int32_t AddRef() { return ++ref_count_; }
+ virtual int32_t Release() { return --ref_count_; }
+ virtual void* GetHandle() { return NULL; }
+
+ int32_t ref_count() { return ref_count_; }
+ private:
+ int32_t ref_count_;
+};
+
+TEST(TestTextureVideoFrame, InitialValues) {
+ NativeHandleImpl handle;
+ TextureVideoFrame frame(&handle, 640, 480, 100, 10);
+ EXPECT_EQ(640, frame.width());
+ EXPECT_EQ(480, frame.height());
+ EXPECT_EQ(100u, frame.timestamp());
+ EXPECT_EQ(10, frame.render_time_ms());
+ EXPECT_EQ(&handle, frame.native_handle());
+
+ EXPECT_EQ(0, frame.set_width(320));
+ EXPECT_EQ(320, frame.width());
+ EXPECT_EQ(0, frame.set_height(240));
+ EXPECT_EQ(240, frame.height());
+ frame.set_timestamp(200);
+ EXPECT_EQ(200u, frame.timestamp());
+ frame.set_render_time_ms(20);
+ EXPECT_EQ(20, frame.render_time_ms());
+}
+
+TEST(TestTextureVideoFrame, RefCount) {
+ NativeHandleImpl handle;
+ EXPECT_EQ(0, handle.ref_count());
+ TextureVideoFrame *frame = new TextureVideoFrame(&handle, 640, 480, 100, 200);
+ EXPECT_EQ(1, handle.ref_count());
+ delete frame;
+ EXPECT_EQ(0, handle.ref_count());
+}
+
+} // namespace webrtc
diff --git a/webrtc/modules/utility/source/video_frames_queue.cc b/webrtc/modules/utility/source/video_frames_queue.cc
index 535660c..d3d37be 100644
--- a/webrtc/modules/utility/source/video_frames_queue.cc
+++ b/webrtc/modules/utility/source/video_frames_queue.cc
@@ -14,6 +14,7 @@
#include <assert.h>
+#include "webrtc/common_video/interface/texture_video_frame.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/system_wrappers/interface/tick_util.h"
#include "webrtc/system_wrappers/interface/trace.h"
@@ -48,6 +49,16 @@
}
int32_t VideoFramesQueue::AddFrame(const I420VideoFrame& newFrame) {
+ if (newFrame.native_handle() != NULL) {
+ _incomingFrames.PushBack(new TextureVideoFrame(
+ static_cast<NativeHandle*>(newFrame.native_handle()),
+ newFrame.width(),
+ newFrame.height(),
+ newFrame.timestamp(),
+ newFrame.render_time_ms()));
+ return 0;
+ }
+
I420VideoFrame* ptrFrameToAdd = NULL;
// Try to re-use a VideoFrame. Only allocate new memory if it is necessary.
if (!_emptyFrames.Empty()) {
@@ -113,12 +124,17 @@
}
int32_t VideoFramesQueue::ReturnFrame(I420VideoFrame* ptrOldFrame) {
- ptrOldFrame->set_timestamp(0);
- ptrOldFrame->set_width(0);
- ptrOldFrame->set_height(0);
- ptrOldFrame->set_render_time_ms(0);
- ptrOldFrame->ResetSize();
- _emptyFrames.PushBack(ptrOldFrame);
+ // No need to reuse texture frames because they do not allocate memory.
+ if (ptrOldFrame->native_handle() == NULL) {
+ ptrOldFrame->set_timestamp(0);
+ ptrOldFrame->set_width(0);
+ ptrOldFrame->set_height(0);
+ ptrOldFrame->set_render_time_ms(0);
+ ptrOldFrame->ResetSize();
+ _emptyFrames.PushBack(ptrOldFrame);
+ } else {
+ delete ptrOldFrame;
+ }
return 0;
}
diff --git a/webrtc/modules/video_render/incoming_video_stream.cc b/webrtc/modules/video_render/incoming_video_stream.cc
index eb602d1..39556d8 100644
--- a/webrtc/modules/video_render/incoming_video_stream.cc
+++ b/webrtc/modules/video_render/incoming_video_stream.cc
@@ -101,7 +101,8 @@
return -1;
}
- if (true == mirror_frames_enabled_) {
+ // Mirroring is not supported if the frame is backed by a texture.
+ if (true == mirror_frames_enabled_ && video_frame.native_handle() == NULL) {
transformed_video_frame_.CreateEmptyFrame(video_frame.width(),
video_frame.height(),
video_frame.stride(kYPlane),
diff --git a/webrtc/modules/video_render/video_render_frames.cc b/webrtc/modules/video_render/video_render_frames.cc
index 80b3d59..be5cac9 100644
--- a/webrtc/modules/video_render/video_render_frames.cc
+++ b/webrtc/modules/video_render/video_render_frames.cc
@@ -12,6 +12,7 @@
#include <assert.h>
+#include "webrtc/common_video/interface/texture_video_frame.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/system_wrappers/interface/tick_util.h"
#include "webrtc/system_wrappers/interface/trace.h"
@@ -47,6 +48,16 @@
return -1;
}
+ if (new_frame->native_handle() != NULL) {
+ incoming_frames_.PushBack(new TextureVideoFrame(
+ static_cast<NativeHandle*>(new_frame->native_handle()),
+ new_frame->width(),
+ new_frame->height(),
+ new_frame->timestamp(),
+ new_frame->render_time_ms()));
+ return incoming_frames_.GetSize();
+ }
+
// Get an empty frame
I420VideoFrame* frame_to_add = NULL;
if (!empty_frames_.Empty()) {
@@ -103,10 +114,7 @@
// This is the oldest one so far and it's OK to render.
if (render_frame) {
// This one is older than the newly found frame, remove this one.
- render_frame->ResetSize();
- render_frame->set_timestamp(0);
- render_frame->set_render_time_ms(0);
- empty_frames_.PushFront(render_frame);
+ ReturnFrame(render_frame);
}
render_frame = oldest_frame_in_list;
incoming_frames_.Erase(item);
@@ -122,10 +130,15 @@
}
int32_t VideoRenderFrames::ReturnFrame(I420VideoFrame* old_frame) {
- old_frame->ResetSize();
- old_frame->set_timestamp(0);
- old_frame->set_render_time_ms(0);
- empty_frames_.PushBack(old_frame);
+ // No need to reuse texture frames because they do not allocate memory.
+ if (old_frame->native_handle() == NULL) {
+ old_frame->ResetSize();
+ old_frame->set_timestamp(0);
+ old_frame->set_render_time_ms(0);
+ empty_frames_.PushBack(old_frame);
+ } else {
+ delete old_frame;
+ }
return 0;
}
diff --git a/webrtc/video_engine/include/vie_render.h b/webrtc/video_engine/include/vie_render.h
index 24e5926..48afc1a 100644
--- a/webrtc/video_engine/include/vie_render.h
+++ b/webrtc/video_engine/include/vie_render.h
@@ -39,7 +39,13 @@
// RTP timestamp in 90kHz.
uint32_t time_stamp,
// Wallclock render time in miliseconds
- int64_t render_time) = 0;
+ int64_t render_time,
+ // Handle of the underlying video frame,
+ void* handle) = 0;
+
+ // Returns true if the renderer supports textures. DeliverFrame can be called
+ // with NULL |buffer| and non-NULL |handle|.
+ virtual bool IsTextureSupported() = 0;
protected:
virtual ~ExternalRenderer() {}
diff --git a/webrtc/video_engine/internal/video_receive_stream.cc b/webrtc/video_engine/internal/video_receive_stream.cc
index 50f4553..6f23e9c 100644
--- a/webrtc/video_engine/internal/video_receive_stream.cc
+++ b/webrtc/video_engine/internal/video_receive_stream.cc
@@ -117,7 +117,8 @@
}
int VideoReceiveStream::DeliverFrame(uint8_t* frame, int buffer_size,
- uint32_t timestamp, int64_t render_time) {
+ uint32_t timestamp, int64_t render_time,
+ void* /*handle*/) {
if (config_.renderer == NULL) {
return 0;
}
@@ -142,6 +143,8 @@
return 0;
}
+bool VideoReceiveStream::IsTextureSupported() { return false; }
+
int VideoReceiveStream::SendPacket(int /*channel*/,
const void* packet,
int length) {
diff --git a/webrtc/video_engine/internal/video_receive_stream.h b/webrtc/video_engine/internal/video_receive_stream.h
index 932776a..cdac6fb 100644
--- a/webrtc/video_engine/internal/video_receive_stream.h
+++ b/webrtc/video_engine/internal/video_receive_stream.h
@@ -46,7 +46,9 @@
virtual int FrameSizeChange(unsigned int width, unsigned int height,
unsigned int /*number_of_streams*/) OVERRIDE;
virtual int DeliverFrame(uint8_t* frame, int buffer_size, uint32_t timestamp,
- int64_t render_time) OVERRIDE;
+ int64_t render_time, void* /*handle*/) OVERRIDE;
+
+ virtual bool IsTextureSupported() OVERRIDE;
virtual int SendPacket(int /*channel*/, const void* packet, int length)
OVERRIDE;
diff --git a/webrtc/video_engine/test/auto_test/primitives/framedrop_primitives.cc b/webrtc/video_engine/test/auto_test/primitives/framedrop_primitives.cc
index e49be53..a79f7d5 100644
--- a/webrtc/video_engine/test/auto_test/primitives/framedrop_primitives.cc
+++ b/webrtc/video_engine/test/auto_test/primitives/framedrop_primitives.cc
@@ -588,7 +588,7 @@
int FrameDropMonitoringRemoteFileRenderer::DeliverFrame(
unsigned char *buffer, int buffer_size, uint32_t time_stamp,
- int64_t render_time) {
+ int64_t render_time, void* /*handle*/) {
// |render_time| provides the ideal render time for this frame. If that time
// has already passed we will render it immediately.
int64_t report_render_time_us = render_time * 1000;
@@ -600,7 +600,7 @@
frame_drop_detector_->ReportFrameState(FrameDropDetector::kRendered,
time_stamp, report_render_time_us);
return ViEToFileRenderer::DeliverFrame(buffer, buffer_size,
- time_stamp, render_time);
+ time_stamp, render_time, NULL);
}
int FrameDropMonitoringRemoteFileRenderer::FrameSizeChange(
diff --git a/webrtc/video_engine/test/auto_test/primitives/framedrop_primitives.h b/webrtc/video_engine/test/auto_test/primitives/framedrop_primitives.h
index 3a60520..b507784 100644
--- a/webrtc/video_engine/test/auto_test/primitives/framedrop_primitives.h
+++ b/webrtc/video_engine/test/auto_test/primitives/framedrop_primitives.h
@@ -223,10 +223,11 @@
// Implementation of ExternalRenderer:
int FrameSizeChange(unsigned int width, unsigned int height,
- unsigned int number_of_streams);
+ unsigned int number_of_streams) OVERRIDE;
int DeliverFrame(unsigned char* buffer, int buffer_size,
uint32_t time_stamp,
- int64_t render_time);
+ int64_t render_time,
+ void* handle) OVERRIDE;
private:
FrameDropDetector* frame_drop_detector_;
};
diff --git a/webrtc/video_engine/test/auto_test/source/vie_autotest_render.cc b/webrtc/video_engine/test/auto_test/source/vie_autotest_render.cc
index c4b8155..c1d2fac 100644
--- a/webrtc/video_engine/test/auto_test/source/vie_autotest_render.cc
+++ b/webrtc/video_engine/test/auto_test/source/vie_autotest_render.cc
@@ -58,7 +58,8 @@
virtual int DeliverFrame(unsigned char* buffer, int bufferSize,
uint32_t time_stamp,
- int64_t render_time) {
+ int64_t render_time,
+ void* /*handle*/) {
if (bufferSize != CalcBufferSize(webrtc::kI420, _width, _height)) {
ViETest::Log("Incorrect render buffer received, of length = %d\n",
bufferSize);
@@ -67,6 +68,8 @@
return 0;
}
+ virtual bool IsTextureSupported() { return false; }
+
public:
virtual ~ViEAutoTestExternalRenderer()
{
diff --git a/webrtc/video_engine/test/libvietest/helpers/vie_to_file_renderer.cc b/webrtc/video_engine/test/libvietest/helpers/vie_to_file_renderer.cc
index a70d6c9..245ddfe 100644
--- a/webrtc/video_engine/test/libvietest/helpers/vie_to_file_renderer.cc
+++ b/webrtc/video_engine/test/libvietest/helpers/vie_to_file_renderer.cc
@@ -123,7 +123,8 @@
int ViEToFileRenderer::DeliverFrame(unsigned char *buffer,
int buffer_size,
uint32_t time_stamp,
- int64_t render_time) {
+ int64_t render_time,
+ void* /*handle*/) {
webrtc::CriticalSectionScoped lock(frame_queue_cs_.get());
test::Frame* frame;
if (free_frame_queue_.empty()) {
@@ -146,6 +147,8 @@
return 0;
}
+bool ViEToFileRenderer::IsTextureSupported() { return false; }
+
int ViEToFileRenderer::FrameSizeChange(unsigned int width,
unsigned int height,
unsigned int number_of_streams) {
diff --git a/webrtc/video_engine/test/libvietest/include/vie_external_render_filter.h b/webrtc/video_engine/test/libvietest/include/vie_external_render_filter.h
index a1aeb4c..f337d17 100644
--- a/webrtc/video_engine/test/libvietest/include/vie_external_render_filter.h
+++ b/webrtc/video_engine/test/libvietest/include/vie_external_render_filter.h
@@ -35,7 +35,8 @@
return renderer_->DeliverFrame(frame_buffer,
size,
time_stamp90KHz,
- webrtc::TickTime::MillisecondTimestamp());
+ webrtc::TickTime::MillisecondTimestamp(),
+ NULL);
}
private:
diff --git a/webrtc/video_engine/test/libvietest/include/vie_to_file_renderer.h b/webrtc/video_engine/test/libvietest/include/vie_to_file_renderer.h
index 32ee1c1..393524a 100644
--- a/webrtc/video_engine/test/libvietest/include/vie_to_file_renderer.h
+++ b/webrtc/video_engine/test/libvietest/include/vie_to_file_renderer.h
@@ -55,12 +55,15 @@
// Implementation of ExternalRenderer:
int FrameSizeChange(unsigned int width, unsigned int height,
- unsigned int number_of_streams);
+ unsigned int number_of_streams) OVERRIDE;
int DeliverFrame(unsigned char* buffer,
int buffer_size,
uint32_t time_stamp,
- int64_t render_time);
+ int64_t render_time,
+ void* handle) OVERRIDE;
+
+ bool IsTextureSupported() OVERRIDE;
const std::string GetFullOutputPath() const;
diff --git a/webrtc/video_engine/vie_channel.cc b/webrtc/video_engine/vie_channel.cc
index 78a9015..7184777 100644
--- a/webrtc/video_engine/vie_channel.cc
+++ b/webrtc/video_engine/vie_channel.cc
@@ -1684,18 +1684,21 @@
}
decoder_reset_ = false;
}
- if (effect_filter_) {
- unsigned int length = CalcBufferSize(kI420,
- video_frame.width(),
- video_frame.height());
- scoped_array<uint8_t> video_buffer(new uint8_t[length]);
- ExtractBuffer(video_frame, length, video_buffer.get());
- effect_filter_->Transform(length, video_buffer.get(),
- video_frame.timestamp(), video_frame.width(),
- video_frame.height());
- }
- if (color_enhancement_) {
- VideoProcessingModule::ColorEnhancement(&video_frame);
+ // Post processing is not supported if the frame is backed by a texture.
+ if (video_frame.native_handle() == NULL) {
+ if (effect_filter_) {
+ unsigned int length = CalcBufferSize(kI420,
+ video_frame.width(),
+ video_frame.height());
+ scoped_array<uint8_t> video_buffer(new uint8_t[length]);
+ ExtractBuffer(video_frame, length, video_buffer.get());
+ effect_filter_->Transform(length, video_buffer.get(),
+ video_frame.timestamp(), video_frame.width(),
+ video_frame.height());
+ }
+ if (color_enhancement_) {
+ VideoProcessingModule::ColorEnhancement(&video_frame);
+ }
}
uint32_t arr_ofCSRC[kRtpCsrcSize];
diff --git a/webrtc/video_engine/vie_frame_provider_base.cc b/webrtc/video_engine/vie_frame_provider_base.cc
index ac05841..1ed0966 100644
--- a/webrtc/video_engine/vie_frame_provider_base.cc
+++ b/webrtc/video_engine/vie_frame_provider_base.cc
@@ -56,7 +56,7 @@
// Deliver the frame to all registered callbacks.
if (frame_callbacks_.size() > 0) {
- if (frame_callbacks_.size() == 1) {
+ if (frame_callbacks_.size() == 1 || video_frame->native_handle() != NULL) {
// We don't have to copy the frame.
frame_callbacks_.front()->DeliverFrame(id_, video_frame, num_csrcs, CSRC);
} else {
diff --git a/webrtc/video_engine/vie_renderer.cc b/webrtc/video_engine/vie_renderer.cc
index 597f49d..35c68aa 100644
--- a/webrtc/video_engine/vie_renderer.cc
+++ b/webrtc/video_engine/vie_renderer.cc
@@ -169,6 +169,21 @@
int32_t ViEExternalRendererImpl::RenderFrame(
const uint32_t stream_id,
I420VideoFrame& video_frame) {
+ if (video_frame.native_handle() != NULL) {
+ NotifyFrameSizeChange(stream_id, video_frame);
+
+ if (external_renderer_->IsTextureSupported()) {
+ external_renderer_->DeliverFrame(NULL,
+ 0,
+ video_frame.timestamp(),
+ video_frame.render_time_ms(),
+ video_frame.native_handle());
+ } else {
+ // TODO(wuchengli): readback the pixels and deliver the frame.
+ }
+ return 0;
+ }
+
VideoFrame* out_frame = converted_frame_.get();
// Convert to requested format.
@@ -218,21 +233,28 @@
break;
}
- if (external_renderer_width_ != video_frame.width() ||
- external_renderer_height_ != video_frame.height()) {
- external_renderer_width_ = video_frame.width();
- external_renderer_height_ = video_frame.height();
- external_renderer_->FrameSizeChange(external_renderer_width_,
- external_renderer_height_, stream_id);
- }
+ NotifyFrameSizeChange(stream_id, video_frame);
if (out_frame) {
external_renderer_->DeliverFrame(out_frame->Buffer(),
out_frame->Length(),
video_frame.timestamp(),
- video_frame.render_time_ms());
+ video_frame.render_time_ms(),
+ NULL);
}
return 0;
}
+void ViEExternalRendererImpl::NotifyFrameSizeChange(
+ const uint32_t stream_id,
+ I420VideoFrame& video_frame) {
+ if (external_renderer_width_ != video_frame.width() ||
+ external_renderer_height_ != video_frame.height()) {
+ external_renderer_width_ = video_frame.width();
+ external_renderer_height_ = video_frame.height();
+ external_renderer_->FrameSizeChange(
+ external_renderer_width_, external_renderer_height_, stream_id);
+ }
+}
+
} // namespace webrtc
diff --git a/webrtc/video_engine/vie_renderer.h b/webrtc/video_engine/vie_renderer.h
index a4a1e22..04295f7 100644
--- a/webrtc/video_engine/vie_renderer.h
+++ b/webrtc/video_engine/vie_renderer.h
@@ -36,6 +36,8 @@
I420VideoFrame& video_frame);
private:
+ void NotifyFrameSizeChange(const uint32_t stream_id,
+ I420VideoFrame& video_frame);
ExternalRenderer* external_renderer_;
RawVideoType external_renderer_format_;
int external_renderer_width_;
