I removed the 'libjingle' target in talk/libjingle.gyp and replaced
all users of it with base/base.gyp:rtc_base. It seems the jsoncpp
and expat dependencies were not used by it's previous references.
The files in talk/media/testdata were uploaded to Google Storage and
added .sha1 files in resources/media instead of simply moving them.
The previously disabled warnings that were inherited from
talk/build/common.gypi are now replaced by target-specific disabling
of only the failing warnings. Additional disabling was needed since the stricter
compilation warnings that applies to code in webrtc/.
License headers will be updated in a follow-up CL in order to not
break Git history.
Other modifications:
* Updated the header guards.
* Sorted the includes using chromium/src/tools/sort-headers.py
except for these files:
talk/app/webrtc/peerconnectionendtoend_unittest.cc
talk/app/webrtc/java/jni/androidmediadecoder_jni.cc
talk/app/webrtc/java/jni/androidmediaencoder_jni.cc
webrtc/media/devices/win32devicemanager.cc.
* Unused GYP reference to libjingle_tests_additional_deps was removed.
* Removed duplicated GYP entries of
webrtc/base/testutils.cc
webrtc/base/testutils.h
The HAVE_WEBRTC_VIDEO and HAVE_WEBRTC_VOICE defines were used by only talk/media,
so they were moved to the media.gyp.
I also checked that none of
EXPAT_RELATIVE_PATH,
FEATURE_ENABLE_VOICEMAIL,
GTEST_RELATIVE_PATH,
JSONCPP_RELATIVE_PATH,
LOGGING=1,
SRTP_RELATIVE_PATH,
FEATURE_ENABLE_SSL,
FEATURE_ENABLE_VOICEMAIL,
FEATURE_ENABLE_PSTN,
HAVE_SCTP,
HAVE_SRTP,
are used by the talk/media code.
For Chromium, the following changes will need to be applied to the roll CL that updates the
DEPS for WebRTC and libjingle: https://codereview.chromium.org/1604303002/
BUG=webrtc:5420
NOPRESUBMIT=True
TBR=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1587193006
Cr-Commit-Position: refs/heads/master@{#11495}
- Reduce maximum pending frames for H.264 decoder to 8.
- Log data for next 2 output frames every time frame drop
happens or decoder drain is triggered.
- When timeout happens for dequeueInputBuffer call try to
drain the decoder and get input buffer one more time.
- Fix PTS values overflow.
Review URL: https://codereview.webrtc.org/1661203002
Cr-Commit-Position: refs/heads/master@{#11492}
- Add extra logging for Android HW codec corner cases
when frames are dropped or resolution is changed.
- Use presentation timestamps for decoded frame logging.
- Enable key frame sending on long frame gap for
H.264 codec.
BUG=b/26504665
Review URL: https://codereview.webrtc.org/1653523003
Cr-Commit-Position: refs/heads/master@{#11452}
Before this fix, it was required that the EGL context used as an argument was kept open until all PeerConnections using the context had been closed. With this fix, that is no longer required.
Also, if a released EGLContext (EGL_NO_CONTEXT) is used, harware codecs will fallback to use byte buffers for encoding and decoding.
BUG=b/26583522
R=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/1615153002 .
Cr-Commit-Position: refs/heads/master@{#11398}
- Similar to HW decoder log timing for first 5 frames
of HW encoder input and output to track correct encoder
start.
- Increment input timestamp when frame is dropped.
- Better calculate average encoding time.
- Use different tags for encoder and decoder modules.
BUG=b/26504665
Review URL: https://codereview.webrtc.org/1587943004
Cr-Commit-Position: refs/heads/master@{#11277}
Plus log first few decoder frames in and out events.
BUG=b/25287910
Review URL: https://codereview.webrtc.org/1423843005
Cr-Commit-Position: refs/heads/master@{#10439}
of decoder factory class.
- Add new Peer connection factory method to initialize shared
EGL context.
This provides an option to use single peer connection factory
in the application and create peer connections from the same
factory and reinitialize shared EGL context for video
decoding HW acceleration.
R=wzh@webrtc.org
Review URL: https://codereview.webrtc.org/1304063011 .
Cr-Commit-Position: refs/heads/master@{#9838}
- Remove an option to use MediaCodec SW decoder from Java layer.
- Better handling Java exceptions in JNI - detect exceptions
and either try to reset the codec or fallback to SW decoder.
- If any error is reported by codec try to fallback to SW
codec for VP8 or reset decoder and continue decoding for H.264.
- Add more logging for error conditions.
R=wzh@webrtc.org
Review URL: https://codereview.webrtc.org/1178943007.
Cr-Commit-Position: refs/heads/master@{#9431}
- Qualcomm H.264 HW decoder on KK and older requires
a few video frames before it can generate output. Increase
maximum allowed pending frames for H.264 decoder to 30.
Plus changes in the logging to track decoder buffers
timestamps.
R=wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36319004
Cr-Commit-Position: refs/heads/master@{#8490}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8490 4adac7df-926f-26a2-2b94-8c16560cd09d
- Allow to configure MediaCodec Java wrapper to use VP8
and H.264 codec.
- Save H.264 config frames with SPS and PPS NALUs and append them to every key frame.
- Correctly handle the case when one encoded frame may generate several output NALUs.
- Add code to find H.264 start codes.
- Add a flag (non configurable yet) to use H.264 in AppRTCDemo.
- Improve MediaCodec logging.
R=wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43379004
Cr-Commit-Position: refs/heads/master@{#8465}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8465 4adac7df-926f-26a2-2b94-8c16560cd09d
