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Magnus Jedvert 31f18e164e Android SurfaceTextureHelper: Avoid crashing if size hasn't been set
SurfaceTextureHelper currently crashes if an OES texture is produced
before setTextureSize() has been called. This is annoying if the texture
size is not easily known beforehand. A real world example is MediaPlayer
that provides the video size with an asynchronous call to
setOnVideoSizeChangedListener(), but that might happen after the first
texture is produced on some devices.

This CL waits with delivering frames until the size has been sent,
rather than crashing.

Bug: webrtc:10709
Change-Id: I5d9ce542e0edaafe1153fd5fe7d64dba86d7e33c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140080
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28151}
2019-06-04 10:40:39 +00:00
api Fix comment typo about degradation preference. 2019-06-04 06:49:32 +00:00
audio Reland "Avoid encrypting empty audio packet." 2019-05-28 12:30:07 +00:00
build_overrides Remove crbug.com/904400 workaround. 2019-03-15 18:36:23 +00:00
call Add support for early loss detection using transport feedback. 2019-05-29 13:21:10 +00:00
common_audio Add ability to play audio in circle for TestAudioDevice wav file capturer 2019-04-16 15:33:03 +00:00
common_video Move H.264 SPS VUI rewriting to FrameEncodeMetadataWriter. 2019-05-29 10:37:22 +00:00
crypto Adding new top-level directory crypto/ 2019-03-08 00:35:05 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
examples Target SDK level 29 in AppRTCMobile. 2019-06-03 12:40:42 +00:00
logging Fix for crash in event log when using scenario tests. 2019-05-22 15:22:49 +00:00
media Check H264 packetization mode when using IsSameCodec 2019-05-31 21:18:21 +00:00
modules Fix bug in neteq_quality_test 2019-06-04 08:53:07 +00:00
p2p Move datagram transport to JsepTransport 2019-06-03 22:24:12 +00:00
pc Move datagram transport to JsepTransport 2019-06-03 22:24:12 +00:00
resources RNN VAD: clean-up unit tests 2019-04-29 12:55:02 +00:00
rtc_base Lockless SwapQueue 2019-06-03 11:49:49 +00:00
rtc_tools Fix BWE simulation graph in event log visualization 2019-05-28 10:40:29 +00:00
sdk Android SurfaceTextureHelper: Avoid crashing if size hasn't been set 2019-06-04 10:40:39 +00:00
stats Implement QualityLimitationReasonTracker and expose "reason". 2019-05-28 16:23:55 +00:00
style-guide Add style guide rule about paired .h and .cc files 2018-03-14 13:02:35 +00:00
system_wrappers Ensure CpuInfo::DetectNumberOfCores is > 0 and thread safe. 2019-05-24 12:59:14 +00:00
test in test/scenario pass TaskQueueFactory explicitly 2019-06-04 09:21:17 +00:00
tools_webrtc mb: Implement 'quiet' flag in mb lookup 2019-05-31 11:51:15 +00:00
video Allowing buffering a LNTF (loss notification) feedback message in RTCPSender 2019-06-03 16:28:34 +00:00
.clang-format Tune ObjC clang-format configuration 2017-05-11 09:14:18 +00:00
.git-blame-ignore-revs Let git-hyper-blame ignore format commit. 2018-06-20 09:26:44 +00:00
.gitignore Add Visual Studio Code project folder to gitignore file. 2019-01-21 18:42:33 +00:00
.gn Remove last mention of ortc from the codebase. 2019-05-25 07:28:05 +00:00
.vpython Add vpython dependencies needed to run presubmit tests on LUCI 2018-05-18 08:10:25 +00:00
abseil-in-webrtc.md Allowing buffering a LNTF (loss notification) feedback message in RTCPSender 2019-06-03 16:28:34 +00:00
AUTHORS Import proto_library.gni when rtc_enable_protobuf is true 2019-02-27 09:56:42 +00:00
BUILD.gn Delete rtc_base/unittest_main.cc 2019-05-21 14:44:11 +00:00
CODE_OF_CONDUCT.md Add code of conduct to WebRTC repo 2017-05-16 12:09:13 +00:00
codereview.settings Don't add webrtc-reviews@ to CC, it can be added globally on Gerrit 2018-10-25 08:19:53 +00:00
common_types.h Move kRtpCsrcSize from common_types.h to rtp_headers.h 2019-05-10 09:43:39 +00:00
DEPS Roll chromium_revision 584b49b1a7..1070231d7d (665633:665750) 2019-06-04 01:44:10 +00:00
ENG_REVIEW_OWNERS Enforce LGTM from owners of depends-on paths in DEPS via presubmit. 2018-09-28 12:49:54 +00:00
LICENSE Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
license_template.txt Update template to follow chromium copyright style 2013-04-24 01:01:28 +00:00
native-api.md Delete unused I420 "codec" 2018-12-18 12:30:58 +00:00
OWNERS Add juberti@ to webrtc root owners 2019-05-17 18:11:58 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
PRESUBMIT.py Do not run CheckNoStreamUsageIsAdded on tests. 2019-05-07 15:52:35 +00:00
presubmit_test.py Fixing py lint errors 2018-07-23 15:28:48 +00:00
presubmit_test_mocks.py Reland: Add presubmit check for changes in 3pp 2018-05-22 13:11:18 +00:00
pylintrc Fixing py lint errors 2018-07-23 15:28:48 +00:00
README.chromium Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
README.md Tell users where they can find the native API headers 2017-11-14 10:36:46 +00:00
style-guide.md Remove rule that discourages passing optional by const reference 2019-02-05 11:58:05 +00:00
WATCHLISTS Add fhernqvist to watchlist. 2019-06-03 10:13:40 +00:00
webrtc.gni Testing no /DUNICODE assumptions with Win more configs bots. 2019-03-28 08:46:37 +00:00
whitespace.txt Whitespace change 2018-02-23 10:34:16 +00:00

WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.

Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.

The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.

Development

See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.

Authoritative list of directories that contain the native API header files.

More info