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Sami Kalliomäki 3aa3ea7913 Improve HardwareVideoDecoder stability. 
Adds a timeout to the dequeue input buffer call. This improves stability
because WebRTC quickly queues frames multiple when the call starts. This
might cause the decoder to run out of input buffers. Waiting for
dequeueOutputBuffers call is no longer necessary.

Bug: webrtc:7760
Change-Id: I503ff1cf44042c4d8610077090148d9dfef169f5
Reviewed-on: https://chromium-review.googlesource.com/548357
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18800}
2017-06-28 07:38:22 +00:00
..
api
Don't forget to support G722 stereo decoding
2017-06-26 11:19:43 +00:00
audio
Disable AudioBweIntegrationTest.NoBandwidthDropAfterDtx - it's flaky
2017-06-26 12:04:12 +00:00
base
Rename safe_cmp::{Eq,Ne,Lt,Le,Ge,Gt} to Safe{Eq,Ne,Lt,Le,Ge,Gt}
2017-06-26 08:31:31 +00:00
call
Reland of Periodically update codec bit/frame rate settings.
2017-06-27 14:06:52 +00:00
common_audio
Put attribute before function name instead of after, as required by GCC
2017-06-26 08:32:40 +00:00
common_video
Return WrappedI444Buffer in VP9Impl
2017-06-23 03:28:06 +00:00
examples
Notify delegates about audio glitches in real time
2017-06-27 09:43:27 +00:00
logging
Support building WebRTC without audio and video.
2017-06-15 19:52:32 +00:00
media
Reland of Periodically update codec bit/frame rate settings.
2017-06-27 14:06:52 +00:00
modules
Ensure Dxgi duplicator works correctly in session 0
2017-06-28 05:04:21 +00:00
ortc
Fixing bad use of std::sort in test method.
2017-06-27 14:52:50 +00:00
p2p
Handle the PROTO_TSL when getting the protocol priority.
2017-06-27 22:11:24 +00:00
pc
Remove unused "crypto_options_" field.
2017-06-26 11:07:52 +00:00
rtc_base
Add webrtc/rtc_base skeleton.
2017-06-21 12:50:22 +00:00
sdk
Improve HardwareVideoDecoder stability.
2017-06-28 07:38:22 +00:00
stats
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
system_wrappers
Prevent data race in GetStaticInstance
2017-05-29 07:01:14 +00:00
test
Reland of Periodically update codec bit/frame rate settings.
2017-06-27 14:06:52 +00:00
tools
Add NetEq delay plotting to event_log_visualizer
2017-06-14 13:09:58 +00:00
video
Fix deadlock in webrtc_perf_tests
2017-06-27 14:21:01 +00:00
voice_engine
Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface.
2017-06-21 08:05:22 +00:00
.gitignore
…
BUILD.gn
Add webrtc/rtc_base skeleton.
2017-06-21 12:50:22 +00:00
codereview.settings
…
common_types.cc
Allow parsing empty RTCP TargetBitrate messages, but stop sending them.
2017-06-22 12:40:25 +00:00
common_types.h
Allow parsing empty RTCP TargetBitrate messages, but stop sending them.
2017-06-22 12:40:25 +00:00
config.cc
Implement timing frames.
2017-06-19 14:18:55 +00:00
config.h
Implement timing frames.
2017-06-19 14:18:55 +00:00
DEPS
Delete old include file webrtc/video_frame.h.
2017-06-13 13:05:05 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
OWNERS
Don't add or rename files in webrtc/ and webrtc/api/ without a proper review
2017-05-08 13:01:54 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
typedefs.h
Make WARN_UNUSED_RESULT a no-op on gcc
2017-04-06 11:32:27 +00:00
video_receive_stream.h
Delete VieRemb class, move functionality to PacketRouter.
2017-04-19 06:38:35 +00:00
video_send_stream.h
Add field trial for balanced degradation preference.
2017-06-14 12:52:21 +00:00
webrtc.gni
Adds detection of audio glitches for playout on iOS (reland)
2017-06-13 16:00:18 +00:00
whitespace.txt
Moving whitespace file up by one folder
2017-01-24 13:19:42 +00:00

README.chromium 

Name: WebRTC
URL: http://www.webrtc.org
Version: 90
License: BSD
License File: LICENSE

Description:
WebRTC provides real time voice and video processing
functionality to enable the implementation of 
PeerConnection/MediaStream.

Third party code used in this project is described 
in the file LICENSE_THIRD_PARTY.