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Jakob Ivarsson 00a6ab568b Check timestamp difference when choosing to extract multiple packets from the jitter buffer. 
This fixes a bug where we sometimes extract an Opus CNG packet and the packet after, even though there was big timestamp gap between the packets, which causes expansion during the next GetAudio calls.

Change-Id: I2409ac08df58afc496f74b91981657b7206e8bb1
Bug: webrtc:10167
Reviewed-on: https://webrtc-review.googlesource.com/c/115419
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26179}
2019-01-09 16:21:11 +00:00
api
Include pacing buffer size in congestion window.
2019-01-09 11:16:58 +00:00
audio
Update refcounting of AudioState to use rtc::RefCountedObject
2019-01-09 13:49:19 +00:00
build_overrides
Disable hermetic toolchain when building on macOS 10.14.
2018-12-07 13:57:28 +00:00
call
Include pacing buffer size in congestion window.
2019-01-09 11:16:58 +00:00
common_audio
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
2018-11-28 18:25:07 +00:00
common_video
Introduce VideoFrame::id to keep track of frames inside application.
2019-01-04 08:59:26 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
examples
Convert peerconnection_client to use VcmCapturer
2019-01-03 10:52:29 +00:00
infra
Re-add MSVC debug bots to CQ
2018-12-05 12:20:56 +00:00
logging
Simplifies RtcEventProcessor interface.
2019-01-08 15:16:19 +00:00
media
Introduce VideoFrame::id to keep track of frames inside application.
2019-01-04 08:59:26 +00:00
modules
Check timestamp difference when choosing to extract multiple packets from the jitter buffer.
2019-01-09 16:21:11 +00:00
p2p
Include absl/memory/memory.h if absl::make_unique is used
2019-01-08 20:08:32 +00:00
pc
Include absl/memory/memory.h if absl::make_unique is used
2019-01-08 20:08:32 +00:00
resources
Base ApmTest.Process on AudioProcessingStats.output_rms_dbfs
2018-12-18 16:45:03 +00:00
rtc_base
Include absl/memory/memory.h if absl::make_unique is used
2019-01-08 20:08:32 +00:00
rtc_tools
Delete NetEqTest::ExtDecoderMap
2019-01-08 16:25:05 +00:00
sdk
Include absl/memory/memory.h if absl::make_unique is used
2019-01-08 20:08:32 +00:00
stats
Add ',' between elements in RTCStatsReport::ToJson
2019-01-08 16:05:23 +00:00
style-guide
Add style guide rule about paired .h and .cc files
2018-03-14 13:02:35 +00:00
system_wrappers
[Fuchsia] Implement detection of available cores.
2018-12-17 19:37:46 +00:00
test
Include absl/memory/memory.h if absl::make_unique is used
2019-01-08 20:08:32 +00:00
tools_webrtc
mb: Turn LUCI desktop perf bots into builder + tester pairs
2018-12-21 13:43:04 +00:00
video
Calculate video quality metrics only for rendered frames.
2019-01-09 14:38:44 +00:00
.clang-format
…
.git-blame-ignore-revs
Let git-hyper-blame ignore format commit.
2018-06-20 09:26:44 +00:00
.gitignore
Reland "Compile frame analyzer for the host machine on perf tests."
2018-09-18 09:51:19 +00:00
.gn
Re-enable gtest absl pretty printers.
2018-08-13 13:54:05 +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
Rebase std::is_trivially_* with absl::is_trivially_*
2018-11-26 19:20:27 +00:00
AUTHORS
Add ',' between elements in RTCStatsReport::ToJson
2019-01-08 16:05:23 +00:00
BUILD.gn
Reland "Default to dlopening the PipeWire."
2019-01-08 08:09:42 +00:00
CODE_OF_CONDUCT.md
…
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
Remove CodecInst pt.3
2018-12-18 07:42:21 +00:00
DEPS
Roll chromium_revision aa595e0ebb..63546ce7e8 (620951:621059)
2019-01-09 06:44:33 +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
…
native-api.md
Delete unused I420 "codec"
2018-12-18 12:30:58 +00:00
OWNERS
Clean up root OWNERS.
2018-11-09 14:23:59 +00:00
PATENTS
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00
presubmit_test_mocks.py
Reland: Add presubmit check for changes in 3pp
2018-05-22 13:11:18 +00:00
presubmit_test.py
Fixing py lint errors
2018-07-23 15:28:48 +00:00
PRESUBMIT.py
Delete WebRtcVideoCapturer and related classes.
2019-01-03 15:15:48 +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
Add a style rule about not using const optional<T>& arguments
2018-11-08 11:57:35 +00:00
WATCHLISTS
Remove likely obsolete entries from WATCHLISTS
2018-10-30 07:46:29 +00:00
webrtc.gni
Reland "Default to dlopening the PipeWire."
2019-01-08 08:09:42 +00:00
whitespace.txt
Whitespace change
2018-02-23 10:34:16 +00:00

README.md

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

Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
Readme BSD-3-Clause 446 MiB
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