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aluebs f03a8d4c4d Unpack different wav files after each INIT event of the aecdump 
Some aecdumps have more than one INIT event. In those cases only the last wav file was unpacked, which sometimes is not the most interesting or desired one.
This CL creates a different wav file after each INIT event.

Review-Url: https://codereview.webrtc.org/2067423002
Cr-Commit-Position: refs/heads/master@{#13196}
2016-06-17 16:41:50 +00:00
..
androidjunit
Reland of Android: Change camera fps range selection (patchset #1 id:1 of https://codereview.webrtc.org/2021233002/ )
2016-06-01 07:44:07 +00:00
api
Partial reland of Delete unused and almost unused frame-related methods. (patchset #1 id:1 of https://codereview.webrtc.org/2076113002/ )
2016-06-17 12:03:09 +00:00
audio
Add AudioReceiveStream::SetGain() method and use that in WVoMC::SetOutputVolume().
2016-06-17 15:30:58 +00:00
base
Android: Add Size class.
2016-06-17 08:02:10 +00:00
build
GN: Add modules_unittests
2016-06-13 07:19:53 +00:00
call
Remove ViEEncoder::Pause / Start
2016-06-17 14:27:23 +00:00
common_audio
GN: Add common_audio_unittests and common_video_unittests
2016-06-08 08:28:15 +00:00
common_video
Partial reland of Delete unused and almost unused frame-related methods. (patchset #1 id:1 of https://codereview.webrtc.org/2076113002/ )
2016-06-17 12:03:09 +00:00
examples
Add RTCEventLog API to ObjC.
2016-06-14 22:03:19 +00:00
libjingle
Adding more detail to MessageQueue::Dispatch logging.
2016-06-10 21:17:33 +00:00
media
Add AudioReceiveStream::SetGain() method and use that in WVoMC::SetOutputVolume().
2016-06-17 15:30:58 +00:00
modules
Unpack different wav files after each INIT event of the aecdump
2016-06-17 16:41:50 +00:00
p2p
Fix header size check in PseudoTcp::parse().
2016-06-17 01:08:19 +00:00
pc
Delete GetExecutablePath and related unused code.
2016-06-16 10:44:44 +00:00
sdk
Partial reland of Delete unused and almost unused frame-related methods. (patchset #1 id:1 of https://codereview.webrtc.org/2076113002/ )
2016-06-17 12:03:09 +00:00
system_wrappers
FileWrapper[Impl] modifications and actually remove the "Impl" class.
2016-06-15 17:30:18 +00:00
test
Add AudioReceiveStream::SetGain() method and use that in WVoMC::SetOutputVolume().
2016-06-17 15:30:58 +00:00
tools
New misc scripts, header_usage.sh and author_line_count.sh.
2016-06-15 07:06:32 +00:00
video
Remove ViEEncoder::Pause / Start
2016-06-17 14:27:23 +00:00
voice_engine
Add AudioReceiveStream::SetGain() method and use that in WVoMC::SetOutputVolume().
2016-06-17 15:30:58 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
audio_receive_stream.h
Add AudioReceiveStream::SetGain() method and use that in WVoMC::SetOutputVolume().
2016-06-17 15:30:58 +00:00
audio_send_stream.h
Add AudioSendStream::SetMuted() method and use it in WVoMC::MuteStream().
2016-06-16 17:53:28 +00:00
audio_sink.h
Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."
2016-02-26 21:46:22 +00:00
audio_state.h
Add comments about the Audio parts of the public Call API being WIP.
2015-12-03 12:06:31 +00:00
BUILD.gn
GN: Add video_engine_tests
2016-06-14 10:53:09 +00:00
call.h
Movable support for VideoReceiveStream::Config and avoid copies.
2016-06-10 15:58:12 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
FileWrapper[Impl] modifications and actually remove the "Impl" class.
2016-06-15 17:30:18 +00:00
common_types.h
Avoid unnecessary HW video encoder reconfiguration
2016-06-16 19:08:11 +00:00
common.gyp
Revert of Add missing headers and fix some missing dependencies (patchset #1 id:20001 of https://codereview.webrtc.org/1990593002/ )
2016-05-18 12:00:55 +00:00
common.h
#include "webrtc/base/constructormagic.h" where appropriate
2016-04-26 15:14:48 +00:00
config.cc
Avoid unnecessary HW video encoder reconfiguration
2016-06-16 19:08:11 +00:00
config.h
Avoid unnecessary HW video encoder reconfiguration
2016-06-16 19:08:11 +00:00
DEPS
Moved creation of AudioDecoderFactory to inside PeerConnectionFactory.
2016-06-13 14:35:01 +00:00
engine_configurations.h
Remove the VoEDtmf interface.
2016-03-08 12:11:00 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Add the Ooura FFT to RealFourier.
2015-03-19 20:07:43 +00:00
OWNERS
Add kwiberg@webrtc.org as root owner.
2016-06-14 09:39:40 +00:00
PATENTS
…
PRESUBMIT.py
PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.
2014-05-23 17:27:18 +00:00
README.chromium
…
rtc_unittests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
rtc_unittests.isolate
Update all .isolate files for the new format.
2014-10-31 18:08:09 +00:00
supplement.gypi
Roll chromium_revision 086802955f..7fa6701bc5 (396351:398458)
2016-06-08 11:05:27 +00:00
transport.h
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
2015-12-09 11:13:40 +00:00
typedefs.h
Remove runtime NEON detection
2016-05-17 17:56:48 +00:00
video_decoder.h
Replace scoped_ptr with unique_ptr in webrtc/common_video/
2016-03-02 11:41:40 +00:00
video_encoder.h
Remove QualityScaler framerate reduction.
2016-04-26 11:37:20 +00:00
video_engine_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
video_engine_tests.isolate
Add audio streams to CallTest and a first A/V call test.
2016-01-07 16:43:31 +00:00
video_frame.h
Partial reland of Delete unused and almost unused frame-related methods. (patchset #1 id:1 of https://codereview.webrtc.org/2076113002/ )
2016-06-17 12:03:09 +00:00
video_receive_stream.h
Movable support for VideoReceiveStream::Config and avoid copies.
2016-06-10 15:58:12 +00:00
video_send_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
webrtc_examples.gyp
Remove metrics_default from rtc_media dependencies.
2016-06-01 12:28:57 +00:00
webrtc_nonparallel_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_nonparallel_tests.isolate
Create isolate files for nonparallel tests.
2015-10-29 14:29:15 +00:00
webrtc_perf_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_perf_tests.isolate
Fix missing resource file in webrtc_perf_tests.isolate
2016-06-17 07:39:49 +00:00
webrtc_tests.gypi
GN: Add SDK tests to rtc_unittests.
2016-06-08 08:14:22 +00:00
webrtc.gyp
Remove webrtc_all target
2016-06-14 09:09:24 +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.