24 Commits

Author SHA1 Message Date
henrika
ba35d05a49 Cleanup of iOS AudioDevice implementation
TBR=tkchin
BUG=webrtc:4789
TEST=modules_unittests --gtest_filter=AudioDeviceTest* and AppRTCDemo

Review URL: https://codereview.webrtc.org/1206783002 .

Cr-Commit-Position: refs/heads/master@{#9578}
2015-07-14 15:04:19 +00:00
Miguel Casas-Sanchez
4765070b8d Rename I420VideoFrame to VideoFrame.
This is a mechanical change since it affects so many
files.
I420VideoFrame -> VideoFrame
and reformatted.

Rationale: in the next CL I420VideoFrame will
get an indication of Pixel Format (I420 for
starters) and of storage type: usually
UNOWNED, could be SHMEM, and in the near
future will be possibly TEXTURE. See
https://codereview.chromium.org/1154153003
for the change that happened in Cr.

BUG=4730, chromium:440843
R=jiayl@webrtc.org, niklas.enbom@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/52629004

Cr-Commit-Position: refs/heads/master@{#9339}
2015-05-30 00:21:56 +00:00
henrika
ee369e4277 Refactoring of AudioTrackJni and AudioRecordJni using new JVM/JNI classes
BUG=NONE
TEST=./webrtc/build/android/test_runner.py gtest -s modules_unittests --gtest_filter=AudioDevice*
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51079004

Cr-Commit-Position: refs/heads/master@{#9271}
2015-05-25 08:11:38 +00:00
Peter Boström
c3f4dbc40b Remove rtp_rtcp/ dump functionality.
Removes RTP dumping from VideoEngine and VoiceEngine.

BUG=1695
R=henrika@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47179004

Cr-Commit-Position: refs/heads/master@{#9234}
2015-05-20 12:10:56 +00:00
henrika
b26198972c Adding support for OpenSL ES output in native WebRTC
BUG=4573,2982,2175,3590
TEST=modules_unittests --gtest_filter=AudioDevice*, AppRTCDemo and WebRTCDemo

Summary:

- Removes dependency of the 'enable_android_opensl' compiler flag.
  Instead, OpenSL ES is always supported, and will enabled for devices that
  supports low-latency output.
- WebRTC no longer supports OpenSL ES for the input/recording side.
- Removes old code and demos using OpenSL ES for audio input.
- Improves accuracy of total delay estimates (better AEC performance).
- Reduces roundtrip audio latency; especially when OpenSL can be used.

Performance verified on: Nexus 5, 6, 7 and 9. Samsung Galaxy S4 and S6.
Android One device.

R=magjed@webrtc.org, phoglund@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51759004

Cr-Commit-Position: refs/heads/master@{#9208}
2015-05-18 14:49:04 +00:00
Thiago Farina
9bfe3daf73 Cleanup: Remove i420_video_frame.h header.
It is just a pass through to webrtc/video_frame.h. Updated the callers
to include webrtc/video_frame.h instead and removed i420_video_frame.h.

This should fix pbos' TODO in i420_video_frame.h.

Tested on Linux with the following command lines:

$ rm -rf out/
$ ./webrtc/build/gyp_webrtc
$ ninja -C out/Debug

BUG=None
TEST=see above
R=magjed@webrtc.org, pbos@webrtc.org, tommi@webrtc.org
TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46819004

Patch from Thiago Farina <tfarina@chromium.org>.

Cr-Commit-Position: refs/heads/master@{#8973}
2015-04-10 10:52:15 +00:00
magjed@webrtc.org
2056ee3e3c Revert "Changed argument occurences of const I420VideoFrame* to const I420VideoFrame& and non-const I420VideoFrame& to I420VideoFrame*."
This reverts commit r8731.

Reason for revert: Breakes Chromium FYI bots.

TBR=hbos, tommi

Review URL: https://webrtc-codereview.appspot.com/40359004

Cr-Commit-Position: refs/heads/master@{#8733}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8733 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-16 13:48:18 +00:00
hbos@webrtc.org
2dc5fa69b2 Changed argument occurences of const I420VideoFrame* to const I420VideoFrame& and non-const I420VideoFrame& to I420VideoFrame*.
R=magjed@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40299004

Cr-Commit-Position: refs/heads/master@{#8731}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8731 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-16 13:02:19 +00:00
tommi@webrtc.org
03054486f5 Adding basic support for posting tasks to a process thread.
BUG=
R=magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41099004

Cr-Commit-Position: refs/heads/master@{#8614}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8614 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-05 13:14:19 +00:00
andresp@webrtc.org
e8f50df6b9 Remove avi recorder and corresponding enable_video flags.
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42099004

Cr-Commit-Position: refs/heads/master@{#8554}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8554 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-02 13:07:44 +00:00
tommi@webrtc.org
3985f0151a ProcessThread improvements.
* Added a way to notify a Module that it's been attached to a ProcessThread.
  The benefit of this is to give the module a way to wake up the thread
  when it needs work to happen on the worker thread, immediately.
  Today, module instances are typically registered with a process thread
  outside the control of the modules themselves.  I.e. they typically
  don't know about the process thread they're attached to.

* Improve ProcessThread's WakeUp algorithm to not call TimeUntilNextProcess
  when a WakeUp call is requested.  This is an optimization for the above
  case which avoids the module having to acquire a lock or do an interlocked
  operation before calling WakeUp(), which would ensure the module's
  TimeUntilNextProcess() implementation would return 0.

BUG=2822
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39239004

Cr-Commit-Position: refs/heads/master@{#8527}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8527 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-27 13:37:25 +00:00
henrika@webrtc.org
62f6e75673 Refactoring WebRTC Java/JNI audio recording in C++ and Java.
This is a big refactoring of the existing C++/JNI/Java support for audio recording in native WebRTC:

- Removes unused code and old WEBRTC logging macros
- Now uses optimal sample rate and buffer size in Java AudioRecord (used hard-coded sample rate before)
- Makes code more inline with the implementation in Chrome
- Adds helper methods for JNI handling to improve readability
- Changes the threading model (high-prio audio thread now lives in Java-land and C++ only works as proxy)
- Adds basic thread checks
- Removes all locks in C++ land
- Removes all locks in Java
- Improves construction/destruction
- Additional cleanup

Tested using AppRTCDemo and WebRTCDemo APKs on N6, N5, N7, Samsung Galaxy S4 and
Samsung Galaxy S4 mini (which uses 44.1kHz as native sample rate).

BUG=NONE
R=magjed@webrtc.org, perkj@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33969004

Cr-Commit-Position: refs/heads/master@{#8325}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8325 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-11 08:39:19 +00:00
tommi@webrtc.org
0c3e12b7bf Revamp the ProcessThreadImpl implementation.
* Add a new WakeUp method that gives a module a chance to be called back right away on the worker thread.
* Wrote unit tests for the class.
* Significantly reduce the amount of locking.
  - ProcessThreadImpl itself does a lot less locking.
  - Reimplemented the way we keep track of when to make calls to Process.
    This reduces the amount of calls to TimeUntilNextProcess and since most implementations of that function grab a lock, this means less locking.
* Renamed ProcessThread::CreateProcessThread to ProcessThread::Create.
* Added thread checks for Start/Stop.  Threading model of other functions is now documented.
* We now log an error if an implementation of TimeUntilNextProcess returns a negative value (some implementations do, but the method should only return a positive nr of ms).
* Removed the DestroyProcessThread method and instead force callers to use scoped_ptr<> to maintain object lifetime.

BUG=2822
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35999004

Cr-Commit-Position: refs/heads/master@{#8261}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8261 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-06 09:44:45 +00:00
pkasting@chromium.org
4591fbd09f Use size_t more consistently for packet/payload lengths.
See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information.

This CL was reviewed and approved in pieces in the following CLs:
https://webrtc-codereview.appspot.com/24209004/
https://webrtc-codereview.appspot.com/24229004/
https://webrtc-codereview.appspot.com/24259004/
https://webrtc-codereview.appspot.com/25109004/
https://webrtc-codereview.appspot.com/26099004/
https://webrtc-codereview.appspot.com/27069004/
https://webrtc-codereview.appspot.com/27969004/
https://webrtc-codereview.appspot.com/27989004/
https://webrtc-codereview.appspot.com/29009004/
https://webrtc-codereview.appspot.com/30929004/
https://webrtc-codereview.appspot.com/30939004/
https://webrtc-codereview.appspot.com/31999004/
Committing as TBR to the original reviewers.

BUG=chromium:81439
TEST=none
TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom

Review URL: https://webrtc-codereview.appspot.com/23129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-20 22:28:14 +00:00
wuchengli@chromium.org
637c55f45b Add support of texture frames for video capturer.
This is a reland of r6252. The video_capture_tests failure on
builder Android Chromium-APK Tests should be flaky.

- Add ViECapturer unittest.
- Add CloneFrame function in I420VideoFrame.
- Encoders do not support texture yet and texture frames
are dropped in ViEEncoder for now.

Corresponding CLs:
https://codereview.chromium.org/277943002
http://cl/66620352

BUG=chromium:362437
TEST=WebRTC video stream forwarding, video_engine_core_unittests,
     common_video_unittests and video_capture_tests_apk.
TBR=fischman@webrtc.org, perkj@webrtc.org, stefan@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6258 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-28 07:00:51 +00:00
wuchengli@chromium.org
89e8ffb395 Revert "Add support of texture frames for video capturer."
This reverts commit 83c89cd003be75d7d06ef9a2b139588f08d280ca.

Reason: The Buildbot has detected a new failure on builder
Android Chromium-APK Tests.

BUG=chromium:362437
TBR=fischman@webrtc.org, perkj@webrtc.org, stefan@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6253 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-27 14:12:58 +00:00
wuchengli@chromium.org
efe15355ee Add support of texture frames for video capturer.
- Add ViECapturer unittest.
- Add CloneFrame function in I420VideoFrame.
- Encoders do not support texture yet and texture frames
  are dropped in ViEEncoder for now.

Corresponding CLs:
https://codereview.chromium.org/277943002
http://cl/66620352

BUG=chromium:362437
TEST=WebRTC video stream forwarding. Run video_engine_core_unittests and common_video_unittests.
R=fischman@webrtc.org, perkj@webrtc.org, stefan@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6252 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-27 12:40:27 +00:00
henrike@webrtc.org
79cf3acc79 Removes usage of ListWrapper from several files.
BUG=2164
R=andrew@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5373 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-13 15:21:30 +00:00
fischman@webrtc.org
4e65e07e41 VideoCaptureAndroid: rewrote the (standalone) implementation of video capture on Android.
Besides being ~40% the size of the previous implementation, this makes it so
that VideoCaptureAndroid can stop and restart capture, which is necessary to
support onPause/onResume reasonably on Android.

BUG=1407
R=henrike@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2334004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4915 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-03 18:23:13 +00:00
pbos@webrtc.org
8b06200802 Include files from webrtc/.. paths in utility/.
BUG=1662
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1786004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4336 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-12 08:28:10 +00:00
pbos@webrtc.org
d900e8bea8 Proper spacing for end-of-namespace comments.
BUG=
R=mflodman@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1760006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4293 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-03 15:12:26 +00:00
pbos@webrtc.org
c75102eba7 WebRtc_Word32 -> int32_t in utility/
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1307005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3797 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-09 13:32:55 +00:00
mikhal@webrtc.org
9fedff7c17 Switching to I420VideoFrame
Review URL: https://webrtc-codereview.appspot.com/922004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2983 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-24 18:33:04 +00:00
andrew@webrtc.org
14b43beb7c Move src/ -> webrtc/
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/915006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-22 18:19:23 +00:00