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Author SHA1 Message Date
tommi@webrtc.org
704eb8fa15 Restructure the video_capture code a bit to make room for a Media Foundation class implementation. 
This change includes the following:

* Skeleton classes for a couple of base MediaFoundation classes (no code yet).  See *_mf.*.
* Renaming the DirectShow based implementation from "_windows" to "_ds".
* Move the VideoCaptureImpl::CreateDeviceInfo() method into video_capture_factory_windows.cc
  The reason for this is that that's where the other VideoCaptureImpl factory function is
  and the factory function won't be implementation specific.
* Removed use of <initguid.h> from a header file to avoid defining the same guids in multiple object files.
  (more info here: http://msdn.microsoft.com/en-us/library/windows/desktop/dd375463(v=vs.85).aspx)
* Moved a couple of global variables from the capture_delay_values_windows.h header and into
  device_info_ds.cc since that's the only file that uses those variables.
* Delete capture_delay_values_windows.h.
* Added a factory function: DeviceInfoDS::Create() that'll create the DirectShow specific implementation.

TEST=This is mostly moving code around.  The code that is added, is currently "dead". No manual testing needed.
BUG=chromium:140545
Review URL: https://webrtc-codereview.appspot.com/967008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3135 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-20 11:27:25 +00:00
mikhal@webrtc.org
0f34fd7660 Updating Memory allocation for rotation and related tests. 
Review URL: https://webrtc-codereview.appspot.com/943004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3132 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-19 21:15:35 +00:00
stefan@webrtc.org
4100b0402e Move SSRC list to RemoteBitrateEstimator. 
BUG=1105

Review URL: https://webrtc-codereview.appspot.com/965027

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3130 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-19 10:09:20 +00:00
tina.legrand@webrtc.org
5ac387c4d1 Allow NetEQ to use real packet durations. 
This is a copy of http://review.webrtc.org/864014/

This adds a FuncDurationEst to each codec instance which estimates
the duration of a packet given the packet contents and the duration
of the previous packet. By default, this simply returns the
duration of the previous packet (which is what is currently assumed
to be the duration of all future packets). This patch also provides
an initial implementation of this function for G.711 which returns
the actual number of samples in the packet.

BUG=issue1015

Review URL: https://webrtc-codereview.appspot.com/935016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3129 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-19 08:02:55 +00:00
kma@webrtc.org
55cd78cfc2 Porting ARM optimization from Android to ios. 
Tested APM and iSAC in Android. Bit-exact with original versions.
Changes include removing or changing some GCC derivatives (e.g. .fnstart, .hword), instruction syntax, etc.
Review URL: https://webrtc-codereview.appspot.com/934009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3124 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-17 00:22:46 +00:00
niklas.enbom@webrtc.org
c79505f95e Add warning comment 
Review URL: https://webrtc-codereview.appspot.com/933012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3122 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-16 22:46:36 +00:00
tina.legrand@webrtc.org
d0acdf6e2b Fix ordered comparison warnings in the RTPtimeshift unit test 
Original CL uploaded here: http://review.webrtc.org/933013/

Removing the checks if (argv[4/5] >= 0), they are not doing anything useful.

BUG=

Review URL: https://webrtc-codereview.appspot.com/933019

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3118 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-16 15:34:32 +00:00
mflodman@webrtc.org
b2f474e8bb Adding ViE CallStats to keep track of call statistics. As a start, only rtt is handled. 
This CL will be followed by another CL connecting the dots.

BUG=769
TEST=New unittest added.

Review URL: https://webrtc-codereview.appspot.com/968006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3117 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-16 13:57:26 +00:00
tommi@webrtc.org
12773eaa8c Minor cleanup of the videocapture code. No "real" code change :) 
Remove declaration of a function that doesn't exist.
Remove call to GetProductId from video_capture_factory_windows that's not necessary.
This was the only dependency on device_info_windows.h outside of video_capture_windows.cc.

TEST=There's no code change, so no manual testing is required.
BUG=none
Review URL: https://webrtc-codereview.appspot.com/936025

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3114 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-16 09:03:12 +00:00
marpan@webrtc.org
d14ff54de2 Removed unnecessary lines in one of the tests and changed one parameter. 
Review URL: https://webrtc-codereview.appspot.com/933018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3113 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-16 01:17:00 +00:00
andrew@webrtc.org
d72b3d6c81 Fix cpplint errors in audio_processing.h 
TBR=leozwang@webrtc.org
BUG=1114

Review URL: https://webrtc-codereview.appspot.com/934015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3108 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-15 21:46:06 +00:00
leozwang@webrtc.org
f3adba499e Add Android include path so that header files can follow google style 
BUG=1011
TEST=bot
Review URL: https://webrtc-codereview.appspot.com/930018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3107 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-15 18:17:40 +00:00
tommi@webrtc.org
2d0b81b8af Fix sorting issues in video_capture.gypi. 
No code change.

TBR=henrikg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/937022

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3104 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-15 10:30:57 +00:00
andrew@webrtc.org
23ec30bdfc Clean up TraceCallback::Print. 
Review URL: https://webrtc-codereview.appspot.com/936024

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3102 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-15 05:33:25 +00:00
kma@webrtc.org
1786436eb2 Pure Neon assembly coding for WebRtcIsacfix_AutocorrNeon() in iSAC-Fix. 
Review URL: https://webrtc-codereview.appspot.com/939018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3098 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-14 18:44:24 +00:00
mikhal@webrtc.org
9e9cc72b53 Relanding r3071 - updates for i420: Making sure that decoded frame is complete and buffer size is sufficient. Re-landing is possible following r3094 - which disabled a problematic test. 
Review URL: https://webrtc-codereview.appspot.com/939019

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3097 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-14 17:56:46 +00:00
brykt@google.com
71fd288b95 Fixed indentation and added the description of how to supply argument with specification of a name for the ouputfile where the contentMetrics etc. are logged. 
Added description for argument to specify filename for output file where feature vectors are stored.

Fixed indentation

BUG=

Review URL: https://webrtc-codereview.appspot.com/966019

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3096 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-14 14:46:52 +00:00
pwestin@webrtc.org
571a1c035b Enable paced sender. 
Review URL: https://webrtc-codereview.appspot.com/965016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3089 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-13 21:12:39 +00:00
stefan@webrtc.org
42aa10eba7 Clarifies the bandwidth estimation interfaces. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/965019

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3087 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-13 15:02:13 +00:00
tina.legrand@webrtc.org
7577ddf27b Refactoring acm_generic_codec 
First patch: updating comments.

BUG=1024

Review URL: https://webrtc-codereview.appspot.com/936019

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3085 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-13 14:09:35 +00:00
asapersson@webrtc.org
1726661ca2 Update parsed non ref frame info. 
Review URL: https://webrtc-codereview.appspot.com/932015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3084 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-13 13:39:51 +00:00
fischman@webrtc.org
b900721472 Fix OpenGL rendering of WebRTCDemo by accounting for stride != width. 
BUG=998

Review URL: https://webrtc-codereview.appspot.com/936021

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3079 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-12 16:39:25 +00:00
henrik.lundin@webrtc.org
af49062a35 Revert 3071 - i420:verify image length 
This CL breaks vie_auto_test, test case
ViEVideoVerificationTest.RunsBaseStandardTestWithoutErrors.

BUG=

Review URL: https://webrtc-codereview.appspot.com/930016

TBR=mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/933014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3078 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-12 09:16:08 +00:00
pwestin@webrtc.org
b518017e71 Adding pacing module, will replace the transmission_bucket in the RTP module. 
TESTED=unittest
Review URL: https://webrtc-codereview.appspot.com/930015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3073 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-09 20:56:23 +00:00
mikhal@webrtc.org
f875fd22c0 i420:verify image length 
BUG=

Review URL: https://webrtc-codereview.appspot.com/930016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3071 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-09 18:48:30 +00:00
mikhal@webrtc.org
701567a864 Capture module: Fixing size computation for u and v planes 
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/932017

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3070 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-09 18:45:12 +00:00
leozwang@webrtc.org
06d72d881f Add Android OWNER files 
Message:
Add OWNER files so I can review and approve changes for Android.
I also should be owner for all .mk file, but it's OK for now,
please review.

BUG=None
TEST=None
Review URL: https://webrtc-codereview.appspot.com/932016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3069 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-09 17:51:55 +00:00
brykt@google.com
e8ef807a2d Added possibility to run quality modes test. Added possibility to input arguments to the test. The test will (for each frame) log the values in contentMetrics to a txt-file. The txt-file can optionally be saved in a specific place. Fixed an issue where video_coding_test crashed if there weren't any parameter submitted to an input argument. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/772005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3068 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-09 16:16:41 +00:00
kma@webrtc.org
fa65c851fe Optimized function AllpassFilter2FixDec16() in isac fix for Android Neon platforms. 
With an offline test, codec cycles were reduced by 4%.
Review URL: https://webrtc-codereview.appspot.com/936007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3066 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-09 00:39:45 +00:00
wu@webrtc.org
2a749d3108 Verify output frame timestamp in VideoProcessingModuleTest.Resampler. 
TEST=unit tests
BUG=1069

Review URL: https://webrtc-codereview.appspot.com/964014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3063 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-08 01:07:21 +00:00
wu@webrtc.org
206532e6e2 Fix a bug in spatial_resampler where we should set the timestamp after Scale. 
TEST=try bots
BUG=1069

Review URL: https://webrtc-codereview.appspot.com/936018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3061 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-07 23:37:41 +00:00
kma@webrtc.org
12454028bc Fixed and enabled ARM assembly code in AECM and NS. 
Review URL: https://webrtc-codereview.appspot.com/860005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3060 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-07 22:34:31 +00:00
mikhal@webrtc.org
055663be0a Updating vp8 tests 
Review URL: https://webrtc-codereview.appspot.com/936017

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3058 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-07 21:36:15 +00:00
stefan@webrtc.org
7096fc0126 Break out unittest helpers for remote_bitrate_estimator. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/934012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3056 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-07 18:35:30 +00:00
mikhal@webrtc.org
ac993fef2c Adding codecType to OnIncomingCapturedEncodedFrame 
partially reverting r3013.

Review URL: https://webrtc-codereview.appspot.com/965010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3055 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-07 17:18:04 +00:00
pwestin@webrtc.org
c66e8b3f31 pre-factor cleanup pre-work. 
Review URL: https://webrtc-codereview.appspot.com/938010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3054 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-07 17:01:04 +00:00
phoglund@webrtc.org
4cebe6cded Made TickTime immutable, rewrote tick utils to be fakeable. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/798004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3053 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-07 13:37:19 +00:00
tina.legrand@webrtc.org
0ad3c1af0a Adding Opus stereo support to WebRTC 
This CL adds support for sending and receiving stereo using the Opus codec.

BUG=issue1013

Review URL: https://webrtc-codereview.appspot.com/930008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3050 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-07 08:07:29 +00:00
asapersson@webrtc.org
e5b49a0472 Update timestamp offset for re-transmitted packets. 
BUG=1059
Review URL: https://webrtc-codereview.appspot.com/930011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3046 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-06 13:09:39 +00:00
kjellander@webrtc.org
f17d7d1dde Using proper GYP references for Strmiids.lib on Windows 
This makes it possible to link properly using Ninja on Windows.

Another issue exists due to the path containing spaces in webrtc/modules/video_capture/windows/direct_show_base_classes.gyp but that can be worked around by copying the baseclasses dir from the SDK into a path without spaces and then overriding the direct_show_dir variable with GYP_DEFINES.

Example:

set GYP_DEFINES=direct_show_dir=C:/WinSDKv7.1_directshow_baseclasses/
gclient runhooks
ninja -C out/Debug

Notice that the ending slash is needed for the direct_show_dir variable.

BUG=none
TEST=local compilation with the baseclasses copied from Windows SDK 7.1 into C:/WinSDKv7.1_directshow_baseclasses.

Review URL: https://webrtc-codereview.appspot.com/936014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3043 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-05 13:13:05 +00:00
tina.legrand@webrtc.org
f7fa6276e2 Reformating files in audio coding module. 
This CL format the ramaining files on the audio coding module. No other changes are done, except for fixing a few long lines and TODOs without owner.

BUG=issue1024

Review URL: https://webrtc-codereview.appspot.com/928012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3042 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-05 09:35:51 +00:00
leozwang@webrtc.org
5f9970fb0b Refactor OpenSL audio driver 
Message:
I want to start this review, the basica framework is almost done.

Description:
This implementation is similar to current one, but
1. followed the design doc at
https://docs.google.com/a/google.com/document/d/1g5q2SVtkFPl2OSjvSF3eeLb_S7sCnbINxiBUES6XLCM/edit
which uses two threads, playout thread and recording thread, uses large
audio buffer, etc.
2. google code style.

What are missing in this cl,
1. a better way to control schedule/thread priority.
2. java/jni interface to better support what cannot be done in OpenSL.

Please take a review, thanks.
Review URL: https://webrtc-codereview.appspot.com/902004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3040 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-02 16:48:03 +00:00
andrew@webrtc.org
e034f21d8d Remove video_capture/test/android 
Review URL: https://webrtc-codereview.appspot.com/937011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3028 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-10-31 05:36:50 +00:00
andrew@webrtc.org
9841d92b8d Reorganize modules/video_render. 
The usual elimination of main/source etc.

Review URL: https://webrtc-codereview.appspot.com/929011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3027 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-10-31 05:22:11 +00:00
andrew@webrtc.org
94caca7105 Reorganize modules/video_capture. 
- The usual elimination of main/source etc.
- Fully qualify "public" includes.
- Use lowercase for all directory names.

Review URL: https://webrtc-codereview.appspot.com/933010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3025 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-10-30 21:58:00 +00:00
mikhal@webrtc.org
91a0340a61 Adding stride alignment 
TEST= common_video_unittests and video_capture_module_test

BUG=985

Review URL: https://webrtc-codereview.appspot.com/965005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3023 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-10-30 19:19:32 +00:00
mikhal@webrtc.org
c381a8487a Fix valgrind issue. 
Code was removed by mistake in r2983.

BUG=1020

Review URL: https://webrtc-codereview.appspot.com/938006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3021 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-10-30 18:22:02 +00:00
stefan@webrtc.org
6dcef360f8 Follow-up CL for r3007. 
Fixes an initialization issue introduced with r3007 and adds a test for
detecting problems with reordering. Also some cleanup for better code.

TEST=remote_bitrate_estimator_unittests

BUG=1009

Review URL: https://webrtc-codereview.appspot.com/937007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3014 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-10-29 16:06:08 +00:00
mikhal@webrtc.org
e83d3111d4 Removing codecType from capture API 
Review URL: https://webrtc-codereview.appspot.com/964005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3013 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-10-29 15:59:40 +00:00
andrew@webrtc.org
8186534111 Only reinitialize AudioProcessing when needed. 
This takes away the burden from the user, resulting in cleaner code.

Review URL: https://webrtc-codereview.appspot.com/941005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3010 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-10-27 00:28:27 +00:00