Commit Graph

1309 Commits

Author SHA1 Message Date
75f8c78d08 Fixing path to ptypes.txt in NetEqRTPplay
The default path to the file ptypes.txt needed by NetEqRTPplay
had gone old. Updating to new repo layout.

Also purging old payload types from the file itself.

Review URL: https://webrtc-codereview.appspot.com/966035

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3243 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-06 15:22:00 +00:00
226db898f7 Dual-stream implementation, not including VoE APIs.
Review URL: https://webrtc-codereview.appspot.com/933015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3230 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-03 22:13:31 +00:00
277ec8e3f5 Fix a bug when iSAC-48kHz was added.
I discovered this by running extended VoE test on "Codecs."

TBR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/973010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3229 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-03 21:16:23 +00:00
f18de86db1 Revert 3227
> vp8 unittest: Adding qcif stride test
> 
> Review URL: https://webrtc-codereview.appspot.com/930030

TBR=mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/929037

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3228 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-03 20:08:57 +00:00
ab83bb39ad vp8 unittest: Adding qcif stride test
Review URL: https://webrtc-codereview.appspot.com/930030

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3227 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-03 19:12:29 +00:00
b0dff12d2b 48 kHz extension to iSAC.
Test:
-manual test with voe_cmd_test.
-manual test with RTPEncode & NetEqRTPPlay.
-manual test with simpleKenny.
-Bit-exact test of iSAC-swb and iSAC-wb with head revision of trunk. The bit-exactness is confirmed on all files generated by running webrtc/modules/audio_coding/codecs/isac/main/test/QA/runiSACLongtest.txt
Review URL: https://webrtc-codereview.appspot.com/937025

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3226 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-03 17:43:52 +00:00
8d0cd07d0c Add test to verify that padding only frames are passing through the RTP module.
Review URL: https://webrtc-codereview.appspot.com/934023

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3224 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-03 14:01:46 +00:00
5b4fe494e7 Changing default bitrate to 64000 bps for Opus.
Default settings for Opus in WebRtc is stereo, but we had default rate to 32 kbps. This is too low for stereo, where we need to encode using 64 kbps to get the quality we like.

BUG=

Review URL: https://webrtc-codereview.appspot.com/974008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3223 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-03 12:08:53 +00:00
ad0f3baf90 Removing redundant codec unittest targets.
The following targets have been merged into audio_coding_unittests:
* cng_unittests
* g711_unittests
* g722_unittests
* isacfix_unittests
* pcm16b_unittests

Some of them were empty and were created with the assumption they were
needed in order to get code coverage (which was actually not needed).

The following test has been removed since it was empty:
* audio_conference_mixer_unittests

BUG=none
TEST=trybots passing (well, except for the unittests that are removed, they're failing until the try server configuration is updated)

Review URL: https://webrtc-codereview.appspot.com/971008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3222 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-03 10:52:29 +00:00
c94f8d4e8f Fix OOB read in padding tests.
BUG=1177

Review URL: https://webrtc-codereview.appspot.com/973009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3220 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-03 08:57:54 +00:00
fc4a7ee807 Fixes chromium build bots.
BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/971014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3213 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-30 16:17:44 +00:00
bd941d3f4c Fixes two bugs related to padding in the jitter buffer.
- Pad packets (empty) were often NACKed even though they were received.
- Padding only frames (empty) were didn't properly update the decoding state,
  and would therefore be NACKed even though they were received.

This is a recommit of r3183. Extensive testing suggest that this may have been caused by virtual machine flakiness.

TBR=mflodman@webrtc.org

BUG=1150

Review URL: https://webrtc-codereview.appspot.com/971011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3200 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-29 14:37:18 +00:00
8552c71290 Fixing neteq_unittests for VS 2012
For Visual Studio versions older than 2012, we are using a
separate reference output file for windows. (All other platforms
share the same generic reference file.) In VS 2012, the output
matches the generic reference, and not the platform-specific one.

Since, the ResourcePath() method cannot change behavior depending
on compiler version, this fix will short-cut ResourcePath() for
VS 2012 or newer (_MSC_VER >= 1700).

Also made NetEqDecodingTest.TestBitExactnes stop on the first diff.
Once there is a difference, the output is no longer bit-exact, and
the test should be declared a failure.

BUG=
TEST=neteq_unittests on VS2012, try bots

Review URL: https://webrtc-codereview.appspot.com/966028

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3199 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-29 12:03:18 +00:00
273ccad59d Fixed standard PSNR/SSIM test.
BUG=1103

Review URL: https://webrtc-codereview.appspot.com/971005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3197 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-29 10:08:16 +00:00
662651ac95 Disable denoise filter for Arm, as it is not optimized enough yet.
BUG=https://code.google.com/p/chrome-os-partner/issues/detail?id=16318
TEST=none
Review URL: https://webrtc-codereview.appspot.com/968008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3195 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-29 09:01:21 +00:00
f826bb6fb2 Fixing a bug related to RCU in NetEQ
RCU was disabled due to that the RCU flag was overwritten with zero
in the packet buffer.

BUG=1156
TEST=trybots, neteq_unittests, audio_coding_module_test, audio_coding_unittests

Review URL: https://webrtc-codereview.appspot.com/969012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3193 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-29 07:32:38 +00:00
f3cefe1104 Added metrics test code for the FEC packet masks.
The test computes metrics (average residual loss) for each mask type and size, 
for a given set of loss models (random and bursty), and verifies various 
behaviour of the codes (including relation/comparison to RS code).

http://webrtc-codereview.appspot.com/748008
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/929034

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3189 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-28 23:27:34 +00:00
c09e779766 Allow for 1 layer case to be set in temporal_layers.
Review URL: https://webrtc-codereview.appspot.com/971007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3188 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-28 22:06:21 +00:00
7d5dacc985 Revert 3183 - Fixes two bugs related to padding in the jitter buffer.
- Pad packets (empty) were often NACKed even though they were received.
- Padding only frames (empty) were didn't properly update the decoding state,
  and would therefore be NACKed even though they were received.

http://webrtc-cb-linux-master.cbf.corp.google.com:8010/builders/Win32Release/builds/1704/steps/video_coding_unittests/logs/stdio

TEST=trybots

BUG=1150

Review URL: https://webrtc-codereview.appspot.com/929031

TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/971010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3187 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-28 22:04:45 +00:00
c244cefe1d Reverting r3185
TBR=marpan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/933029

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3186 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-28 21:00:36 +00:00
993494764d Added metrics test code for the FEC packet masks.
The test computes metrics (average residual loss) for each mask type and size, 
for a given set of loss models (random and bursty), and verifies various 
behaviour of the codes (including relation/comparison to RS code).
Review URL: https://webrtc-codereview.appspot.com/748008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3185 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-28 19:43:58 +00:00
e4fb44c29d Fixes two bugs related to padding in the jitter buffer.
- Pad packets (empty) were often NACKed even though they were received.
- Padding only frames (empty) were didn't properly update the decoding state,
  and would therefore be NACKed even though they were received.

TEST=trybots

BUG=1150

Review URL: https://webrtc-codereview.appspot.com/929031

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3183 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-28 19:36:20 +00:00
891d55eb35 Revert 3181 - Fixes two bugs related to padding in the jitter buffer.
- Pad packets (empty) were often NACKed even though they were received.
- Padding only frames (empty) didn't properly update the decoding state,
  and would therefore be NACKed even though they were received.

Broke [Builder Win32Debug] (http://webrtc-cb-linux-master.cbf.corp.google.com:8010/builders/Win32Debug/builds/1728)

TEST=trybots

BUG=1150

Review URL: https://webrtc-codereview.appspot.com/966026

TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/939031

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3182 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-28 17:45:01 +00:00
d42e51ce7c Fixes two bugs related to padding in the jitter buffer.
- Pad packets (empty) were often NACKed even though they were received.
- Padding only frames (empty) didn't properly update the decoding state,
  and would therefore be NACKed even though they were received.

TEST=trybots

BUG=1150

Review URL: https://webrtc-codereview.appspot.com/966026

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3181 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-28 16:40:28 +00:00
c4590580e8 Opus mono/stereo on the same payloadtype, and fix of memory bug
During call setup Opus should always be signaled as a 48000 Hz stereo codec, not depending on what we plan to send, or how we plan to decode received packets.
The previous implementation had different payload types for mono and stereo, which breaks the proposed standard.

While working on this CL I ran in to the problem reported earlier, that we could get a crash related to deleting decoder memory. This should now be solved in Patch Set 3.

BUG=issue1013, issue1112

Review URL: https://webrtc-codereview.appspot.com/933022

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3177 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-28 12:23:29 +00:00
81fb7bfd8b Adding video_coding_integrationtests test.
These changes makes it possible to run this tool with some gtest additions in an automated manner on the buildbots.

This test was previously known as video_coding_test, which is an
integration test that is mostly used as a development tool.

Parts of this test should be extracted and kept as a separate
development tool, but that's something for a future CL.

I also refactored the old command line parsing to use gflags instead.

Previous code from the following tests were merged into
video_coding_integrationtests and video_coding_unittests:
* video_codecs_test_framework_integrationtests
* video_codecs_test_framework_unittests
So these targets are now gone.

BUG=none
TEST=trybots passing + Executing video_coding_integrationtests on Linux, Mac and Windows since it's not currently added to the trybots. I ran with a couple of different combinations of settings.

Review URL: https://webrtc-codereview.appspot.com/933026

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3176 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-28 08:40:16 +00:00
8049608226 VP8 wrapper: updating raw image allocation.
As we set the pointers to the data, there is no need to allocate that memory.

Review URL: https://webrtc-codereview.appspot.com/964021

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3175 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-27 17:06:10 +00:00
b43502e388 Revert 3170 - Added performance benchmarking in APM and iSAC-fix for Buildbots.
Review URL: https://webrtc-codereview.appspot.com/929022

TBR=kma@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/969009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3172 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-26 23:57:38 +00:00
4cd8f1f182 Added performance benchmarking in APM and iSAC-fix for Buildbots.
Review URL: https://webrtc-codereview.appspot.com/929022

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3170 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-26 22:02:47 +00:00
ef90c3227e Will now correctly identify the first-ever received packet as the first packet in its frame.
We used to flag the _second_ packet in the first frame as the first. Subsequent frames worked as intended.

BUG=1103
TEST=vie_auto_test --automated, rtp_rtcp_unittests

Review URL: https://webrtc-codereview.appspot.com/964020

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3164 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-26 16:30:40 +00:00
7c894b7cc7 Wire up CallStats to provide modules with correct RTT.
BUG=769
TEST=Manual test since there is no ViE APi to get RTT for receive channels.

Review URL: https://webrtc-codereview.appspot.com/937027

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3163 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-26 12:40:15 +00:00
5ba3decc94 Ensures that we can build using VS 2012 on Windows.
See more details at https://code.google.com/p/webrtc/issues/detail?id=1146&

TBR=Niklas
BUG=1146

Review URL: https://webrtc-codereview.appspot.com/939028

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3162 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-26 09:12:02 +00:00
418443c531 Remove operator overloading from RTPFragmentationHeader.
Instead supply a CopyFrom() method.

TEST=vie_auto_test

Review URL: https://webrtc-codereview.appspot.com/972004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3158 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-23 19:17:23 +00:00
6ba79a88e5 Condition for DirectX variable on Windows
The directx_sdk_path GYP variable got the value $(DXSDK_DIR) on non-windows platforms which is normally an uninitialized environment variable, causing an error during GYP generation.
Putting this include within a condition for Windows resolves this.

This was only triggered when GYP_GENERATORS=ninja and not for the default on Linux (make), so the bots haven't noticed this.

BUG=none
TEST=All default trybots passing. Successfully generating projects on Linux and Mac for make and ninja (plus XCode on Mac). Successful compile on Windows without DirectX SDK installed (but with files located in third_party/directxsdk/files).

Review URL: https://webrtc-codereview.appspot.com/936031

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3156 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-23 09:41:42 +00:00
97dcf36a7b Adding Direct X SDK include directory.
This makes it possible to keep a copy of the Direct X SDK in third_party/directxsdk/files and get it automatically used instead of having to install it manually on the system.

BUG=none
TEST=Compilation with SDK files in third_party/directxsdk/files  and uninstalled Direct X SDK on Windows.
Review URL: https://webrtc-codereview.appspot.com/937028

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3153 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-23 08:54:07 +00:00
1c61196095 Removed not used include.
TEST=Compiles.

Review URL: https://webrtc-codereview.appspot.com/966025

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3150 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-22 09:37:27 +00:00
4c4d01dd3d Setting capture stride to width
Review URL: https://webrtc-codereview.appspot.com/935021

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3148 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-21 22:18:32 +00:00
4b97793f91 Ensure opus_demo has a targets block.
TBR=leozwang@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/933023

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3147 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-21 20:16:53 +00:00
8cd18c55b8 Add winsdk_samples to provide directshow_baseclasses.
Builds locally on Windows (and passes try), and a gyp run succeeds with
include_internal_video_capture=0. This should ensure it won't impact
Chromium.

Review URL: https://webrtc-codereview.appspot.com/937026

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3146 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-21 18:46:45 +00:00
cfcadab8d4 Build opus_demo
BUG=1082
TEST=trybots

With this cl we can build opus_demo.
Review URL: https://webrtc-codereview.appspot.com/936029

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3144 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-21 18:13:46 +00:00
3263a7a616 Setting correct stride for VP8 encoder
BUG=1137

Review URL: https://webrtc-codereview.appspot.com/929024

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3140 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-21 00:15:19 +00:00
8187877402 Reland 3135 - Previous failure was bot flakiness.
*****

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

TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/934017

TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/934018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3137 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-20 13:35:33 +00:00
951b6c404a Revert 3135 - This broke the Mac bots somehow. Here's the error:
[ RUN      ] ViEStandardIntegrationTest.RunsBaseTestWithoutErrors
2012-11-20 13:06:59.625 vie_auto_test[4001:f07] An uncaught exception was raised
2012-11-20 13:06:59.625 vie_auto_test[4001:f07] Error (1000) creating CGSWindow on line 259
2012-11-20 13:06:59.670 vie_auto_test[4001:f07] (
	0   CoreFoundation                      0x9a4e912b __raiseError + 219
	1   libobjc.A.dylib                     0x95ee252e objc_exception_throw + 230
	2   CoreFoundation                      0x9a448bbb +[NSException raise:format:] + 139
	3   AppKit                              0x996b4757 _NSCreateWindowWithOpaqueShape2 + 302
	4   AppKit                              0x996b2f40 -[NSWindow _commonAwake] + 1823
	5   AppKit                              0x9966fa77 -[NSWindow _commonInitFrame:styleMask:backing:defer:] + 1652
	6   AppKit                              0x9966eb3f -[NSWindow _initContent:styleMask:backing:defer:contentView:] + 1063
	7   AppKit                              0x9966e704 -[NSWindow initWithContentRect:styleMask:backing:defer:] + 70
	8   vie_auto_test                       0x0015c232 -[TestCocoaUi createWindows:] + 338
	9   libobjc.A.dylib                     0x95eef5d3 -[NSObject performSelector:withObject:] + 70
	10  Foundation                          0x969050c0 __NSThreadPerformPerform + 395
	11  CoreFoundation                      0x9a3bf66f __CFRUNLOOP_IS_CALLING_OUT_TO_A_SOURCE0_PERFORM_FUNCTION__ + 15
	12  CoreFoundation                      0x9a3bf099 __CFRunLoopDoSources0 + 233
	13  CoreFoundation                      0x9a3e4e46 __CFRunLoopRun + 934
	14  CoreFoundation                      0x9a3e463a CFRunLoopRunSpecific + 378
	15  CoreFoundation                      0x9a3e44ab CFRunLoopRunInMode + 123
	16  Foundation                          0x9690d946 -[NSRunLoop(NSRunLoop) runMode:beforeDate:] + 278
	17  vie_auto_test                       0x002e490a main + 522
	18  vie_auto_test                       0x000b1835 start + 53

*****


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

TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/934017

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3136 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-20 12:17:05 +00:00
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
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
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
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
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
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
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