wu@webrtc.org
d6b827a28e
Fix for the build broken on Windows.
...
Review URL: http://webrtc-codereview.appspot.com/335017
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1341 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-04 22:38:05 +00:00
mikhal@webrtc.org
a58888d382
Updating capture module following latest libyuv api changes
...
Review URL: http://webrtc-codereview.appspot.com/337009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1338 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-04 19:23:24 +00:00
mikhal@webrtc.org
7d5ca2be1f
Updating render module following latest libyuv api changes.
...
Review URL: http://webrtc-codereview.appspot.com/331019
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1337 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-04 19:01:48 +00:00
kma@webrtc.org
746f9e31c0
Changed build settings for ARMv5 in Android.
...
I found some issues in building ARMv5 with ICM. This CL includes fixes,
and a design change which now will exclude any NEON libraries unless
the build is for dynamic detection or for Neon specifically.
Review URL: http://webrtc-codereview.appspot.com/330021
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1335 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-04 17:47:57 +00:00
pwestin@webrtc.org
6c1d41583a
Fix for RTP extension audio level.
...
Review URL: http://webrtc-codereview.appspot.com/339002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1334 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-04 17:04:51 +00:00
andrew@webrtc.org
d77a6614fa
Consts can't be used as C array size initializers.
...
(Unless you happen to be using clang...)
TBR=bjornv@webrtc.org
TEST=build on gcc
Review URL: http://webrtc-codereview.appspot.com/333029
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1333 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-04 16:22:24 +00:00
henrik.lundin@webrtc.org
d047b2e7f6
Enabling NetEQ unittest for more platforms
...
Removing platform limitations for NetEqDecodingTest:TestBitExactness
and NetEqDecodingTest:TestNetworkStatistics. New reference files
where provided in revision 6 of the resources, which allows us
to enable these tests.
BUG=
TEST=neteq_unittests linux32/64, win32/64, mac32
Review URL: http://webrtc-codereview.appspot.com/329027
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1332 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-04 16:10:23 +00:00
andrew@webrtc.org
3905b0c45d
Protect against divide-by-zeros in AGC.
...
TEST=audioproc_unittest
Review URL: http://webrtc-codereview.appspot.com/333024
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1331 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-04 15:47:20 +00:00
pwestin@webrtc.org
c450a19669
Removed Version function from all modules.
...
TBR=henrik_a
Review URL: http://webrtc-codereview.appspot.com/329023
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1330 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-04 15:00:12 +00:00
henrik.lundin@webrtc.org
d439870473
Adding two new network metrics to NetEQ
...
Clock-drift (in parts-per-million) and peaky-jitter mode status.
Both metrics are propagated to the VoE API. Tests are added
in the NetEQ unittests, and to some extent in ACM unittests
and VoE tests.
Introducing a proper translation between structs NetworkStatistics
and ACMNetworkStatistics.
Note: The file neteq_network_stats.dat in resources must be updated
for the unittests to pass.
Review URL: http://webrtc-codereview.appspot.com/337005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1328 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-04 13:09:55 +00:00
bjornv@webrtc.org
80d28b22b9
Changed to new ring buffer in AECM.
...
Replaced the old ring buffer in AECM with the new one. Also removed the old one from ring_buffer.
Changes are bit exact according to audioproc_unittest fixed.
TEST=audioproc_unittest
Review URL: http://webrtc-codereview.appspot.com/331022
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1327 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-04 09:55:09 +00:00
kjellander@webrtc.org
cc33737a80
Changing all PSNR/SSIM calculations to use libyuv.
...
Removed old PSNR/SSIM implementations in:
* test/testsupport/metrics/video_metrics.cc
* src/modules/video_coding/codecs/test_framework/test.cc
The functions in video_metrics.cc is now using code in libyuv instead. Old code in test.cc is using the same functions.
The code for video_metrics.h had to be moved into a separate GYP file to avoid circular dependency error on Mac (see issue 160 for more details). The reason for this is that libyuv's unittest target depends on test_support_main.
BUG=
TEST=metrics_unittests in Debug+Release on Linux, Mac and Windows.
Review URL: http://webrtc-codereview.appspot.com/333025
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1325 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-04 08:09:32 +00:00
turaj@webrtc.org
a574b1c617
The inline implementation of WebRtcIsac_lrint(), which was implemented in several files, is now os_specific_inline.h. Define guards are modified according to WebRtc OS macros.
...
This resolves BUG=issue137.
Review URL: http://webrtc-codereview.appspot.com/269014
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1323 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-04 02:26:23 +00:00
mikhal@webrtc.org
cd64886a2f
video_coding: Updating NACK functions naming
...
Review URL: http://webrtc-codereview.appspot.com/329018
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1322 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-03 23:59:42 +00:00
punyabrata@webrtc.org
8fa31bc4e5
Truncated messages, need a %S instead of $s for a double byte TCHAR
...
Review URL: http://webrtc-codereview.appspot.com/333002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1321 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-03 22:34:15 +00:00
mikhal@webrtc.org
77c425b976
video_coding: Checking/updating seq num for an old packet regardless of size.
...
Review URL: http://webrtc-codereview.appspot.com/330028
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1318 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-03 20:35:25 +00:00
henrik.lundin@webrtc.org
6c877363f7
Fix formatting for some NetEQ test tools
...
Format and lint for RTPchange.cc, RTPcat.cc and RTPanalyze.cc.
Review URL: http://webrtc-codereview.appspot.com/329024
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1313 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-03 10:03:19 +00:00
kjellander@webrtc.org
a643d5c4ef
Integration test for videoprocessor
...
Added temporal layers number flag for video_quality_measurement tool.
This tool now also uses webrtc::VideoCodingModule::Codec() to get its
VideoCodec struct configuration instead of filling it in manually.
Updated paths for header files to use full directory paths.
Tested in Debug+Release on Linux, Mac and Windows. Passes Valgrind memcheck on Linux.
BUG=
TEST=video_codecs_test_framework_integrationtests. Also executed out/Debug/video_quality_measurement --input_filename=resources/foreman_cif.yuv --width=352 --height=288
Review URL: http://webrtc-codereview.appspot.com/339001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1310 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-30 14:44:07 +00:00
mikhal@webrtc.org
62665b8cd3
video_coding: Adding a unit test to the decodableState class
...
Review URL: http://webrtc-codereview.appspot.com/315001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1309 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-29 18:09:58 +00:00
mikhal@webrtc.org
9eeafbef3c
Updating the frame buffer return value in InsertPacket: Return NoError when a packet is inserted to a frame which is being decoded.
...
Review URL: http://webrtc-codereview.appspot.com/330027
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1308 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-29 17:38:56 +00:00
mikhal@webrtc.org
bed34a341a
video_coding: Updating seq number for old zero size packets. Updating function name to reflect zero size packets and not empty packets.
...
Review URL: http://webrtc-codereview.appspot.com/333009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1307 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-29 17:37:39 +00:00
bjornv@webrtc.org
70adcd46b2
Delay estimator improvements.
...
Robustness improvements to the delay estimator used in AECM and AEC. In AEC only for logging. Faster convergence.
TEST=audioproc_unittest + offline file tests.
output_data_fixed.pb updated despite unverified changes in r1112.
Review URL: http://webrtc-codereview.appspot.com/337006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1306 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-29 14:51:21 +00:00
stefan@webrtc.org
efd0a48c61
Add error resilient mode options to the VP8 specific VideoCodec struct.
...
It is useful to disable error resilience when we know we won't decode
with errors.
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/329015
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1305 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-29 10:12:35 +00:00
andrew@webrtc.org
6d6a43d6e3
Use char as ring-buffer data type.
...
- Avoids a bunch of char* casts.
- Use enum type rather than char.
TEST=audioproc_unittest on Linux (float and fixed), build on Windows
Review URL: http://webrtc-codereview.appspot.com/336010
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1303 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-28 22:40:15 +00:00
bjornv@webrtc.org
267d0133ff
Fixed pointer operations on void.
...
This should fix the error on Win where pointer arithmetics are done on void pointers. Type cast to char to interpret a size.
Review URL: http://webrtc-codereview.appspot.com/329019
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1300 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-28 10:26:17 +00:00
bjornv@webrtc.org
7270a6bcc2
Merged apm-buffer branch [r1293] back to trunk.
...
This CL includes a larger structural change in how we handle buffers in AEC. We now perform FFT at once and move within blocks to compensate for system delays.
TEST=audioproc_unittest(float and fix), voe_auto_test
Review URL: http://webrtc-codereview.appspot.com/335012
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1299 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-28 08:44:17 +00:00
mikhal@webrtc.org
e39de16fa5
Moving video type convert functionality to libyuv. deleting vplibConversions as it is no longer needed.
...
Review URL: http://webrtc-codereview.appspot.com/338002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1298 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-27 23:45:30 +00:00
stefan@webrtc.org
f6c6b1c5b5
Include the media packet FEC headers in the video bitrate.
...
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/328014
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1296 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-23 10:33:39 +00:00
stefan@webrtc.org
39670f6aa6
Only reset the last decoded sequence number after flushing until key frame.
...
We can't reset the complete last decoded state when we recycle until a
key frame because that will allow any delta frame to be decoded afterwards,
and since the decoder isn't reset we will get decode errors.
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/330003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1295 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-23 09:08:51 +00:00
mflodman@webrtc.org
1ce66e4dfb
Don't report error when failing to send RTCP BYE.
...
Review URL: http://webrtc-codereview.appspot.com/337002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1293 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 18:40:15 +00:00
stefan@webrtc.org
6a4bef4e65
Implements selective retransmissions.
...
Default is set to not retransmit VP8 non-base layer packets or FEC packets.
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/323010
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1290 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 12:52:41 +00:00
pwestin@webrtc.org
f4d3b9d5a1
Cleaned up leaky symbols in NS.
...
Review URL: http://webrtc-codereview.appspot.com/337001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1288 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 12:33:08 +00:00
pwestin@webrtc.org
ebcb6421b1
Cleaned up leaky symbols in G722.
...
Review URL: http://webrtc-codereview.appspot.com/333017
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1287 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 12:20:06 +00:00
pwestin@webrtc.org
d8f8b32521
Cleaned up leaky symbols in iSAC.
...
Review URL: http://webrtc-codereview.appspot.com/329014
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1286 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 12:19:43 +00:00
mflodman@webrtc.org
84dc3d134d
Add REMB functionality to ViE.
...
This CL only adds support for encoding one stream, but receiving multiple streams.
BUG=
TEST=video_engine_core_unittest + auto_test/loopback
Review URL: http://webrtc-codereview.appspot.com/333011
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1284 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 10:26:13 +00:00
pwestin@webrtc.org
093ffad26b
Removed unused function messing up the symbols.
...
Review URL: http://webrtc-codereview.appspot.com/336006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1283 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 09:48:48 +00:00
henrik.lundin@webrtc.org
1e28d3c2e1
Change VP8 packetizer to use a single max payload size
...
The packetizer class is changed so that the max payload size is
provided on construction of the class rather than for each packet.
The tests are re-written to comply with the new design.
Also fixing a few errors in the tests.
Review URL: http://webrtc-codereview.appspot.com/335010
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1280 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 08:49:31 +00:00
stefan@webrtc.org
f5edb923b1
Remove unused variable.
...
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/333016
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1279 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 08:34:31 +00:00
pwestin@webrtc.org
8edb39db30
Prevent sending empty RTCP packet.
...
Review URL: http://webrtc-codereview.appspot.com/331009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1277 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 07:40:33 +00:00
henrik.lundin@webrtc.org
4a19030131
New VCM robustness API
...
This CL defines and starts to implement a new robustness API for
video coding module. The API is partly implemented. Some of the
modes and methods are still TBD.
Also including a new unittest with mocking of decoder and callbacks,
and faking of system clock.
Review URL: http://webrtc-codereview.appspot.com/333006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1276 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-21 20:38:37 +00:00
andrew@webrtc.org
697bc43b67
Restore item deletions in Windows UDP.
...
TEST=voe_auto_test on Windows.
Review URL: http://webrtc-codereview.appspot.com/331013
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1275 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-21 19:58:11 +00:00
andrew@webrtc.org
71571c5446
Remove unneeded variables from windows UDP.
...
TEST=build on Windows.
Review URL: http://webrtc-codereview.appspot.com/329013
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1274 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-21 18:30:59 +00:00
mallinath@webrtc.org
03532b5f41
Fixing the double delete problem in UdpSocket2ManagerWindow. PopFront deletes the items, to there is no need to delete item explicitly.
...
Review URL: http://webrtc-codereview.appspot.com/333014
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1268 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-21 15:36:44 +00:00
henrik.lundin@webrtc.org
7d8c72e2db
Re-implement dependency injection of TickTime into VCM and tests
...
This change basicly re-enables the change of r1220, which was
reverted in r1235 due to Clang issues.
The difference from r1220 is that the TickTimeInterface was
renamed to TickTimeClass, and no longer inherits from TickTime.
Review URL: http://webrtc-codereview.appspot.com/335006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1267 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-21 15:24:01 +00:00
kjellander@webrtc.org
5490c71a1b
Converted to gtest, writing output files properly and no longer uses exceptions.
...
This test now runs and fails as a gtest should (previously it always
exited with 0 even if the tests failed).
The audio_coding_module_test target no longer uses exceptions in the generated project.
Output files are written to our global output folder, using
testsupport/fileutils.h.
BUG=
TEST=audio_coding_module_test on all platforms, in Debug+Release
Review URL: http://webrtc-codereview.appspot.com/334004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1266 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-21 13:34:18 +00:00
stefan@webrtc.org
898f881e32
Make sure the next frame to be decoded is cleaned up if it's empty.
...
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/332001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1261 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-21 09:13:14 +00:00
niklas.enbom@webrtc.org
6c9be123ef
Letting strncpy do its job. Landing and extending http://webrtc-codereview.appspot.com/329010/ on behalf of tbreisacher.
...
Review URL: http://webrtc-codereview.appspot.com/335009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1260 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-21 08:59:31 +00:00
stefan@webrtc.org
8c5d24266e
Fix VP8 layer 2 sync dependencies.
...
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/333010
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1259 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-21 08:56:04 +00:00
henrik.lundin@webrtc.org
00e730730e
Refactoring RtpFormatVp8Test
...
This is the first change in a series of changes to get new functionality
into the VP8 packetizer.
This first refactors the RtpFormatVp8Test class, without changing the
operation of the tested RtpFormatVp8 class. A test helper class
RtpFormatVp8TestHelper is introduced to reduce code duplication.
Review URL: http://webrtc-codereview.appspot.com/304009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1258 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-21 08:51:36 +00:00
mikhal@webrtc.org
61045a4a03
video_coding/jitter_buffer: Account for layer info when searching for the next frame
...
Review URL: http://webrtc-codereview.appspot.com/328003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1256 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-20 23:24:19 +00:00
andrew@webrtc.org
a38ce09919
Fix last Mac/clang compile error.
...
Fixes "receiver is a forward class and corresponding @interface may
not exist" error.
TEST=build on Mac with -Werror enabled.
TBR=zakkhoyt@webrtc.org
Review URL: http://webrtc-codereview.appspot.com/333012
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1255 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-20 22:23:21 +00:00
pwestin@webrtc.org
061fa5b828
Changed handling of padding data.
...
Review URL: http://webrtc-codereview.appspot.com/331008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1252 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-20 15:56:17 +00:00
henrik.lundin@webrtc.org
dbba1f969f
Packet waiting-time statistics
...
Adding new statistics API to NetEQ, reporting the waiting time
for each frame. The output is raw waiting time for the frames
that have been decoded since the last statistics report (or
maximum 100 frames). The statistics are reset on each query.
Implemented functionality in ACM to query NetEQ for the raw
waiting times, and process it to produce max, average and
median.
Updating common_types.h and VoiceEngine tests to include the
new metrics.
Unit tests are also added for NetEQ and AcmNetEq.
Review URL: http://webrtc-codereview.appspot.com/328011
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1251 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-20 15:45:05 +00:00
henrik.lundin@webrtc.org
219acc6cec
Including Brighten function in namespace VideoProcessing
...
This change is in response to Issue 173.
BUG=http://code.google.com/p/webrtc/issues/detail?id=173
Review URL: http://webrtc-codereview.appspot.com/328012
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1250 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-20 15:33:49 +00:00
stefan@webrtc.org
62fdc42e9c
Fix build issue with clang.
...
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/330009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1244 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-20 09:27:09 +00:00
stefan@webrtc.org
8dc9e4760e
Fixes for selective NACKing.
...
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/332007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1243 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-20 09:12:50 +00:00
tina.legrand@webrtc.org
5efcad1758
We used the wrong syntax for "new", which generated a warning/error building with clang.
...
Review URL: http://webrtc-codereview.appspot.com/336003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1241 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-20 09:05:55 +00:00
mikhal@webrtc.org
0e7d9d862a
Adding layer info consideration when applying FEC protection. In this first version, we hard code protection zero for non-base layer frames. As a future enhancement, an array should be passed from mediaOpt to set the protection per layer. A TODO was added in MediaOpt.
...
Review URL: http://webrtc-codereview.appspot.com/330005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1238 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-19 19:04:49 +00:00
mikhal@webrtc.org
190e88a6d3
video_coding: When in hybrid mode, don't NACK non-base layer packets
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Review URL: http://webrtc-codereview.appspot.com/334002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1237 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-19 18:57:14 +00:00
mikhal@webrtc.org
884d8e7f4b
video_coding: Updating sync state based on the layer flag
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Review URL: http://webrtc-codereview.appspot.com/333004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1236 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-19 18:53:05 +00:00
henrik.lundin@webrtc.org
303158588b
Revert "Inject TickTimeInterface into VCM and tests"
...
This CL reverts r1220.
Review URL: http://webrtc-codereview.appspot.com/336002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1235 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-19 17:55:45 +00:00
henrika@webrtc.org
e32c08a5a6
Removes usage of default parameters and fixes a bug which was found
...
using Clang on Linux.
BUG=none
TEST=none
TBR=pwestin
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1234 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-19 17:39:48 +00:00
stefan@webrtc.org
b33f9dccd6
Correction to how the VP8 wrapper generates picture ids.
...
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/329006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1229 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-19 10:24:02 +00:00
tina.legrand@webrtc.org
398af2337b
Solving issue 178, errorbuild warnings on Mac.
...
This CL continues the work of solving issue 178. A small change in one file.
Review URL: http://webrtc-codereview.appspot.com/330006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1227 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-19 07:36:07 +00:00
henrike@webrtc.org
cf5bcd1fd2
Removed usage of the deprecated critical section constructor in audio_conference_mixer.
...
Review URL: http://webrtc-codereview.appspot.com/320007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1226 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-16 23:00:30 +00:00
andrew@webrtc.org
8a44259ea8
Move static consts out of class.
...
Still causing a gtest error on non-Win platforms. This should fix it...
TBR=asapersson@webrtc.org
TEST=build on Linux
Review URL: http://webrtc-codereview.appspot.com/332006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1225 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-16 21:24:30 +00:00
andrew@webrtc.org
41192469f6
Switch enums to consts to fix gtest error.
...
TBR=asapersson@webrtc.org
TEST=build on Windows
Review URL: http://webrtc-codereview.appspot.com/330008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1224 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-16 20:55:46 +00:00
henrike@webrtc.org
105e07193e
Removed usage of the deprecated critical section constructor in modules/utility.
...
Review URL: http://webrtc-codereview.appspot.com/321006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1223 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-16 19:53:46 +00:00
marpan@webrtc.org
57353a33f1
FEC Receiver: Fix to how old packets (e.g., re-tranmitted packets in hybird NACK-FEC mode) are treated.
...
This change avoids having old packets end up on the current packet list for FEC decoding, and so they are immediately sent out to jitter buffer.
The current list of packets for FEC decoding are sent out only when new packet arrives (with time-stamp greater than current).
Review URL: http://webrtc-codereview.appspot.com/322009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1222 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-16 17:21:09 +00:00
henrik.lundin@webrtc.org
e7d8c56c56
Fix for dual decoder in VCM receiver
...
In VCMReceiver::FrameForDecoding, one of the if-cases could sometimes
extract an incomplete frame without first copying the state to the
dual decoder.
Review URL: http://webrtc-codereview.appspot.com/328006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1221 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-16 15:40:52 +00:00
henrik.lundin@webrtc.org
a70f945086
Inject TickTimeInterface into VCM and tests
...
The purpose of this change is to introduce dependency injection
of the timer into the video coding module.
Review URL: http://webrtc-codereview.appspot.com/332003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1220 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-16 14:40:05 +00:00
asapersson@webrtc.org
5249cc8f77
Review URL: http://webrtc-codereview.appspot.com/295010
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1219 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-16 14:31:37 +00:00
tina.legrand@webrtc.org
9775a30859
Added variable to catch return value.
...
Review URL: http://webrtc-codereview.appspot.com/329004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1218 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-16 11:15:46 +00:00
kjellander@webrtc.org
08dec7f449
Now using fileutils.h OutputPath to write output to the right directory and ResourcePath to read resource files from the resource bundle.
...
Removed some Valgrind warnings by closing output files. There are still some Valgrind warnings left, that needs to be fixed by a developer with more insight.
Updated all include paths to contain full paths to header files.
Tested in Debug+Release on Linux, Mac and Windows.
All tests ran successfully except the VideoProcessingModuleTest.ContentAnalysis test that fails on Windows with the following error:
unknown file: error: SEH exception with code 0xc0000005
thrown in the test body.
Fixing that is out of scope for this CL.
Review URL: http://webrtc-codereview.appspot.com/266011
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1217 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-16 10:31:38 +00:00
tina.legrand@webrtc.org
554ae1ad4e
Changes to solve warnings on Mac, issue #178 .
...
Review URL: http://webrtc-codereview.appspot.com/320005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1216 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-16 10:09:04 +00:00
henrike@webrtc.org
7136990a3f
Removed usage of the deprecated critical section constructor in udp_transport.
...
Review URL: http://webrtc-codereview.appspot.com/321005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1211 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-15 19:17:28 +00:00
leozwang@webrtc.org
0c839fe873
Add new source file to makefile
...
Review URL: http://webrtc-codereview.appspot.com/322015
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1209 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-15 19:10:24 +00:00
henrik.lundin@webrtc.org
0a10e3c4b2
Fix order of include and guard in tick_time_interface.h
...
Review URL: http://webrtc-codereview.appspot.com/331002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1207 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-15 16:08:36 +00:00
henrik.lundin@webrtc.org
c74b2861f3
Fix the include in fake_tick_timer_interface.h
...
The include was in error.
Review URL: http://webrtc-codereview.appspot.com/330002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1204 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-15 11:28:31 +00:00
kma@webrtc.org
ee36b9587d
corrected android makefile for isac build.
...
Review URL: http://webrtc-codereview.appspot.com/321013
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1200 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-15 00:18:45 +00:00
andrew@webrtc.org
59ccd5c71f
Rename _windows.h -> _win.h in system_wrappers.
...
- Also rename _dummy -> no_op which states its purpose more clearly.
- Always use exclusion lists (i.e. sources! instead of sources)
TEST=builds and passes system_wrapper_unittest on Linux, Mac, Win
Review URL: http://webrtc-codereview.appspot.com/317007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1199 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-15 00:17:43 +00:00
kma@webrtc.org
6a17340db5
Review URL: http://webrtc-codereview.appspot.com/318014
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1197 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-14 22:16:57 +00:00
kma@webrtc.org
a30093bb85
Added one file associated with check in in r1192.
...
Review URL: http://webrtc-codereview.appspot.com/320012
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1194 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-14 19:57:23 +00:00
leozwang@webrtc.org
9aa9f44ebc
Add new source files because of r1174
...
Review URL: http://webrtc-codereview.appspot.com/320011
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1193 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-14 19:08:33 +00:00
kma@webrtc.org
f0a964dc0a
Optimized WebRtcIsacfix_NormLatticeFilterMa() function for iSAC fix for ARM Neon
...
architecture with intrinsics and assembly code. The total iSAC codec speech improved
about 3~5%.
Notes
(1) The Neon version after this optimization is not bit-exact with the generic
C version. The out quality, however, is not worse as verified by test vectors ouput,
and undertandably in theory (32bit x 32bit in Neon is more accurate than the approximation
C code in the generic version).
(2) In Android, a isac neon library will be built. Along with some new function structures,
it is partly for preparation of introducing a run time detection of Neon architecture soon.
Review URL: http://webrtc-codereview.appspot.com/268016
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1192 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-14 18:59:43 +00:00
kma@webrtc.org
6601902504
Introduced WebRtcSpl_SatW32ToW16 to iSAC fix, for Android platforms.
...
Review URL: http://webrtc-codereview.appspot.com/315005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1190 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-14 18:41:07 +00:00
leozwang@webrtc.org
f147bbc878
Change codec test app lib dependency from webrtc lib to codec library
...
Review URL: http://webrtc-codereview.appspot.com/317009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1189 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-14 18:22:41 +00:00
henrike@webrtc.org
7cdcde3460
Removed usage of the deprecated critical section constructor in media_file.
...
Review URL: http://webrtc-codereview.appspot.com/321004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1187 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-14 17:27:58 +00:00
stefan@webrtc.org
780a07a843
Fix infinite loop bug introduced in r1174.
...
Merges CleanUpSizeZeroFrames with CleanUpOldFrames, and changes the
behavior to go through all frames looking for empty frames.
TBR=mikhals
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/318013
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1186 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-14 15:55:19 +00:00
pwestin@webrtc.org
9fe3d51372
Set the new layer sync bit in the VP8 info struct.
...
Review URL: http://webrtc-codereview.appspot.com/324010
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1185 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-14 15:13:04 +00:00
henrik.lundin@webrtc.org
fbf5af444b
Adding a mockable wrapper class for TickTime in VCM
...
The class is called TickTimeInterface, with a fake implementation in FakeTickTime.
Review URL: http://webrtc-codereview.appspot.com/323012
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1183 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-14 10:36:10 +00:00
stefan@webrtc.org
ef5247b5b1
Fix session_info_unittest error.
...
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/324009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1182 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-14 10:25:38 +00:00
stefan@webrtc.org
0c40d3315f
Fixes an assert triggered in jitter_buffer_test and disables deblocking.
...
When deblocking is enabled the first frames can include uninitialized
memory. Disabling for now.
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/320010
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1181 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-14 09:39:30 +00:00
andrew@webrtc.org
6d609b59f3
Fix crashes due to static_instance.
...
- Initialize a needed critsect in the constructor of
UdpSocket2ManagerWindows.
- Don't return NULL when creating a static instance.
TEST=voe_auto_test on Windows.
Review URL: http://webrtc-codereview.appspot.com/324008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1177 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-14 02:36:33 +00:00
andrew@webrtc.org
5ae19de3ec
Fix error in RtpDump::Start due to r1156.
...
- r1156 fixed a check on the _text member of FileWrapper. Turns out this
was incompatibile with the RTP dumps, which want to write both binary
and text data. Writing text data to a file open as "b" isn't actually
an error, so I simply removed the check.
- Also cleans up the interface, most notably removing all WebRtc types.
TEST=vie_auto_test
Review URL: http://webrtc-codereview.appspot.com/317005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1175 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-13 22:59:33 +00:00
mikhal@webrtc.org
832cacacff
video-coding: Adding a decoded state to the JB logic (JB refactor).
...
This new class stores the last decoded info, including temporal info.
Review URL: http://webrtc-codereview.appspot.com/300005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1174 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-13 21:15:05 +00:00
henrike@webrtc.org
65573f2922
Removed usage of the deprecated critical section constructor in rtp_rtcp.
...
Review URL: http://webrtc-codereview.appspot.com/315004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1173 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-13 19:17:27 +00:00
stefan@webrtc.org
f4c8286222
Pass NACK and FEC overhead rates through the ProtectionCallback to VCM.
...
These overhead rates are used by the VCM to compensate the source
coding rate for NACK and FEC.
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/323003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1171 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-13 15:38:14 +00:00
henrik.lundin@webrtc.org
1ced840893
Fixing a nit in the unittest
...
This caused some of the build bots to fail.
Review URL: http://webrtc-codereview.appspot.com/324005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1170 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-13 14:59:00 +00:00
henrik.lundin@webrtc.org
eda86dc76b
Adding a LayerSync bit to VP8 RTP header
...
Updated RtpFormatVp8, ModuleRTPUtility, VP8Encoder and VP8Decoder
to support a new LayerSync ("Y") bit. Note, in VP8Encoder the bit
must be used together with a non-negative value for temporalIdx.
Fixing the plumbing between RTP module and and from VP8 wrapper.
Updating unit tests; all pass.
The new bit is yet to be used by the VP8 wrapper.
Review URL: http://webrtc-codereview.appspot.com/323008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1169 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-13 14:11:06 +00:00
henrik.lundin@webrtc.org
4aae0e489f
Shaping up formatting of rtp_utility_test.cc
...
Preparations for future work in this file.
Review URL: http://webrtc-codereview.appspot.com/318011
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1168 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-13 09:15:21 +00:00
stefan@webrtc.org
076fa6e674
The second step towards a list based SessionInfo.
...
Added unittests for most of public functions of SessionInfo.
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/301014
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1166 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-13 07:54:56 +00:00
mikhal@webrtc.org
352ade7023
video_coding: Allocating encoded buffer based on length and not size
...
Review URL: http://webrtc-codereview.appspot.com/318010
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1163 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-13 00:54:28 +00:00
stefan@webrtc.org
1480f02faf
Fix VCM test build warnings on Mac with clang.
...
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/318008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1160 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-12 13:45:59 +00:00
stefan@webrtc.org
7889a9b49a
Remove use of CriticalSectionScoped(CriticalSectionWrapper& critsect) in VCM.
...
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/318005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1159 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-12 08:18:24 +00:00
mikhal@webrtc.org
ea71440aec
video_coding: Adding the non reference flag to the receive side logic.
...
Review URL: http://webrtc-codereview.appspot.com/323005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1157 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-12 02:29:34 +00:00
henrike@webrtc.org
315282c01a
Fixes a compiler warning related to dynamically allocated static memory. the fix is to leak the memory since the OS will clean it up anyways. This will not add noise to memory tools so it's ok. The issue is reported here: http://code.google.com/p/webrtc/issues/detail?id=147 .
...
Review URL: http://webrtc-codereview.appspot.com/267023
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1150 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-09 17:46:20 +00:00
xians@webrtc.org
0744ee563d
Disable API tests on ALSA since the tests don't work for all the alsa devices.
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Review URL: http://webrtc-codereview.appspot.com/317004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1147 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-09 14:05:29 +00:00
henrik.lundin@webrtc.org
6198624815
Remove warnings on Mac (Issue 178)
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Remove an if-else that can never execute the else statement.
Remove double parenthesis.
BUG=http://code.google.com/p/webrtc/issues/detail?id=178
TEST=
Review URL: http://webrtc-codereview.appspot.com/318004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1146 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-09 13:58:17 +00:00
mflodman@webrtc.org
5cc4dc9e0c
Remove warnings in VideoEngine, capture module and render module.
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BUG=164, 176, 180
Review URL: http://webrtc-codereview.appspot.com/303004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1145 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-09 10:12:57 +00:00
henrikg@webrtc.org
af225d6bf6
The change http://webrtc-codereview.appspot.com/299001 (commit 1062) does not do what it intends (exclude codecs from Chromium build). This is a fix for that. webrtc.gyp is not pulled in Chromium, hence it has no effect putting a define there. Moving it to src/build/common.gypi.
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Review URL: http://webrtc-codereview.appspot.com/315002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1143 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-09 09:58:39 +00:00
mikhal@webrtc.org
2ab104e6be
Switching WebRtc to LibYuv.
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General Notes:
1. In general, API structure was not modified and is based on VPLIB.
2. Modification to API: Return values are based on libyuv, i.e. 0 if ok, a negative value in case of an error (instead of length).
3. All scaling (inteprolation) is now done via the scale interface. Crop/Pad is not being used.
4. VPLIB was completely removed. All tests are now part of the libyuv unit test (significantly more comprehensive and based on gtest).
5. JPEG is yet to be implemented in LibYuv and therefore existing implementation remains.
Review URL: http://webrtc-codereview.appspot.com/258001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1140 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-09 02:46:22 +00:00
mallinath@webrtc.org
7766e2a82d
- This issue raised by the chromium team when clang compiler is used. This was not an error as in this case we were accessing IPV6 address with IPV4 struct which is defined as 14 bytes in the header file, but we had the runtime check to determine the address space.
...
Now the solution is to use IPV6 structures instead of IPV4 when address space is determined.
I haven't put the new solution behind AF_INET6 flag, as i don't think it's necessary.
Review URL: http://webrtc-codereview.appspot.com/291014
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1138 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-08 21:37:19 +00:00
andrew@webrtc.org
b0be7aa7ae
Remove deprecated OS X Core Audio APIs.
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We no longer support the 10.4 SDK, so we can remove the weak-leaking
feature and exclusively use the added-in-10.5 APIs.
BUG=issue143
TEST=voe_auto_test
Review URL: http://webrtc-codereview.appspot.com/322001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1136 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-08 20:15:36 +00:00
marpan@webrtc.org
63b50f60d6
test_fec: Fix to valgrind warnings.
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Review URL: http://webrtc-codereview.appspot.com/304002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1135 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-08 19:05:39 +00:00
mikhal@webrtc.org
f5ee1dc3e6
video_coding: Adding temporal layer info support to receive side
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Review URL: http://webrtc-codereview.appspot.com/303005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1134 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-08 19:04:47 +00:00
henrik.lundin@webrtc.org
d03718d1e4
Use ResourcePath in NetEQ unittest
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Review URL: http://webrtc-codereview.appspot.com/320001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1130 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-08 11:42:43 +00:00
kjellander@webrtc.org
7de6e10410
Fixing compilation error on Linux 64-bit
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Problem was introduced in http://webrtc-codereview.appspot.com/311001/ because I had projects generated with Valgrind configuration, which is more forgiving about these implicit conversions.
BUG=
TEST=Compiling in Debug+Release on Linux, Mac and Windows.
Review URL: http://webrtc-codereview.appspot.com/318002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1127 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-08 08:39:13 +00:00
kjellander@webrtc.org
5b97b1216f
Splitted FileHandler into FrameReader and FrameWriter classes and moved them to testsupport in test.gyp.
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Fixed unit tests so they don't use ASSERT_DEATH since that doesn't work with Valgrind.
Fixed all Valgrind warnings except the one caused by CriticalSectionWrapper in system_wrappers.
Reworked all includes and GYP include paths to use full directory paths.
Removed util.h for logging, since it rendered warnings in Valgrind because of gflags. Replaced it with a verbose flag and a new function in video_quality_measurement.cc
BUG=
TEST=Passed test_support_unittests and video_codecs_test_framework_unittests on Linux, Mac and Windows.
Review URL: http://webrtc-codereview.appspot.com/311001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1126 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-08 07:42:18 +00:00
henrike@webrtc.org
441b3fe2a1
Made some global statics have function scope so that the global static count is 0 for the rtp_rtcp module.
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Review URL: http://webrtc-codereview.appspot.com/316001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1125 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-08 02:03:49 +00:00
stefan@webrtc.org
cc7b649474
Add trace for the situation when the min bitrate > available bandwidth.
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BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/312001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1123 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-07 13:22:06 +00:00
henrik.lundin@webrtc.org
598ad06432
Fixing compiler warning in NetEQ
...
With some compiler settings, a warning was issued for NetEQ,
saying that pw16_randVec was accessed out of bounds.
This did never happen in practice, but this change makes the
compiler understand this.
Review URL: http://webrtc-codereview.appspot.com/309001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1121 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-07 11:52:09 +00:00
stefan@webrtc.org
b3bd1cd5f1
Fixes Valgrind warnings in the default VCM tests.
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BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/299010
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1120 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-07 11:48:09 +00:00
henrik.lundin@webrtc.org
bf86c33b0e
Removing OutputDebugString from rtp_rtcp module
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This is in response to WebRTC issue 167.
BUG=http://code.google.com/p/webrtc/issues/detail?id=167
Review URL: http://webrtc-codereview.appspot.com/301013
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1119 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-07 10:44:05 +00:00
henrik.lundin@webrtc.org
44ef3774ce
Fixing a compiler error in NetEQ
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This error would only arise when compiling without support for
DTMF (which is not the default config).
Review URL: http://webrtc-codereview.appspot.com/310001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1118 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-07 10:43:25 +00:00
phoglund@webrtc.org
5b343aedcc
Added missing .h files to .gypi files so they will show up in xcode / vc projects.
...
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/304008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1117 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-07 10:34:35 +00:00
stefan@webrtc.org
58927e8d8f
Disable deblocking temporarily due to Valgrind warnings.
...
Also corrects the copying of the decoded image data for frames
with odd width or height.
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/307002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1116 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-07 08:13:31 +00:00
marpan@webrtc.org
1d34212a45
FEC: Update to packets masks (FEC generator matrix) in fec_private_tables.h
...
A set of the packet masks (up 10x10 size) are modified for the following reasons:
1) have more even column and row degree (number of 1 bits), when possible.
2) if cases where the column degree cannot be constant across source packets, placed the extra 1 bit in the first packet column (so little more protection on 1st partition), as opposed to having some ~middle source packet have the extra bit.
3) in some cases, made the mask a little more sparse/reduced the overlap.
Overall the average recovery is a little better with these masks.
Mask sizes above 10 will be updated in future changelist.
Review URL: http://webrtc-codereview.appspot.com/305001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1113 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-06 18:50:11 +00:00
kma@webrtc.org
4a8b1eaf6e
In NS, replaced a divide calculatoin by shifting, and thus saved the MIPS by 5%(ARMv7) and 10%(ARMv7-Neon). Bit is not exact with the original. Quality is similar.
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Review URL: http://webrtc-codereview.appspot.com/298004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1112 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-06 18:04:48 +00:00
henrik.lundin@webrtc.org
b6e58eb5a1
Fix formatting of rtp_format_vp8*
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Sorting out all lint issues and fixing indentation.
Review URL: http://webrtc-codereview.appspot.com/301011
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1111 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-06 15:56:18 +00:00
stefan@webrtc.org
c7e2bffb66
Fix header/lib mismatch caused by a constant not defined for header file.
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BUG=http://code.google.com/p/webrtc/issues/detail?id=170
TEST=
Review URL: http://webrtc-codereview.appspot.com/300008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1110 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-06 13:44:25 +00:00
xians@webrtc.org
eff3c8905f
this patch fixes the valgrind warnings in the adm api test for pulseaudio in linux.
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Review URL: http://webrtc-codereview.appspot.com/301012
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1108 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-06 10:02:56 +00:00
mikhal@webrtc.org
a5e980a906
Updating jitter buffer test following latest changes.
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Review URL: http://webrtc-codereview.appspot.com/294002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1106 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-05 18:27:31 +00:00
perkj@webrtc.org
ec7759a8c4
Fix broken vie_capture_module_test on mac.
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BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/303006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1101 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-05 12:17:10 +00:00
perkj@webrtc.org
8627adc158
Refactored Video capture Unit test to use gtest.
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Fix Valgrind warnings on Linux.
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/296009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1100 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-05 09:58:55 +00:00
stefan@webrtc.org
0ae71b9ccb
Disable temporal layers when building with Chromium.
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BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/301010
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1099 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-05 08:42:52 +00:00
andrew@webrtc.org
c9cc3750cf
Add missing system_wrappers dependency.
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TBR=kma@webrtc.org
Review URL: http://webrtc-codereview.appspot.com/301009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1097 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-03 20:51:20 +00:00
kma@webrtc.org
b59c031660
For Android ARMv7 platforms, added a feature of dynamically detecting the existence of Neon,
...
and when it's present, switch to some functions optimized for Neon at run time.
Review URL: http://webrtc-codereview.appspot.com/268002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1096 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-03 18:34:50 +00:00
andrew@webrtc.org
ae7017d588
Fix missing dependency in audioproc.
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TBR=bjornv@webrtc.org
Review URL: http://webrtc-codereview.appspot.com/300006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1095 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-03 01:43:29 +00:00
andrew@webrtc.org
7bf2646e4d
Make protobuf use optional.
...
- By default, disable the AudioProcessing protobuf usage in the Chromium
build. The standalone build is unaffected.
- Add a test for the AudioProcessing debug dumps.
TEST=audioproc_unittest
Review URL: http://webrtc-codereview.appspot.com/303003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1094 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-03 00:03:31 +00:00
perkj@webrtc.org
6b1bfd6c5e
Changed webrtc::ACMCodecDB::neteq_decoders_ to a const array.
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BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/304003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1092 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-02 12:48:19 +00:00
pwestin@webrtc.org
db221d2b81
Fixes to temporal layers, Henrika please review src/common_types.h
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Review URL: http://webrtc-codereview.appspot.com/286001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1091 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-02 11:31:08 +00:00
henrik.lundin@webrtc.org
e26aad4a9e
Disable NetEQ unittest for Windows
...
Disable NetEqDecodingTest::TestNetworkStatistics for Windows.
It was never tested for Windows. Something is causing it to
fail, probably need different set of test vectors.
Review URL: http://webrtc-codereview.appspot.com/302003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1089 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-02 10:27:14 +00:00
stefan@webrtc.org
9cb2b56b65
Corrected a fread verification.
...
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/301006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1088 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-02 10:22:29 +00:00
perkj@webrtc.org
38ca4f2953
Fix code review comments.
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@1086 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-02 09:34:10 +00:00
perkj@webrtc.org
d3eac4158c
Fixed webrtc::perm variable.
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@1085 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-02 09:34:01 +00:00
perkj@webrtc.org
1b72fcd27b
Fix symbol RTPFILE_VERSION.
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@1084 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-02 09:33:51 +00:00
stefan@webrtc.org
772d70bcd2
Fix release build error.
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BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/304005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1083 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-02 09:18:49 +00:00
stefan@webrtc.org
a4a88f90c4
Implemented NACK based reference picture selection.
...
This CL implements NACK based reference picture selection for VP8. A separate
class is used for keeping track of the references and managing the VP8 encode
flags. Appropriate tests have also been added.
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/284002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1082 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-02 08:34:05 +00:00
henrike@webrtc.org
4b00560a6e
Fixes build error in rtp_rtc module introduced in r1076.
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Review URL: http://webrtc-codereview.appspot.com/301005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1081 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-02 00:32:24 +00:00
punyabrata@webrtc.org
c1ed87602a
Adding some error handling functionality in the windows audio core implementation to
...
stop rendering automatically and throw a playout-error callback when RequestPlayoutData
fails
Review URL: http://webrtc-codereview.appspot.com/300003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1080 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-01 17:55:35 +00:00
kjellander@webrtc.org
5f4f69ac57
Removing sleeps from vp8_test.
...
These sleeps were remains from earlier tests that required them to work with some codecs. Removing these sleep calls cut the execution time from 90s to 30s on my machine.
Review URL: http://webrtc-codereview.appspot.com/304004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1077 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-01 15:50:04 +00:00
pwestin@webrtc.org
0644b1dc35
Introduce a mockable RtpRtcpClock interface replacing ModuleRTPUtility time functions
...
A new RtpRtcpClock interface has been added to rtp_rtcp_defines.h
and provides time facilities used by an RTP/RTCP module. Also,
NTP constants have been made public in the
webrtc::ModuleRTPUtility namespace to make implementation of
external clocks easier.
An overloaded version of CreateRtpRtcp() accepts a clock argument. By
default, if no clock is provided, the module uses the system clock
(old ModuleRTPUtility implementation).
Throughout the RTP/RTCP module code, calls to TickTime and
ModuleRTPUtility time functions have been replaced with calls to time
methods on a clock object.
The following classes take a clock object in their constructor and
hold a _clock field (either directly, or inherited from a parent):
Bitrate
ModuleRtpRtcpImpl
RTCPReceiver
RTCPSender
RTPReceiver
RTPSender
RTPSenderAudio
RTPSenderVideo
Methods from other classes that do not derive any of those and
require a time take an additional nowMS parameter, that should be
the result of calling GetTimeInMS() on a clock object.
Review URL: http://webrtc-codereview.appspot.com/268017
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1076 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-01 15:42:31 +00:00
bjornv@webrtc.org
132feb1270
Made tables static.
...
In this CL global tables have been moved to where they are actually used. If for some reason they need to be available in a larger scope we can add them again at that point.
Review URL: http://webrtc-codereview.appspot.com/303002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1075 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-01 15:40:50 +00:00
kjellander@webrtc.org
4c4b7f500f
Converting vp8_test to use fileutils and gtest
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Review URL: http://webrtc-codereview.appspot.com/289012
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1074 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-01 15:24:36 +00:00
tina.legrand@webrtc.org
f64162c335
Adding const to a number of constant tables. Setting some tables to static.
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Patch set 2: Renaming static const tables. They no longer need the prefix WebRtc_Isac...
Review URL: http://webrtc-codereview.appspot.com/301001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1073 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-01 13:01:39 +00:00
zakkhoyt@webrtc.org
a7e70b43e2
When entering fullscreen mode, the CocoaRenderView is attached as a subview to a new full screen window.
...
When the class is torn down, the view was not being attached back to it's original NSView. I added a
new class variable to remember the original superview and then reattach it at the appropriate time.
Review URL: http://webrtc-codereview.appspot.com/290009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1070 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-30 22:30:50 +00:00
mikhal@webrtc.org
b9db43e1b6
video_coding/jitter buffer: Reduce delay on a complete frame: No need for the next frame when current frame is already complete.
...
Review URL: http://webrtc-codereview.appspot.com/289007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1069 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-30 18:38:01 +00:00
andrew@webrtc.org
587c844741
Query the capture volume immediately on Win Core.
...
This allows the capture volume to be queried immediately at capture
startup, rather than waiting the usual one second interval.
Review URL: http://webrtc-codereview.appspot.com/297003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1064 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-30 17:43:05 +00:00
henrik.lundin@webrtc.org
524eb48081
Removing deprecated NetEQ APIs
...
Removing WebRtcNetEQ_GetPreferredBufferSize and
WebRtcNetEQ_GetCurrentDelay and all dependent APIs.
Review URL: http://webrtc-codereview.appspot.com/289006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1063 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-30 16:21:22 +00:00
xians@webrtc.org
0dffc6449a
To be able to get webrtc into chrome, we need to reduce the size of the binary and the usage of memory.
...
This patch disbale some codecs which are not considered necessary.
Review URL: http://webrtc-codereview.appspot.com/299001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1062 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-30 15:35:44 +00:00
stefan@webrtc.org
0c2adf0b75
Fix bug introduced when enabling VP8 frame dropping.
...
Also fixes two unit test mismatches.
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/299002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1061 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-30 14:41:58 +00:00
stefan@webrtc.org
ac2c677bf6
Make all video_coding tests use the resources and output directories.
...
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/298001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1060 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-30 14:23:39 +00:00
andrew@webrtc.org
268257475b
Fix one more Objective-C clang error.
...
(Analogous to r1056).
BUG=issue78
Review URL: http://webrtc-codereview.appspot.com/297004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1058 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-30 00:54:04 +00:00
punyabrata@webrtc.org
c9801465b6
Adding a check to ensure that the memcpy does not exceed bounds of the arrays.
...
Review URL: http://webrtc-codereview.appspot.com/290007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1055 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-29 18:49:54 +00:00
andrew@webrtc.org
1e91693fe2
Move stream_delay check to ProcessStream().
...
- was_stream_delay_set_ was being incorrectly reset in
AnalyzeReverseStream().
- Added tests to catch this case.
BUG=
TEST=audioproc_unittest
Review URL: http://webrtc-codereview.appspot.com/291011
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1054 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-29 18:28:57 +00:00
henrik.lundin@webrtc.org
fc9b903fbe
Enable NetEQ statistics unit testing
...
Adding test target NetEqDecodingTest::TestNetworkStatistics.
Update neteq_unittest to get files from resources folder.
Update DEPS file to get resources revision 2.
Review URL: http://webrtc-codereview.appspot.com/291013
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1050 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-29 15:38:27 +00:00
henrik.lundin@webrtc.org
2d8125dd1a
Testing NetEQ network statistics
...
Implementing helper function for new unit test
NetEqDecodingTest::TestNetworkStatistics. The test itself
remains to be defined. (Will be added in a coming CL.)
This change required some refactoring of the test code
to avoid excessive code duplication.
Review URL: http://webrtc-codereview.appspot.com/295009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1049 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-29 14:30:28 +00:00
stefan@webrtc.org
932ab18d32
Default to always NACKing residual losses when having both FEC and NACK.
...
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/296002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1047 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-29 11:33:31 +00:00
bjornv@webrtc.org
4b80eb4fcd
Name change resampler.c/h to aec_resampler.c/h.
...
To avoid possible conflict with resampler in common_audio.
Review URL: http://webrtc-codereview.appspot.com/296006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1046 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-29 08:44:01 +00:00
marpan@webrtc.org
9d8bec6f76
FEC: Fix to valgrind warning.
...
Review URL: http://webrtc-codereview.appspot.com/292009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1042 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-28 22:10:05 +00:00
andrew@webrtc.org
400ad6928e
Fix compile warning in NS.
...
BUG=issue151
TEST=audioproc_unittest
Review URL: http://webrtc-codereview.appspot.com/290005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1041 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-28 21:33:42 +00:00
marpan@webrtc.org
d1b7932adf
VP8: Setting non-zero (conservative) threshold for frame dropper.
...
Review URL: http://webrtc-codereview.appspot.com/291001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1040 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-28 19:20:31 +00:00
andrew@webrtc.org
1e39bc80dc
Handle debug files from multiple AEC instances.
...
Review URL: http://webrtc-codereview.appspot.com/295006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1036 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-27 23:46:23 +00:00
andrew@webrtc.org
a919d3a643
Don't return a zero delay with insufficient data.
...
For estimating a delay over a long segment (e.g. a file) this can
throw off the estimate. Better not to add values to the AEC's histogram
until they're reliable.
BUG=
TEST=audiproc, audioproc_unittest
Review URL: http://webrtc-codereview.appspot.com/292004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1035 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-27 23:40:58 +00:00
stefan@webrtc.org
94a8c03141
Slightly increased bandwidth adaptation at both receive- and send-side.
...
The send-side increase factor is increased to better follow the pace
of the receive-side estimate, while the receive-side factor is
increased to speed up adaptation.
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/297002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1030 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-25 14:09:37 +00:00
xians@webrtc.org
8738d277a1
Valgrind detects that there are racing conditions in RTPReceiver::PacketTimeout and RTPSender
...
This CL fixes two of them.
Review URL: http://webrtc-codereview.appspot.com/295005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1029 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-25 13:43:53 +00:00
henrik.lundin@webrtc.org
0fcc2eb368
Cleaning up neteq_unittest
...
- Conforming to testing standards.
- Fixing a way of generating new reference output files.
- ifdef the test to run only on linux 64-bit
- Renaming unittest source file.
- Renaming test vectors
Review URL: http://webrtc-codereview.appspot.com/296007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1028 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-25 13:43:42 +00:00
henrik.lundin@webrtc.org
789da89d37
Fix a valgrind warning in NetEQ
...
The special cases for packet sizes <= 10 ms (one case for each
sample rate) resulted in reading outside of the pw16_decoded
vector. This is now fixed by making sure that WebRtcSpl_DownsampleFast
gets correct input and output vector lengths.
Review URL: http://webrtc-codereview.appspot.com/295008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1027 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-25 12:35:31 +00:00
stefan@webrtc.org
0ee8ba1929
Remove WebRTC dependency on libvpx_lib and libvpx_include.
...
Removes dependencies on libvpx_lib and libvpx_include targets when
building with Chromium.
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/293004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1026 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-25 12:12:43 +00:00
henrik.lundin@webrtc.org
859626570a
VP8 RTP work
...
Fixing the plumbing to get the KEYIDX between VP8 wrapper and
rtp_rtcp module. Also fixing a missing pipe for temporalIdx
Review URL: http://webrtc-codereview.appspot.com/295004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1024 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-25 10:17:00 +00:00
braveyao@webrtc.org
0a18522e1b
Add support to 96kHz sampling rate to Windows CoreAudio interface.
...
Review URL: http://webrtc-codereview.appspot.com/295003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1018 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-25 02:45:39 +00:00
mflodman@webrtc.org
26b9777e62
Only trigger one call to OnNetworkChanged for each incoming RTCP packet.
...
Review URL: http://webrtc-codereview.appspot.com/289004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1016 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-24 15:22:33 +00:00
henrik.lundin@webrtc.org
9af365d3c5
Fixing VP8 RTP parser bug
...
Missing one initialization of new struct variable hasKeyIdx.
TBR=stefan@webrtc.org
Review URL: http://webrtc-codereview.appspot.com/296004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1014 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-24 13:28:29 +00:00
henrik.lundin@webrtc.org
6f2c0168f0
Updating to VP8 RTP spec rev -02
...
Updating the VP8 packetizer class (RtpFormatVp8) and VP8 parser
(in class RTPPayloadParser) to follow the -02 revision of the spec.
See http://tools.ietf.org/html/draft-ietf-payload-vp8-02 .
Updating the unit tests, too. Finally, updating the tests to
follow the recommendations from the test team; specifically
including the test code in the webrtc namespace, and omitting
the main function at the end of each test file.
Review URL: http://webrtc-codereview.appspot.com/296003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1013 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-24 12:52:40 +00:00
kjellander@webrtc.org
d492f72e43
Added empty unit tests to get code coverage measured.
...
In order to get code coverage recorded, there must be an executing test that is linked to the code to measure.
These projects are currently not showing up in the code coverage.
Review URL: http://webrtc-codereview.appspot.com/293002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1010 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-24 07:20:00 +00:00
andrew@webrtc.org
ba028a31c9
Fix sample rate printout in process_test.
...
TBR=bjornv
Review URL: http://webrtc-codereview.appspot.com/292005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1008 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-23 20:37:12 +00:00
henrik.lundin@webrtc.org
4257790d2d
NetEQ-related bug in ACM
...
Fixing a bug when creating new NetEQ slave instances in ACM.
The old code called WebRtcNetEQ_GetCurrentDelay() for the
master instance to get a delay value for WebRtcNetEQ_SetExtraDelay().
This is wrong, since WebRtcNetEQ_GetCurrentDelay() reports on the
current total buffer length, while WebRtcNetEQ_SetExtraDelay() is
the extra delay that is desired to in order to sync with video.
The fix includes keeping the extra delay value in a member variable
in the ACMNetEQ class.
Review URL: http://webrtc-codereview.appspot.com/295001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1001 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-23 13:04:05 +00:00
kjellander@webrtc.org
543c3eaa46
Fixing Release compilation errors
...
Review URL: http://webrtc-codereview.appspot.com/267026
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1000 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-23 12:20:35 +00:00
henrik.lundin@webrtc.org
89ab652250
Cleaning up NetEQ statistics
...
Removed struct MCUStats_t and all references to it.
Removed totalDiscardedPackets and totalFlushedPackets
from the PacketBuf_t struct.
Review URL: http://webrtc-codereview.appspot.com/293001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@999 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-23 11:06:05 +00:00
henrik.lundin@webrtc.org
df10de4b27
Removing statistics API from NetEQ
...
Removing WebRtcNetEQ_GetJitterStatistics(),
WebRtcNetEQ_ResetJitterStatistics(), and the associated
struct WebRtcNetEQ_JitterStatistics. The change ripples
through all the way to the VoiceEngine API.
Review URL: http://webrtc-codereview.appspot.com/285002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@998 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-23 09:36:23 +00:00
braveyao@webrtc.org
7d3e9498bc
This CL is to support certain audio devices which don't offer volume control. Try to be more compatible to those rare cases.
...
Review URL: http://webrtc-codereview.appspot.com/276011
git-svn-id: http://webrtc.googlecode.com/svn/trunk@997 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-23 03:35:42 +00:00
mikhal@webrtc.org
2b838b4121
video_coding: updating the session info unit test following recent changes
...
Review URL: http://webrtc-codereview.appspot.com/290002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@996 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-23 00:20:19 +00:00
mikhal@webrtc.org
425b377973
video_coding: Updating internal_defines to resolve latest build error. Refers to JB flush update.
...
Review URL: http://webrtc-codereview.appspot.com/289001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@995 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-22 23:41:29 +00:00
mikhal@webrtc.org
f13388f134
video_coding: Requesting a key frame after a JB flush
...
Review URL: http://webrtc-codereview.appspot.com/280006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@994 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-22 22:57:51 +00:00
mikhal@webrtc.org
6b9a7f8704
video_coding: Allowing for a decodable state independent of selective nacking
...
Review URL: http://webrtc-codereview.appspot.com/263001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@993 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-22 22:48:20 +00:00
andrew@webrtc.org
828af1b4b9
Add lookahead to the delay estimator.
...
TEST=audioproc_unittest
Review URL: http://webrtc-codereview.appspot.com/279014
git-svn-id: http://webrtc.googlecode.com/svn/trunk@992 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-22 22:40:27 +00:00
andrew@webrtc.org
5a529395aa
Make DMO init safe when not supported.
...
BUG=issue133
TEST=voe_auto_test
Review URL: http://webrtc-codereview.appspot.com/284001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@990 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-22 18:04:26 +00:00
andrew@webrtc.org
8594f7688b
Add a gyp variable for AEC debug dumps.
...
TEST=process_test.cc
Review URL: http://webrtc-codereview.appspot.com/276012
git-svn-id: http://webrtc.googlecode.com/svn/trunk@987 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-22 00:51:41 +00:00
kma@webrtc.org
a249f35203
Correct several makefile errors for Android build.
...
Review URL: http://webrtc-codereview.appspot.com/267024
git-svn-id: http://webrtc.googlecode.com/svn/trunk@986 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-21 22:16:10 +00:00
kjellander@webrtc.org
274c2efbc1
Adding empty test method required to get code coverage
...
Review URL: http://webrtc-codereview.appspot.com/279008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@983 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-21 09:36:28 +00:00
marpan@webrtc.org
3caa327af0
VP8 wrapper: Turn on some mild amount of deblocking in post-processing.
...
Review URL: http://webrtc-codereview.appspot.com/268015
git-svn-id: http://webrtc.googlecode.com/svn/trunk@982 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-19 01:08:09 +00:00
kma@webrtc.org
ced118636d
Changed keyword __restrict__ to __restrict.
...
Review URL: http://webrtc-codereview.appspot.com/279011
git-svn-id: http://webrtc.googlecode.com/svn/trunk@978 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-18 17:51:19 +00:00
kjellander@webrtc.org
543611a77a
Reverting r972 due to compilation error on Windows Release build.
...
TBR=kma
Review URL: http://webrtc-codereview.appspot.com/282003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@976 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-18 13:25:13 +00:00
bjornv@webrtc.org
2f047ccede
Removed unnecessary variable to avoid compiler error on Win.
...
Review URL: http://webrtc-codereview.appspot.com/267021
git-svn-id: http://webrtc.googlecode.com/svn/trunk@975 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-18 12:03:25 +00:00
henrik.lundin@webrtc.org
ba74924043
Remove use of exceptions in NetEQ test code
...
Replaced the exceptions thrown when codec instance creation failed
with simple exit(EXIT_FAILURE). There is no point in continuing
if creating the codec fails.
Review URL: http://webrtc-codereview.appspot.com/282002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@974 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-18 09:55:01 +00:00
bjornv@webrtc.org
6a9835d59c
Delay estimator structural changes.
...
Improved the way we handle different data types (float vs fixed) and reduced the complexity by nearly 50%.
We now have a generic struct for both float and fixed delay estimators and a core struct for the binary spectrum based delay estimator. All wrapper codes (for both fixed and float) are gathered in delay_estimator_wrappers.*.
Moved out the far end history buffer to AEC(M).
Added a union to handle difference types when create.
Review URL: http://webrtc-codereview.appspot.com/277004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@973 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-18 08:30:34 +00:00
kma@webrtc.org
fa9b016fb5
Optimized WebRtcIsacfix_AutocorrFix() function for iSAC fix.
...
(1) For generic platforms, code was changed to remove the shifting within loops.
Basically, it's just change a loop from
for() {
sum += (a*b) >> scale;
}
to:
for() {
sum += (a*b);
}
sum >> scale;
Type int64_t is used for sum to make sure no information is not lost.
Performance is about the same as before the change. Bits are not exact,
although in theory the change should have preserved more information. The purpose
of this change is to make the generic code and ARM code bit exact, simpify the code,
while keep the speech quality at least not lower. (Some speech tests might be good.)
(2) For ARM platform, used assembly to optimize the performance. iSAC runs faster
with this change. (Reduced run time of an offline file test from 10.16ms to 8.81ms)
Review URL: http://webrtc-codereview.appspot.com/267014
git-svn-id: http://webrtc.googlecode.com/svn/trunk@972 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-18 02:50:55 +00:00
braveyao@webrtc.org
f556b9d1f4
This modification is supposed to fix the webrtc issue 144/145. With this fix, people could set/get mic volume before StartSend().
...
Review URL: http://webrtc-codereview.appspot.com/277007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@971 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-18 02:17:28 +00:00
kjellander@webrtc.org
cd7b57ef9e
Fixing release compilation error
...
Review URL: http://webrtc-codereview.appspot.com/279007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@968 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-17 14:26:21 +00:00
kjellander@webrtc.org
3f1cb8e546
Restructuring and adding dummy unit test target.
...
Empty test added to get code coverage recorded.
Review URL: http://webrtc-codereview.appspot.com/269018
git-svn-id: http://webrtc.googlecode.com/svn/trunk@967 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-17 13:56:54 +00:00
kjellander@webrtc.org
cc2ecb3c2e
Restructuring and adding dummy unit test target.
...
Empty test added to get code coverage recorded.
Review URL: http://webrtc-codereview.appspot.com/267019
git-svn-id: http://webrtc.googlecode.com/svn/trunk@966 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-17 13:48:36 +00:00
kjellander@webrtc.org
b72268e147
Restructuring and adding dummy unit test target.
...
Empty test added to get code coverage recorded.
Review URL: http://webrtc-codereview.appspot.com/280004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@965 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-17 13:39:15 +00:00
kjellander@webrtc.org
64a897a772
Restructuring and adding dummy unit test target.
...
Empty test added to get code coverage recorded.
Review URL: http://webrtc-codereview.appspot.com/282001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@964 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-17 13:33:11 +00:00
kjellander@webrtc.org
c05b56a38b
Fixing compilation error
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Review URL: http://webrtc-codereview.appspot.com/276010
git-svn-id: http://webrtc.googlecode.com/svn/trunk@961 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-17 08:59:48 +00:00
kjellander@webrtc.org
0403ef419f
Restructuring and adding unit test targets on project level instead of in common_audio.
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Review URL: http://webrtc-codereview.appspot.com/280001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@959 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-17 08:35:47 +00:00
phoglund@webrtc.org
337dc68992
Included modules in webrtc.gyp and fixed build errors.
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Removed TODO from webrtc.gyp since it is done.
Tabs -> spaces.
Tabs -> spaces.
Tabs -> spaces.
Fixed compilation on Windows.
Added missing file.
Merge branch 'master' into fix_mac_modules
Fixed compilation errors for the modules.gyp on Mac. This included some pretty large refactorings.
Please enter the commit message for your changes. Lines starting
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/269005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@957 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-16 15:36:44 +00:00
stefan@webrtc.org
fcf33eb7e0
Limit number of send-side BWE increases to one per second.
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Also report 0 losses if not enough expected packets since
previous receiver report.
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/270009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@954 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-16 07:58:31 +00:00
punyabrata@webrtc.org
81d4499dee
Microphone volume on Mac not being printed properly due
...
to a mismatch in variable type. Additionally, now printing
a volume that will range from 0 - 255
Review URL: http://webrtc-codereview.appspot.com/267016
git-svn-id: http://webrtc.googlecode.com/svn/trunk@951 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-16 02:06:49 +00:00
andrew@webrtc.org
755b04a06e
Add RMS computation for the RTP level indicator.
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- Compute RMS over a packet's worth of audio to be sent in Channel, rather than the captured audio in TransmitMixer.
- We now use the entire packet rather than the last 10 ms frame.
- Restore functionality to LevelEstimator.
- Fix a bug in the splitting filter.
- Fix a number of bugs in process_test related to a poorly named
AudioFrame member.
- Update the unittest protobuf and float reference output.
- Add audioproc unittests.
- Reenable voe_extended_tests, and add a real function test.
- Use correct minimum level of 127.
TEST=audioproc_unittest, audioproc, voe_extended_test, voe_auto_test
Review URL: http://webrtc-codereview.appspot.com/279003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@950 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-15 16:57:56 +00:00
andrew@webrtc.org
6a85b17a0a
Potential fix for crash after Mac sleep.
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When a Mac goes to sleep, the OS pauses the IO threads. If a
subsequent StopSend/Playout happens, we time out waiting for the IO
threads, but didn't ensure they were shut down.
BUG=
TEST=voe_cmd_test, voe_auto_test
Review URL: http://webrtc-codereview.appspot.com/269013
git-svn-id: http://webrtc.googlecode.com/svn/trunk@949 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-15 16:23:41 +00:00
kjellander@webrtc.org
85596d5bf4
Setting completeFrame to true for all created encoded images.
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Review URL: http://webrtc-codereview.appspot.com/276008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@948 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-15 13:45:25 +00:00
henrik.lundin@webrtc.org
bc91d5af86
NetEQ tests
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Adding capability to parse RED payloads to the RTPanalyze tool.
Also adding a method to scramble an RTP payload (currently not
used).
Review URL: http://webrtc-codereview.appspot.com/276006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@945 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-15 10:16:01 +00:00
mflodman@webrtc.org
a02ef1ace2
Fix broken tree.
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Review URL: http://webrtc-codereview.appspot.com/267015
git-svn-id: http://webrtc.googlecode.com/svn/trunk@943 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-15 07:50:50 +00:00
mflodman@webrtc.org
1f69c03739
Added size sanity check for copying app specific RTCP data.
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Similar check as done in RTCPUtility::RTCPParserV2::ParseAPPItem.
Review URL: http://webrtc-codereview.appspot.com/277002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@942 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-15 06:12:39 +00:00
henrik.lundin@webrtc.org
33df5335bf
Change luminance of all pixels by a specified value.
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Modeled on color_enhancement.
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/269004
Patch from SriRam <tvnsriram@google.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@941 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-14 15:30:26 +00:00
andrew@webrtc.org
0db7dc6e18
Add file-playing channels to voe_cmd_test.
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Fix file reading and writing.
TEST=voe_cmd_test
Review URL: http://webrtc-codereview.appspot.com/279001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@938 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-13 01:34:05 +00:00
andrew@webrtc.org
cd8243807e
Unpack the full set of audioproc data.
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Review URL: http://webrtc-codereview.appspot.com/276004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@937 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-11 19:13:36 +00:00
kma@webrtc.org
d71d480487
Fixed a build error of audio conference mixer in Android.
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Review URL: http://webrtc-codereview.appspot.com/267009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@936 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-11 17:14:23 +00:00
mflodman@webrtc.org
fd3a0efd15
RTP bw estimate fix.
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Review URL: http://webrtc-codereview.appspot.com/279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@932 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-11 10:55:26 +00:00
phoglund@webrtc.org
1144ba2268
Base and codec tests now run verify output and render to file instead of to screen.
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Rewrote the codec test to render to file and do video comparisons.
Refactored the coded tests somewhat. I still need to figure out how to do comparison in the automated case.
Added video analysis to the test. This will make sure that the system output roughly the right thing.
Moved the video metrics library into the test_support library. Made the metrics library available in the automated tests.
Made sure no one passes in too large YUV videos into the autotest.
The standard test's output now gets captured for both the left and right windows.
Wrote a rendering device which just writes the raw frames to file, for analysis. Updated the base standard test to dump its left window output to file. We don't do anything with it yet though.
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/249001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@931 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-11 09:01:03 +00:00
kma@webrtc.org
13318ef422
(1) Corrected the makefile for testing iLBC in Android, and changed the location of the test makefile to make it consistent with audio_processing.
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(2) Added a makefile for testing fiexed point iSAC in Android.
Review URL: http://webrtc-codereview.appspot.com/266005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@927 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-10 18:00:22 +00:00
mflodman@webrtc.org
7a4eb2837a
Calculate the available bandwidth before sending a TMMBR
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Also changed the way TMMBR was processed since it did not match the new bandwidth estimator.
Review URL: http://webrtc-codereview.appspot.com/270003
Patch from pwestin1 <pwestin@webrtc.org>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@925 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-10 12:54:46 +00:00
mflodman@webrtc.org
637a59e68e
jitter buffer update: waiting for key frame when Nack is enabled and continuity cannot be determined.
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Review URL: http://webrtc-codereview.appspot.com/266010
Patch from mikhals <mikhal@webrtc.org>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@924 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-10 12:50:48 +00:00
tina.legrand@webrtc.org
855a77c972
Audio Coding Module: Fixing a bug that prevented the encoder from being re-initialized when changing codec from mono to stereo.
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Solving issue 130 reported by Niklas.
Reviewer: Turaj
Review URL: http://webrtc-codereview.appspot.com/268007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@921 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-10 08:17:08 +00:00
andrew@webrtc.org
c4f129f97c
Improve the mixing saturation protection scheme.
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A single participant is not processed at all. With multiple
participants, we divide-by-2 as before when mixing. Afterwards,
the mixed signal is limited by the AGC to -7 dBFS and then doubled to
restore the original level.
This preserves the level while guaranteeing good saturation protection.
Add a test to voe_auto_test. Hijack and improve the existing mixing test
for this.
TEST=voe_auto_test, voe_cmd_test
Review URL: http://webrtc-codereview.appspot.com/241013
git-svn-id: http://webrtc.googlecode.com/svn/trunk@920 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-10 03:41:22 +00:00
andrew@webrtc.org
4b13fc9c09
Add delay modification to process_test.
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Review URL: http://webrtc-codereview.appspot.com/266007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@916 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-09 19:27:11 +00:00
henrike@webrtc.org
2f32b5c8a7
Fixes an issue where file playing could happen at a lower sampling frequency than the file.
...
Details:
The mixer looks at all the participants desired frequency and concludes the highest desired mixing frequency. This is the frequency that the mixer will mix at. Participants that are always mixed are in a separate list and the function concluding the highest desired mixing frequency did not look at that list and therefore always conclude that the lowest mixing frequency is sufficient.
Review URL: http://webrtc-codereview.appspot.com/277003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@915 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-09 19:02:17 +00:00
mikhal@webrtc.org
eb4ef17bbd
Removing vplib include and VideoInterpolator when not needed
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Review URL: http://webrtc-codereview.appspot.com/268004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@914 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-09 18:11:02 +00:00
kjellander@webrtc.org
488ed92c3b
Removing exceptions since not used
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Review URL: http://webrtc-codereview.appspot.com/267003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@912 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-09 16:12:40 +00:00
kjellander@webrtc.org
c3a4dcd101
Removing exceptions since not used
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Review URL: http://webrtc-codereview.appspot.com/266008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@911 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-09 16:11:44 +00:00
kjellander@webrtc.org
ad79d6f164
Removing exceptions since not used
...
Review URL: http://webrtc-codereview.appspot.com/267002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@910 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-09 16:11:14 +00:00
mflodman@webrtc.org
03a9eb1526
RTP module: Make sure payloadName is null terminated.
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Review URL: http://webrtc-codereview.appspot.com/268006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@908 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-09 14:51:18 +00:00
kjellander@webrtc.org
9dcab8fb14
Restoring Android.mk
...
This is the last file left from 256006 that I forgot to restore according to your comments.
The other Android.mk you fixed in 266004.
Review URL: http://webrtc-codereview.appspot.com/268003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@905 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-09 08:59:13 +00:00
henrikg@webrtc.org
c58ef08da2
Removes system CPU measurement for Chrome build.
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It does not work on Chrome Windows, and is anyway not needed for Chrome.
Review URL: http://webrtc-codereview.appspot.com/243006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@902 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-08 08:44:17 +00:00
henrik.lundin@webrtc.org
f15fbc379d
Change in RTP module SendVP8
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Changing how the max payload length is calculated. Instead
of handling RTP and FEC header overhead explicitly, call the
MaxDataPayloadLength method which already does it. Avoid redundant code. Had to move MaxDataPayloadLength to the
RTPSenderInterface.
Review URL: http://webrtc-codereview.appspot.com/269002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@901 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-08 08:23:47 +00:00
kma@webrtc.org
9b813510eb
Changes for building audio coding in anroid. Only makefiles are touched.
...
Review URL: http://webrtc-codereview.appspot.com/266004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@899 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-07 23:30:01 +00:00
henrike@webrtc.org
26d3667a26
Fix for broken test after r897
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Review URL: http://webrtc-codereview.appspot.com/274001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@898 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-07 23:24:40 +00:00
henrike@webrtc.org
e2a34f8275
Removes the API for setting RX VAD since the RX vad should always be on anyways.
...
Review URL: http://webrtc-codereview.appspot.com/264001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@897 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-07 21:33:24 +00:00
mflodman@webrtc.org
5ae9f5ed6c
Adding logs in RTPSender::ReSendToNetwork.
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Review URL: http://webrtc-codereview.appspot.com/273001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@896 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-07 20:03:00 +00:00
kjellander@webrtc.org
bf483844af
Restructuring and removing neteq_tests.gypi according to project structure discussed with Andrew. We want to flatten out the hierarchy and minimize the number of GYP files.
...
I also fixed compilation on Mac (by enabling exceptions for the NetEqTestTools target). Executing the test fails on Mac, but I assume this is because it checks bit exactness, similar to the issue we had with audio_coding_module (see issue 114)
Review URL: http://webrtc-codereview.appspot.com/255004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@895 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-07 16:05:19 +00:00
kjellander@webrtc.org
36e1ad9b5d
Restructuring and removing ilbc_test.gypi.
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According to project structure discussed with Andrew. We want to flatten out the hierarchy and minimize the number of GYP files.
No changes at all are being made in the source files; they are just moved.
The only modified files are the GYP file and Android.mk
Kevin: I updated relative paths in Android.mk so please verify it is correct, since I don't know how to build that.
Review URL: http://webrtc-codereview.appspot.com/256006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@894 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-07 15:27:11 +00:00
vikasmarwaha@webrtc.org
a5c4c1f1d4
Fix for WebRTC issue 64, removed the screenupdate thread and events from start render as they are already created in the ctor.
...
Review URL: http://webrtc-codereview.appspot.com/253008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@890 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-04 23:22:51 +00:00
marpan@webrtc.org
040cb71e0a
Fix windows compilation errors and warning for test_fec. Disabled VERBOSE_OUTPUT.
...
Review URL: http://webrtc-codereview.appspot.com/253005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@889 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-04 22:57:56 +00:00
tina.legrand@webrtc.org
731e9aea79
Fixes ACM API test to build on 32-bits machines.
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Changing counters from unsigned int64 to int.
Review URL: http://webrtc-codereview.appspot.com/256010
git-svn-id: http://webrtc.googlecode.com/svn/trunk@887 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-04 07:34:22 +00:00
kjellander@webrtc.org
20a370e875
Changing the namespace of TestSuite to webrtc::test.
...
Adding gmock initialization into main test runner class
Review URL: http://webrtc-codereview.appspot.com/254004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@885 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-04 01:19:16 +00:00
kjellander@webrtc.org
1a8d08ad76
Changing usage of gtest_main target, to use test_support_main instead.
...
Review URL: http://webrtc-codereview.appspot.com/252002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@884 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-03 23:28:47 +00:00
andrew@webrtc.org
89088b963e
Fix the path to protoc.gypi.
...
It was mistakenly pointing to the trunk/build rather than the
trunk/src/build copy, causing the Chrome build to fail.
TEST=./build/gyp_chromium in Chrome
Review URL: http://webrtc-codereview.appspot.com/253006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@883 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-03 20:43:45 +00:00
tina.legrand@webrtc.org
2475a1953a
Committing a file that was part of CL 175002, but for wome reason weren't uploaded correctly.
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@882 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-03 17:54:27 +00:00
tina.legrand@webrtc.org
fb389e3b02
This CL is divided in several patches, to make review easier.
...
Patch Set 1: Removing blanks at end of lines.
Patch Set 2: Removing tabs.
Patch Set 3: Fixing include-guards.
Patch Set 4-7: Formatting files in the list.
Patch Set 8: Formatting CNG.
Patch Set 9:
* Fixing comments from code review
* Fixing formating in acm_dtmf_playout.cc
* Started fixing formating of acm_g7221.cc. More work needed, so don't spend too much time reviewing.
* Refactored constructor of ACMGenericCodec. Rest of file still to be fixed.
* Fixing break; after return ...; in several files.
Patch Set 10:
* Chaning from reintepret_cast to static_cast in three files, acm_amr.cc, acm_cng.cc and acm_g722.cc
NOTE! Not all files have the right format. That work will continue in separate CLs.
Review URL: http://webrtc-codereview.appspot.com/175002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@881 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-03 17:20:10 +00:00
mikhal@webrtc.org
e203de7ba2
jitter_buffer updates:
...
1. Determining continuity based on pictureId and not seq. numbers when available.
2. Hybrid bug fix: Don't set to decodable when the nack list is empty.
Review URL: http://webrtc-codereview.appspot.com/255001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@878 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-03 00:42:52 +00:00
pwestin@webrtc.org
7232ad78b2
reverted back the sanity and changed the test
...
Review URL: http://webrtc-codereview.appspot.com/254006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@877 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-03 00:36:32 +00:00
pwestin@webrtc.org
cfc1070586
Fixed sanity for min length
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Review URL: http://webrtc-codereview.appspot.com/259003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@876 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-03 00:15:44 +00:00
pwestin@webrtc.org
075e91fa27
Added parsing of width and height from VP8 header
...
Review URL: http://webrtc-codereview.appspot.com/241012
git-svn-id: http://webrtc.googlecode.com/svn/trunk@875 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-02 23:14:58 +00:00
henrik.lundin@webrtc.org
679cb07980
Fix build error for release build
...
Review URL: http://webrtc-codereview.appspot.com/252003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@874 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-02 19:52:27 +00:00
henrik.lundin@webrtc.org
baf6db5ead
Making dual decoder work again in VCM
...
Changing the assignment operator in VCMJitterBuffer to a named
function (CopyFrom) instead, since it is not a straight
assignment. Also fixing two bugs in the jitter copy function.
Bug fix in VCMEncodedFrame: The copy constructor did not
copy _length.
In VCM codec database, make sure that the callback object is
preserved when copying back from secondary to primary decoder.
In VP8 wrapper, adding code to copy the _decodedImage to the
Copy() method.
Bugfix in video_coding_test rtp_player:
The retransmissions where made in reverse order. Now new items are
appended to the end of the LostPackets list, which makes the order
correct when retransmitting.
Handling the case when cloning an unused decoder state:
When the decoder has not successfully decoded a frame yet,
it cannot be cloned. A NULL pointer will be returned all
the way out to VideoCodingModuleImpl::Decode(). When this
happens, the VCM will call Reset() for the dual receiver,
in order to reset the state to kPassive.
Review URL: http://webrtc-codereview.appspot.com/239010
git-svn-id: http://webrtc.googlecode.com/svn/trunk@873 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-02 18:58:39 +00:00
kjellander@webrtc.org
d292b9c9da
Unit tests now compile and run at all platforms.
...
Cosmetic changes to mocks.h.
Review URL: http://webrtc-codereview.appspot.com/253003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@871 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-02 16:34:52 +00:00
henrik.lundin@webrtc.org
895870b68f
Adding marker bit to RTPanalyze results
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Review URL: http://webrtc-codereview.appspot.com/254005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@867 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-02 08:44:42 +00:00
mikhal@webrtc.org
bb8dfbdee2
updating vpm unit_test following r858
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Review URL: http://webrtc-codereview.appspot.com/255005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@865 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-01 22:07:16 +00:00
turaj@webrtc.org
7395d3d8e9
Addressing issue 115 http://code.google.com/p/webrtc/issues/detail?id=115
...
Review URL: http://webrtc-codereview.appspot.com/261002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@864 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-01 17:33:06 +00:00
turaj@webrtc.org
fac5316856
Address the problem that iSAC could not go 16 kHz. It was addressed in P4 but not moved to svn.
...
Review URL: http://webrtc-codereview.appspot.com/261001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@863 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-01 17:32:46 +00:00
turaj@webrtc.org
9116cf7c9b
Have a guard on computing nrg to avoid wrap-around. This is discovered in a release test. During entropy coding of spectrum the value of "nrg" was too large and after shifting it became negative, resulting in decoder error.
...
Review URL: http://webrtc-codereview.appspot.com/239016
git-svn-id: http://webrtc.googlecode.com/svn/trunk@862 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-01 17:29:34 +00:00
mflodman@webrtc.org
29d75b3f7d
Only allow increasing capture time.
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Review URL: http://webrtc-codereview.appspot.com/259001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@861 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-01 17:10:49 +00:00
andrew@webrtc.org
18ee6ec8e9
Use __inline in NS-fixed.
...
The use of "inline" was failing to build on Windows.
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/255003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@860 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-01 17:07:46 +00:00
andrew@webrtc.org
3119ecfec8
Fix audioproc build errors on Windows.
...
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/254003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@859 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-01 17:00:18 +00:00
mikhal@webrtc.org
c4ab8706f4
video_processing: Adding logic to avoid a memcpy when not required
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Review URL: http://webrtc-codereview.appspot.com/255002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@858 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-01 16:44:24 +00:00
punyabrata@webrtc.org
0ab521f754
Resolving a crash related to strncopy followed by a strcat
...
call. strncopy will not explicity copy or add a "\0" therefore
strcat did not know where to append the "\n" which was causing
an out of bounds crash.
Because we are checking the length, strcpy should be good enough
as it also copies the "\0". Please note that that I am pre-emptively
adding 2 instead of 1 to the length to take into account of the \n
that will be added later.
Review URL: http://webrtc-codereview.appspot.com/253004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@857 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-01 15:19:44 +00:00
kjellander@webrtc.org
d6837709cf
Fixing VPMUnitTest compilation error on Windows.
...
It tried to include Visual Leak Detector which is not a tool that is installed/configured by default in the build.
Review URL: http://webrtc-codereview.appspot.com/257002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@854 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-01 01:51:10 +00:00
henrike@webrtc.org
b37c628ae4
Fixes crash due to r841.
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Review URL: http://webrtc-codereview.appspot.com/256004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@853 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-31 23:53:04 +00:00
kma@webrtc.org
e9f909b575
Move the SetAndroidObjects to VideoCaptureFactory so that ViE can get access to it.
...
Review URL: http://webrtc-codereview.appspot.com/244002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@852 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-31 22:24:57 +00:00
kma@webrtc.org
27957508a3
Changed Android makefile to make the lastest video render code run.
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Review URL: http://webrtc-codereview.appspot.com/247005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@849 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-31 21:29:50 +00:00
henrike@webrtc.org
066f9e5a2f
Ray, please verify that this cl fixes the issue. Once the verification has been made, please review:
...
Henrik A: VoE
Andrew: audio_conference_mixer
Thanks!
Review URL: http://webrtc-codereview.appspot.com/241010
git-svn-id: http://webrtc.googlecode.com/svn/trunk@841 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-28 23:15:47 +00:00
henrike@webrtc.org
731ecba47d
Review URL: http://webrtc-codereview.appspot.com/251002
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@840 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-28 22:49:24 +00:00
braveyao@webrtc.org
1f6d740571
This CL is about to manually reset the ShutdownRenderEvent at StopPlayout().
...
It could happen that if you want to restart playout, the new sponsored Render thread would catch this event
if the previous Render thread quits before this event is set.
With this modification, the device plugging out/in during talking would be supported well.
Review URL: http://webrtc-codereview.appspot.com/248002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@839 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-28 21:30:30 +00:00
stefan@webrtc.org
f960211f8b
Fixes two jitter buffer bugs related to NACK.
...
Avoid decoding delta frames after a Flush() and after requesting
a key frame due to full NACK list.
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/247011
git-svn-id: http://webrtc.googlecode.com/svn/trunk@837 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-28 16:00:49 +00:00
stefan@webrtc.org
eb65860720
Reverts the workaround in r823 and solves a macro bug.
...
The macro bug caused frames to be dropped after being grabbed
for decoding.
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/248004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@831 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-28 12:25:34 +00:00
tina.legrand@webrtc.org
8b1f621e3a
Updated gypi for tests to work on osx.
...
Review URL: http://webrtc-codereview.appspot.com/250002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@830 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-28 08:57:34 +00:00
mikhal@webrtc.org
5200a05500
video_coding/jitter_buffer Updating condition on which we return a frame.
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Review URL: http://webrtc-codereview.appspot.com/240011
git-svn-id: http://webrtc.googlecode.com/svn/trunk@825 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-27 16:54:54 +00:00
mikhal@webrtc.org
30f6376802
VP8: Updating codec version: VP8 version will now return the libvpx version used.
...
Review URL: http://webrtc-codereview.appspot.com/247009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@824 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-27 16:45:00 +00:00
stefan@webrtc.org
2d28aff785
Workaround for an issue where frames are grabbed for decoding prematurely.
...
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/240013
git-svn-id: http://webrtc.googlecode.com/svn/trunk@823 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-27 16:13:18 +00:00
stefan@webrtc.org
fbea4e555d
Solves two bandwidth estimation issues and measures the sent video bitrate.
...
Issues solved:
1. Possible overflow when reducing the bandwidth estimate at the send-side
2. A burst of loss reports could make us reduce the rate way too far since
we reduced the rate relative the current estimate and not the actual
rate sent.
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/244011
git-svn-id: http://webrtc.googlecode.com/svn/trunk@822 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-27 16:08:29 +00:00
mflodman@webrtc.org
7e4269e9ee
Changed VP8 qp min and added noise reduction.
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Review URL: http://webrtc-codereview.appspot.com/248003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@821 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-27 12:59:47 +00:00
kjellander@webrtc.org
6b7799021c
Fixing build errors on Windows platform. Minor changes...
...
Review URL: http://webrtc-codereview.appspot.com/241004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@819 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-26 02:38:09 +00:00
andrew@webrtc.org
cb18121990
Add an unpacker tool for audioproc debug files.
...
- It only unpacks audio data at the moment.
- Also switch to Chrome's protoc.gypi for protobuf targets. This reduces
the complexity of our targets.
Review URL: http://webrtc-codereview.appspot.com/241009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@817 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-26 00:27:17 +00:00
frkoenig@google.com
fc9bcef8c7
Data alignment fix for SSIM.
...
WebRtc_UWord64[2] wasn't always aligned to 128 bytes, which
is necessary for _mm_store_si128. By declaring the
variable as __m128i it will always be 128 bytes aligned.
Incorrect include files.
__m128i is defined in emmintrin.h for visual studio. Extra include on mac and linux is not a problem.
Review URL: http://webrtc-codereview.appspot.com/239013
git-svn-id: http://webrtc.googlecode.com/svn/trunk@816 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-26 00:07:32 +00:00
stefan@webrtc.org
d855c1a4e8
Reverts r807 and fixes the real issue in the VCM.
...
This fixes an issue in the VCM where we don't wait for a packet to arrive
if the jitter buffer is empty. This also fixes an issue where an old
packet can trigger a packet event signal for a future frame.
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/248001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@814 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-25 11:52:48 +00:00
henrika@webrtc.org
bdb55c806f
This CL is an attempt to remove a crash we can see when closing down VoiceEgine.
...
It can happen that the capture thread tries to access an invalid object after StopPlayout has been called.
I have also extended the usage of the new ScopedCOMInitializer to all threads. See this step as code cleanup.
Review URL: http://webrtc-codereview.appspot.com/239014
git-svn-id: http://webrtc.googlecode.com/svn/trunk@813 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-25 11:03:28 +00:00
henrika@webrtc.org
a6c23357c0
Solves crash in ADM API unit test for Core Audio on Windows
...
Review URL: http://webrtc-codereview.appspot.com/244009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@812 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-25 08:31:33 +00:00
henrika@webrtc.org
5423bc2d0b
Adds correct absolute paths to all input files in ADM functional unit tests.
...
Files are now read and played out correctly.
Review URL: http://webrtc-codereview.appspot.com/246006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@811 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-25 08:24:20 +00:00
kma@webrtc.org
ca325ececd
Corrected a linux build error introduced in issue 246005.
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Review URL: http://webrtc-codereview.appspot.com/246008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@809 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-25 02:36:09 +00:00
wjia@webrtc.org
f0cd394a2e
Put fwrite calls under corresponding macros since they shouldn't show up in release build.
...
This also make chromeos build happy.
BUG=none
TEST=compile
Review URL: http://webrtc-codereview.appspot.com/247006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@808 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-25 00:40:43 +00:00
mikhal@webrtc.org
f31826e17b
adding a wait on the decode thread when no frames are available
...
Review URL: http://webrtc-codereview.appspot.com/246009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@807 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-25 00:20:54 +00:00
mikhal@webrtc.org
a412924c0e
VP8:Setting number of cores based on image size
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Review URL: http://webrtc-codereview.appspot.com/242010
git-svn-id: http://webrtc.googlecode.com/svn/trunk@806 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-25 00:02:43 +00:00
kma@webrtc.org
913644b92d
For commiting changes in CL 277002, due to file structure changes introduced during
...
the review of the code.
Review URL: http://webrtc-codereview.appspot.com/246005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@805 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-24 21:36:33 +00:00
andrew@webrtc.org
537096a5c1
Remove unnecessary objective-c compiler flags.
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Review URL: http://webrtc-codereview.appspot.com/239011
git-svn-id: http://webrtc.googlecode.com/svn/trunk@802 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-24 15:07:08 +00:00
henrika@webrtc.org
bf478faebb
Ensures that ADM unit tests builds on all platforms.
...
Review URL: http://webrtc-codereview.appspot.com/240009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@800 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-24 10:31:02 +00:00
stefan@webrtc.org
5eb64f06be
Fix BitrateSent() API when having a default RTP module.
...
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/242004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@796 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-21 13:42:50 +00:00
stefan@webrtc.org
158f496030
Fixes a rate control bug in the VP8 wrapper.
...
Changes how we signal frame rate and frame durations to the encoder. Rather
than changing the time base, we now only modify the frame durations, while
keeping the timebase constant. The frame duration is currently calculated
from the average input frame rate. Ideally, the frame duration should
be calculated as the timestamp diff, which is the real duration of a
frame, but the encoder doesn't seem to like too varying durations.
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/247001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@795 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-21 13:15:16 +00:00
stefan@webrtc.org
ead87b5051
Fix potential issue where frame buffers might be freed while being decoded.
...
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/243004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@791 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-21 06:46:37 +00:00
stefan@webrtc.org
2b0f094c8f
Avoid reallocating the decodedImage for every decoded frame.
...
Also made sure the right size is allocated.
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/240004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@790 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-21 06:39:03 +00:00
mikhal@webrtc.org
ee3dfa6f43
Review URL: http://webrtc-codereview.appspot.com/241007
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@789 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-21 00:46:09 +00:00
mikhal@webrtc.org
1af915d8ae
video_coding: vp8: Updating error propagation threshold
...
Review URL: http://webrtc-codereview.appspot.com/246002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@788 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-20 18:19:18 +00:00
kma@webrtc.org
d75889e2eb
Change of Android makefiles to build latest video coding code.
...
Review URL: http://webrtc-codereview.appspot.com/239008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@786 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-20 16:28:56 +00:00
henrika@webrtc.org
cedbb036d1
[Issue 101] Solves memory leak on Windows
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@784 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-20 12:11:45 +00:00
stefan@webrtc.org
c4d1983b7b
Changes in rtp_format_vp8_unittest to match the changes in CL 774.
...
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/241006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@782 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-20 08:19:34 +00:00
kjellander@webrtc.org
81f25f9ff8
Fixing build errors on Windows platform. Minor changes...
...
Review URL: http://webrtc-codereview.appspot.com/241004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@779 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-19 20:06:56 +00:00
wu@webrtc.org
f3f2f6abdb
* Add include_internal_video_capture and include_internal_video_render to include/exclude the internal VCM and VRM.
...
* Split the WEBRTC_VIDEO_EXTERNAL_CAPTURE_AND_RENDER into WEBRTC_INCLUDE_INTERNAL_VIDEO_CAPTURE and WEBRTC_INCLUDE_INTERNAL_VIDEO_RENDER.
* Add DummyDeviceInfo for the case when WEBRTC_INCLUDE_INTERNAL_VIDEO_CAPTURE is not defined.
Review URL: http://webrtc-codereview.appspot.com/224005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@778 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-19 18:42:17 +00:00
henrike@webrtc.org
509c9c5d09
operator + is evaluated before ?:
...
Parenthesis ensures the intended behavior.
Review URL: http://webrtc-codereview.appspot.com/239003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@777 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-19 18:31:01 +00:00
henrike@webrtc.org
4df8c9a2ed
Review URL: http://webrtc-codereview.appspot.com/243001
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@776 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-19 18:30:25 +00:00
stefan@webrtc.org
ffd28f95c5
Request key frames to battle error propagation.
...
The VP8 decoder wrapper will request key frames 30 frames after seeing
a packet loss, if it hasn't received a state refresh (only possible
through key frames in this version).
For this to be possible the jitter buffer has been made aware of
picture ids to be able to detect frame losses. Legacy JB code to
handle streams without marker bits was also removed since it
conflicts with streams with FEC.
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/239002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@774 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-19 15:55:39 +00:00
mikhal@webrtc.org
d0752c370d
video_coding: Update to hybrid mode: Set FEC values for zero below a threshold.
...
Review URL: http://webrtc-codereview.appspot.com/245001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@773 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-19 15:48:30 +00:00
bjornv@webrtc.org
4c636764b7
Updated the AEC delay logging to output values in ms. PB output updated.
...
Review URL: http://webrtc-codereview.appspot.com/223003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@770 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-19 08:47:40 +00:00
mflodman@webrtc.org
ce8813da4e
Using id instead of name when setting Mac/QTKit capture device.
...
Review URL: http://webrtc-codereview.appspot.com/241002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@768 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-19 06:45:16 +00:00
andrew@webrtc.org
4d5d5c1267
Reorganize the audio_processing source.
...
- Remove main and source directories.
- Change .gyp, .gypi and Android.mk files correspondingly. No other
source changes.
Review URL: http://webrtc-codereview.appspot.com/241001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@767 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-19 01:40:33 +00:00
wu@webrtc.org
8fd93d4d96
Move DeliverCapturedFrame from private to protected.
...
Review URL: http://webrtc-codereview.appspot.com/246001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@765 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-19 00:16:36 +00:00
stefan@webrtc.org
5b15cfc6dd
Fix BWE unit test build issue
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@762 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-18 07:22:33 +00:00
kjellander@webrtc.org
61f07c3184
I have made a small fix so it will execute properly from the default working directory location (trunk), finding its resource files.
...
The ApmTest.Process test is still failing and needs to be resolved.
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/194002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@761 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-18 06:54:58 +00:00
wu@webrtc.org
76aea651ff
When _audioConfigured, should not try to use the _video.
...
Review URL: http://webrtc-codereview.appspot.com/224004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@758 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-17 21:40:32 +00:00
wu@webrtc.org
f10ea31211
Add IncomingFrameI420 to ViEExternalCapture interface to take captured video frame buffer as 3 planes.
...
Review URL: http://webrtc-codereview.appspot.com/219004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@753 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-14 17:16:04 +00:00
marpan@webrtc.org
14aaaf116a
Some re-organization of the fec-uep code: updated protection modes, comments, and some variable/function re-naming.
...
Review URL: http://webrtc-codereview.appspot.com/231001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@752 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-14 16:28:02 +00:00
wu@webrtc.org
55c39f0940
Add mallinath@webrtc.org and wu@webrtc.org as the capture owner for US office.
...
Review URL: http://webrtc-codereview.appspot.com/230001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@751 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-14 15:34:19 +00:00
wu@webrtc.org
58691ebb97
Remove the DestroyDeviceInfo for mac video capture. (This is missed in r731.)
...
Review URL: http://webrtc-codereview.appspot.com/229001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@750 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-14 15:13:16 +00:00
stefan@webrtc.org
d0bdab0128
Adding API to get sent total bitrate, FEC bitrate and NACK bitrate.
...
Also adding tests for this in vie_auto_test.
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/199001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@749 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-14 14:24:54 +00:00
marpan@webrtc.org
5a3e20f678
Removed unused variables (build error) for test_fec.
...
Review URL: http://webrtc-codereview.appspot.com/223001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@738 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-13 16:59:24 +00:00
pwestin@webrtc.org
1da1ce0da5
First implementation of simulcast, adds VP8 simulcast to video engine.
...
Changed API to RTP module
Expanded Auto test with a test for simulcast
Made the video codec tests compile
Added the vp8_simulcast files to this cl
Added missing auto test file
Review URL: http://webrtc-codereview.appspot.com/188001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@736 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-13 15:19:55 +00:00
stefan@webrtc.org
4c059d87b3
Add metric for number of packets discarded by JB due to not being decodable
...
Also fixes a couple of bugs related to sequence number wrap found while
testing.
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/218001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@732 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-13 07:35:37 +00:00
wu@webrtc.org
77d7d5455e
Replace the DestroyDeviceInfo with a virtual destructor.
...
Review URL: http://webrtc-codereview.appspot.com/212005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@731 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-12 16:57:53 +00:00
wu@webrtc.org
ea89922b56
Add VideoCaptureFactory so that we don't need to expose VideoCaptureImpl.
...
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/213002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@727 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-11 17:13:51 +00:00
andrew@webrtc.org
199f4defd3
Rename all .cc files which include Objective-C headers to .mm.
...
This allows the Mac Make build to pass. We were hacking it in XCode with "-x objective-c++", but gyp/Make doesn't seem to accept that flag.
Also switch Objective-C #includes to #imports.
There is one file missing from this: vie_autotest_main.cc, because it's required on multiple platforms. I'm not immediately sure what the best approach is there, but the Objective-C headers should be somehow hidden.
Review URL: http://webrtc-codereview.appspot.com/153005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@726 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-11 15:43:35 +00:00
stefan@webrtc.org
791eec7424
Add API to get the number of packets discarded by the video jitter buffer due to being too late.
...
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/200001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@723 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-11 07:53:43 +00:00
stefan@webrtc.org
06887aebae
Fixes two bugs when decoding with packet losses.
...
Disable _missingFrame bit since we can't set it correctly with FEC.
No longer return more than one decoded frame per Decode() call.
This is a work-around for a bug where the frame info map was popped more often than items were added to the map.
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/215001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@722 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-10 14:17:46 +00:00
kjellander@webrtc.org
25e0b8e3a0
Python output flag and keyframe interval flags.
...
Refactored main method into using 6 helper methods for better overview.
Review URL: http://webrtc-codereview.appspot.com/207001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@710 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-07 07:52:00 +00:00
kjellander@webrtc.org
a31b254084
Python output flag and keyframe interval flags.
...
Refactored main method into using 6 helper methods for better overview.
Review URL: http://webrtc-codereview.appspot.com/207001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@709 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-07 06:50:22 +00:00
henrike@webrtc.org
bf54ef9bb7
Removed code under a non-existing define.
...
Review URL: http://webrtc-codereview.appspot.com/193006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@706 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-06 18:14:25 +00:00
andrew@webrtc.org
b2d4921f3b
Remove trailing whitespace in AudioDevice.
...
(That I introduced...)
Review URL: http://webrtc-codereview.appspot.com/198002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@703 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-06 16:34:36 +00:00
kjellander@webrtc.org
35a1756502
First version of video quality measurement program and test framework.
...
See https://docs.google.com/a/google.com/document/d/1w6Nrxw6yTg_sDu18Ux8oZPEMo5F_R-zt62udrmmTeOc/edit?hl=en_US
for background, details and additional instructions on usage.
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/175001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@700 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-06 06:44:54 +00:00
kma@webrtc.org
af57de006a
Some code style changes in audio_processing/ns/main/source/ by Astyle,
...
with a little manual modification.
Review URL: http://webrtc-codereview.appspot.com/201002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@698 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-05 23:36:01 +00:00
henrik.lundin@webrtc.org
01ca01f6e6
Adding neteq_tests to modules tests
...
Also moving neteq_tests.gyp and renaming to gypi. Cleaning up a
little in neteq_tests.gypi.
Review URL: http://webrtc-codereview.appspot.com/191004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@696 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-05 20:38:19 +00:00
kma@webrtc.org
bbc1f10187
Changed modules/audio_processing/utility/Android.mk, to correct a build error in
...
Android with the change from version r674.
Review URL: http://webrtc-codereview.appspot.com/197003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@694 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-05 18:09:02 +00:00
kma@webrtc.org
bf39ff4271
Some general optimization in NS.
...
No big effort in introducing new style.
Speed improved ~2%.
Bit exact.
Will introduce mulpty-and-accumulate and sqrt_floor next, which increase speed another 2% or so.
Note: In function WebRtcNsx_DataAnalysis, did the block separation because I found one "if" case is more frequent than "else" within a for loop; rest is kind of code re-aligning.
Review URL: http://webrtc-codereview.appspot.com/181002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@692 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-05 17:10:06 +00:00
stefan@webrtc.org
4b6f747373
Fixes a newly introduced bug in the jitter buffer where buffer reallocation
...
causes corrupt pointers.
Review URL: http://webrtc-codereview.appspot.com/186003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@688 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-04 06:58:39 +00:00
stefan@webrtc.org
93d216c23f
Fixed bug in jitter buffer which caused the missingFrames bit to never be set.
...
Also updated the VP8 wrapper to return fully concealed frames (for rendering).
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/190003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@687 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-04 06:48:11 +00:00
stefan@webrtc.org
61b4abf1f8
Proper use of frame rate argument in generic_codec_test.
...
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/181005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@686 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-04 06:40:21 +00:00
mikhal@webrtc.org
e06be4f678
video coding tests: Adding ssimFrame to interface
...
Review URL: http://webrtc-codereview.appspot.com/188004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@685 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-03 22:54:43 +00:00
mikhal@webrtc.org
ae7a0522c5
video_coding robustness: Updating hybrid mode's settings
...
1. Disabling adjustment factor - temporary update.
2. Enabling a windowed filtered loss for the hybrid mode.
Review URL: http://webrtc-codereview.appspot.com/192003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@684 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-03 22:54:34 +00:00
marpan@google.com
f1f3fb33b5
Update to rate-mismatch factor in media_opt_util.
...
Review URL: http://webrtc-codereview.appspot.com/193003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@678 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-03 19:09:45 +00:00
stefan@webrtc.org
5b91464edf
Allow an aggregated partition to spill over to a new packet.
...
Adds support for the case where the partition 0 and parts of partition 1
are transmitted in packet 1, and the end of partition 2 is transmitted
in packet 2.
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/181003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@675 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-03 10:26:12 +00:00
bjornv@google.com
1ba3dbecbb
Adds possibility to log delay estimates in AEC.
...
Review URL: http://webrtc-codereview.appspot.com/178001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@674 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-03 08:18:10 +00:00
kma@google.com
c611b1a950
Bit-exact with non-Neon version.
...
Review URL: http://webrtc-codereview.appspot.com/180002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@660 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-28 16:03:38 +00:00
andrew@webrtc.org
18421f2063
Remove unnecessary include from NS interface.
...
http://code.google.com/p/webrtc/issues/detail?id=46
Review URL: http://webrtc-codereview.appspot.com/183001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@656 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-27 19:50:52 +00:00
mikhal@webrtc.org
848fad23c6
video_coding: Updating media opt test - fixing call to protection callback.
...
Review URL: http://webrtc-codereview.appspot.com/179003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@653 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-27 16:30:59 +00:00
bjornv@google.com
a2c6ea09b0
Removed a segmentation fault error when processing near_file only.
...
Review URL: http://webrtc-codereview.appspot.com/174001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@650 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-27 08:04:45 +00:00
mikhal@webrtc.org
e185e9f68a
video_coding: updates to jitter buffer logic: Make sure that every frame is inserted only once to the list.
...
Review URL: http://webrtc-codereview.appspot.com/165001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@648 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-23 22:02:40 +00:00
turajs@google.com
cf136186f5
Deleting matlab files
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@647 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-23 21:49:25 +00:00
turajs@google.com
13335ccd7a
Deleting matlab files
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@646 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-23 21:47:25 +00:00
turajs@google.com
610f478705
Deleting matlab files
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@645 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-23 21:45:34 +00:00
turajs@google.com
53439d9982
Deleting matlab files
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@644 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-23 21:44:00 +00:00
mikhal@webrtc.org
105ff39dec
video coding: updating offline tests.
...
Additional clean-up to the offline test: Placing test callbacks in a designated file.
Review URL: http://webrtc-codereview.appspot.com/167002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@642 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-23 16:41:11 +00:00
turajs@google.com
496ef8aca8
To fix warnings in test files.
...
Review URL: http://webrtc-codereview.appspot.com/169001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@641 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-23 15:45:48 +00:00
bjornv@google.com
8e9e83b530
This CL adds guards against division by zero, that should fix http://b/issue?id=5278531
...
In addition a read outside memory event has been detected and removed.
Also an improper noise weighting has been corrected.
Review URL: http://webrtc-codereview.appspot.com/152001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@640 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-23 12:39:47 +00:00
bjornv@google.com
dc743a8bba
Replaces a use of log2.
...
I've replaced a log2 operation so it works on Windows.
Review URL: http://webrtc-codereview.appspot.com/171002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@637 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-23 08:13:53 +00:00
wu@webrtc.org
221b522118
Return the number of /dev/video* without trying to open it.
...
Consider the case when there're /dev/video0 and /dev/video1. But for somereason the video0 is not in a correct state and can't be open. As a result, current NumberOfDevices will return 1, which is fine. However, we will then never be able to get the device we really want - /dev/video1. Consider the code below, the GetCaptureDevice will fail because it calls into DeviceInfoLinux::GetDeviceName(0, ...) which will again try to open the /dev/video0. So the root cause is the mismatching of the NumberOfDevices and GetDeviceName.
Since we will open the device in DeviceInfoLinux::GetDeviceName anyway, I think we should return the number of /dev/video* in DeviceInfoLinux::NumberOfDevices without trying to open it. Otherwise the DeviceInfoLinux::NumberOfDevices should return more information like which /dev/video* is valid which is not.
bool found = false;
for (int i = 0; i < vie_capture->NumberOfCaptureDevices(); ++i) {
if (vie_capture->GetCaptureDevice(i, ...) == 0) {
found = true;
break;
}
}
Review URL: http://webrtc-codereview.appspot.com/148004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@635 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-21 16:57:15 +00:00
bjornv@google.com
65e6ab31eb
Temporary log2 remove to build in chrome
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@633 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-21 11:56:46 +00:00
pwestin@webrtc.org
741da942ec
Added support for new RTCP message REMB (remote estimated max bitrate)
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Review URL: http://webrtc-codereview.appspot.com/149001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@628 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-20 13:52:04 +00:00
andrew@webrtc.org
86b85db67e
Add missing intrinsic casts for VS 2005.
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Allows re-enabling SSE optimization on Windows.
Review URL: http://webrtc-codereview.appspot.com/161003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@623 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-19 18:48:25 +00:00
leozwang@google.com
522f42bb80
Add kPlatformAndroid to platform check function
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Review URL: http://webrtc-codereview.appspot.com/161002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@622 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-19 17:39:05 +00:00
andrew@webrtc.org
ed083d4079
Modify the _vadActivity member of the AudioFrame passed to AudioProcessing.
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This saves the user from having to explicitly check stream_has_voice(). It will allow typing detection to function, which relies on this behaviour.
Review URL: http://webrtc-codereview.appspot.com/144004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@621 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-19 15:28:51 +00:00
andrew@webrtc.org
94c7413b0d
Allow echo metrics to be enabled in process_test.
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Review URL: http://webrtc-codereview.appspot.com/155002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@620 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-19 15:17:57 +00:00
henrik.lundin@webrtc.org
4c36d3b424
Fixing windows warnings in rtp_utility
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Adding explicit casting to bool to avoid warnings when compiling
in windows.
Review URL: http://webrtc-codereview.appspot.com/140002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@619 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-19 08:16:20 +00:00
andrew@webrtc.org
d02dc6e682
Removing bwe_standalone from modules.gyp
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Review URL: http://webrtc-codereview.appspot.com/144003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@614 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-17 00:44:23 +00:00
wjia@google.com
fdaee9c014
include build/common.gypi directly
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Review URL: http://webrtc-codereview.appspot.com/153006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@613 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-17 00:06:08 +00:00
andrew@webrtc.org
7b7c045b75
Fix MSVC issues in AEC to enable SSE2 optimization on Windows.
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Variables now declared at top of scope and replacing C casts with intrinsic cast functions.
Review URL: http://webrtc-codereview.appspot.com/160001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@611 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-16 22:51:57 +00:00
leozwang@google.com
b37ec71dbd
Remove delay_estimator_float.c from android build
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Review URL: http://webrtc-codereview.appspot.com/161001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@610 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-16 21:50:36 +00:00
leozwang@google.com
ce95069ade
Fix buidling error
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Review URL: http://webrtc-codereview.appspot.com/151002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@603 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-15 22:28:08 +00:00
andrew@webrtc.org
4537c2a464
Remove the UNCONSTR code path from AEC.
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Leave the unconstrained filter adaptation in a commented out function. Consider using this for a low-complexity mode.
Review URL: http://webrtc-codereview.appspot.com/146001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@601 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-15 18:25:14 +00:00
tommi@webrtc.org
8dc3985a10
Fix windows build.
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Review URL: http://webrtc-codereview.appspot.com/150001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@600 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-15 15:01:04 +00:00
bjornv@google.com
b47d4b287d
This CL includes a move of the fixed point delay estimator from aecm to apm/utility. There has also been a code change that makes it possible to enable/disable the far end alignment, so that we save complexity when used as a quality metrics.
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Review URL: http://webrtc-codereview.appspot.com/135014
git-svn-id: http://webrtc.googlecode.com/svn/trunk@599 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-15 12:27:36 +00:00
henrik.lundin@webrtc.org
29fd9a5f30
Removing warnings in all NetEQ test targets
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Now all targets in neteq.gypi builds again. Also added payload type to
the log produced by RTPanalyze.
Review URL: http://webrtc-codereview.appspot.com/148001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@598 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-15 08:25:45 +00:00
andrew@webrtc.org
b524f441d0
Correct some comment spelling errors. Skipping review.
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Review URL: http://webrtc-codereview.appspot.com/144002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@594 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-13 18:04:30 +00:00
andrew@webrtc.org
a3c6d61c44
Integrate the built-in WASAPI AEC DMO to VoE.
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Review URL: http://webrtc-codereview.appspot.com/108006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@592 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-13 17:17:49 +00:00
leozwang@google.com
b1b3e67c97
Fix compilation errors
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Review URL: http://webrtc-codereview.appspot.com/142002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@591 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-13 17:16:24 +00:00
andrew@webrtc.org
2cef36fa98
Fix Windows gyp run.
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On Windows, gyp seems to require valid source files. The matlab_plotting_test target was missing its one source file, so I removed the target.
Also moving bwe_standalone.gypi to the test include list.
Review URL: http://webrtc-codereview.appspot.com/143001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@589 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-13 17:03:15 +00:00
andrew@webrtc.org
f5fb095bf9
Fix audio processing tests gypi after recent changes.
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Review URL: http://webrtc-codereview.appspot.com/137025
git-svn-id: http://webrtc.googlecode.com/svn/trunk@588 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-13 01:04:59 +00:00
marpan@google.com
45fa141f0a
qm_select: changed default settings for uep.
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Review URL: http://webrtc-codereview.appspot.com/132015
git-svn-id: http://webrtc.googlecode.com/svn/trunk@584 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-12 16:53:19 +00:00
henrik.lundin@webrtc.org
9f710d08e1
Switch to new sqrt in NetEQ
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Switched to WebRtcSpl_SqrtFloor instead of WebRtcSpl_Sqrt in
NetEQ. The output is not bit-exact, but subjective listening
tests show no audible difference. Analysis shows that almost
all of the difference is in changed delay.
The reference file for NetEQ's unit test was updated.
Review URL: http://webrtc-codereview.appspot.com/139019
git-svn-id: http://webrtc.googlecode.com/svn/trunk@583 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-12 16:44:37 +00:00
kjellander@webrtc.org
f0a8464b74
Added more statistics during SSIM/PSNR calculation, including calculation of min/max value.
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Moved video_metrics.h into a GYP library so it can be used from other projects.
Review URL: http://webrtc-codereview.appspot.com/132013
git-svn-id: http://webrtc.googlecode.com/svn/trunk@582 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-12 13:45:39 +00:00
xians@google.com
d3185fe219
refactor the gyp file to gypi file.
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Basically, the gypi file is a copy of gyp file, but has some difference on the
path of the dependencies.
Review URL: http://webrtc-codereview.appspot.com/137020
git-svn-id: http://webrtc.googlecode.com/svn/trunk@581 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-12 12:24:39 +00:00
perkj@webrtc.org
0cc68dc38a
Change Video capture module to be reference counting. Also prevent the module from beeing deleted using the interface.
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Furthermore remove all static module creation and deletion functions.
Review URL: http://webrtc-codereview.appspot.com/133012
git-svn-id: http://webrtc.googlecode.com/svn/trunk@580 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-12 08:53:36 +00:00
tina.legrand@webrtc.org
31c6b60456
Adding calls to Version functions for external codecs.
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Also clarified in comments where to put interface files for external codecs.
Review URL: http://webrtc-codereview.appspot.com/135017
git-svn-id: http://webrtc.googlecode.com/svn/trunk@579 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-12 07:18:37 +00:00
zakkhoyt@google.com
c6e8b72c83
Removing qualifiers on include path
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Review URL: http://webrtc-codereview.appspot.com/132014
git-svn-id: http://webrtc.googlecode.com/svn/trunk@576 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-09 17:41:13 +00:00