599 Commits

Author SHA1 Message Date
kjellander
28c99bc44a iOS: Include legacy objc API in all.gyp + fix H264 libyuv dependency
The legacy objc API is not included in the GYP generation if include_tests=0.
This causes problems downstream in some cases, so it's changed in this CL.

The libyuv dependency needs to be possible to disable using the build_libyuv
GYP variable.

NOTRY=True

Review URL: https://codereview.webrtc.org/1705733002

Cr-Commit-Position: refs/heads/master@{#11652}
2016-02-17 13:38:35 +00:00
kjellander
0206000a66 iOS: Add resource files for tests and implement OutputPath
With this change the following tests have been successfully
passing in the iOS Simulator for iPhone 5 and iOS 9:
* audio_decoder_unittests
* common_video_unittests
* modules_tests
* rtc_api_objc_tests
* rtc_pc_unittests
* system_wrappers_unittests
* voice_engine_unittests

The modules_unittests and common_audio_unittests are
handled in https://codereview.webrtc.org/1698033002/

BUG=webrtc:4755
NOTRY=True

Review URL: https://codereview.webrtc.org/1694353003

Cr-Commit-Position: refs/heads/master@{#11646}
2016-02-17 06:06:17 +00:00
Peter Boström
b72dada927 Remove Reset from conditionally-compiled decoders.
Reset is no longer called but removal of them was missed in a previous
commit.

BUG=webrtc:5475
TBR=hbos@webrtc.org,tkchin@webrtc.org

Review URL: https://codereview.webrtc.org/1690193003 .

Cr-Commit-Position: refs/heads/master@{#11603}
2016-02-12 14:57:03 +00:00
hbos
0715a83a07 Avoid OpenH264 encoder bug for #threads > 1 on Mac and Chromium+Sandbox.
Until the bug has been further investigated, we're limiting the number
of threads to 1 to avoid problems. See crbug.com/583348.

BUG=chromium:500605, chromium:468365, chromium:583348

Review URL: https://codereview.webrtc.org/1677543002

Cr-Commit-Position: refs/heads/master@{#11536}
2016-02-09 10:34:34 +00:00
kjellander
f6b5509229 Fix GYP and GN references that are invalid in Chromium builds.
There were a couple of GN and GYP references that were incorrect in Chromium builds:
- GN references between WebRTC targets must be using relative paths, not absolute.
- GYP references between WebRTC targets must be using the <(webrtc_root)v variable
  in order to be expanded to the correct path in a Chromium build.

NOTRY=True
TBR=hjon@webrtc.org, hbos@webrtc.org

Review URL: https://codereview.webrtc.org/1681493002

Cr-Commit-Position: refs/heads/master@{#11521}
2016-02-08 07:04:33 +00:00
hbos
900f97534b H264: Improve FFmpeg decoder performance by using I420BufferPool.
Had to update I420BufferPool to allow zero-initializing buffers.

BUG=chromium:500605, chromium:468365, webrtc:5428

Review URL: https://codereview.webrtc.org/1645543003

Cr-Commit-Position: refs/heads/master@{#11505}
2016-02-05 16:08:39 +00:00
Peter Boström
1d04ac6f29 Untangle ViEChannel and ViEEncoder.
Extracts shared members outside the two objects, removing PayloadRouter
from receivers and the VCM for ViEChannel from senders.

Removes Start/StopThreadsAndSetSharedMembers that was used to set the
shared state between them.

Also adding DCHECKs to document what's only used by the
sender/receiver side.

BUG=webrtc:5494
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1654913002 .

Cr-Commit-Position: refs/heads/master@{#11500}
2016-02-05 10:25:52 +00:00
Peter Boström
e449915455 Measure encoding time on encode callbacks.
Permits measuring encoding time even when performed on another thread,
typically for hardware encoding, instead of assuming that encoding is
blocking the calling thread.

Permitted encoding time is increased for hardware encoders since they
can be timed to keep 30fps, for instance, without indicating overload.

Merges EncodingTimeObserver into EncodedFrameObserver to have one post-encode
callback.

BUG=webrtc:5042, webrtc:5132
R=asapersson@webrtc.org, mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1569853002 .

Cr-Commit-Position: refs/heads/master@{#11499}
2016-02-05 10:13:41 +00:00
sprang
2ddb8bd359 Avoid undefined behavior in vp8 screenshare_layers
active_layer_ could be dereferenced while being -1...
Also added som DCHECKs

BUG=webrtc:5490

Review URL: https://codereview.webrtc.org/1656233002

Cr-Commit-Position: refs/heads/master@{#11486}
2016-02-04 11:59:57 +00:00
hbos
9dc5928eb2 Ability to disable the effects of |rtc_use_h264| with DisableRtcUseH264.
Renamed the WEBRTC_THIRD_PARTY_H264 macro to WEBRTC_USE_H264 to match flag name.

The idea is to be able to turn off H264 from chromium with this function because...
1) The Chromium trybots will soon use this flag, we want to temporarily disable H264 from chromium even if flag is set in case something is broken. That way when we are ready to flip the switch the trybots will run our test code then and not after it is already enabled.
2) If feature is launched and we discover major problems we can easily disable H264 and merge with beta/stable.
3) Or, if feature is behind a *runtime* flag, this is how we would control if it is used or not.

The idea is to call DisableRtcUseH264 in chromium's PeerConnectionDependencyFactory.

BUG=chromium:500605, chromium:468365
NOTRY=True
NOPRESUBMIT=True

Review URL: https://codereview.webrtc.org/1657273002

Cr-Commit-Position: refs/heads/master@{#11474}
2016-02-03 13:09:40 +00:00
Peter Boström
ed3277bf14 Deprecate VideoDecoder::Reset() and remove calls.
Removes calls to decoder reset and instead drops delta frames and
requests keyframes until one arrives.

BUG=webrtc:5475
R=stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1647163002 .

Cr-Commit-Position: refs/heads/master@{#11460}
2016-02-02 14:40:13 +00:00
hbos
c5a39c2591 H264: Thread-safe InitializeFFmpeg. Flag to control if InitializeFFmpeg should be called.
New flag: rtc_initialize_ffmpeg, default value = !build_with_chromium.

In WebRTC standalone we initialize FFmpeg by default, in Chromium we don't by default.
Chromium is an external project that also use FFmpeg. If both projects do FFmpeg initialization code things will break. The flag makes it possible for other external projects than chromium to decide whether or not WebRTC should initialize FFmpeg.

BUG=chromium:500605, chromium:468365, webrtc:5427

Review URL: https://codereview.webrtc.org/1639273002

Cr-Commit-Position: refs/heads/master@{#11456}
2016-02-02 10:30:57 +00:00
Peter Boström
74451a5ea9 Prevent zero division in VCMJitterEstimator.
BUG=webrtc:5124
R=sprang@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1652903002 .

Cr-Commit-Position: refs/heads/master@{#11445}
2016-02-01 15:31:17 +00:00
Peter Boström
5e8351b325 Prevent division-by-zero in VCMFecMethod.
Clamps frameRate to at least 1.0 to prevent a zero division.

BUG=webrtc:5124
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1642903002 .

Cr-Commit-Position: refs/heads/master@{#11421}
2016-01-28 22:55:45 +00:00
Peter Boström
fb152707ed Replace const-reference with pointer in SendData.
This argument is never used as a reference and the pointer that's bound
to the const reference may be nullptr. This is undefined behavior and
barks under UBSan.

BUG=webrtc:5124
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1642863003 .

Cr-Commit-Position: refs/heads/master@{#11418}
2016-01-28 18:12:09 +00:00
hbos
bab934bffe H.264 video codec support using OpenH264 (http://www.openh264.org/) for encoding and FFmpeg (https://www.ffmpeg.org/) for decoding.
It works on all platforms except Android and iOS (FFmpeg limitation).

Implemented behind compile time flags, off by default.
The plan is to have it enabled in Chrome (see bug), but not in Chromium/webrtc by default.

Flags to turn it on:
- rtc_use_h264 = true
- ffmpeg_branding = "Chrome" (or other brand that includes H.264 decoder)

Tests using H264:
- video_loopback --codec=H264
- screenshare_loopback --codec=H264
- video_engine_tests (EndToEndTest.SendsAndReceivesH264)

NOTRY=True
BUG=500605, 468365
BUG=https://bugs.chromium.org/p/webrtc/issues/detail?id=5424

Review URL: https://codereview.webrtc.org/1306813009

Cr-Commit-Position: refs/heads/master@{#11390}
2016-01-27 09:36:07 +00:00
asapersson
7fd881743c Fix type of local encoded length variable from uint32_t to size_t.
BUG=chromium:571594

Review URL: https://codereview.webrtc.org/1635083002

Cr-Commit-Position: refs/heads/master@{#11383}
2016-01-26 15:26:12 +00:00
Peter Boström
59b2d3ebe7 Remove zero-divide in VCMContentMetricsProcessing.
Protects against zero-framerate in UpdateFrameRate.

BUG=webrtc:5124
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1636943003 .

Cr-Commit-Position: refs/heads/master@{#11382}
2016-01-26 15:18:56 +00:00
asapersson
ffa3fdc8d6 Reallocate encoded buffer size if needed for VP8. Initially set to the input image size.
Issue may occur for very small input images (e.g. 4x4) when encoded image length > input image size.

BUG=chromium:571594

Review URL: https://codereview.webrtc.org/1626373002

Cr-Commit-Position: refs/heads/master@{#11376}
2016-01-26 09:56:35 +00:00
Peter Boström
4fd6cda067 Add tracing to VCMGenericEncoder::Release.
This call can take time, especially on mobile HW encoders. Make sure it
shows up in tracing.

BUG=webrtc:5167
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1606413002 .

Cr-Commit-Position: refs/heads/master@{#11374}
2016-01-26 09:20:03 +00:00
asapersson
86956ded42 Small cleanup in VP9EncoderImpl::GetEncodedLayerFrame.
BUG=

Review URL: https://codereview.webrtc.org/1616843004

Cr-Commit-Position: refs/heads/master@{#11373}
2016-01-26 09:05:28 +00:00
Peter Boström
85b22e2306 Remove vp8_factory.{cc,h}.
Removes use of global VP8EncoderFactory::use_simulcast_adapter which is
thread-unsafe. Also the code wasn't in use.

BUG=
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1598803005 .

Cr-Commit-Position: refs/heads/master@{#11370}
2016-01-25 16:58:08 +00:00
asapersson
28ba92731d Switch to use new implementation in metrics.h.
Sparse macro replaced for all video histograms that have a constant name.

BUG=webrtc:5283

Review URL: https://codereview.webrtc.org/1616153005

Cr-Commit-Position: refs/heads/master@{#11368}
2016-01-25 13:58:27 +00:00
pbos
5ad935cb56 Remove mutable from rtc::CriticalSection members.
rtc::CriticalSection is now lockable from const methods and no longer
need to remain mutable.

BUG=
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1613643004

Cr-Commit-Position: refs/heads/master@{#11367}
2016-01-25 11:52:53 +00:00
hbos
3fe2c6a161 VideoProcessorImpl using EncodedImage::GetBufferPaddingBytes.
BUG=https://bugs.chromium.org/p/webrtc/issues/detail?id=5424

Review URL: https://codereview.webrtc.org/1618853002

Cr-Commit-Position: refs/heads/master@{#11353}
2016-01-22 08:07:17 +00:00
hbos
d664836efa Added EncodedImage::GetBufferPaddingBytes.
The FFmpeg video decoder requires up to 8 additional bytes to be allocated for its encoded image buffer input, due to optimized byte readers over-reading on some platforms.
We plan to use FFmpeg for a soon-to-land H.264 enc/dec.

This CL adds support for padding encoded image buffers based on codec type, and makes sure calls to VCMEncodedFrame::VerifyAndAllocate use the padding.

All padding constants are 0 but making H.264 pad with 8 bytes will be a one-line change.

Also, added -framework CoreFoundation to webrtc_h264_video_toolbox which was missing.

BUG=chromium:468365
BUG=https://bugs.chromium.org/p/webrtc/issues/detail?id=5424
NOTRY=True

Review URL: https://codereview.webrtc.org/1602523004

Cr-Commit-Position: refs/heads/master@{#11337}
2016-01-21 13:43:18 +00:00
hbos
902c03e724 rtc_use_h264 flag (replacing use_third_party_h264 flag) for building OpenH264/FFmpeg, false by default but can be overridden in supplement.gypi and build_overrides/webrtc.gni.
BUG=468365
NOTRY=True

Review URL: https://codereview.webrtc.org/1601813005

Cr-Commit-Position: refs/heads/master@{#11333}
2016-01-21 11:34:47 +00:00
asapersson
d9f641e341 Reallocate encoded buffer size if needed. Initially set to the input image size.
Issue may occur for very small input images (e.g. 4x4) when encoded image length > input image size.

BUG=chromium:578193

Review URL: https://codereview.webrtc.org/1603643006

Cr-Commit-Position: refs/heads/master@{#11329}
2016-01-21 09:11:43 +00:00
Peter Boström
a5dec16b42 Name SimulcastEncoderApdater on InitEncode.
Provides a better string (provides names of all implementations), but
also fixes a crash when accessing the ImplementationName() of
SimulcastEncoderAdapter where InitEncode has failed.

BUG=chromium:577932, webrtc:4897
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1599353003 .

Cr-Commit-Position: refs/heads/master@{#11321}
2016-01-20 14:54:02 +00:00
Peter Boström
7b971e728b Remove extra_options from VideoCodec.
Constructing default options is racy when initializing multiple VP8
encoders in parallel. This is only used for VP8 temporal layers. Adding
TemporalLayerFactory to VP8 codec specifics instead of generic options.

Removes the last webrtc::Config uses/includes from video code.

Also removes VideoCodec equality operators which are no longer in use.

BUG=webrtc:5410
R=stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1606613003 .

Cr-Commit-Position: refs/heads/master@{#11307}
2016-01-19 15:26:24 +00:00
Peter Boström
233bfd2da4 Move keyframe requests outside encoder mutex.
Enables faster keyframe requests since they are no longer blocked by
calls to the encoder.

BUG=webrtc:5410
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1600553003 .

Cr-Commit-Position: refs/heads/master@{#11294}
2016-01-18 19:23:51 +00:00
aluebs
688e308a35 Re-land: "Use an explicit identifier in Config"
This let's us use them to configure them when using WebRTC as an external library. One use case where this is necessary is in the Android OS.

Original CL: https://codereview.webrtc.org/1538643004/

TBR=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1589573004

Cr-Commit-Position: refs/heads/master@{#11248}
2016-01-14 12:32:51 +00:00
tommi
fca54f41ad Revert of Use an explicit identifier in Config (patchset #4 id:60001 of https://codereview.webrtc.org/1538643004/ )
Reason for revert:
Reverting due to problem with roll:

/b/build/slave/linux/build/src/buildtools/linux64/gn gen //out/Release '--args=ffmpeg_branding="Chrome" proprietary_codecs=true is_debug=false is_component_build=false use_goma=true goma_dir="/b/build/goma" symbol_level=1 dcheck_always_on=true' --check --runtime-deps-list-file=/b/build/slave/linux/build/src/out/Release/runtime_deps
  -> returned 1
ERROR at //third_party/webrtc/BUILD.gn:245:18: Item not found
    configs -= [ "//build/config/clang:find_bad_constructs" ]
                 ^-----------------------------------------
You were trying to remove "//build/config/clang:find_bad_constructs"
from the list but it wasn't there.
GN gen failed: 1
step returned non-zero exit code: 1
@@@STEP_FAILURE@@@

Original issue's description:
> Use an explicit identifier in Config
>
> This let's us use them to configure them when using WebRTC as an external library. One use case where this is necessary is in the Android OS.
>
> Committed: https://crrev.com/25249d92d3cf105bcc7b684c8924ccdbc9afcb93
> Cr-Commit-Position: refs/heads/master@{#11231}

TBR=henrik.lundin@webrtc.org,stefan@webrtc.org,tommi@chromium.org,aluebs@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review URL: https://codereview.webrtc.org/1586563003

Cr-Commit-Position: refs/heads/master@{#11239}
2016-01-13 16:12:07 +00:00
aluebs
25249d92d3 Use an explicit identifier in Config
This let's us use them to configure them when using WebRTC as an external library. One use case where this is necessary is in the Android OS.

Review URL: https://codereview.webrtc.org/1538643004

Cr-Commit-Position: refs/heads/master@{#11231}
2016-01-13 02:50:31 +00:00
hbos
a9a1d2acaf H.264: Default flags and pulling in openh264 and ffmpeg.
Defining use_third_party_h264 directly, and indirectly defining use_openh264 (from third_party/openh264) and ffmpeg_branding (from third_party/ffmpeg).
These will be used in a follow-up CL that adds an encoder and decoder implementation.
The flags are added in this CL so that they can be used by trybots/waterfall bots in GN without "Build argument had no effect" errors. Equivalent GYP changes are also added.

BUG=468365

Review URL: https://codereview.webrtc.org/1575913003

Cr-Commit-Position: refs/heads/master@{#11204}
2016-01-11 18:19:06 +00:00
Marco
69387930e0 vp9 tests: Adjust some parameters and re-enable the tests.
Tests were failing on android with new libvpx.
vp9 speed setting was changed to 8 recently and some recent changes
in libvpx require update for the tests to pass.

TBR=stefan@webrtc.org
BUG=webrtc:5401

Review URL: https://codereview.webrtc.org/1569903002 .

Cr-Commit-Position: refs/heads/master@{#11173}
2016-01-07 20:00:44 +00:00
pbos
ecd21b481f Add ImplementationName to SimulcastEncoderAdapter.
BUG=webrtc:4897
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1555673002

Cr-Commit-Position: refs/heads/master@{#11170}
2016-01-07 16:03:13 +00:00
Peter Boström
7776e782d6 Remove unused methods in VideoCodingModule.
Also voids ::Codec which always passed.

BUG=
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1464313004 .

Cr-Commit-Position: refs/heads/master@{#11167}
2016-01-07 14:42:58 +00:00
kjellander
44cc795016 Roll chromium_revision 4df108a..2a70cb1 (367307:367468)
Mac 32-bit support has been gone in Chromium for a long time, but was
removed in https://codereview.chromium.org/1557823002. This called
for finally removing our Mac 32-bit builds, which was done in
http://crbug.com/574320.

Change log: 4df108a..2a70cb1
Full diff: 4df108a..2a70cb1

Changed dependencies:
* src/third_party/libvpx_new/source/libvpx: ecb8dff..a9dd8a7
* src/third_party/nss: aee1b12..225bfc3
DEPS diff: 4df108a..2a70cb1/DEPS

No update to Clang.

TBR=marpan@webrtc.org, stefan@webrtc.org,
BUG=webrtc:5401, webrtc:5402
NOTRY=True

Review URL: https://codereview.webrtc.org/1556273002

Cr-Commit-Position: refs/heads/master@{#11159}
2016-01-07 06:12:36 +00:00
Peter Boström
e2976c87f7 Remove DISABLED_ON_ macros.
Macro incorrectly displays DISABLED_ON_ANDROID in test names for
parameterized tests under --gtest_list_tests, causing tests to be
disabled on all platforms since they contain the DISABLED_ prefix rather
than their expanded variants.

This expands the macro variants to inline if they're disabled or not,
and removes building some tests under configurations where they should
fail, instead of building them but disabling them by default.

The change also removes gtest_disable.h as an unused include from many
other files.

BUG=webrtc:5387, webrtc:5400
R=kjellander@webrtc.org, phoglund@webrtc.org
TBR=henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1547343002 .

Cr-Commit-Position: refs/heads/master@{#11150}
2016-01-04 21:44:16 +00:00
philipel
5908c71128 Lint fix for webrtc/modules/video_coding PART 3!
Trying to submit all changes at once proved impossible since there were
too many changes in too many files. The changes to PRESUBMIT.py
will be uploaded in the last CL.
(original CL: https://codereview.webrtc.org/1528503003/)

BUG=webrtc:5309
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1540243002

Cr-Commit-Position: refs/heads/master@{#11105}
2015-12-21 16:23:29 +00:00
philipel
9d3ab61325 Lint fix for webrtc/modules/video_coding PART 2!
Trying to submit all changes at once proved impossible since there were
too many changes in too many files. The changes to PRESUBMIT.py
will be uploaded in the last CL.
(original CL: https://codereview.webrtc.org/1528503003/)

BUG=webrtc:5309
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1543503002

Cr-Commit-Position: refs/heads/master@{#11102}
2015-12-21 12:12:45 +00:00
philipel
cce46fc108 Lint fix for webrtc/modules/video_coding PART 1!
Trying to submit all changes at once proved impossible since there were
too many changes in too many files. The changes to PRESUBMIT.py
will be uploaded in the last CL.
(original CL: https://codereview.webrtc.org/1528503003/)

BUG=webrtc:5309
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1541803002

Cr-Commit-Position: refs/heads/master@{#11100}
2015-12-21 11:04:57 +00:00
asapersson
53805324c0 Rename RTC_HISTOGRAM_* macros to RTC_HISTOGRAM_*_SPARSE_* to indicate that these are for infrequent updates.
This implementation will be replaced by a faster one and sparse will be removed.

BUG=webrtc:5283

Review URL: https://codereview.webrtc.org/1530913002

Cr-Commit-Position: refs/heads/master@{#11099}
2015-12-21 09:46:25 +00:00
Peter Boström
b7d9a97ce4 Expose codec implementation names in stats.
Used to distinguish between software/hardware encoders/decoders and
other implementation differences. Useful for tracking quality
regressions related to specific implementations.

BUG=webrtc:4897
R=hta@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1406903002 .

Cr-Commit-Position: refs/heads/master@{#11084}
2015-12-18 15:01:23 +00:00
Marco
002f0d09c9 VP9: Set speed setting to 8 for ARM.
At speed 8, vp9 on ARM is currently ~2x times slower than vp8 on ARM (speed -12).

Update some parameters in videoprocessor_integrationtest.cc
to make tests pass on android (which uses the new speed setting).

TBR=stefan@webrtc.org
BUG=

Review URL: https://codereview.webrtc.org/1526973004 .

Cr-Commit-Position: refs/heads/master@{#11072}
2015-12-17 17:49:39 +00:00
kwiberg
0eb15ed7b8 Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector
We can now use std::move instead!

This CL leaves the Pass methods in place; a follow-up CL will add deprecation annotations to them.

Review URL: https://codereview.webrtc.org/1460043002

Cr-Commit-Position: refs/heads/master@{#11064}
2015-12-17 11:04:24 +00:00
kjellander@webrtc.org
361888c324 OWNERS: Add * to .gyp{i,} everywhere.
Also convert DOS->Unix line endings in two of the OWNERS files.

NOTRY=True
NOPRESUBMIT=True
R=niklas.enbom@webrtc.org

Review URL: https://codereview.webrtc.org/1530003003 .

Cr-Commit-Position: refs/heads/master@{#11056}
2015-12-16 19:44:39 +00:00
perkj
796cfaf7f7 Add VideoCodec::PreferDecodeLate
The purpose is so that a decoder (Android) that only have a limited number of output buffers can make sure that decoding is done just before the frame is needed.

Removed unused iSupportsRenderTiming and the settings structs since it was not used.
Added VCMReceiver::FrameForDecoding unit test for the case when PreferDecodeLate is set.

Note that this does not change the current behaviour. We actually currently always decode frames late. This cl is to make sure the behaviour is kept for Android, if the default behaviour is changed.

Review URL: https://codereview.webrtc.org/1428293003

Cr-Commit-Position: refs/heads/master@{#10974}
2015-12-10 17:27:45 +00:00
Åsa Persson
ff24c04c73 Add tests for vp9 (non-flexible mode) using different spatial and temporal configurations.
Specify kf_min_dist to get correct key frame interval in svc mode.

Also set QP-max/min per temporal and spatial layer (was previously only allowed to be set per spatial layer).

BUG=chromium:500602
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1492633005 .

Cr-Commit-Position: refs/heads/master@{#10890}
2015-12-04 09:58:23 +00:00