307 Commits

Author SHA1 Message Date
asapersson
118ef00594 Add histogram stats for average QP per frame for VP8 (for sent video streams):
- "WebRTC.Video.Encoded.Qp.Vp8"
- "WebRTC.Video.Encoded.Qp.Vp8.S0"
- "WebRTC.Video.Encoded.Qp.Vp8.S1"
- "WebRTC.Video.Encoded.Qp.Vp8.S2"

BUG=

Review URL: https://codereview.webrtc.org/1523293002

Cr-Commit-Position: refs/heads/master@{#12174}
2016-03-31 07:00:25 +00:00
Peter Boström
d53c389550 Shorten single-stream VP8 HW implementation names.
Removes "SimulcastEncoderAdapter" from single-stream HW VP8 even though
they are wrapped in a SimulcastEncoderAdapter.

BUG=
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1827553002 .

Cr-Commit-Position: refs/heads/master@{#12161}
2016-03-30 15:04:04 +00:00
asapersson
58d992e025 Add macros for ability to log samples that are added to histograms (RTC_LOGGED_*).
Adds logging of:
- video stats that are recorded when a stream is removed
- bitrate stats that are recorded at the end of a call
- initial bwe rampup stats

BUG=

Review URL: https://codereview.webrtc.org/1788783002

Cr-Commit-Position: refs/heads/master@{#12133}
2016-03-29 09:15:11 +00:00
kjellander@webrtc.org
94a23f04af Reland "Add check_deps rules in DEPS files."
Relanding https://codereview.webrtc.org/1796413002/
without the change to the openmax_dl include path
(which broke downstream code).

TBR=tommi@webrtc.org
BUG=webrtc:5623
TESTED=Passing runs using:
buildtools/checkdeps/checkdeps.py --root=. talk
buildtools/checkdeps/checkdeps.py --root=. webrtc

Review URL: https://codereview.webrtc.org/1804333002 .

Cr-Commit-Position: refs/heads/master@{#12031}
2016-03-17 11:05:50 +00:00
kjellander
56cf60e717 Revert of Add check_deps rules in DEPS files. (patchset #2 id:60001 of https://codereview.webrtc.org/1796413002/ )
Reason for revert:
The openmax_dl include change breaks downstream projects.

Original issue's description:
> Add check_deps rules in DEPS files.
>
> Add fine-grained check_deps rules for all of WebRTC.
> This will help both maintaining sane dependencies and provides a way
> to visualize dependency graphs using the buildtools/checkdeps/graphdeps.py script.
>
> Example:
> buildtools/checkdeps/graphdeps.py --root=. --format=png \
> --out=./webrtc.png --incl='^webrtc/modules/bitrate_controller->' \
> --excl='chromium|base|external|testing|webrtc/test|\.h$|\.cc$'
>
> will produce a neat webrtc.png image showcasing the dependencies
> (according to the DEPS file) for the bitrate_controller module.
> Some dependencies are filtered out for readability.
>
> BUG=webrtc:5623
> TESTED=Passing runs using:
> buildtools/checkdeps/checkdeps.py --root=. talk
> buildtools/checkdeps/checkdeps.py --root=. webrtc
>
> R=tommi@webrtc.org
>
> Committed: https://crrev.com/086f851b7b9b4bcbd4fe507c3bf83b760bd7f4d9
> Cr-Commit-Position: refs/heads/master@{#12008}

TBR=tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5623

Review URL: https://codereview.webrtc.org/1808573002

Cr-Commit-Position: refs/heads/master@{#12009}
2016-03-16 00:41:04 +00:00
kjellander@webrtc.org
086f851b7b Add check_deps rules in DEPS files.
Add fine-grained check_deps rules for all of WebRTC.
This will help both maintaining sane dependencies and provides a way
to visualize dependency graphs using the buildtools/checkdeps/graphdeps.py script.

Example:
buildtools/checkdeps/graphdeps.py --root=. --format=png \
--out=./webrtc.png --incl='^webrtc/modules/bitrate_controller->' \
--excl='chromium|base|external|testing|webrtc/test|\.h$|\.cc$'

will produce a neat webrtc.png image showcasing the dependencies
(according to the DEPS file) for the bitrate_controller module.
Some dependencies are filtered out for readability.

BUG=webrtc:5623
TESTED=Passing runs using:
buildtools/checkdeps/checkdeps.py --root=. talk
buildtools/checkdeps/checkdeps.py --root=. webrtc

R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1796413002 .

Cr-Commit-Position: refs/heads/master@{#12008}
2016-03-16 00:22:53 +00:00
kjellander
e26e78784b Roll chromium_revision ee31124..508edd3 (378158:379249)
This includes renaming back libvpx_new to libvpx in
https://codereview.chromium.org/1765703002

Add symlink to src/mojo as workaround while figuring out how to fix
this upstream in Chromium. See webrtc:5629.

Change log: ee31124..508edd3
Full diff: ee31124..508edd3

Changed dependencies:
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/6d49157..708db16
* src/third_party/libvpx_new/source/libvpx: https://chromium.googlesource.com/webm/libvpx.git/+log/89cc682..None
* src/tools/swarming_client: https://chromium.googlesource.com/external/swarming.client.git/+log/a72f46e..df6e95e
DEPS diff: https://chromium.googlesource.com/chromium/src/+/ee31124..508edd3/DEPS

No update to Clang.

BUG=webrtc:5629
TBR=marpan@webrtc.org, stefan@webrtc.org,
NOTRY=True

Review URL: https://codereview.webrtc.org/1766643002

Cr-Commit-Position: refs/heads/master@{#11879}
2016-03-04 22:39:32 +00:00
sprang
b0fdfea9e8 Add stats (histograms) for vp8 screenshare layers
BUG=

Review URL: https://codereview.webrtc.org/1734793003

Cr-Commit-Position: refs/heads/master@{#11830}
2016-03-01 13:51:20 +00:00
kwiberg
3f55dea259 Replace scoped_ptr with unique_ptr in webrtc/modules/video_coding/
BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1721353002

Cr-Commit-Position: refs/heads/master@{#11814}
2016-02-29 13:52:06 +00:00
Alex Glaznev
a9d0892946 Add initial bitrate and frame resolution parameters to quality scaler.
- Scale down to VGA immediately if call starts with HD resolution
and bitrate below 500 kbps.
- Adjust QP threshold for HW VP8 encoder to scale down faster.

BUG=b/26504665
R=mflodman@webrtc.org, pbos@webrtc.org, sprang@google.com, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1672173002 .

Cr-Commit-Position: refs/heads/master@{#11692}
2016-02-19 23:24:12 +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
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
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
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
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
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
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
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
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
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
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
kjellander@webrtc.org
b7ce96470b modules/video_coding/utility: Remove include
This makes it clearer this code not meant to be used as an API.
I could not find any use of this in downstream code.

BUG=webrtc:5095
TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
R=stefan@webrtc.org
TBR=magjed@webrtc.org

Review URL: https://codereview.webrtc.org/1440873005 .

Cr-Commit-Position: refs/heads/master@{#10699}
2015-11-18 22:04:20 +00:00
Henrik Kjellander
2557b86e76 modules/video_coding refactorings
The main purpose was the interface-> include rename, but other files
were also moved, eliminating the "main" dir.

To avoid breaking downstream, the "interface" directories were copied
into a new "video_coding/include" dir. The old headers got pragma
warnings added about deprecation (a very short deprecation since I plan
to remove them as soon downstream is updated).

Other files also moved:
video_coding/main/source -> video_coding
video_coding/main/test -> video_coding/test

BUG=webrtc:5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc

R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1417283007 .

Cr-Commit-Position: refs/heads/master@{#10694}
2015-11-18 21:00:33 +00:00
pbos
d9eec762ce Trace encoding/decoding time in a generic way.
Removes VP8::Encode trace in favor of VCMGenericEncoder ones and adds
one to InitEncode. Also adds an instant event to ::Encoded since this
can be done on a different thread.

Also adds the corresponding traces to VCMGenericDecoder.

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

Review URL: https://codereview.webrtc.org/1412573010

Cr-Commit-Position: refs/heads/master@{#10674}
2015-11-17 14:03:52 +00:00
kjellander
6f8ce060a2 common_video: rename interface -> include
To avoid breaking downstream, the "interface" directories were copied
into a new "common_video/include" dir. The old headers got pragma
warnings added about deprecation (a very short deprecation since I plan
to remove them as soon downstream is updated).
The header guards are also identical to avoid mixing them up in the transition.

BUG=webrtc:5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc

Review URL: https://codereview.webrtc.org/1418913006

Cr-Commit-Position: refs/heads/master@{#10659}
2015-11-16 21:52:31 +00:00
Per
327d8babc8 Add DecodedImageCallback::Decoded() function with custom decode time value.
On Android, we would like to use MediaCodec output buffers to hold decoded frames until they can be rendered to a texture. There can only be one texture buffer used at the same time and therefore the calculated decode time in VCMTiming will be wrong since that calculation will also include the time where the decoder waited for the upper layers (that depend on network jitter and actual render time) to release the frame.

This new method will be used in
https://codereview.webrtc.org/1422963003/

BUG=webrtc:4993
R=stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1414693006 .

Cr-Commit-Position: refs/heads/master@{#10576}
2015-11-10 13:00:45 +00:00
Henrik Kjellander
ff761fba82 modules: more interface -> include renames
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface

To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
                "use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)

BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc

R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1417683006 .

Cr-Commit-Position: refs/heads/master@{#10500}
2015-11-04 07:32:04 +00:00
Henrik Kjellander
98f53510b2 system_wrappers: rename interface -> include
BUG=webrtc:5095
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1413333002 .

Cr-Commit-Position: refs/heads/master@{#10438}
2015-10-28 17:17:50 +00:00
Peter Boström
49e196af40 Remove VideoFrameType aliases for FrameType.
No longer used in Chromium, so these can now be removed.

BUG=webrtc:5042
R=mflodman@webrtc.org
TBR=magjed@webrtc.org

Review URL: https://codereview.webrtc.org/1415693002 .

Cr-Commit-Position: refs/heads/master@{#10390}
2015-10-23 13:58:27 +00:00
tommi
e4f96501fc Remove system_wrappers/interface/trace_event.h
BUG=

Review URL: https://codereview.webrtc.org/1417773002

Cr-Commit-Position: refs/heads/master@{#10346}
2015-10-21 06:00:57 +00:00
asapersson
da535c4055 Add histogram for percentage of sent frames that are limited in resolution due to bandwidth:
- "WebRTC.Video.BandwidthLimitedResolutionInPercent"

If the frame is bandwidth limited, the average number of disabled resolutions is logged:
- "WebRTC.Video.BandwidthLimitedResolutionsDisabled"

BUG=

Review URL: https://codereview.webrtc.org/1311533012

Cr-Commit-Position: refs/heads/master@{#10333}
2015-10-20 06:32:48 +00:00
pbos
22993e1a0c Unify FrameType and VideoFrameType.
Prevents some heap allocation and frame-type conversion since interfaces
mismatch. Also it's less confusing to have one type for this.

BUG=webrtc:5042
R=magjed@webrtc.org, mflodman@webrtc.org, henrik.lundin@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1371043003

Cr-Commit-Position: refs/heads/master@{#10320}
2015-10-19 09:39:15 +00:00
asapersson
4306fc70d7 Add histogram for percentage of sent frames that are limited in resolution due to quality:
- "WebRTC.Video.QualityLimitedResolutionInPercent"

and if a frame is downscaled, the average number of times the frame is downscaled:
- "WebRTC.Video.QualityLimitedResolutionDownscales"

BUG=

Review URL: https://codereview.webrtc.org/1325153009

Cr-Commit-Position: refs/heads/master@{#10319}
2015-10-19 07:35:27 +00:00
pbos
65e15bafaa Add native-handle support for single VP8 streams.
Implements SupportsNativeHandle() in SimulcastEncoderAdapter which works
when there's only a single encoder.

BUG=webrtc:5060
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1397653004

Cr-Commit-Position: refs/heads/master@{#10291}
2015-10-15 17:52:21 +00:00
Peter Boström
0c4e06b4c6 Use suffixed {uint,int}{8,16,32,64}_t types.
Removes the use of uint8, etc. in favor of uint8_t.

BUG=webrtc:5024
R=henrik.lundin@webrtc.org, henrikg@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/1362503003 .

Cr-Commit-Position: refs/heads/master@{#10196}
2015-10-07 10:23:32 +00:00
Peter Boström
5d0379da2c Remove kSkipFrame usage.
Since padding is no longer sent on Encoded() callbacks, dummy callbacks
aren't required to generate padding. This skip-frame behavior can then
be removed to get rid of dummy callbacks though nothing was encoded. As
frames don't have to be generated for frames that don't have to be sent
we skip encoding frames that aren't intended to be sent either, reducing
CPU load.

BUG=
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1369923005 .

Cr-Commit-Position: refs/heads/master@{#10181}
2015-10-06 12:05:03 +00:00
sprang
fb30c1b5d1 Update VP8 settings to avoid spending bitrate on static areas. PERF NOTE
This CL changes the threshold where we consider a block to be static and
of sufficient quality to not spend bits/CPU encoding it.

Perf note: This change may result in a minor degradation of PSNR/SSIM
and available send bitrate. CPU usage and bitrate sent should however
be greately reduced.

BUG=webrtc:5015

Review URL: https://codereview.webrtc.org/1383533002

Cr-Commit-Position: refs/heads/master@{#10134}
2015-10-01 13:26:16 +00:00
kjellander
d6024e3c34 Roll chromium_revision 310ea93..8cf53d6 (349094:351112)
Our perf test suite webrtc_perf_tests timed out, which caused most
of the delay landing this (https://crbug.comn/535973 and
https://codereview.chromium.org/1370133004).

Other problems with executing Android tests also needed to be
resolved in order to land this (http://crbug.com/534849).

Libvpx has moved from third_party/libvpx to third_party/libvpx_new
as of https://codereview.chromium.org/1323333002/

Android GN was blocking this roll due to a problem that ended up
being caused by a bug (http://crbug.com/534849).

Relevant changes:
* src/buildtools: f7310ee..8d89c1b
* src/third_party/boringssl/src: 1d128f3..4c60d35
* src/third_party/icu: 6b3ce81..423fc7e
* src/third_party/libjpeg_turbo: 631e2dd..e4e7503
* src/third_party/libvpx: ac1772e..70db223
* src/third_party/libyuv: fcacbfb..62c49dc
* src/tools/gyp: 5d01a8c..01528c7
* src/tools/swarming_client: 77f720b..6e5d2b2
Details: 310ea93..8cf53d6/DEPS

Clang version changed 245965:247874
Details: 310ea93..8cf53d6/tools/clang/scripts/update.sh

BUG=481034, 535973
TBR=marpan@webrtc.org

Review URL: https://codereview.webrtc.org/1355083002

Cr-Commit-Position: refs/heads/master@{#10101}
2015-09-29 04:16:53 +00:00
Peter Boström
1741770742 Implement a high-QP threshold for Android H.264.
Android hardware H.264 seems to keep a steady high-QP flow instead of
dropping frames, so framedrops aren't sufficient to detect a bad state
where downscaling would be beneficial.

BUG=webrtc:4968
R=magjed@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1364253002 .

Cr-Commit-Position: refs/heads/master@{#10078}
2015-09-25 15:03:37 +00:00
henrikg
91d6edef35 Add RTC_ prefix to (D)CHECKs and related macros.
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.

Alternative solutions:
* Check if we already have defined e.g. CHECK, and don't define them in that case. This makes us depend on include order in Chromium, which is not acceptable.
* Don't allow using the macros in WebRTC headers. Error prone since if someone adds it there by mistake it may compile fine, but later break if a header in added or order is changed in Chromium. That will be confusing and hard to enforce.
* Ensure that headers that are included by an embedder don't include our macros. This would require some heavy refactoring to be maintainable and enforcable.
* Changes in Chromium for this is obviously not an option.

BUG=chromium:468375
NOTRY=true

Review URL: https://codereview.webrtc.org/1335923002

Cr-Commit-Position: refs/heads/master@{#9964}
2015-09-17 07:24:51 +00:00
Asa Persson
110443c1ec Fix for frame resolution in encoded frame callback.
Scaled resolution for down scaled frames by the quality scaler is not used.

BUG=webrtc:4966
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1317463005 .

Cr-Commit-Position: refs/heads/master@{#9873}
2015-09-07 13:04:00 +00:00
sprang
ef7228cfa0 Selectable number of TL screenshare loopback test. Also contains some tweaks to make a single TL perform better.
BUG=

Review URL: https://codereview.webrtc.org/1242043002

Cr-Commit-Position: refs/heads/master@{#9676}
2015-08-05 09:02:09 +00:00
pbos
ef35f069e7 Remove webrtc::Config from ViEChannelGroup.
Also removing webrtc/experiments.h which is no longer used.

BUG=webrtc:1695
R=stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1250513006

Cr-Commit-Position: refs/heads/master@{#9642}
2015-07-27 15:37:14 +00:00
jackychen
6e2ce6e1ae Allow for framerate reduction for HW encoder.
R=pbos@webrtc.org, stefan@webrtc.org
TBR=glaznev@google.com

Review URL: https://webrtc-codereview.appspot.com/51159004 .

Cr-Commit-Position: refs/heads/master@{#9573}
2015-07-13 23:26:40 +00:00
Erik Språng
2c4c914819 In screenshare mode, suppress VP8 bitrate overshoot and increase quality
This change includes several improvements:

* VP8 configured with new rate control
* Detection of frame dropping, with qp bump for next frame
* Increased target and TL0 bitrates
* Reworked rate control (TL allocation) in screenshare_layers

A note on performance: PSNR and SSIM is expected to get slightly worse with this cl. Frame drops and delays should however improve.

BUG=4171
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1193513006.

Cr-Commit-Position: refs/heads/master@{#9495}
2015-06-24 09:24:50 +00:00
Peter Boström
6a688f5265 Add default downscale threshold to QualityScaler.
Prevents downscaling below 160x90 or 90x160 to gain more quality.

BUG=4625
R=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1160403004.

Cr-Commit-Position: refs/heads/master@{#9480}
2015-06-22 06:03:07 +00:00
Peter Boström
1b9add9df9 Prevent bitrate overshoot for HD layer in VP8.
BUG=chromium:487648
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/55469004

Cr-Commit-Position: refs/heads/master@{#9394}
2015-06-08 20:52:42 +00:00