Commit Graph

9 Commits

Author SHA1 Message Date
a669a3a0dc Revert "Revert of Use sps and pps to determine decodability of H.264 frames. (patchset #4 id:60001 of https://codereview.webrtc.org/2341713002/ )"
This reverts commit 3cdfcd88a14449a9b116cb6149e1348d3a1e4cb2.

NOPRESUBMIT=true
BUG=webrtc:6208

Review-Url: https://codereview.webrtc.org/2385143002
Cr-Commit-Position: refs/heads/master@{#14551}
2016-10-06 12:04:59 +00:00
3cdfcd88a1 Revert of Use sps and pps to determine decodability of H.264 frames. (patchset #4 id:60001 of https://codereview.webrtc.org/2341713002/ )
Reason for revert:
Broke browser_tests, e.g., WebRtcVideoQualityBrowserTests/WebRtcVideoQualityBrowserTest.MANUAL_TestVideoQualityH264

Original issue's description:
> Use sps and pps to determine decodability of H.264 frames.
>
> NOPRESUBMIT=true
> BUG=webrtc:6208
> R=philipel@webrtc.org
>
> Committed: https://crrev.com/17b02633666f2f5d7e78767ad5674c728d639c26
> Cr-Commit-Position: refs/heads/master@{#14458}

TBR=philipel@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6208

Review-Url: https://codereview.webrtc.org/2381233004
Cr-Commit-Position: refs/heads/master@{#14460}
2016-09-30 16:06:43 +00:00
17b0263366 Use sps and pps to determine decodability of H.264 frames.
NOPRESUBMIT=true
BUG=webrtc:6208
R=philipel@webrtc.org

Review URL: https://codereview.webrtc.org/2341713002 .

Cr-Commit-Position: refs/heads/master@{#14458}
2016-09-30 13:24:26 +00:00
6f112cc136 Delete unused support for vp8 partitions.
This also makes it possible to drop the RTPFragmentationHeader from
the class VCMEncodedFrame.

BUG=None

Review-Url: https://codereview.webrtc.org/2380933003
Cr-Commit-Position: refs/heads/master@{#14455}
2016-09-30 10:43:07 +00:00
414dda1a10 Change VCMFrameBuffer and VCMEncodedFrame to use rotation from base class EncodedImage.
BUG=webrtc:5687

Review-Url: https://codereview.webrtc.org/2037633002
Cr-Commit-Position: refs/heads/master@{#13376}
2016-07-04 08:45:28 +00:00
6b4b5f3770 Add sender controlled playout delay limits
This CL adds support for an extension on RTP frames to allow the sender
to specify the minimum and maximum playout delay limits.

The receiver makes a best-effort attempt to keep the capture-to-render delay
within this range. This allows different types of application to specify
different end-to-end delay goals. For example gaming can support rendering
of frames as soon as received on receiver to minimize delay. A movie playback
application can specify a minimum playout delay to allow fixed buffering
in presence of network jitter.

There are no tests at this time and most of testing is done with chromium
webrtc prototype.

On chromoting performance tests, this extension helps bring down end-to-end
delay by about 150 ms on small frames.

BUG=webrtc:5895

Review-Url: https://codereview.webrtc.org/2007743003
Cr-Commit-Position: refs/heads/master@{#13059}
2016-06-08 07:24:30 +00:00
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
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
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