cce46fc108a70336f0477fd58d41f38e547eeb25 |
|
21-Dec-2015 |
philipel <philipel@webrtc.org> |
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}
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
b7d9a97ce41022e984348efb5f28bf6dd6c6b779 |
|
18-Dec-2015 |
Peter Boström <pbos@webrtc.org> |
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}
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
2557b86e7648ffebc5781df9f093ca5a84efc219 |
|
18-Nov-2015 |
Henrik Kjellander <kjellander@google.com> |
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}
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
d9eec762ce8bdc90bd18a387cacae8cab90440f1 |
|
17-Nov-2015 |
pbos <pbos@webrtc.org> |
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}
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
ff761fba8274d93bd73e76c8b8a1f2d0776dd840 |
|
04-Nov-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
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}
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
98f53510b222f71fdd8b799b2f33737ceeb28c61 |
|
28-Oct-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
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}
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
49e196af4060624d620297a6bc017699daa33550 |
|
23-Oct-2015 |
Peter Boström <pbos@webrtc.org> |
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}
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
e4f96501fc5b3e6de0d1ccd262372afcda1f5b4f |
|
21-Oct-2015 |
tommi <tommi@webrtc.org> |
Remove system_wrappers/interface/trace_event.h BUG= Review URL: https://codereview.webrtc.org/1417773002 Cr-Commit-Position: refs/heads/master@{#10346}
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
da535c405597864b8396b2029dec70ab9fb76e8b |
|
20-Oct-2015 |
asapersson <asapersson@webrtc.org> |
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}
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
22993e1a0c114122fc1b9de0fc74d4096ec868bd |
|
19-Oct-2015 |
pbos <pbos@webrtc.org> |
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}
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
4306fc70d778887d8a2ea71b6f4bc6a12d1d9447 |
|
19-Oct-2015 |
asapersson <asapersson@webrtc.org> |
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}
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
5d0379da2cbdcce6f8494209c7ab559cd6de076e |
|
06-Oct-2015 |
Peter Boström <pbos@webrtc.org> |
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}
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
fb30c1b5d1effcc82a96fdf40814a03baf0727bf |
|
01-Oct-2015 |
sprang <sprang@webrtc.org> |
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}
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
17417707428816ffb88d9c71dcc8a5d492cf9fdf |
|
25-Sep-2015 |
Peter Boström <pbos@webrtc.org> |
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}
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
91d6edef35e7275879c30ce16ecb8b6dc73c6e4a |
|
17-Sep-2015 |
henrikg <henrikg@webrtc.org> |
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}
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
110443c1ec40b764db81a6b9679885aec3e4a218 |
|
07-Sep-2015 |
Asa Persson <asapersson@chromium.org> |
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}
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
ef35f069e739feaae16fdfcc815d7af5cb05e9ae |
|
27-Jul-2015 |
pbos <pbos@webrtc.org> |
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}
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
6e2ce6e1ae41d8eeb0f233cbd26087daa03ab702 |
|
14-Jul-2015 |
jackychen <jackychen@webrtc.org> |
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}
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
2c4c9148191a10c0e82c9a209d454c6b1ebbaf20 |
|
24-Jun-2015 |
Erik Språng <sprang@webrtc.org> |
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}
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
6a688f5265040c64e49dd16244ca5bac03327453 |
|
22-Jun-2015 |
Peter Boström <pbos@webrtc.org> |
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}
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
1b9add9df9cdab004a11e75900a341a346ce2049 |
|
08-Jun-2015 |
Peter Boström <pbos@webrtc.org> |
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}
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
4765070b8d6f024509c717c04d9b708750666927 |
|
30-May-2015 |
Miguel Casas-Sanchez <mcasas@webrtc.org> |
Rename I420VideoFrame to VideoFrame. This is a mechanical change since it affects so many files. I420VideoFrame -> VideoFrame and reformatted. Rationale: in the next CL I420VideoFrame will get an indication of Pixel Format (I420 for starters) and of storage type: usually UNOWNED, could be SHMEM, and in the near future will be possibly TEXTURE. See https://codereview.chromium.org/1154153003 for the change that happened in Cr. BUG=4730, chromium:440843 R=jiayl@webrtc.org, niklas.enbom@webrtc.org, pthatcher@webrtc.org Review URL: https://webrtc-codereview.appspot.com/52629004 Cr-Commit-Position: refs/heads/master@{#9339}
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
e87d48719f8478148fc6ec28c3b3663709d585a4 |
|
26-May-2015 |
Stefan Holmer <stefan@webrtc.org> |
Fix ARM64 detection for VP8 and VP9 wrappers. BUG=4702 R=marpan@google.com, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/57459004 Cr-Commit-Position: refs/heads/master@{#9287}
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
98d8cf58ee8c7c7b0672ce7955313a31824d6f3a |
|
21-May-2015 |
jackychen <jackychen@webrtc.org> |
Hardware VP8 encoding: Use QP as metric for resize. Add vp8 frame header parser to get QP from vp8 bitstream. BUG= 4273 R=glaznev@webrtc.org, marpan@google.com, pbos@webrtc.org TBR=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/49259004 Cr-Commit-Position: refs/heads/master@{#9256}
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
b302ad4eab114eb76c3d4531bd92ae8a4e454761 |
|
21-May-2015 |
Peter Boström <pbos@webrtc.org> |
Remove unused VideoDecoder methods. Removing VideoDecoder::Copy() and VideoDecoder::SetCodecConfigParameters(). Also adding override to VP8DecoderImpl. BUG= R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/55409004 Cr-Commit-Position: refs/heads/master@{#9244}
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
143cec1cc68b9ba44f3ef4467f1422704f2395f0 |
|
28-Apr-2015 |
Erik Språng <sprang@google.com> |
Set correct encoder-specific settings for vpx in the new API. Also, make VideoEncoderConfig::ContentType an enum class. BUG=4569 R=mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/46069004 Cr-Commit-Position: refs/heads/master@{#9093}
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
61b4d518af7c2e4156056931d3512a49032b827d |
|
22-Apr-2015 |
jackychen <jackychen@webrtc.org> |
Dynamic resolution change for VP8 HW encode. Off by default for now. BUG= R=glaznev@webrtc.org, stefan@webrtc.org TBR=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/45849004 Cr-Commit-Position: refs/heads/master@{#9045}
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
a3209a2b27b7bf2059f8119a126a1b1be9f0377f |
|
20-Mar-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Release buffer pool in Vp8DecoderImpl::Release(). Permits reusing an external VP8DecoderImpl instance from another VideoReceiveStream without a thread-checker DCHECK blowing up. Also releases buffers that would've been kept in memory even though the decoder isn't configured. BUG= R=magjed@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50449004 Cr-Commit-Position: refs/heads/master@{#8807} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8807 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
e155dbeae972f788b8766b4d0d2cefde2e952a10 |
|
17-Mar-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
VP8/9EncoderImpl::Encode: Check resolution of input I420VideoFrame This CL adds checks in Encode to guard against memory reads out of bounds. R=pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/46429008 Cr-Commit-Position: refs/heads/master@{#8750} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8750 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
73d763e71f5c77cac82aa65d737e64d893f190a0 |
|
17-Mar-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
Add I420 buffer pool to avoid unnecessary allocations Now when we don't use SwapFrame consistently anymore, we need to recycle allocations with a buffer pool instead. This CL adds a buffer pool class, and updates the vp8 decoder to use it. If this CL lands successfully I will update the other video producers as well. BUG=1128 R=stefan@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/41189004 Cr-Commit-Position: refs/heads/master@{#8748} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8748 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
2056ee3e3c7683ae4b2c4b12da99c3105c4f46a9 |
|
16-Mar-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
Revert "Changed argument occurences of const I420VideoFrame* to const I420VideoFrame& and non-const I420VideoFrame& to I420VideoFrame*." This reverts commit r8731. Reason for revert: Breakes Chromium FYI bots. TBR=hbos, tommi Review URL: https://webrtc-codereview.appspot.com/40359004 Cr-Commit-Position: refs/heads/master@{#8733} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8733 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
93d9d6503e2bf2526af2b1c2cc46ef242b9843aa |
|
16-Mar-2015 |
hbos@webrtc.org <hbos@webrtc.org> |
I420VideoFrame.CreateFrame: Removed unnecessary buffer size arguments. R=magjed@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/45629004 Cr-Commit-Position: refs/heads/master@{#8732} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8732 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
2dc5fa69b2baef2ece158c9e1285516087faaa53 |
|
16-Mar-2015 |
hbos@webrtc.org <hbos@webrtc.org> |
Changed argument occurences of const I420VideoFrame* to const I420VideoFrame& and non-const I420VideoFrame& to I420VideoFrame*. R=magjed@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/40299004 Cr-Commit-Position: refs/heads/master@{#8731} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8731 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
6daacbc8aea2e3e14a320b28502c4dc1db21d175 |
|
04-Mar-2015 |
marpan@webrtc.org <marpan@webrtc.org> |
Set cpu_speed parameter for low resolutions, for non-simulcast. Allow for setting different cpu_speed setting based on resolution, for non-simulcast. Use the existing low resolution simulcast cpu_speed setting for the non-simulcast case. No change to simulcast behavior, unless top/highest layer stream is also below CIF resolution, (in which case all layers will use lower the cpu_speed setting =-4). BUG= R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/37319004 Cr-Commit-Position: refs/heads/master@{#8603} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8603 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
16825b1a828bb4ff40f7682040e43a239b7b8ca3 |
|
12-Jan-2015 |
pkasting@chromium.org <pkasting@chromium.org> |
Use int64_t more consistently for times, in particular for RTT values. Existing code was inconsistent about whether to use uint16_t, int, unsigned int, or uint32_t, and sometimes silently truncated one to another, or truncated int64_t. Because most core time-handling functions use int64_t, being consistent about using int64_t unless otherwise necessary minimizes the number of explicit or implicit casts. BUG=chromium:81439 TEST=none R=henrik.lundin@webrtc.org, holmer@google.com, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31349004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@8045 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
9115cde6c9c8f5d749b349d7d10a570e4cb32803 |
|
09-Dec-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Merge VP8 changes. R=stefan@webrtc.org BUG= Review URL: https://webrtc-codereview.appspot.com/35389004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7841 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
4591fbd09f9cb6e83433c49a12dd8524c2806502 |
|
20-Nov-2014 |
pkasting@chromium.org <pkasting@chromium.org> |
Use size_t more consistently for packet/payload lengths. See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information. This CL was reviewed and approved in pieces in the following CLs: https://webrtc-codereview.appspot.com/24209004/ https://webrtc-codereview.appspot.com/24229004/ https://webrtc-codereview.appspot.com/24259004/ https://webrtc-codereview.appspot.com/25109004/ https://webrtc-codereview.appspot.com/26099004/ https://webrtc-codereview.appspot.com/27069004/ https://webrtc-codereview.appspot.com/27969004/ https://webrtc-codereview.appspot.com/27989004/ https://webrtc-codereview.appspot.com/29009004/ https://webrtc-codereview.appspot.com/30929004/ https://webrtc-codereview.appspot.com/30939004/ https://webrtc-codereview.appspot.com/31999004/ Committing as TBR to the original reviewers. BUG=chromium:81439 TEST=none TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom Review URL: https://webrtc-codereview.appspot.com/23129004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
94ff92ceec7a2fe3e1d0e244a06e71e7dcc28256 |
|
23-Sep-2014 |
johannkoenig@google.com <johannkoenig@google.com> |
Use VPX_IMG_FMT_*/VPX_PLANE_* defines The compatibility layer has been removed upstream: https://gerrit.chromium.org/gerrit/gitweb?p=webm%2Flibvpx.git;a=commit;h=9cdaa3d72eade9ad162ef8f78a93bd8f85c6de10 BUG=webrtc:3839 R=marpan@google.com, marpan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/22729004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7279 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
a0d7827b1631fa8e2cf8957a76736ab159cb962a |
|
12-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Add ability to downscale content to improve quality. BUG=3712 R=marpan@google.com, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/18169004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7164 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
9c09e6ee2bc8af49af649a9ab107eb2201d47272 |
|
17-Jun-2014 |
niklas.enbom@webrtc.org <niklas.enbom@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add high perf mode to VP8 R=marpan@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/14709004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6470 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
6c75c989648018a1537bb29d41fbcb730b143c15 |
|
15-Apr-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Propagate capture ntp timestamp from rtp to renderer. Mostly the interface changes, the real implementation of ntp timestamp will come in a follow up cl. TEST=new tests and try bots BUG=3111 R=niklas.enbom@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/11469004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5911 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
1faef7d0843b386d070e2840565e774f251a6b71 |
|
20-Mar-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Use codec width/height as the encoded_image width/height. The raw_->w and raw_->h which are the stored image width/height may not be the encoded image size in the case when the incoming frame has a odd size. R=marpan@google.com, marpan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/10289004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5739 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
7f52a6ef2b5579f05e6d66a77f4dba8c1c0e8985 |
|
25-Feb-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Split the implementation of VP8Encoder|Decoder::Create into a seperated file (vp8_factory.cc). R=fischman@webrtc.org, marpan@google.com, marpan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8589004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5606 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
5500d93fe55702af0b254e5718fcc085298c99ab |
|
06-Sep-2013 |
andresp@webrtc.org <andresp@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add temporal layer factory. R=marpan@google.com, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2180004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4691 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
b2c28c3699f701e2e3e51ea1d361a2f66cf5f4e2 |
|
23-Aug-2013 |
mikhal@webrtc.org <mikhal@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Relanding 4597 - Don't force key frame when decoding with errors. Makes sure that incomplete key frame or delta frames will be released from the JB when decoding with errors. The decoder in turn will trigger a PLI until a complete key frame is received in order to start a session. TBR=stefan@webrtc.org BUG= Review URL: https://webrtc-codereview.appspot.com/2097004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4607 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
ceea41d135d320e185b36957f30b6fe352af4fbf |
|
23-Aug-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 4597 "Don't force key frame when decoding with errors" > Don't force key frame when decoding with errors > > BUG=2241 > R=stefan@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/2036004 TBR=mikhal@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2093004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4600 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
44af55cc44bf8764cd6912dae0c8398cddcb1b1d |
|
23-Aug-2013 |
mikhal@webrtc.org <mikhal@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Don't force key frame when decoding with errors BUG=2241 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2036004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4597 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
d2102afa2a8c61aec350edff0351fe9261284543 |
|
17-Jul-2013 |
tnakamura@webrtc.org <tnakamura@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Undo libvpx include changes in r4348 to fix build. A longer term fix is needed, but this at least quickly unblocks the build. R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1816005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4367 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
a4407329d4e16eab3d2e87cde0824188c06acb5a |
|
16-Jul-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Include files from webrtc/.. paths in video_coding/. BUG=1662 R=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1783006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4348 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
d7148c86c5b020f4add340e68185335f761743fd |
|
19-Jun-2013 |
fbarchard@google.com <fbarchard@google.com@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Use 3 threads for higher than 720p resolutions BUG=1893 TEST=untested R=ajm@google.com, andrew@webrtc.org, dingkai@google.com, marpan@google.com, marpan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1684004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4243 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
b3e5acfb663c43a34825ffda205bde29ee2aa468 |
|
16-May-2013 |
hclam@chromium.org <hclam@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Cleanup traces in WebRTC Remove some unused traces and add a trace counter for encoded video size. R=holmer@google.com, mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1476004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4050 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
1e3c7946880fa9435b3dfcace56a347f0b502711 |
|
09-May-2013 |
fbarchard@google.com <fbarchard@google.com@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Use 2 threads for HD, or 1 for VGA or less. BUG=1739 TEST=try bots Review URL: https://webrtc-codereview.appspot.com/1438005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3996 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
d25b602dc0e70282e1f0223e5909534fa81053f0 |
|
23-Apr-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
VP8: Avoid copying the codec struct on Reset(). BUG= Review URL: https://webrtc-codereview.appspot.com/1319013 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3887 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
48c5882f2a3da8c81c26eb47378c0fcea0d81046 |
|
17-Apr-2013 |
andresp@webrtc.org <andresp@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove libvpx pre-processor conditions and conditional compile of default temporal layers files. R=stefan@webrtc.org,marpan@webrtc.org BUG=201 Review URL: https://webrtc-codereview.appspot.com/1323005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3864 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
557e92515d56b2e5d8f354921e8b037349518ac2 |
|
09-Apr-2013 |
marpan@webrtc.org <marpan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Reapply the reverted r3747. https://code.google.com/p/webrtc/source/detail?r=3747 r3747 timed-out on a tsan test. Verified that it passes the test and reduced the execution time of that test (r3782). TBR=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1292006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3807 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
806dc3b0e62ec68f594e9aadab601b2db7e6c6d5 |
|
09-Apr-2013 |
hclam@chromium.org <hclam@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
More trace events The goal of this change is to unify tracing events styles and add trace events for all RTP traffic. BUG=1555 Review URL: https://webrtc-codereview.appspot.com/1290007 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3806 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
cfc07c943f4657be3cb55c59e6bb06bf38c0809d |
|
02-Apr-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert of r3747. TBR=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1277005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3752 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
747c4cc96e66aee42069f4c33fd2a11ce8bcaca1 |
|
02-Apr-2013 |
fbarchard@google.com <fbarchard@google.com@4adac7df-926f-26a2-2b94-8c16560cd09d> |
For VGA (640x360), currently 1 thread is used. This change increases it to 2 threads. For HD, 4 threads are enabled. BUG=none TEST=run a hangout and screencast high framerate, high resolution windows of youtube. Observe that 1 cpu is insufficient to maintain high framerate with complex content. Review URL: https://webrtc-codereview.appspot.com/1203006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3747 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
84cd8e39cf2a998248daa16cae7b7713f79da0d8 |
|
07-Mar-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Disable frame dropper for screenshare mode. BUG=1466 Review URL: https://webrtc-codereview.appspot.com/1170004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3629 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
eb91792cfd98a76192a855365bb785c0f581b94b |
|
18-Feb-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Refactoring temporal layers implementation and adding VideoCodecMode for easier control of codec settings. Review URL: https://webrtc-codereview.appspot.com/1105007 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3528 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
3d305c64b47a55ce728fb9675118ce5841a52080 |
|
10-Feb-2013 |
mikhal@webrtc.org <mikhal@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Updates to send side streaming mode: 1. Disabling frame-droppers from the vie encoder and not the channel. 2. Accounting for qpMax in the VP8 wrapper. Review URL: https://webrtc-codereview.appspot.com/1101007 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3492 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
e07c661a29058aac858f99c5d0701a52969dfbe4 |
|
31-Jan-2013 |
mikhal@webrtc.org <mikhal@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
VP8: Making key frame interval a tunnable parameter Review URL: https://webrtc-codereview.appspot.com/1070006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3444 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
f222a0088149f5904c2bcb2c7440c003fb04c1f1 |
|
11-Dec-2012 |
hclam@chromium.org <hclam@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Use TRACE_EVENT to track time spent in VP8 encoding Using the TRACE_EVENT macro to log VP8 encoding events. Review URL: https://webrtc-codereview.appspot.com/968011 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3264 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
fc4a7ee8079736923daf0cce6407dec41ce545c6 |
|
30-Nov-2012 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fixes chromium build bots. BUG=N/A Review URL: https://webrtc-codereview.appspot.com/971014 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3213 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
662651ac958ccc785c638f7bcf44e08f11eb95a8 |
|
29-Nov-2012 |
fbarchard@google.com <fbarchard@google.com@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Disable denoise filter for Arm, as it is not optimized enough yet. BUG=https://code.google.com/p/chrome-os-partner/issues/detail?id=16318 TEST=none Review URL: https://webrtc-codereview.appspot.com/968008 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3195 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
c09e779766957e5fa2fdd288d060718137fee0ea |
|
28-Nov-2012 |
marpan@webrtc.org <marpan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Allow for 1 layer case to be set in temporal_layers. Review URL: https://webrtc-codereview.appspot.com/971007 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3188 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
8049608226193aafcb48e946b6e9bfce87ab7aaf |
|
27-Nov-2012 |
mikhal@webrtc.org <mikhal@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
VP8 wrapper: updating raw image allocation. As we set the pointers to the data, there is no need to allocate that memory. Review URL: https://webrtc-codereview.appspot.com/964021 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3175 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
3263a7a61610c9029ef9bcec6dddf50fcacd1c27 |
|
21-Nov-2012 |
mikhal@webrtc.org <mikhal@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Setting correct stride for VP8 encoder BUG=1137 Review URL: https://webrtc-codereview.appspot.com/929024 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3140 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
9fedff7c17a8d3dc46ed5b3207220f59a22391d6 |
|
24-Oct-2012 |
mikhal@webrtc.org <mikhal@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Switching to I420VideoFrame Review URL: https://webrtc-codereview.appspot.com/922004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2983 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|
14b43beb7ce4440b30dcea31196de5b4a529cb6b |
|
22-Oct-2012 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Move src/ -> webrtc/ TBR=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/915006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
|