Commit graph

28 commits

Author SHA1 Message Date
Ilya Nikolaevskiy
f5d877847f Reland "[VP9] Shift spatial layers on RTP level to always start from 0."
This reverts commit 2181228624.

Reason for revert: Reland without changes as it's not the root cause.

Original change's description:
> Revert "[VP9] Shift spatial layers on RTP level to always start from 0."
> 
> This reverts commit 2e73a3d1e9.
> 
> Reason for revert: Fuzzer found some issues.
> 
> Original change's description:
> > [VP9] Shift spatial layers on RTP level to always start from 0.
> > 
> > This CL uses |width| and |height| in RTPVideoHeaderVP9 to pass information
> > about enabled layers from encoder to packetizer.
> > 
> > Bug: webrtc:11319
> > Change-Id: Idc1c337f8dfb3f7631506acb784d2a634b41b955
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167724
> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30428}
> 
> TBR=danilchap@webrtc.org,ilnik@webrtc.org,nisse@webrtc.org
> 
> # Not skipping CQ checks because original CL landed > 1 day ago.
> 
> Bug: webrtc:11319
> Change-Id: I27a7e82737fa604b8ab769ce6503fa93e46f4e86
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168123
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30448}

TBR=danilchap@webrtc.org,ilnik@webrtc.org,nisse@webrtc.org

Change-Id: Ibcd9b6a075ee08c9402de8b0b9d99d77bf59d0ef
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11319
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168185
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30450}
2020-02-04 10:06:44 +00:00
Ilya Nikolaevskiy
2181228624 Revert "[VP9] Shift spatial layers on RTP level to always start from 0."
This reverts commit 2e73a3d1e9.

Reason for revert: Fuzzer found some issues.

Original change's description:
> [VP9] Shift spatial layers on RTP level to always start from 0.
> 
> This CL uses |width| and |height| in RTPVideoHeaderVP9 to pass information
> about enabled layers from encoder to packetizer.
> 
> Bug: webrtc:11319
> Change-Id: Idc1c337f8dfb3f7631506acb784d2a634b41b955
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167724
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30428}

TBR=danilchap@webrtc.org,ilnik@webrtc.org,nisse@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:11319
Change-Id: I27a7e82737fa604b8ab769ce6503fa93e46f4e86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168123
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30448}
2020-02-03 14:15:44 +00:00
Ilya Nikolaevskiy
2e73a3d1e9 [VP9] Shift spatial layers on RTP level to always start from 0.
This CL uses |width| and |height| in RTPVideoHeaderVP9 to pass information
about enabled layers from encoder to packetizer.

Bug: webrtc:11319
Change-Id: Idc1c337f8dfb3f7631506acb784d2a634b41b955
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167724
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30428}
2020-01-30 16:07:32 +00:00
Danil Chapovalov
5c35f2fb1b Delete RtpDepacketizerVp9 in favor of VideoRtpDepacketizerVp9
Bug: webrtc:11152
Change-Id: Ic50f2dc49ca420b3406d4dea11ed20328aa59136
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165382
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30195}
2020-01-09 13:07:44 +00:00
Danil Chapovalov
32fe4ef967 Move vp9 rtp depacketization to VideoRtpDepacketizerVp9
Bug: webrtc:11152
Change-Id: I560d4cd62fabae093e3df592f0e7cc4001c10657
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162420
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30102}
2019-12-16 17:11:13 +00:00
Kuang-che Wu
75acef3962 Reject invalid spatial index
We should reject invalid values explicitly in order to prevent DCHECK
failures later, which affect fuzzing progress.

Bug: chromium:1009172, chromium:1009073
Change-Id: I7f0dc417ecac7aab076a652143f5face2ff98da2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156340
Commit-Queue: Kuang-che Wu <kcwu@google.com>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29459}
2019-10-14 12:24:01 +00:00
Niels Möller
abbc50e9b2 Move frame_type member from RtpDepacketizer::ParsedPayload to RTPVideoHeader
The latter is also a member of the former. This cleanup is also
a preparation for dropping WebRtcRTPHeader::frameType (or deleting
WebRtcRTPHeader right away), now that it's a video-specific member.


Tbr: kwiberg@webrtc.org # Comment change in modules/include/
Bug: None
Change-Id: I5c1f3f981f0d750713fc9b9b145278150fe32b5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133024
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27740}
2019-04-24 13:13:04 +00:00
Niels Möller
9d8eaac4ee Delete unneeded direct includes of common_types.h
And delete corresponding dependencies on :webrtc_common. After this
change, common_types.h is included directly only from code in the
following directories:

api/
api/video/
api/video_codecs/
common_video/libyuv/include/
media/base/
modules/remote_bitrate_estimator/
modules/rtp_rtcp/source/
modules/video_coding/codecs/vp9/

There remains plenty of indirect dependencies on the types declared in
common_types.h, but the fewer direct dependencies should make it
easier to find the proper place for each type.

Bug: webrtc:5876
Change-Id: I93e8f214025ecb613c19fdec2015bd3f96c59aae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130501
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27376}
2019-04-01 07:18:13 +00:00
Niels Möller
8f7ce222e7 Make VideoFrameType an enum class, and move to separate file and target
Bug: webrtc:5876, webrtc:6883
Change-Id: I1435cfa9e8e54c4ba2978261048ff3fbb993ce0e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126225
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27239}
2019-03-22 12:44:51 +00:00
Steve Anton
10542f21c8 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
Mechanically generated by running this command:

tools_webrtc/do-renames.sh update all-renames.txt && git cl format

Then manually updating:

tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833
Reviewed-on: https://webrtc-review.googlesource.com/c/115653
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 17:11:39 +00:00
Yves Gerey
988cc0870b [Cleanup] Add missing #include. Remove useless ones.
This CL is the result of running include-what-you-use tool on part
of the code base (audio target and dependencies) plus manual fixes.

bug: webrtc:8311
Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604
Reviewed-on: https://webrtc-review.googlesource.com/c/106280
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25311}
2018-10-23 11:32:56 +00:00
Danil Chapovalov
fcebe0e1ca in RtpPacketizers separate case 'frame fits into single packet'.
Assumption extra needed bytes for single packet needs is sum
of extra bytes for first and last packet
moved up to RTPSenderVideo from individual packetizers.
There it can be fixed.

Bug: webrtc:9868
Change-Id: I24c80ffa5c174afd3fe3e92fa86ef75560bb961e
Reviewed-on: https://webrtc-review.googlesource.com/c/105662
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25160}
2018-10-15 08:46:27 +00:00
Danil Chapovalov
0b445c6271 Cleanup RtpPacketizerVp9
Merge SetPayloadData into constructor.
Reuse payload size split function

Bug: webrtc:9680
Change-Id: If230a4ea901b5cdd6a376f8dd2db48e94d6dca36
Reviewed-on: https://webrtc-review.googlesource.com/98866
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24635}
2018-09-07 22:01:46 +00:00
Danil Chapovalov
f7f8a1f176 Cleanup RtpPacketizer interface
merge construction and call to SetPayloadData
Add NumPackets instead of SetPayloadData
Remove virtual ToString() as unused
move CHECK(rtp_video_header) from RtpPacketizer::Create to RtpSenderVideo::SendVideo

Bug: webrtc:9680
Change-Id: I074644e048c797eb836f79979df363fe1ea0075e
Reviewed-on: https://webrtc-review.googlesource.com/96543
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24474}
2018-08-29 08:44:08 +00:00
philipel
29d8846df9 Remove RTPVideoHeader::vp9() accessors.
TBR=stefan@webrtc.org

Bug: none
Change-Id: Ia2f728ea3377754a16a0b081e25c4479fe211b3e
Reviewed-on: https://webrtc-review.googlesource.com/93024
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24243}
2018-08-09 10:53:28 +00:00
philipel
5ab67a5d71 Add accessors to the types in the RTPVideoTypeHeader in RTPVideoHeader.
This CL is in preparation to change the RTPVideoTypeHeader into an absl::variant.

Bug: none
Change-Id: I1672d866df0395f3417d8e278cc67f017ab0ff98
Reviewed-on: https://webrtc-review.googlesource.com/87261
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23856}
2018-07-05 14:29:07 +00:00
philipel
cb96ad8f0e Add ParsedPayload::video_header() accessor.
Preparation CL to remove RTPTypeHeader.

Bug: none
Change-Id: I695acf20082b94744a2f6c7692f1b2128932cd79
Reviewed-on: https://webrtc-review.googlesource.com/86132
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23835}
2018-07-04 09:31:21 +00:00
Yves Gerey
665174fdbb Reformat the WebRTC code base
Running clang-format with chromium's style guide.

The goal is n-fold:
 * providing consistency and readability (that's what code guidelines are for)
 * preventing noise with presubmit checks and git cl format
 * building on the previous point: making it easier to automatically fix format issues
 * you name it

Please consider using git-hyper-blame to ignore this commit.

Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
2018-06-19 14:00:39 +00:00
Niels Möller
520ca4e3b8 Delete enum RtpVideoCodecTypes, replaced with VideoCodecType.
Bug: webrtc:8995
Change-Id: I0b44aa26f2f6a81aec7ca1281b8513d8e03228b8
Reviewed-on: https://webrtc-review.googlesource.com/79561
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23507}
2018-06-04 11:53:17 +00:00
Sergey Silkin
bc0f0d3ded Rename end_of_superframe to end_of_picture.
For consistency with the VP9 RTP spec which uses term "picture" for set
of frames which belong to the same time instance.

Bug: none
Change-Id: I30e92d5debb008feb58f770b63fe10c2e0029267
Reviewed-on: https://webrtc-review.googlesource.com/72180
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23040}
2018-04-26 15:47:17 +00:00
Sergey Silkin
c5a131a5fb Add non_ref_for_inter_layer_pred to VP9 RTP.
This converts the reserved bit in VP9 RTP payload descriptor into the
flag which indicates whether current frame can be used for prediction
of next spatial layer or not.

VP9 encoder wrapper sets non_ref_for_inter_layer_pred=false for all
frames for now.

Bug: none
Change-Id: I32f68868686475905fb09173cffd2b6e1bedcb7c
Reviewed-on: https://webrtc-review.googlesource.com/71080
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23010}
2018-04-24 19:20:30 +00:00
Sergey Silkin
2a1f183e99 Set marker bit on last encoded spatial layer.
In order to handle per-layer frame dropping both VP9 encoder wrapper
and RTP packetizer were modified.

- Encoder wrapper buffers last encoded frame and passes it to
packetizer after frame of next layer is encoded or encoding of
superframe is finished.
- Encoder wrapper sets end_of_superframe flag on last encoded frame of
superframe before passing it to packetizer.
- If end_of_superframe is True then packetizer sets marker bit on last
packet of frame.

Bug: webrtc:9066
Change-Id: I1d45319fbe6bc63d01721ea67bfb7440d4c29275
Reviewed-on: https://webrtc-review.googlesource.com/65540
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22722}
2018-04-04 10:40:19 +00:00
Mirko Bonadei
675513b96a Stop using LOG macros in favor of RTC_ prefixed macros.
This CL has been generated with the following script:

for m in PLOG \
  LOG_TAG \
  LOG_GLEM \
  LOG_GLE_EX \
  LOG_GLE \
  LAST_SYSTEM_ERROR \
  LOG_ERRNO_EX \
  LOG_ERRNO \
  LOG_ERR_EX \
  LOG_ERR \
  LOG_V \
  LOG_F \
  LOG_T_F \
  LOG_E \
  LOG_T \
  LOG_CHECK_LEVEL_V \
  LOG_CHECK_LEVEL \
  LOG
do
  git grep -l $m | xargs sed -i "s,\b$m\b,RTC_$m,g"
done
git checkout rtc_base/logging.h
git cl format

Bug: webrtc:8452
Change-Id: I1a53ef3e0a5ef6e244e62b2e012b864914784600
Reviewed-on: https://webrtc-review.googlesource.com/21325
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20617}
2017-11-09 11:56:32 +00:00
Sergey Silkin
3f2634eadc Reland "Keep spatial_idx=kNoSpatialIdx(255) if there is no layer indices."
This is a reland of 47836b4ebb

Internal tests are synced with the fix.

Original change's description:
> Keep spatial_idx=kNoSpatialIdx(255) if there is no layer indices.
> 
> spatial_idx is not present in RTP header if there is no temporal or
> spatial layering. But the parser sets spatial_idx to 0 in this case.
> When reflector repacketizes such packets it writes layering indices
> into outgoing packets. When packets arrive to receiver it thinks that
> it deals with multi layer stream and passes it through special path
> in Vp9 reference frame finder which never outputs inter frames.
> 
> I modified the parser such that it keeps spatial_idx=kNoSpatialIdx(255)
> when there is no layer indices in RTP header. Related unit tests have
> been modified as well.
> 
> Bug: none
> Change-Id: I14498cafb4e57797577dc873298c35b243479f88
> Reviewed-on: https://webrtc-review.googlesource.com/17980
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20560}

TBR=brandtr@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org

Bug: none
Change-Id: I6087a8b20a926296b30432d69251670120b2a20c
Reviewed-on: https://webrtc-review.googlesource.com/20940
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20591}
2017-11-07 16:34:20 +00:00
Ivo Creusen
ae29428489 Revert "Keep spatial_idx=kNoSpatialIdx(255) if there is no layer indices."
This reverts commit 47836b4ebb.

Reason for revert: This breaks internal tests, reverting to check if they recover.

Original change's description:
> Keep spatial_idx=kNoSpatialIdx(255) if there is no layer indices.
> 
> spatial_idx is not present in RTP header if there is no temporal or
> spatial layering. But the parser sets spatial_idx to 0 in this case.
> When reflector repacketizes such packets it writes layering indices
> into outgoing packets. When packets arrive to receiver it thinks that
> it deals with multi layer stream and passes it through special path
> in Vp9 reference frame finder which never outputs inter frames.
> 
> I modified the parser such that it keeps spatial_idx=kNoSpatialIdx(255)
> when there is no layer indices in RTP header. Related unit tests have
> been modified as well.
> 
> Bug: none
> Change-Id: I14498cafb4e57797577dc873298c35b243479f88
> Reviewed-on: https://webrtc-review.googlesource.com/17980
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20560}

TBR=brandtr@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,ssilkin@webrtc.org

Change-Id: I67d083cf769974d8df8bd5d70942af97db578db9
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/20501
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20565}
2017-11-06 15:27:48 +00:00
Sergey Silkin
47836b4ebb Keep spatial_idx=kNoSpatialIdx(255) if there is no layer indices.
spatial_idx is not present in RTP header if there is no temporal or
spatial layering. But the parser sets spatial_idx to 0 in this case.
When reflector repacketizes such packets it writes layering indices
into outgoing packets. When packets arrive to receiver it thinks that
it deals with multi layer stream and passes it through special path
in Vp9 reference frame finder which never outputs inter frames.

I modified the parser such that it keeps spatial_idx=kNoSpatialIdx(255)
when there is no layer indices in RTP header. Related unit tests have
been modified as well.

Bug: none
Change-Id: I14498cafb4e57797577dc873298c35b243479f88
Reviewed-on: https://webrtc-review.googlesource.com/17980
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20560}
2017-11-06 12:15:16 +00:00
Mirko Bonadei
92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 05:02:56 +00:00
Mirko Bonadei
bb547203bf Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 04:25:06 +00:00
Renamed from webrtc/modules/rtp_rtcp/source/rtp_format_vp9.cc (Browse further)