Commit graph

7 commits

Author SHA1 Message Date
Danil Chapovalov
e7e156d7a6 Support first packet reduction in H264 packetizer
Bug: webrtc:9680
Change-Id: I73c9a5acecdf8dd82347be602bbfd7412c9610c5
Reviewed-on: https://webrtc-review.googlesource.com/99804
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24823}
2018-09-25 11:43:29 +00:00
Danil Chapovalov
fd5fbd0b58 Cleanup RtpPacketizerH264 constructor
Merge SetPayloadData into constructor.
Add TODO to support first_packet_reduction_len

Bug: webrtc:9680
Change-Id: I65e771848e0ffe8968cd084840e77afc0152caeb
Reviewed-on: https://webrtc-review.googlesource.com/99505
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24702}
2018-09-12 11:11:18 +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
Ilya Nikolaevskiy
1d037ae704 Don't crash in SingleNalu packetization for h264 if no space in packet
Also, pass correct max payload data size to encoders: now accounting for
rtp headers.

Bug: chromium:819259
Change-Id: I586924e9246218fab6072e05eca894925cfe556e
Reviewed-on: https://webrtc-review.googlesource.com/61425
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22460}
2018-03-15 15:42:57 +00:00
Danil Chapovalov
2a5ce2bcf8 Fix clang style errors in rtp_rtcp and dependant targets
Mark functions with override instead of virtual.
Add explicit non-trivial constructors/assign operators/destructors.
Define them in .cc files instead of inlining
use auto* instead of auto when deduced type is raw pointer

Bug: webrtc:163
Change-Id: I4d8a05d6a64fcc2ca16d02c5fcf9488fda832a6d
Reviewed-on: https://webrtc-review.googlesource.com/48781
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21927}
2018-02-07 09:48:28 +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_h264.h (Browse further)