9 Commits

Author SHA1 Message Date
eladalon
9addbebf42 Remove RtpDemuxer tweak for preventing multiple RSID inspections
We have a tweak preventing multiple deep-examinations of packets; packets with a given SSRC are only inspected deeply (RSID) once (only the first received packet). Once we move to many-to-one stream-to-sink associations, this becomes less useful, and is better removed.

BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2955373002
Cr-Commit-Position: refs/heads/master@{#18859}
2017-06-30 13:26:54 +00:00
eladalon
c3e3e60f59 nit: Rename RtpDemuxer::sink_ to RtpDemuxer::ssrc_sinks_
Rationale:
1. sinks_ is not properly differentiated from rsid_sinks_.
2. Consistency with RtcpDemuxer.

BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2958283002
Cr-Commit-Position: refs/heads/master@{#18813}
2017-06-28 15:18:51 +00:00
eladalon
a52722fac4 Reland of Create RtcpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2957763002/ )
Reason for revert:
About to fix problem and reland.

Original issue's description:
> Revert of Create RtcpDemuxer (patchset #13 id:240001 of https://codereview.webrtc.org/2943693003/ )
>
> Reason for revert:
> Breaks Chromium FYI bots.
>
> The problem is in the BUILD.gn file.
>
> Sample failure:
> https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Builder/builds/17829
>
> Sample logs:
> use_goma = true
> """ to /b/c/b/Linux_Builder/src/out/Release/args.gn.
>
> /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check
>   -> returned 1
> ERROR at //third_party/webrtc/call/BUILD.gn:46:5: Can't load input file.
>     "//webrtc/base:rtc_base_approved",
>     ^--------------------------------
>
> Original issue's description:
> > Create RtcpDemuxer. Capabilities:
> > 1. Demux RTCP messages according to the sender-SSRC.
> > 2. Demux RTCP messages according to the RSID (resolved to an SSRC, then compared to the sender-RTCP).
> > 3. Allow listening in on all RTCP messages passing through the demuxer ("broadcast sinks").
> >
> > BUG=webrtc:7135
> >
> > Review-Url: https://codereview.webrtc.org/2943693003
> > Cr-Commit-Position: refs/heads/master@{#18763}
> > Committed: cb83bdf01f
>
> BUG=webrtc:7135
>
> Review-Url: https://codereview.webrtc.org/2957763002
> Cr-Commit-Position: refs/heads/master@{#18764}
> Committed: 0e7e7869e7

BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2960623002
Cr-Commit-Position: refs/heads/master@{#18768}
2017-06-26 18:23:54 +00:00
guidou
0e7e7869e7 Revert of Create RtcpDemuxer (patchset #13 id:240001 of https://codereview.webrtc.org/2943693003/ )
Reason for revert:
Breaks Chromium FYI bots.

The problem is in the BUILD.gn file.

Sample failure:
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Builder/builds/17829

Sample logs:
use_goma = true
""" to /b/c/b/Linux_Builder/src/out/Release/args.gn.

/b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check
  -> returned 1
ERROR at //third_party/webrtc/call/BUILD.gn:46:5: Can't load input file.
    "//webrtc/base:rtc_base_approved",
    ^--------------------------------

Original issue's description:
> Create RtcpDemuxer. Capabilities:
> 1. Demux RTCP messages according to the sender-SSRC.
> 2. Demux RTCP messages according to the RSID (resolved to an SSRC, then compared to the sender-RTCP).
> 3. Allow listening in on all RTCP messages passing through the demuxer ("broadcast sinks").
>
> BUG=webrtc:7135
>
> Review-Url: https://codereview.webrtc.org/2943693003
> Cr-Commit-Position: refs/heads/master@{#18763}
> Committed: cb83bdf01f

TBR=stefan@webrtc.org,danilchap@webrtc.org,nisse@webrtc.org,kjellander@webrtc.org,holmer@google.com,eladalon@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2957763002
Cr-Commit-Position: refs/heads/master@{#18764}
2017-06-26 13:28:36 +00:00
eladalon
cb83bdf01f Create RtcpDemuxer. Capabilities:
1. Demux RTCP messages according to the sender-SSRC.
2. Demux RTCP messages according to the RSID (resolved to an SSRC, then compared to the sender-RTCP).
3. Allow listening in on all RTCP messages passing through the demuxer ("broadcast sinks").

BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2943693003
Cr-Commit-Position: refs/heads/master@{#18763}
2017-06-26 12:56:34 +00:00
eladalon
dea075c7a6 Log an error in RtpDemuxer::FindSsrcAssociations() if kMaxProcessedSsrcs exceeded
BUG=None

Review-Url: https://codereview.webrtc.org/2941513002
Cr-Commit-Position: refs/heads/master@{#18569}
2017-06-13 14:57:31 +00:00
eladalon
d0244c21cd Add RSID-based demuxing to RtpDemuxer
Make RtpDemuxer able to demux RTP packets according to RSID (RTP Stream ID), as well as the (pre-existing) ability to demux according to SSRC.

BUG=None

Review-Url: https://codereview.webrtc.org/2920993002
Cr-Commit-Position: refs/heads/master@{#18495}
2017-06-08 11:19:13 +00:00
nisse
d76b7b294a New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender.
BUG=webrtc:7135
TBR=sprang@webrtc.org

Review-Url: https://codereview.webrtc.org/2913143003
Cr-Commit-Position: refs/heads/master@{#18371}
2017-06-01 11:02:35 +00:00
nisse
e4bcd6d02a New class RtpDemuxer and RtpPacketSinkInterface, use in Call.
BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2867943003
Cr-Commit-Position: refs/heads/master@{#18160}
2017-05-16 11:47:04 +00:00