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Author SHA1 Message Date
pbos
2169d8bc68 Reland of move audio/video distinction for probe packets. (patchset #1 id:1 of https://codereview.webrtc.org/2086633002/ ) 
Reason for revert:
Fix already landed in google3, this revert actually breaks the import.

Original issue's description:
> Revert of Remove audio/video distinction for probe packets. (patchset #2 id:20001 of https://codereview.webrtc.org/2061193002/ )
>
> Reason for revert:
> Revert this because it broke the google3 import build.
> http://webrtc-buildbot-master.mtv.corp.google.com:21000/builders/WebRTC%20google3%20Importer%20%28Shem%20TOT%29/builds/67/steps/blaze_regular_tests/logs/stdio
>
> Original issue's description:
> > Remove audio/video distinction for probe packets.
> >
> > Allows detecting large-enough audio packets as part of a probe,
> > speculative fix for a rampup-time regression in M50. These packets are
> > accounted on the send side when probing.
> >
> > BUG=webrtc:5985
> > R=mflodman@webrtc.org, philipel@webrtc.org
> >
> > Committed: https://crrev.com/a7d88d38448f6a5677a017562765ab505b89d468
> > Cr-Commit-Position: refs/heads/master@{#13210}
>
> TBR=mflodman@webrtc.org,philipel@webrtc.org,pbos@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5985
>
> Committed: https://crrev.com/17bde8c96ee8b5a7e496a7dc98828b84f9756925
> Cr-Commit-Position: refs/heads/master@{#13221}

TBR=mflodman@webrtc.org,philipel@webrtc.org,honghaiz@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5985

Review-Url: https://codereview.webrtc.org/2085653002
Cr-Commit-Position: refs/heads/master@{#13223}
 2016-06-20 18:53:09 +00:00
honghaiz
17bde8c96e Revert of Remove audio/video distinction for probe packets. (patchset #2 id:20001 of https://codereview.webrtc.org/2061193002/ ) 
Reason for revert:
Revert this because it broke the google3 import build.
http://webrtc-buildbot-master.mtv.corp.google.com:21000/builders/WebRTC%20google3%20Importer%20%28Shem%20TOT%29/builds/67/steps/blaze_regular_tests/logs/stdio

Original issue's description:
> Remove audio/video distinction for probe packets.
>
> Allows detecting large-enough audio packets as part of a probe,
> speculative fix for a rampup-time regression in M50. These packets are
> accounted on the send side when probing.
>
> BUG=webrtc:5985
> R=mflodman@webrtc.org, philipel@webrtc.org
>
> Committed: https://crrev.com/a7d88d38448f6a5677a017562765ab505b89d468
> Cr-Commit-Position: refs/heads/master@{#13210}

TBR=mflodman@webrtc.org,philipel@webrtc.org,pbos@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5985

Review-Url: https://codereview.webrtc.org/2086633002
Cr-Commit-Position: refs/heads/master@{#13221}
 2016-06-20 18:47:25 +00:00
Peter Boström
a7d88d3844 Remove audio/video distinction for probe packets. 
Allows detecting large-enough audio packets as part of a probe,
speculative fix for a rampup-time regression in M50. These packets are
accounted on the send side when probing.

BUG=webrtc:5985
R=mflodman@webrtc.org, philipel@webrtc.org

Review URL: https://codereview.webrtc.org/2061193002 .

Cr-Commit-Position: refs/heads/master@{#13210}
 2016-06-20 08:51:20 +00:00
stefan
1112b2bc68 Fix bug when the BWE times out due to no incoming packets. 
Both InterArrival and OveruseEstimator should be timed out at the same time since otherwise the overuse filter may take a long time to converge.

BUG=webrtc:5773

Review URL: https://codereview.webrtc.org/1886783002

Cr-Commit-Position: refs/heads/master@{#12364}
 2016-04-14 15:08:20 +00:00
kwiberg
92931b15d8 Replace scoped_ptr with unique_ptr in webrtc/modules/remote_bitrate_estimator/ 
BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1750533002

Cr-Commit-Position: refs/heads/master@{#11829}
 2016-03-01 13:32:39 +00:00
Stefan Holmer
58c664c13d Clean up of CongestionController. 
Removes unused methods and moves out ViERemb to Call.

R=pbos@webrtc.org, solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/1663413003 .

Cr-Commit-Position: refs/heads/master@{#11527}
 2016-02-08 13:31:53 +00:00
terelius
8f09f170e6 Simple CL to fix lint errors in webrtc/modules/remote_bitrate_estimator. Added the lint check for the folder to the presubmit script. 
BUG=webrtc:5310

Review URL: https://codereview.webrtc.org/1520513003

Cr-Commit-Position: refs/heads/master@{#11021}
 2015-12-15 08:52:03 +00:00
Henrik Kjellander
ff761fba82 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}
 2015-11-04 07:32:04 +00:00
stefan
4fbd145dce Fix suspend below min bitrate in new API by making it possible to set min bitrate at the receive-side. 
In addition to this the ramp-up tests are refactored to use a receive call instead of only a remote bitrate estimator, and to make use of BaseTest.

BUG=webrtc:4836

Review URL: https://codereview.webrtc.org/1368943002

Cr-Commit-Position: refs/heads/master@{#10087}
 2015-09-28 10:57:23 +00:00
stefan
b947f287a6 Add pcap support to bwe tools. Allow filtering on SSRCs. 
Also switches the command line interface to gflags.

Review URL: https://codereview.webrtc.org/1235433005

Cr-Commit-Position: refs/heads/master@{#9599}
 2015-07-17 12:27:27 +00:00
Erik Språng
468e62a974 Remove MimdRateControl and factories for RemoteBitrateEstimor. 
BUG=
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1208083002.

Cr-Commit-Position: refs/heads/master@{#9541}
 2015-07-06 08:51:01 +00:00
Stefan Holmer
ff4ea9310e Only use paced packets for estimating bitrate probes. 
BUG=4778
R=mflodman@webrtc.org, solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/1188823007.

Cr-Commit-Position: refs/heads/master@{#9463}
 2015-06-18 14:01:43 +00:00
kwiberg@webrtc.org
00b8f6b364 Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away 
BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36229004

Cr-Commit-Position: refs/heads/master@{#8517}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8517 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-26 14:43:50 +00:00
pkasting@chromium.org
0b1534c52e Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess. 
This fixes a variety of MSVC warnings about value truncations when implicitly
storing the 64-bit values we get back from e.g. TimeTicks in 32-bit objects, and
removes the need for a number of explicit casts.

This also moves a number of constants so they're declared right where they're used, which is easier to read and maintain, and makes some of them of integral type rather than using the "enum hack".

BUG=chromium:81439
TEST=none
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7905 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-15 22:09:40 +00:00
stefan@webrtc.org
0b38478885 Add support for parsing header only RTP dumps with bwe_rtp_play. 
Also adds support for printing the original_length in rtp_to_text.

R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7812 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-04 15:43:49 +00:00
henrik.lundin@webrtc.org
91d928e737 Rename RtpFileReader::Packet to RtpPacket and move out of RtpFileReader 
This is in preparation for creating a new class RtpFileWriter which
will use the same RtpPacket struct.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7749 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-26 15:50:30 +00:00
pkasting@chromium.org
4591fbd09f 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
 2014-11-20 22:28:14 +00:00
pbos@webrtc.org
4b5625e5ac RTP video playback tool using Call APIs. 
Plays back rtpdump files from Wireshark in realtime as well as save the
resulting raw video to file. Unlike the RTP playback tool it doesn't
support faster-than-realtime playback/rendering, but it instead utilizes
the same path as production code and also contains support for playing
back FEC.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6838 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-06 16:26:56 +00:00
solenberg@webrtc.org
b1f5010075 VoE changes to allow forwarding of packets from VoE to ViE BWE. 
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5757 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-24 10:38:25 +00:00
stefan@webrtc.org
af839b28b0 Add AIMD option to BWE API. 
TEST=trybots
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10319005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5755 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-24 09:42:08 +00:00
stefan@webrtc.org
9b5f4d8a84 Fix build breakage introduce with r5665. 
TBR=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5666 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-10 09:38:39 +00:00
stefan@webrtc.org
f9e7c9d865 Add option to bwe_rtp_to_text to output arrival times only in nanoseconds. 
R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5665 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-10 09:11:21 +00:00
stefan@webrtc.org
1dd9b4d98e Add BWE tools for parsing RTP files. 
bwe_rtp_play feeds packets from an RTP file into the BWE and prints the estimates.
bwe_rtp_to_text parses an RTP file and outputs the result to a text file.

R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7689006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5466 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-31 09:15:48 +00:00
stefan@webrtc.org
f5d7c5891c Revert r4934 "Add a tool for parsing an RTP file and outputting the BWE relevant fields." 
Revert r4935 "Fix build error in r4934."

TBR=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2364004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4936 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-07 08:42:46 +00:00
stefan@webrtc.org
611e5141cb Fix build error in r4934. 
TBR=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2363004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4935 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-07 08:35:36 +00:00
stefan@webrtc.org
bc99bcfa6f Add a tool for parsing an RTP file and outputting the BWE relevant fields. 
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2237005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4934 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-07 08:21:24 +00:00