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Starred by 5 users
Status: Fixed
Owner:
Closed: Nov 7
Cc:
Components:
NextAction: ----
OS: ----
Pri: 3
Type: Enhancement

Blocked on:
issue 414



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Clean up NetEq test tools
Project Member Reported by henrik.lundin@webrtc.org, Dec 2 2013 Back to list
The test tools for NetEq are diverging. They are currently found in two places: .../neteq/test/ and .../neteq4/test/.

The ones in neteq/test should mostly be deprecated.

The ones in neteq4/test should be cleaned up, changed to correct format, and so on.

 
Project Member Comment 1 by henrik.lundin@webrtc.org, Dec 4 2013
Blockedon: webrtc:414
Project Member Comment 2 by henrik.lundin@webrtc.org, Dec 4 2013
Labels: hotlist-neteq-14Q1
Project Member Comment 3 by henrik.lundin@webrtc.org, Mar 4 2014
Labels: neteq
Project Member Comment 4 by bugdroid1@chromium.org, Apr 2 2014
The following revision refers to this bug:
  http://code.google.com/p/webrtc/source/detail?r=5832

------------------------------------------------------------------
r5832 | henrik.lundin@webrtc.org | 2014-04-02T20:56:17.017967Z

Changed paths:
   D http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/neteq4/test/RTPanalyze.cc&spec=svn5832&r_previous=5831&r=5832&format=side
   A http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/neteq4/tools/rtp_analyze.cc&spec=svn5832&r_previous=5831&r=5832&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/neteq4/neteq_tests.gypi&spec=svn5832&r_previous=5831&r=5832&format=side
   D http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/neteq/test/RTPanalyze.cc&spec=svn5832&r_previous=5831&r=5832&format=side

Rename RTPanalyze to rtp_analyze and remove old version

The tool RTPanalyze (used to process an input RTP dump into a
text file of RTP header info) was present in both the neteq and
neteq4 folders. This change pulls in changes from the old to the new
and renames the source file and tool to rtp_analyze.

Removing special code for dummy-rtp files (it is supported without
special code), and making the RED payload type settable using flags.

Moving from test/ to tools/ folder.

BUG= 2692 
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10789004
-----------------------------------------------------------------
Project Member Comment 5 by bugdroid1@chromium.org, Apr 14 2014
The following revision refers to this bug:
  http://code.google.com/p/webrtc/source/detail?r=5901

------------------------------------------------------------------
r5901 | henrik.lundin@webrtc.org | 2014-04-14T18:42:23.744929Z

Changed paths:
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/neteq4/tools/rtp_analyze.cc&spec=svn5901&r_previous=5900&r=5901&format=side
   A http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/neteq4/tools/packet.h&spec=svn5901&r_previous=5900&r=5901&format=side
   A http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/neteq4/tools/packet_source.h&spec=svn5901&r_previous=5900&r=5901&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/modules.gyp&spec=svn5901&r_previous=5900&r=5901&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/neteq4/neteq.gypi&spec=svn5901&r_previous=5900&r=5901&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/neteq4/neteq_tests.gypi&spec=svn5901&r_previous=5900&r=5901&format=side
   A http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/neteq4/tools/rtp_file_source.cc&spec=svn5901&r_previous=5900&r=5901&format=side
   A http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/neteq4/tools/packet_unittest.cc&spec=svn5901&r_previous=5900&r=5901&format=side
   A http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/neteq4/tools/rtp_file_source.h&spec=svn5901&r_previous=5900&r=5901&format=side
   A http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/neteq4/tools/packet.cc&spec=svn5901&r_previous=5900&r=5901&format=side

New Packet and PacketSource classes for NetEq tests

These new classes are intended to replace the old NETEQTEST_RTPpacket
classes. The code in rtp_analyze.cc has been updated to use the new
classes; other test applications will follow.

BUG= 2692 
R=andrew@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11769004
-----------------------------------------------------------------
Project Member Comment 6 by bugdroid1@chromium.org, Aug 11 2014
The following revision refers to this bug:
  http://code.google.com/p/webrtc/source/detail?r=6862

------------------------------------------------------------------
r6862 | henrik.lundin@webrtc.org | 2014-08-11T12:29:38.980664Z

Changed paths:
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/neteq/neteq_tests.gypi&spec=svn6862&r_previous=6861&r=6862&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc&spec=svn6862&r_previous=6861&r=6862&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/neteq/tools/packet.cc&spec=svn6862&r_previous=6861&r=6862&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc&spec=svn6862&r_previous=6861&r=6862&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/neteq/tools/packet.h&spec=svn6862&r_previous=6861&r=6862&format=side

Use test::Packet test::PacketSource classes in neteq_rtpplay

This change replaces the old NETEQTEST_RTPpacket and
NETEQTEST_DummyRTPpacket with the new test::Packet class. Note that the
Packet class automatically handles "dummy" packets (i.e., packets for
which only the header and a length field was stored to file)
automatically. There is no need to explicitly signal this to the
application any longer. The RTP input file is now handled as a
test::PacketSource object.

Also adding a new ConvertHeader method to the Packet class. This is
needed to extract the header information as an alternative data type.

Finally, some dead code was deleted from rtp_analyze.cc (unrelated to
the reset of this change).

BUG= 2692 
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21139004
-----------------------------------------------------------------
Project Member Comment 7 by tina.legrand@webrtc.org, Aug 13 2014
Labels: Area-SignalProcessing
Project Member Comment 8 by bugdroid1@chromium.org, Oct 2 2014
The following revision refers to this bug:
  http://code.google.com/p/webrtc/source/detail?r=7367

------------------------------------------------------------------
r7367 | henrik.lundin@webrtc.org | 2014-10-02T08:19:38.001998Z

Changed paths:
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc&spec=svn7367&r_previous=7366&r=7367&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/neteq/tools/packet_source.h&spec=svn7367&r_previous=7366&r=7367&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/neteq/neteq.gypi&spec=svn7367&r_previous=7366&r=7367&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc&spec=svn7367&r_previous=7366&r=7367&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h&spec=svn7367&r_previous=7366&r=7367&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc&spec=svn7367&r_previous=7366&r=7367&format=side

Let RtpFileSource use RtpFileReader

RtpFileSource used to implement it's own RTP dump file reader, but
with this change it simply uses RtpFileReader. One benefit is that
pcap files are now also supported.

All NetEq test tools that use RtpFileSource are updated.

BUG= 2692 
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22839004
-----------------------------------------------------------------
Project Member Comment 9 by bugdroid1@chromium.org, Oct 7 2014
The following revision refers to this bug:
  http://code.google.com/p/webrtc/source/detail?r=7380

------------------------------------------------------------------
r7380 | henrik.lundin@webrtc.org | 2014-10-07T05:30:04.078011Z

Changed paths:
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc&spec=svn7380&r_previous=7379&r=7380&format=side

Add an SSRC filter to neteq_rtpplay

This allows the user to set the desired SSRC if the input file
contains multiple streams.

BUG= 2692 
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30609004
-----------------------------------------------------------------
Project Member Comment 10 by bugdroid1@chromium.org, Oct 7 2014
The following revision refers to this bug:
  http://code.google.com/p/webrtc/source/detail?r=7382

------------------------------------------------------------------
r7382 | henrik.lundin@webrtc.org | 2014-10-07T07:18:36.704577Z

Changed paths:
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc&spec=svn7382&r_previous=7381&r=7382&format=side

Fix neteq_rtpplay so that empty SSRC is valid

In r7380, the command line flag --ssrc was added to neteq_rtpplay.
However, it was not possible to omit that flag, since the validation
did not accept an empty string. This CL fixes that.

TBR=kwiberg@webrtc.org
BUG= 2692 

Review URL: https://webrtc-codereview.appspot.com/24869004
-----------------------------------------------------------------
Project Member Comment 11 by bugdroid1@chromium.org, Oct 28 2014
The following revision refers to this bug:
  http://code.google.com/p/webrtc/source/detail?r=7542

------------------------------------------------------------------
r7542 | henrik.lundin@webrtc.org | 2014-10-28T09:47:13.210489Z

Changed paths:
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/neteq/neteq.gypi&spec=svn7542&r_previous=7541&r=7542&format=side

Use neteq_unittest_tools in audio_decoder_unittests

With the recent move of RtpFileReader to the rtp_test_utils target
(in r7536), it is now possible to let audio_decoder_unittests depend
on neteq_unittest_tools without breaking the Android build.

BUG= 2692 
TBR=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30819004
-----------------------------------------------------------------
Project Member Comment 12 by tina.legrand@webrtc.org, Oct 30 2014
Labels: EngTriaged IceBox
Project Member Comment 14 by bugdroid1@chromium.org, Nov 17 2014
The following revision refers to this bug:
  http://code.google.com/p/webrtc/source/detail?r=7709

------------------------------------------------------------------
r7709 | henrik.lundin@webrtc.org | 2014-11-17T09:08:38.750622Z

Changed paths:
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/modules.gyp&spec=svn7709&r_previous=7708&r=7709&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/neteq/tools/packet_source.h&spec=svn7709&r_previous=7708&r=7709&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/neteq/neteq_unittest.cc&spec=svn7709&r_previous=7708&r=7709&format=side

Use RtpFileSource in NetEqDecodingTest

This CL removes the dependency on the old NETEQTEST_RTPpacket class
from the NetEqDecodingTest code, and also removes the dependency
from the modules_unittests target to neteq_test_tools.

BUG= 2692 
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24269004
-----------------------------------------------------------------
Project Member Comment 15 by bugdroid1@chromium.org, Nov 24 2014
The following revision refers to this bug:
  http://code.google.com/p/webrtc/source/detail?r=7740

------------------------------------------------------------------
r7740 | henrik.lundin@webrtc.org | 2014-11-24T14:50:53.570872Z

Changed paths:
   A http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/neteq/tools/output_wav_file.h&spec=svn7740&r_previous=7739&r=7740&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/neteq/neteq.gypi&spec=svn7740&r_previous=7739&r=7740&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc&spec=svn7740&r_previous=7739&r=7740&format=side

Add wav output capability to neteq_rtpplay

This CL makes neteq_rtpplay capable of writing to wav files as well as
pcm files. This is done through the new class OutputWavFile, which
wraps a WavWriter object in an AudioSink interface.

BUG= 2692 
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32139004
-----------------------------------------------------------------
Project Member Comment 16 by bugdroid1@chromium.org, Dec 1 2014
The following revision refers to this bug:
  http://code.google.com/p/webrtc/source/detail?r=7768

------------------------------------------------------------------
r7768 | henrik.lundin@webrtc.org | 2014-12-01T11:25:04.534202Z

Changed paths:
   A http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/test/rtp_file_writer_unittest.cc&spec=svn7768&r_previous=7767&r=7768&format=side
   A http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/test/rtp_file_writer.cc&spec=svn7768&r_previous=7767&r=7768&format=side
   A http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/test/rtp_file_writer.h&spec=svn7768&r_previous=7767&r=7768&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/neteq/test/RTPcat.cc&spec=svn7768&r_previous=7767&r=7768&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/test/rtp_file_reader.cc&spec=svn7768&r_previous=7767&r=7768&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/test/webrtc_test_common.gyp&spec=svn7768&r_previous=7767&r=7768&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/test/test.gyp&spec=svn7768&r_previous=7767&r=7768&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/neteq/neteq_tests.gypi&spec=svn7768&r_previous=7767&r=7768&format=side

Adding a new test helper RtpFileWriter and use it in RTPcat

This new helper class writes RTP packets to file in rtpdump format.
A unit test is also included.

The new test class is used while re-writing the test tool RTPcat.

BUG= 2692 
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28099004
-----------------------------------------------------------------
Project Member Comment 18 by bugdroid1@chromium.org, Dec 3 2014
The following revision refers to this bug:
  http://code.google.com/p/webrtc/source/detail?r=7796

------------------------------------------------------------------
r7796 | henrik.lundin@webrtc.org | 2014-12-03T13:28:53.759315Z

Changed paths:
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc&spec=svn7796&r_previous=7795&r=7796&format=side

Adding a duration printout to neteq_rtpplay

BUG= 2692 
TBR=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30339004
-----------------------------------------------------------------
Project Member Comment 19 by bugdroid1@chromium.org, Apr 17 2015
The following revision refers to this bug:
  https://chromium.googlesource.com/external/webrtc.git/+/b0b54259c3c793cc21695d77c2cc4caeab11bb7b

commit b0b54259c3c793cc21695d77c2cc4caeab11bb7b
Author: Henrik Lundin <henrik.lundin@webrtc.org>
Date: Fri Apr 17 09:47:07 2015

Let rtp_analyze parse absolute sender time

Also change to use virtual_packet_length_bytes in order to print the
actual packet size of the complete packet even when the RTP file only
contains RTP headers.

BUG= 2692 
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51559004

Cr-Commit-Position: refs/heads/master@{#9025}

[modify] http://crrev.com/b0b54259c3c793cc21695d77c2cc4caeab11bb7b/webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc

Project Member Comment 20 by bugdroid1@chromium.org, Apr 22 2015
Project Member Comment 21 by bugdroid1@chromium.org, May 8 2015
The following revision refers to this bug:
  https://chromium.googlesource.com/external/webrtc.git/+/83b5c053b9687813c5fc9c08b3beee4d464f7950

commit 83b5c053b9687813c5fc9c08b3beee4d464f7950
Author: Henrik Lundin <henrik.lundin@webrtc.org>
Date: Fri May 08 08:33:57 2015

Modify NetEqQualityTest

- Take input sample rate as parameter - provides resampling when needed.
- Add support for wav output.

BUG= 2692 
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49699004

Cr-Commit-Position: refs/heads/master@{#9158}

[modify] http://crrev.com/83b5c053b9687813c5fc9c08b3beee4d464f7950/webrtc/modules/audio_coding/neteq/interface/neteq.h
[modify] http://crrev.com/83b5c053b9687813c5fc9c08b3beee4d464f7950/webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc
[modify] http://crrev.com/83b5c053b9687813c5fc9c08b3beee4d464f7950/webrtc/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc
[modify] http://crrev.com/83b5c053b9687813c5fc9c08b3beee4d464f7950/webrtc/modules/audio_coding/neteq/tools/input_audio_file.h
[modify] http://crrev.com/83b5c053b9687813c5fc9c08b3beee4d464f7950/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc
[modify] http://crrev.com/83b5c053b9687813c5fc9c08b3beee4d464f7950/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h
[modify] http://crrev.com/83b5c053b9687813c5fc9c08b3beee4d464f7950/webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.cc
[modify] http://crrev.com/83b5c053b9687813c5fc9c08b3beee4d464f7950/webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h

Project Member Comment 23 by bugdroid1@chromium.org, May 12 2015
Project Member Comment 24 by bugdroid1@chromium.org, May 22 2015
Project Member Comment 25 by bugdroid1@chromium.org, May 28 2015
The following revision refers to this bug:
  https://chromium.googlesource.com/external/webrtc.git/+/8051832a9d2d3a18dabeaf6fd09103a4055aceee

commit 8051832a9d2d3a18dabeaf6fd09103a4055aceee
Author: Henrik Lundin <henrik.lundin@webrtc.org>
Date: Thu May 28 10:37:46 2015

Adding a new Matlab tool rtpAnalyze

The purpose of the tool is to analyze the output from the command line
tool rtp_analyze. That is, starting with an rtpdump or pcap file, it
is processed through rtp_analyze to produce a text output, which is
then used as input to this new Matlab function.

BUG= 2692 
TBR=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47339004

Cr-Commit-Position: refs/heads/master@{#9306}

[add] http://crrev.com/8051832a9d2d3a18dabeaf6fd09103a4055aceee/tools/matlab/rtpAnalyze.m

Project Member Comment 26 by bugdroid1@chromium.org, Jun 3 2015
The following revision refers to this bug:
  https://chromium.googlesource.com/external/webrtc.git/+/60508f8621ae9188871c68a7317413a274071550

commit 60508f8621ae9188871c68a7317413a274071550
Author: Henrik Lundin <henrik.lundin@webrtc.org>
Date: Wed Jun 03 07:38:23 2015

Small changes to rtpAnalyze Matlab script

These changes are in response to post-commit comments in
https://webrtc-codereview.appspot.com/47339004/.

BUG= webrtc:2692 
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50189004

Cr-Commit-Position: refs/heads/master@{#9358}

[modify] http://crrev.com/60508f8621ae9188871c68a7317413a274071550/tools/matlab/rtpAnalyze.m

Project Member Comment 27 by bugdroid1@chromium.org, Jun 16 2015
The following revision refers to this bug:
  https://chromium.googlesource.com/external/webrtc.git/+/76381d921f817888d9c050d5f2fcbe3de412e0d7

commit 76381d921f817888d9c050d5f2fcbe3de412e0d7
Author: Henrik Lundin <henrik.lundin@webrtc.org>
Date: Tue Jun 16 07:28:09 2015

Update rtpAnalyze matlab tool to handle reordered packets

With this change, the tool will find and mark reordered packets in the
plot. Furthermore, the instantaneous send bitrate will be correct even
for reordered packets.

BUG= webrtc:2692 
R=tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/1172533004.

Cr-Commit-Position: refs/heads/master@{#9443}

[modify] http://crrev.com/76381d921f817888d9c050d5f2fcbe3de412e0d7/tools/matlab/rtpAnalyze.m

Project Member Comment 28 by bugdroid1@chromium.org, Aug 18 2015
The following revision refers to this bug:
  https://chromium.googlesource.com/external/webrtc.git/+/d84dcbd2ecd28bd57e1f17bc00dc6390cbbd0cda

commit d84dcbd2ecd28bd57e1f17bc00dc6390cbbd0cda
Author: henrik.lundin <henrik.lundin@webrtc.org>
Date: Tue Aug 18 11:46:46 2015

rtpAnalyze matlab tool: filter out RTCP packets

This change relates to the matlab tool rtpAnalyze. With this change,
RTP packets with payload types 72 through 76 are removed. In IETF
RFC3551, section "Payload Type Definitions", this range is marked as
reserved so that RTCP and RTP packets can be reliably distinguished.

BUG= webrtc:2692 
TBR=tina.legrand@webrtc.org
NOTRY=true

Review URL: https://codereview.webrtc.org/1284423006

Cr-Commit-Position: refs/heads/master@{#9724}

[modify] http://crrev.com/d84dcbd2ecd28bd57e1f17bc00dc6390cbbd0cda/tools/matlab/rtpAnalyze.m

Project Member Comment 31 by bugdroid1@chromium.org, May 24 2016
The following revision refers to this bug:
  https://chromium.googlesource.com/external/webrtc.git/+/ce5570e54ebd03cf51f08ccd25dd02b4eccd2900

commit ce5570e54ebd03cf51f08ccd25dd02b4eccd2900
Author: henrik.lundin <henrik.lundin@webrtc.org>
Date: Tue May 24 13:14:57 2016

Move neteq_rtpplay.cc inside webrtc::test namespace

This simplifies the code.

BUG= webrtc:2692 

Review-Url: https://codereview.webrtc.org/2006723002
Cr-Commit-Position: refs/heads/master@{#12873}

[modify] https://crrev.com/ce5570e54ebd03cf51f08ccd25dd02b4eccd2900/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc

Project Member Comment 32 by bugdroid1@chromium.org, May 25 2016
The following revision refers to this bug:
  https://chromium.googlesource.com/external/webrtc.git/+/46ba49c622d305dff756d4d228d9ec953a2aea4e

commit 46ba49c622d305dff756d4d228d9ec953a2aea4e
Author: henrik.lundin <henrik.lundin@webrtc.org>
Date: Wed May 25 05:50:47 2016

Let PacketSource::NextPacket() return an std::unique_ptr

The return type of PacketSource::NextPacket() is changed from a naked
pointer to an std::uniqe_ptr. The interface contract was and still is
that the ownership is passed from the callee to the caller, but a
unique_ptr makes this explicit.

BUG= webrtc:2692 

Review-Url: https://codereview.webrtc.org/2005873002
Cr-Commit-Position: refs/heads/master@{#12884}

[modify] https://crrev.com/46ba49c622d305dff756d4d228d9ec953a2aea4e/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc
[modify] https://crrev.com/46ba49c622d305dff756d4d228d9ec953a2aea4e/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.cc
[modify] https://crrev.com/46ba49c622d305dff756d4d228d9ec953a2aea4e/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.h
[modify] https://crrev.com/46ba49c622d305dff756d4d228d9ec953a2aea4e/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
[modify] https://crrev.com/46ba49c622d305dff756d4d228d9ec953a2aea4e/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
[modify] https://crrev.com/46ba49c622d305dff756d4d228d9ec953a2aea4e/webrtc/modules/audio_coding/neteq/tools/constant_pcm_packet_source.cc
[modify] https://crrev.com/46ba49c622d305dff756d4d228d9ec953a2aea4e/webrtc/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h
[modify] https://crrev.com/46ba49c622d305dff756d4d228d9ec953a2aea4e/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
[modify] https://crrev.com/46ba49c622d305dff756d4d228d9ec953a2aea4e/webrtc/modules/audio_coding/neteq/tools/packet_source.h
[modify] https://crrev.com/46ba49c622d305dff756d4d228d9ec953a2aea4e/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
[modify] https://crrev.com/46ba49c622d305dff756d4d228d9ec953a2aea4e/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h
[modify] https://crrev.com/46ba49c622d305dff756d4d228d9ec953a2aea4e/webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc
[modify] https://crrev.com/46ba49c622d305dff756d4d228d9ec953a2aea4e/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
[modify] https://crrev.com/46ba49c622d305dff756d4d228d9ec953a2aea4e/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h

Project Member Comment 33 by bugdroid1@chromium.org, Jun 22 2016
The following revision refers to this bug:
  https://chromium.googlesource.com/external/webrtc.git/+/e8a77e330942dd98ee86c09ac22850e0d4225944

commit e8a77e330942dd98ee86c09ac22850e0d4225944
Author: henrik.lundin <henrik.lundin@webrtc.org>
Date: Wed Jun 22 13:34:03 2016

Refactor neteq_rtpplay

This change is a major refactoring of the neteq_rtpplay tool. It
consists of the following parts:

- NetEqTest class: Breaks out the main simulation loop from
  neteq_rtpplay into a separate class with well defined inputs and
  outputs.
- NetEqInput: Interface class for the input to NetEqTest.
- NetEqPacketSourceInput: Implementation of NetEqInput that provides a
  PacketSource objects with a NetEqInput interface. This has two
  subclasses; one for RtpFileSource and one for RtcEventLogSource.
- NetEqReplacementInput: An object that modifies the packets provided by
  another NetEqInput object, and replaces the packet payloads with meta
  data readable by a FakeDecodeFromFile decoder.
- FakeDecodeFromFile: An AudioDecoder implementation that produces
  "decoded" data by reading from an audio file.

BUG= webrtc:2692 , webrtc:5447

Review-Url: https://codereview.webrtc.org/2020363003
Cr-Commit-Position: refs/heads/master@{#13252}

[modify] https://crrev.com/e8a77e330942dd98ee86c09ac22850e0d4225944/webrtc/modules/audio_coding/BUILD.gn
[modify] https://crrev.com/e8a77e330942dd98ee86c09ac22850e0d4225944/webrtc/modules/audio_coding/neteq/neteq.gypi
[modify] https://crrev.com/e8a77e330942dd98ee86c09ac22850e0d4225944/webrtc/modules/audio_coding/neteq/neteq_tests.gypi
[add] https://crrev.com/e8a77e330942dd98ee86c09ac22850e0d4225944/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc
[add] https://crrev.com/e8a77e330942dd98ee86c09ac22850e0d4225944/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.h
[add] https://crrev.com/e8a77e330942dd98ee86c09ac22850e0d4225944/webrtc/modules/audio_coding/neteq/tools/neteq_input.h
[add] https://crrev.com/e8a77e330942dd98ee86c09ac22850e0d4225944/webrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.cc
[add] https://crrev.com/e8a77e330942dd98ee86c09ac22850e0d4225944/webrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.h
[add] https://crrev.com/e8a77e330942dd98ee86c09ac22850e0d4225944/webrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.cc
[add] https://crrev.com/e8a77e330942dd98ee86c09ac22850e0d4225944/webrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.h
[modify] https://crrev.com/e8a77e330942dd98ee86c09ac22850e0d4225944/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
[add] https://crrev.com/e8a77e330942dd98ee86c09ac22850e0d4225944/webrtc/modules/audio_coding/neteq/tools/neteq_test.cc
[add] https://crrev.com/e8a77e330942dd98ee86c09ac22850e0d4225944/webrtc/modules/audio_coding/neteq/tools/neteq_test.h

Project Member Comment 35 by bugdroid1@chromium.org, Aug 10 2016
Project Member Comment 36 by bugdroid1@chromium.org, Aug 24 2016
The following revision refers to this bug:
  https://chromium.googlesource.com/external/webrtc.git/+/d1a10a0f7795213210f9a9f5720167f97bade8c9

commit d1a10a0f7795213210f9a9f5720167f97bade8c9
Author: henrik.lundin <henrik.lundin@webrtc.org>
Date: Wed Aug 24 17:58:54 2016

Make FakeDecodeFromFile handle codec-internal CNG

This implementation interprets payloads of size 1 as codec-internal SID
frames, marking the start of a CNG period. Changes were made to other
parts of the test payload chain, since it had to make use of the virtual
payload size in the case of header-only RTP files.

BUG= webrtc:2692 

Review-Url: https://codereview.webrtc.org/2275903002
Cr-Commit-Position: refs/heads/master@{#13901}

[modify] https://crrev.com/d1a10a0f7795213210f9a9f5720167f97bade8c9/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc
[modify] https://crrev.com/d1a10a0f7795213210f9a9f5720167f97bade8c9/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.h
[modify] https://crrev.com/d1a10a0f7795213210f9a9f5720167f97bade8c9/webrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.cc
[modify] https://crrev.com/d1a10a0f7795213210f9a9f5720167f97bade8c9/webrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.cc
[modify] https://crrev.com/d1a10a0f7795213210f9a9f5720167f97bade8c9/webrtc/modules/audio_coding/neteq/tools/packet.cc

Project Member Comment 38 by bugdroid1@chromium.org, Sep 6 2016
The following revision refers to this bug:
  https://chromium.googlesource.com/external/webrtc.git/+/d4ec970a795dec643130af7fe766fbbb5bd101c0

commit d4ec970a795dec643130af7fe766fbbb5bd101c0
Author: henrik.lundin <henrik.lundin@webrtc.org>
Date: Tue Sep 06 08:22:45 2016

neteq_rtpplay: Add an error message for unmatched SSRC

If neteq_rtpplay is invoked with the --ssrc option to select packets
matching a specific SSRC, but no matching packets are found, this CL
provides a meaningful error message.

BUG= webrtc:2692 
NOTRY=True
TBR=ivoc@webrtc.org

Review-Url: https://codereview.webrtc.org/2318503002
Cr-Commit-Position: refs/heads/master@{#14083}

[modify] https://crrev.com/d4ec970a795dec643130af7fe766fbbb5bd101c0/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc

Project Member Comment 39 by henrik.lundin@webrtc.org, Oct 5 2016
Components: Audio
Project Member Comment 40 by henrik.lundin@webrtc.org, Oct 5 2016
Components: -SignalProcessing
Project Member Comment 41 by tina.legrand@webrtc.org, Nov 7
Status: Archived
[Bulk edit] This issue hasn't been modified the last twelve months -> archiving.

If this is still a valid issue that should be open, please reopen again.
Project Member Comment 42 by bugdroid1@chromium.org, Nov 24
Project Member Comment 43 by bugdroid1@chromium.org, Nov 24
The following revision refers to this bug:
  https://webrtc.googlesource.com/src.git/+/81af414ffb103c645bf50c77413096de3eb4e532

commit 81af414ffb103c645bf50c77413096de3eb4e532
Author: Henrik Lundin <henrik.lundin@webrtc.org>
Date: Fri Nov 24 13:38:59 2017

Replacing the legacy tool RTPjitter with a new rtp_jitter

This new tool provides the similar functionality as the legacy tool, but
is implemented using less legacy helpers. It also replaces RTPtimeshift
and RTPchange. The most significant change versus the old RTPjitter tool
is that the new tool takes the timing data in the form of integers in a
text file (instead of the binary data file used by the old tool). This
should make it easier to create custom timing files when needed.

Bug:  webrtc:2692 
Change-Id: I5e46fe7abdd9ca8c04a04de87555204fca36e287
Reviewed-on: https://webrtc-review.googlesource.com/25700
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20868}
[modify] https://crrev.com/81af414ffb103c645bf50c77413096de3eb4e532/modules/audio_coding/BUILD.gn
[delete] https://crrev.com/0e1d7989a54e146974eec758056f4af97273d2b7/modules/audio_coding/neteq/test/NETEQTEST_DummyRTPpacket.cc
[delete] https://crrev.com/0e1d7989a54e146974eec758056f4af97273d2b7/modules/audio_coding/neteq/test/NETEQTEST_DummyRTPpacket.h
[delete] https://crrev.com/0e1d7989a54e146974eec758056f4af97273d2b7/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.cc
[delete] https://crrev.com/0e1d7989a54e146974eec758056f4af97273d2b7/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h
[delete] https://crrev.com/0e1d7989a54e146974eec758056f4af97273d2b7/modules/audio_coding/neteq/test/RTPchange.cc
[delete] https://crrev.com/0e1d7989a54e146974eec758056f4af97273d2b7/modules/audio_coding/neteq/test/RTPjitter.cc
[delete] https://crrev.com/0e1d7989a54e146974eec758056f4af97273d2b7/modules/audio_coding/neteq/test/RTPtimeshift.cc
[add] https://crrev.com/81af414ffb103c645bf50c77413096de3eb4e532/modules/audio_coding/neteq/tools/rtp_jitter.cc

Project Member Comment 44 by henrik.lundin@webrtc.org, Nov 24
Status: Fixed
This work is now done.
Project Member Comment 45 by bugdroid1@chromium.org, Nov 28
The following revision refers to this bug:
  https://webrtc.googlesource.com/src.git/+/32f64d2ef947a0026cd2eba7d6a4bb81d961ac3f

commit 32f64d2ef947a0026cd2eba7d6a4bb81d961ac3f
Author: Henrik Lundin <henrik.lundin@webrtc.org>
Date: Tue Nov 28 12:35:38 2017

rtp_encode: Fixing bug related to DTX

Bug:  webrtc:2692 
Change-Id: I7b884b22cab21b9dce77e5599f43431bbc899f5d
Reviewed-on: https://webrtc-review.googlesource.com/26027
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20909}
[modify] https://crrev.com/32f64d2ef947a0026cd2eba7d6a4bb81d961ac3f/modules/audio_coding/neteq/tools/rtp_encode.cc

Project Member Comment 46 by bugdroid1@chromium.org, Dec 7
The following revision refers to this bug:
  https://webrtc.googlesource.com/src.git/+/f1061c2d90ba966257e06473ab6a227811e0a8a8

commit f1061c2d90ba966257e06473ab6a227811e0a8a8
Author: Henrik Lundin <henrik.lundin@webrtc.org>
Date: Thu Dec 07 09:43:17 2017

rtp_encode: Unify the encoder configs somewhat

For uniformity. Uniformity is nice.

Bug:  webrtc:2692 
Change-Id: Id85e54fa31bf3cc79e73a72805e57d5e3164252f
Reviewed-on: https://webrtc-review.googlesource.com/27400
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21135}
[modify] https://crrev.com/f1061c2d90ba966257e06473ab6a227811e0a8a8/modules/audio_coding/neteq/tools/rtp_encode.cc

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