Commit graph

19 commits

Author SHA1 Message Date
Joachim Reiersen
5075cb4a60 Expose AudioLevel as an absl::optional struct in api/rtp_headers.h
Start migrating away from `hasAudioLevel`, `voiceActivity`, `audioLevel` fields in RTPHeaderExtension and switch usages to a more modern absl::optional<AudioLevel> accessor instead.

The old fields are preserved for compatibility with downstream projects, but will be removed in the future.

Bug: webrtc:15788
Change-Id: I76599124fd68dd4d449f850df3b9814d6a002f5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336303
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41947}
2024-03-22 10:07:47 +00:00
Joachim Reiersen
4a97488714 Rename AudioLevel to AudioLevelExtension in rtp_header_extensions.h
To prepare for a new AudioLevel struct to be added to the public WebRTC API, rename the internal RTP extension reader/writer class to AudioLevelExtension. A temporary alias is provided to avoid breaking downstream projects.

Bug: webrtc:15788
Change-Id: Ie231668f25932fd9b539229114128b1d0b949a6e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339887
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41787}
2024-02-22 23:12:52 +00:00
Harald Alvestrand
d43af9172b Remove internal overrides using old SendRtp and SendRtcp interfaces.
This CL takes away all usages except for Android code.

Low-Coverage-Reason: Refactoring old code
Bug: webrtc:15410
Change-Id: I66bed6a4a2787b4177a82e599b48623ca67cd235
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315940
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40554}
2023-08-15 13:20:21 +00:00
Artem Titov
8a9f3a8f53 Reland "Remove dependency of video_replay on TestADM."
This reverts commit f9e3bdd2ce.

Reason for revert: reland with fix

Original change's description:
> Revert "Remove dependency of video_replay on TestADM."
>
> This reverts commit 01716663a9.
>
> Reason for revert:  breaking CallPerfTest
> https://ci.chromium.org/ui/p/webrtc/builders/perf/Perf%20Android32%20(R%20Pixel5)/967/overview 
>
> Original change's description:
> > Remove dependency of video_replay on TestADM.
> >
> > This should remove requirement to build TestADM in chromium build.
> >
> > Bug: b/272350185, webrtc:15081
> > Change-Id: Iceb8862aa81099c22bd378ae692229f01ab3314c
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302380
> > Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Commit-Queue: Artem Titov <titovartem@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#39934}
>
> Bug: b/272350185, webrtc:15081
> Change-Id: I73aa0fd3c3d8c244d20e5f29f5792a4c7d7e4165
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303160
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Owners-Override: Jeremy Leconte <jleconte@google.com>
> Commit-Queue: Jeremy Leconte <jleconte@google.com>
> Cr-Commit-Position: refs/heads/main@{#39939}

Bug: b/272350185, webrtc:15081
Change-Id: I360ef3e140e60fc21d622480d1f3326e40a76f58
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303400
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Auto-Submit: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39946}
2023-04-25 09:39:22 +00:00
Jeremy Leconte
f9e3bdd2ce Revert "Remove dependency of video_replay on TestADM."
This reverts commit 01716663a9.

Reason for revert:  breaking CallPerfTest
https://ci.chromium.org/ui/p/webrtc/builders/perf/Perf%20Android32%20(R%20Pixel5)/967/overview 

Original change's description:
> Remove dependency of video_replay on TestADM.
>
> This should remove requirement to build TestADM in chromium build.
>
> Bug: b/272350185, webrtc:15081
> Change-Id: Iceb8862aa81099c22bd378ae692229f01ab3314c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302380
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39934}

Bug: b/272350185, webrtc:15081
Change-Id: I73aa0fd3c3d8c244d20e5f29f5792a4c7d7e4165
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303160
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#39939}
2023-04-24 19:02:23 +00:00
Artem Titov
01716663a9 Remove dependency of video_replay on TestADM.
This should remove requirement to build TestADM in chromium build.

Bug: b/272350185, webrtc:15081
Change-Id: Iceb8862aa81099c22bd378ae692229f01ab3314c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302380
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39934}
2023-04-24 13:17:45 +00:00
Markus Handell
f4f22872d0 CallTest: migrate timeouts to TimeDelta.
Bug: webrtc:13756
Change-Id: I1b6675dfd1f0b9f3868c0db81d24e9a80d90657d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271483
Auto-Submit: Markus Handell <handellm@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37794}
2022-08-16 12:06:54 +00:00
Tommi
3176ef79e9 Rename AudioReceiveStream to AudioReceiveStreamInterface
Bug: webrtc:7484
Change-Id: I22eaa7a9e082fc575cf7471d7a2f4f706564d54f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262805
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36965}
2022-05-23 08:44:26 +00:00
Jakob Ivarsson
47a03e8743 Default enable sending transport sequence numbers on audio packets.
This enables send side bandwidth estimation for audio and removes field
trial "WebRTC-Audio-SendSideBwe" which this was controlled through.

Transport-cc extension still needs to be negotiated.

Bug: webrtc:12222
Change-Id: Ie2268fad13703eeb0f0d38fcf484baaa29715b7c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/194142
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32681}
2020-11-24 09:19:54 +00:00
Danil Chapovalov
1b4e4bf42e Migrate several call tests from legacy RtpHeaderParser to RtpPacket parsing.
Bug: None
Change-Id: I96f11922b3cd66eb02093fa7e6e4d21774c19575
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161323
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30166}
2020-01-07 13:02:52 +00:00
Elad Alon
d8d3248d95 Reland "Delete test/constants.h"
This reverts commit 4f36b7a478.

Reason for revert: Failing tests fixed.

Original change's description:
> Revert "Delete test/constants.h"
>
> This reverts commit 389b1672a3.
>
> Reason for revert: Causes failure (and empty result list) in CallPerfTest.PadsToMinTransmitBitrate
>
> Original change's description:
> > Delete test/constants.h
> >
> > It's not possible to use constants.h for all RTP extensions
> > after the number of extensions exceeds 14, which is the maximum
> > number of one-byte RTP extensions. This is because some extensions
> > would have to be assigned a number greater than 14, even if the
> > test only involves 14 extensions or less.
> >
> > For uniformity's sake, this CL also edits some files to use an
> > enum as the files involved in this CL, rather than free-floating
> > const-ints.
> >
> > Bug: webrtc:10288
> > Change-Id: Ib5e58ad72c4d3756f4c4f6521f140ec59617f3f5
> > Reviewed-on: https://webrtc-review.googlesource.com/c/123048
> > Commit-Queue: Elad Alon <eladalon@webrtc.org>
> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#26728}
>
> TBR=danilchap@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org
>
> Bug: webrtc:10288, chromium:933127
> Change-Id: If1de0bd8992137c52bf0b877b3cb0a2bafc809d4
> Reviewed-on: https://webrtc-review.googlesource.com/c/123381
> Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26744}

TBR=danilchap@webrtc.org,oprypin@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org

Change-Id: I65e391325d3a6df6db3c0739185e2002e70fb954
Bug: webrtc:10288, chromium:933127
Reviewed-on: https://webrtc-review.googlesource.com/c/123384
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26750}
2019-02-19 08:51:20 +00:00
Oleh Prypin
4f36b7a478 Revert "Delete test/constants.h"
This reverts commit 389b1672a3.

Reason for revert: Causes failure (and empty result list) in CallPerfTest.PadsToMinTransmitBitrate

Original change's description:
> Delete test/constants.h
>
> It's not possible to use constants.h for all RTP extensions
> after the number of extensions exceeds 14, which is the maximum
> number of one-byte RTP extensions. This is because some extensions
> would have to be assigned a number greater than 14, even if the
> test only involves 14 extensions or less.
>
> For uniformity's sake, this CL also edits some files to use an
> enum as the files involved in this CL, rather than free-floating
> const-ints.
>
> Bug: webrtc:10288
> Change-Id: Ib5e58ad72c4d3756f4c4f6521f140ec59617f3f5
> Reviewed-on: https://webrtc-review.googlesource.com/c/123048
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26728}

TBR=danilchap@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org

No-Presubmit: True
Bug: webrtc:10288, chromium:933127
Change-Id: If1de0bd8992137c52bf0b877b3cb0a2bafc809d4
Reviewed-on: https://webrtc-review.googlesource.com/c/123381
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26744}
2019-02-18 18:09:22 +00:00
Elad Alon
389b1672a3 Delete test/constants.h
It's not possible to use constants.h for all RTP extensions
after the number of extensions exceeds 14, which is the maximum
number of one-byte RTP extensions. This is because some extensions
would have to be assigned a number greater than 14, even if the
test only involves 14 extensions or less.

For uniformity's sake, this CL also edits some files to use an
enum as the files involved in this CL, rather than free-floating
const-ints.

Bug: webrtc:10288
Change-Id: Ib5e58ad72c4d3756f4c4f6521f140ec59617f3f5
Reviewed-on: https://webrtc-review.googlesource.com/c/123048
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26728}
2019-02-17 21:47:41 +00:00
Per Kjellander
914351de5c Reland "Always offer transport sequence number header extension for audio""
(reverted in https://webrtc-review.googlesource.com/c/src/+/123182/1)

Original cl description:
Always offer transport sequence number header extension for audio

If the extension is negotiated, it will only be used if
the field trial WebRTC-Audio-SendSideBwe is enabled.
This allows simpler experimentation if it should be used or not.

Patchset 3 contain the only change:
  Add the field trial WebRTC-Audio-SendSideBwe to  call/rampup_tests.cc

TBR: srte@webrtc.org,ossu@webrtc.org
Bug: webrtc:10309 webrtc:10286
Change-Id: I2c1224e8a9fab52c1030369c1364686322e88a0f
Reviewed-on: https://webrtc-review.googlesource.com/c/123183
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26706}
2019-02-15 10:57:38 +00:00
Ying Wang
397c06fe9d Revert "Always offer transport sequence number header extension for audio"
This reverts commit fd965c008c.

Reason for revert: Cause test failure.

Original change's description:
> Always offer transport sequence number header extension for audio
> 
> If the extension is negotiated, it will only be used if
> the field trial WebRTC-Audio-SendSideBwe is enabled.
> This allows simpler experimentation if it should be used or not.
> 
> Bug: webrtc:10309 webrtc:10286
> Change-Id: I797e6f14c06d46189e40f6d09805c2e09afc015b
> Reviewed-on: https://webrtc-review.googlesource.com/c/122542
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26689}

TBR=ossu@webrtc.org,sprang@webrtc.org,srte@webrtc.org,perkj@webrtc.org

Change-Id: I1b7d3fa5c282a5bf049ca54695ad16c8278a2698
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10309 webrtc:10286
Reviewed-on: https://webrtc-review.googlesource.com/c/123182
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26703}
2019-02-15 08:53:25 +00:00
Per Kjellander
fd965c008c Always offer transport sequence number header extension for audio
If the extension is negotiated, it will only be used if
the field trial WebRTC-Audio-SendSideBwe is enabled.
This allows simpler experimentation if it should be used or not.

Bug: webrtc:10309 webrtc:10286
Change-Id: I797e6f14c06d46189e40f6d09805c2e09afc015b
Reviewed-on: https://webrtc-review.googlesource.com/c/122542
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26689}
2019-02-14 15:28:07 +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
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
solenberg
18f5427e4c Remove voe_auto_test and add new tests to cover the missing cases.
BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/3007383002
Cr-Commit-Position: refs/heads/master@{#19865}
2017-09-15 16:56:08 +00:00