Commit graph

13 commits

Author SHA1 Message Date
Yury Yarashevich
87e74f9fb7 Remove unused combined_audio_video_bwe.
Bug: None
Change-Id: Ie539351f98b7a0ebb5f08e0df5c5759a2bcb5588
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306520
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Yury Yarashevich <yura.yaroshevich@gmail.com>
Cr-Commit-Position: refs/heads/main@{#40160}
2023-05-26 15:56:00 +00:00
Tommi
1def899931 Remove legacy (unused) config param: jitter_buffer_enable_rtx_handling
Bug: none
Change-Id: I14164546950cc63c37e54544cdc80bfd4eddf211
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262962
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36955}
2022-05-21 23:06:21 +00:00
Sam Zackrisson
eeffb6aaf7 Remove deprecated audio option residual_echo_detector
The feature is now enabled in other ways. See PSA or linked Monorail issue for details.
https://groups.google.com/g/discuss-webrtc/c/mJV5cDysBDI/m/7PTPBjVHCgAJ

Bug: webrtc:11539
Change-Id: I0f5816baf2bfa1508a1c85ddbd7b775417434c62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260107
Auto-Submit: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36742}
2022-05-03 07:07:58 +00:00
Alessio Bazzica
d7fdb95346 Remove typing detection
Motivation: never used.
Follow-up CL of https://webrtc-review.googlesource.com/c/src/+/250680.

Tested on a custom chromium build by making a video call and with
keyboard activity. The expected logs from the APM transient suppressor
sub-module were shown.

Bug: webrtc:11226
Change-Id: I4186994412dd8ba2e71ed8f9dcc9cf8f8e40fbd3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250667
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36300}
2022-03-23 10:23:54 +00:00
Alessio Bazzica
93348d89bc Remove unused audio options and corresponding media constraints
- experimental AGC (aka googAutoGainControl2) removed in [1]
- experimental NS (aka googNoiseSuppression2) removed in [2]
- typing noise detection (aka googTypingNoiseDetection)
  removed in [3]
- cricket::AudioOptions::tx_agc_ are unused

[1] https://webrtc-review.googlesource.com/c/src/+/219463
[2] https://webrtc-review.googlesource.com/c/src/+/232128
[3] https://chromium-review.googlesource.com/c/chromium/src/+/1617352

Bug: webrtc:11226
Change-Id: Id1ecef3d3e193c210fc11832e16db4f84d866d14
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250680
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35987}
2022-02-14 10:50:20 +00:00
Xavier Lepaul
1e12f2a800 Add an option to avoid early initialization of audio capture
This can cause issues on Android S if this initialization happens when
the app doesn't have permission to access the microphone.

Bug: b/197461765
Change-Id: Iebccff9d15f5bb12a7b2c78e1c373e379b37a127
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246104
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Xavier Lepaul‎ <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35689}
2022-01-13 17:06:09 +00:00
Per Åhgren
f40a340756 Remove deprecated code related to AEC2
This CL removes code related to the usage of the delay agnostic and
extended filter modes in AEC2.

Bug: webrtc:8671
Change-Id: I1a2c7a9eba54b03f5a015df3adb617785f52a939
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133912
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28789}
2019-08-07 10:09:36 +00:00
Jakob Ivarsson
53eae87bf8 Add PeerConnection option to enable RTX handling in the audio jitter buffer.
Bug: webrtc:10178
Change-Id: I70abce0c7b74124d2b1978d9a5eb8216b6233d1a
Reviewed-on: https://webrtc-review.googlesource.com/c/116784
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26203}
2019-01-10 16:28:43 +00:00
Yves Gerey
3e70781361 [Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
This is a follow-up to
https://webrtc-review.googlesource.com/c/src/+/106280.
This time the whole code base is covered.
Some files may have not been fixed though, whenever the IWYU tool
was breaking the build.

Bug: webrtc:8311
Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef
Reviewed-on: https://webrtc-review.googlesource.com/c/111965
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 18:25:07 +00:00
Jakob Ivarsson
10403ae87c Add PeerConnection option to configure minimum audio jitter buffer delay.
Note that this value will override the minimum delay that is used for audio/video sync.

Bug: webrtc:10053
Change-Id: Ia129f6c9ee9da5d00a3d955afaaa6e8f0c2bee33
Reviewed-on: https://webrtc-review.googlesource.com/c/112121
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25805}
2018-11-27 19:49:48 +00:00
Sam Zackrisson
be65d4886a Remove AECM comfort noise setting from API
The internal functionality has already been disabled.
The default - no comfort noise - is now the only option.

Bug: webrtc:9535
Change-Id: Idcf233625857c0120c7b355048e24ef3124196c1
Reviewed-on: https://webrtc-review.googlesource.com/c/102560
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25199}
2018-10-16 09:42:16 +00:00
Danil Chapovalov
2165233874 Uninline non-trivial AudioOptions functions
reimplement ToString using rtc::SimpleStringBuilder instead of std::ostringstream
Side effect: ToString converts booleans as 0/1 instead of false/true.

Bug: None
Change-Id: I8a57d208b016d3af5a09f7dc2e2ec4e5634446fa
Reviewed-on: https://webrtc-review.googlesource.com/95080
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24512}
2018-08-31 16:25:48 +00:00
Paulina Hensman
11b34f4d08 Remove chromium clang style errors affecting sdk/android/media_jni
Bug: webrtc:163
Change-Id: I1e98174817ca032ee13f9a6a386803382843389d
Reviewed-on: https://webrtc-review.googlesource.com/67360
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22796}
2018-04-09 13:55:49 +00:00