Commit Graph

12 Commits

Author SHA1 Message Date
Tommi
254994f071 Update SyncBuffer::GetNextAudioInterleaved() to use InterleavedView
* Remove redundant error handling in the GetAudio path following
  recent stricter error handling changes in NetEqImpl.
* GetAudioInternal() now depends on SyncBuffer's
  GetNextAudioInterleaved either reading what's requested or
  returning an error. Before there was a check to see when a partial
  read happened. Instead we now checks if a read was done and the
  internal SyncBuffer read index is not changed.
* Also, minor consistency updates to neteq_->GetAudio() call sites.
  Don't set AudioFrame properties before issuing the call.
  NetEq always sets these fields as per design. Call sites that set
  the properties themselves, might mask a bug if that were to regress.

Bug: chromium:335805780
Change-Id: I18afd3cbae1ff8ba2782ad7677b1dbccb1e1f646
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391620
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#44851}
2025-06-06 02:34:43 -07:00
Tommi
ad180cf625 Make AudioVector classes and SyncBuffer final, non virtual unless needed
Bug: chromium:335805780
Change-Id: I28bd4451c7e94e044174e751eacbca49df9898d5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392321
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#44677}
2025-05-19 02:58:50 -07:00
Philipp Hancke
c4fe8256f3 IWYU modules/audio_coding
using
  find modules/audio_coding -name "*.h" -o -name "*.cc" | grep -v mock_debug_dump_writer.h | xargs tools_webrtc/iwyu/apply-include-cleaner
followed by
  tools_webrtc/gn_check_autofix.py -C out/Default/
and git cl format

Manual changes:
* controller_manager.h: add api/array_view.h include
* RTPFile.cc / dtmf_buffer_unittest.cc:
  use rtc_base/ip_address.h instea of netinet/in.h
* neteq_delay_analyzer.cc: add TODO for stringstream to ostream include
* opus: fixup paths and make third_party/opus includes go via opus_interface.h by using IWYU export pragmas

The mock_debug_dump_writer has issues with protobuf includes so was
ignored for this round.

BUG=webrtc:42226242

Change-Id: I5b8613053da6cc0a2a44d1d59bd2efe11e501681
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/387640
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#44546}
2025-05-08 05:10:13 -07:00
Evan Shrubsole
a0ea43ed2b Use webrtc name specifier instead of rtc/cricket in modules/audio_coding
WebRTC has unified all namespaces to webrtc, and the rtc:: and cricket::
name specifiers need to be replaced with webrtc::. This was generated using
a combination of clang AST rewriting tools and sed.

This CL was uploaded by git cl split.

Bug: webrtc:42232595
Change-Id: I41a494333ec6720bffb0d9f0489faa99893847ca
No-Iwyu: LSC
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/386720
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#44395}
2025-04-16 03:05:51 -07:00
Artem Titov
d3251968d1 Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED
Add implementation of RTC_DCHECK_NOTREACHED equal to the RTC_NOTREACHED.
The new macros will replace the old one when old one's usage will be
removed. The idea of the renaming to provide a clear signal that this
is debug build only macros and will be stripped in the production build.

Bug: webrtc:9065
Change-Id: I4c35d8b03e74a4b3fd1ae75dba2f9c05643101db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237802
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35348}
2021-11-15 21:44:59 +00:00
Artem Titov
d00ce747c7 Use backticks not vertical bars to denote variables in comments for /modules/audio_coding
Bug: webrtc:12338
Change-Id: I02613d9fca45d00e2477f334b7a0416e7912e26b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227037
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34621}
2021-08-02 10:45:40 +00:00
Mirko Bonadei
25ab3228f3 Replace assert() with RTC_DCHECK().
CL partially auto-generated with:

git grep -l "\bassert(" | grep "\.[c|h]" | \
  xargs sed -i 's/\bassert(/RTC_DCHECK(/g'

And with:

git grep -l "RTC_DCHECK(false)" |  \
  xargs sed -i 's/RTC_DCHECK(false)/RTC_NOTREACHED()/g'

With some manual changes to include "rtc_base/checks.h" where
needed.

A follow-up CL will remove assert() from Obj-C code as well
and remove the #include of <assert.h>.

The choice to replace with RTC_DCHECK is because assert()
is because RTC_DCHECK has similar behavior as assert()
based on NDEBUG.

This CL also contains manual changes to switch from
basic RTC_DCHECK to other (preferred) versions like
RTC_DCHECK_GT (and similar).

Bug: webrtc:6779
Change-Id: I00bed8886e03d685a2f42324e34aef2c9b7a63b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224846
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34442}
2021-07-09 07:49:43 +00:00
Jonas Olsson
a4d873786f Format almost everything.
This CL was generated by running

git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \
grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \
grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \
grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \
grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \
| xargs clang-format -i ; git cl format

Most of these changes are clang-format grouping and reordering includes
differently.

Bug: webrtc:9340
Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28505}
2019-07-08 13:45:15 +00:00
Henrik Lundin
00eb12a20c Let NetEq use the PLC output from a decoder
This change enables NetEq to use the packet concealment audio (aka
PLC) produced by a decoder. The change also includes a new API to the
AudioDecoder interface, which lets the decoder implementation generate
and deliver concealment audio.

Bug: webrtc:9180
Change-Id: Icaacebccf645d4694b0d2d6310f6f2c7132881c4
Reviewed-on: https://webrtc-review.googlesource.com/96340
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24738}
2018-09-14 07:05:20 +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
92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 05:02:56 +00:00
Mirko Bonadei
bb547203bf Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 04:25:06 +00:00