10 Commits

Author SHA1 Message Date
Tommi
5d3e6805f2 Add audio view classes
From the new header file:
* MonoView<>: A single channel contiguous buffer of samples.
* InterleavedView<>: Channel samples are interleaved (side-by-side) in
  the buffer. A single channel InterleavedView<> is the same thing as a
  MonoView<>
* DeinterleavedView<>: Each channel's samples are contiguous within the
  buffer. Channels can be enumerated and accessing the
  individual channel data is done via MonoView<>.

There are also a few utility functions that offer a unified way to check
the properties regardless of what view type is in use.

Bug: chromium:335805780
Change-Id: I28196f8f4ded4fadc72ee32b62af304c62f4fc47
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349300
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42377}
2024-05-24 18:08:37 +00:00
Tommi
1a436f7e9e Remove AudioFrameOperations::Add, ApplyHalfGain and Scale.
These methods are unused.

Bug: none
Change-Id: If1499c7c0bc925c2504b7a1318b2d7c4fc4240b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349500
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42214}
2024-05-02 19:39:20 +00:00
Tommi
57b09eca64 Update AudioFrameOperations to require ArrayView
Bug: chromium:335805780
Change-Id: I14d97315f4cffa21bcc11b063e86c5adcebe78ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/348800
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42204}
2024-04-30 21:26:56 +00:00
Tommi
909f3a5339 Rename several more tests that use EXPECT_DEATH to *DeathTest.
Bug: webrtc:11577
Change-Id: I0397ee933464496e4885bb0f8030f3d669e5e612
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175641
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31309}
2020-05-18 16:10:04 +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
Alex Loiko
b4977de306 Receive-side ready for multiple channels.
Made path from NetEq to AudioTransport ready for many-channel audio.
If there is one stream, we can handle anything that fits in an
AudioFrame. For many streams, the current limit is 6.

Some multi-channel combinations are not supported: e.g. if we get
stereo audio and attempt to play out 6 channels.

Changes:
* AudioFrameOperations - replaced the MonoTo* and *ToMono methods by
  UpmixChannels & DownmixChannels.
* AudioMixer: removed DCHECKs for <= 2 channels and tweaked the mixing
  algorithm to handle many channels.

Bug: webrtc:8649
Change-Id: Ib83e16d463694e35658caa09c27849e853d508fb
Reviewed-on: https://webrtc-review.googlesource.com/c/106040
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26446}
2019-01-29 12:43:23 +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
Fredrik Solenberg
bbf21a3fd6 Remove dependencies on modules:module_api from AudioProcessing.
- Directly include api/audio/audio_frame.h everywhere AudioFrame is used.
- This *will* remove transient dependencies on libjpeg and a bunch of other things from the e.g. APM.
- audio_frame.h still included from module_common_types.h for backwards compatibility with clients.

Bug: webrtc:9139, webrtc:7504
Change-Id: Id96f9268c01667fbcc29a01f5c1dd25a37836897
Reviewed-on: https://webrtc-review.googlesource.com/62464
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22845}
2018-04-12 22:05:27 +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