9 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
1f3679884c Start using ArrayView in AudioFrame, update PushResampler
Start introducing ArrayView to AudioFrame and code that flows down
from there.  In this first step:
* Add `data_view()` that returns a read-only ArrayView for the
  audio buffer. When AudioFrame is not initialized however, data_view()
  will return a nullptr whereas the current data() method never returns
  nullptr.
* Add `mutable_data()` that requires two arguments for properly setting
  the samples per channel and number of channels that's required for
  accurately reserving the returned mutable ArrayView.
  A notable behavior change is that if the requested number of channels
  is larger than supported or the calculated buffer size is too large,
  the function will trigger a check.
* Add TODOs for following work.

Bug: chromium:335805780
Change-Id: I2937de800422589ebe6a3840b3caadf3d9ff8b00
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347982
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42202}
2024-04-30 15:33:08 +00:00
Jakob Ivarsson
4a97d7281f Remove NetEq extra delay option.
Bug: b/156734419
Change-Id: I787e6961ad283990d633029c0cf296e10b825875
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237403
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35326}
2021-11-09 17:25:46 +00:00
Henrik Lundin
c49e9c253f Adding a delay line to NetEq's output
This change adds an optional delay to NetEq's output. Note, this is not
equivalent to increasing the jitter buffer with the same extra length.

Bug: b/156734419
Change-Id: I8b70b6b3bffcfd3da296ccf29853864baa03d6bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175110
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31343}
2020-05-25 12:03:39 +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
henrika
2250b05778 Adding support for channel mixing between different channel layouts.
Two new classes are added to WebRTC from Chrome: ChannelMixer and
ChannelMixingMatrix but they are not yet utilized in the audio path for
WebRTC.

The idea is to utilize these new classes when adding support for multi-
channel encoding/decoding in WebRTC/Chrome.

Adds support for a new enumerator call webrtc::ChannelLayout and some
helper methods which maps between channel layout and number of channels.
These parts are also copied from Chrome.

Minor (cosmetic) changes are also done on the AudioFrame to prepare
for upcoming work.

Bug: webrtc:10783
Change-Id: I6cd7a13a3bc1c8bbfa19bc974c7a011d22d19197
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141674
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28482}
2019-07-04 10:10:54 +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
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