webrtc/modules/audio_coding
Tommi 19c51ea537 Use std::array<> consistently for reusable audio buffers.
This is a minor change for places where we use
AudioFrame::kMaxDataSizeSamples sized intermediary buffers. The change
uses `std::array<>` instead of C style arrays which allows for use
of utility templates that incorporate type based buffer size checking.
Also adding `ClearSamples()` method, which complements CopySamples.

Bug: chromium:335805780
Change-Id: I813feb32937e020ceb9ca4b00632dc90907c93fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351681
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42400}
2024-05-29 09:20:36 +00:00
..
acm2 Use std::array<> consistently for reusable audio buffers. 2024-05-29 09:20:36 +00:00
audio_network_adaptor Remove expired field trial UseTwccPlrForAna 2024-04-15 14:26:33 +00:00
codecs Remove expired WebRTC-Audio-OpusSetSignalVoiceWithDtx 2024-04-05 07:49:33 +00:00
g3doc Reland "Migrate WebRTC documentation to new renderer" 2023-01-31 09:30:04 +00:00
include Format /modules 2023-04-20 02:02:45 +00:00
neteq Revert "Propagate arrival time inside NetEq" 2024-05-27 17:17:04 +00:00
test Reland "Unify access to SDP codec parameters" 2024-01-03 12:03:11 +00:00
audio_coding.gni Reland "[ACM] iSAC audio codec removed" 2022-11-17 12:52:35 +00:00
BUILD.gn Revert "Propagate arrival time inside NetEq" 2024-05-27 17:17:04 +00:00
DEPS Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
OWNERS Add jakobi to modules/audio_coding OWNERS 2021-06-18 11:52:58 +00:00