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webrtc_m130/modules/audio_coding/codecs/opus
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Jared Siskin c018bae807 Format /modules 
git ls-files | grep -e  "\(\.h\|\.cc\)$" | grep -e  "^modules/" | xargs clang-format -i ; git cl format
after landing: add to .git-blame-ignore-revs

Bug: webrtc:15082
Change-Id: I2c3cd28740062794f8c10e39d8406aadb9e9a35a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301620
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Jared Siskin <jtsiskin@meta.com>
Cr-Commit-Position: refs/heads/main@{#39901}
2023-04-20 02:02:45 +00:00
..
test
Remove dependency on rtc_base_approved from most targets
2022-04-25 12:15:30 +00:00
audio_coder_opus_common.cc
Adopt absl::string_view in modules/audio_coding/
2022-07-20 13:34:23 +00:00
audio_coder_opus_common.h
Adopt absl::string_view in modules/audio_coding/
2022-07-20 13:34:23 +00:00
audio_decoder_multi_channel_opus_impl.cc
Always call IsOk() to ensure audio codec configuration is valid when negotiating.
2021-11-26 10:11:21 +00:00
audio_decoder_multi_channel_opus_impl.h
Remove RTC_DISALLOW_COPY_AND_ASSIGN from modules/
2022-01-24 11:50:20 +00:00
audio_decoder_multi_channel_opus_unittest.cc
Always call IsOk() to ensure audio codec configuration is valid when negotiating.
2021-11-26 10:11:21 +00:00
audio_decoder_opus.cc
AudioDecoderOpus: Add support for 16 kHz output sample rate
2019-05-29 12:42:38 +00:00
audio_decoder_opus.h
Remove RTC_DISALLOW_COPY_AND_ASSIGN from modules/
2022-01-24 11:50:20 +00:00
audio_encoder_multi_channel_opus_impl.cc
Always call IsOk() to ensure audio codec configuration is valid when negotiating.
2021-11-26 10:11:21 +00:00
audio_encoder_multi_channel_opus_impl.h
Remove RTC_DISALLOW_COPY_AND_ASSIGN from modules/
2022-01-24 11:50:20 +00:00
audio_encoder_multi_channel_opus_unittest.cc
Always call IsOk() to ensure audio codec configuration is valid when negotiating.
2021-11-26 10:11:21 +00:00
audio_encoder_opus_unittest.cc
Remove default enabled field trial WebRTC-SendSideBwe-WithOverhead
2022-11-30 20:19:36 +00:00
audio_encoder_opus.cc
Remove default enabled field trial WebRTC-SendSideBwe-WithOverhead
2022-11-30 20:19:36 +00:00
audio_encoder_opus.h
Remove default enabled field trial WebRTC-SendSideBwe-WithOverhead
2022-11-30 20:19:36 +00:00
DEPS
Don't add libopus to public_deps, its headers are only used directly
2022-06-28 19:13:14 +00:00
opus_bandwidth_unittest.cc
Update Opus tests for Opus 1.3
2020-03-05 08:53:37 +00:00
opus_complexity_unittest.cc
Migrate audio perf tests on new perf metrics export API
2022-09-25 18:55:50 +00:00
opus_fec_test.cc
Delete rtc_base/format_macros.h
2022-05-09 12:03:21 +00:00
opus_inst.h
Don't add libopus to public_deps, its headers are only used directly
2022-06-28 19:13:14 +00:00
opus_interface.cc
Format /modules
2023-04-20 02:02:45 +00:00
opus_interface.h
Avoiding the noise pumping during DTX regions by just forwarding the refresh DTX packets that decrease the comfort noise level at the decoder.
2021-02-05 10:05:25 +00:00
opus_speed_test.cc
WebRTC Opus C interface: Add support for non-48 kHz decode sample rate
2019-05-29 10:33:03 +00:00
opus_unittest.cc
Stop setting OPUS_SIGNAL_VOICE when DTX is enabled.
2022-12-20 11:06:48 +00:00