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webrtc/modules/audio_coding/codecs/opus
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Jakob Ivarsson 918eb19303 Fix crash when Opus maxptime < 20ms. 
A follow up cl will be created to better handle nullopt frame length range in AudioSendStream.

Note that maxptime is still not used for setting the frame length (only ptime is).

Bug: chromium:1109337
Change-Id: Id21fd8c76a6c4a0c85719a955116f8d16001a3d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284501
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38702}
2022-11-22 01:21:24 +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.cc Fix crash when Opus maxptime < 20ms. 2022-11-22 01:21:24 +00:00
audio_encoder_opus.h Adopt absl::string_view in modules/audio_coding/ 2022-07-20 13:34:23 +00:00
audio_encoder_opus_unittest.cc Fix crash when Opus maxptime < 20ms. 2022-11-22 01:21:24 +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 Use backticks not vertical bars to denote variables in comments for /modules/audio_coding 2021-08-02 10:45:40 +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 Use backticks not vertical bars to denote variables in comments 2021-08-10 10:40:03 +00:00