Artem Titov 3ecec176a8 Extract third party part of g711 codec into separate target ...

Bug: webrtc:8366
Change-Id: I34c7ea707213e0c1a50826896da01f70c072eae5
Reviewed-on: https://webrtc-review.googlesource.com/84741
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23724}

2018-06-25 11:26:59 +00:00

..

cng

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

g711

Extract third party part of g711 codec into separate target

2018-06-25 11:26:59 +00:00

g722

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

ilbc

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

isac

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

opus

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

pcm16b

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

red

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

tools

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

audio_decoder.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

audio_encoder.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

audio_format_conversion.cc

Replace rtc::Optional with absl::optional in modules/audio_coding

2018-06-19 12:46:20 +00:00

audio_format_conversion.h

Adding NOLINT for typedefs.h and common_types.h

2017-09-15 13:03:51 +00:00

builtin_audio_decoder_factory_unittest.cc

Replace rtc::Optional with absl::optional in modules/audio_coding

2018-06-19 12:46:20 +00:00

builtin_audio_encoder_factory_unittest.cc

Replace rtc::Optional with absl::optional in modules/audio_coding

2018-06-19 12:46:20 +00:00

legacy_encoded_audio_frame_unittest.cc

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

legacy_encoded_audio_frame.cc

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

legacy_encoded_audio_frame.h

Replace rtc::Optional with absl::optional in modules/audio_coding

2018-06-19 12:46:20 +00:00

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