5 Commits

Author SHA1 Message Date
Arun Raghavan
c6abf6cd3f Bump to WebRTC M120 release
Some API deprecation -- ExperimentalAgc and ExperimentalNs are gone.
We're continuing to carry iSAC even though it's gone upstream, but maybe
we'll want to drop that soon.
2024-12-24 11:05:39 -05:00
Martin Jansa
cdec109331 file_utils.h: Fix build with gcc-13
* add missing include as reported by gcc-13:
webrtc/modules/audio_processing/transient/file_utils.cc:11:
../webrtc-audio-processing-1.0/webrtc/modules/audio_processing/transient/file_utils.h:36:35: error: 'uint8_t' does not name a type
   36 | int ConvertByteArrayToFloat(const uint8_t bytes[4], float* out);
      |                                   ^~~~~~~
webrtc/modules/audio_processing/transient/file_utils.h:17:1: note: 'uint8_t' is defined in header '<cstdint>'; did you forget to '#include <cstdint>'?
   16 | #include "rtc_base/system/file_wrapper.h"
  +++ |+#include <cstdint>
   17 |

Signed-off-by: Martin Jansa <Martin.Jansa@gmail.com>
2023-05-25 18:13:04 -04:00
Arun Raghavan
bcec8b0b21 Update to current webrtc library
This is from the upstream library commit id
3326535126e435f1ba647885ce43a8f0f3d317eb, corresponding to Chromium
88.0.4290.1.
2020-10-23 13:30:23 -04:00
Arun Raghavan
34abadd258 Update code to current Chromium master
This corresponds to:

Chromium: 6555f9456074c0c0e5f7713564b978588ac04a5d
webrtc: c8b569e0a7ad0b369e15f0197b3a558699ec8efa
2015-11-04 13:11:30 +05:30
Arun Raghavan
753eada3aa Update audio_processing module
Corresponds to upstream commit 524e9b043e7e86fd72353b987c9d5f6a1ebf83e1

Update notes:

 * Pull in third party license file

 * Replace .gypi files with BUILD.gn to keep track of what changes
   upstream

 * Bunch of new filse pulled in as dependencies

 * Won't build yet due to changes needed on top of these
2015-10-15 16:18:45 +05:30