mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-12 21:30:45 +01:00
17887eb04a
This is a reland of commit b46c4bf27b
Original change's description:
> [ACM] iSAC audio codec removed
>
> Note: this CL has to leave behind one part of iSAC, which is its VAD
> currently used by AGC1 in APM. The target visibility has been
> restricted and the VAD will be removed together with AGC1 when the
> time comes.
>
> Tested: see https://chromium-review.googlesource.com/c/chromium/src/+/4013319
>
> Bug: webrtc:14450
> Change-Id: I69cc518b16280eae62a1f1977cdbfa24c08cf5f9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282421
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38652}
Bug: webrtc:14450
Change-Id: Ia22c4d7724b6022238235fede93e36e570a49376
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283843
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38665}
25 lines
854 B
Text
25 lines
854 B
Text
# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
|
|
#
|
|
# Use of this source code is governed by a BSD-style license
|
|
# that can be found in the LICENSE file in the root of the source
|
|
# tree. An additional intellectual property rights grant can be found
|
|
# in the file PATENTS. All contributing project authors may
|
|
# be found in the AUTHORS file in the root of the source tree.
|
|
|
|
import("../../webrtc.gni")
|
|
|
|
audio_codec_defines = []
|
|
if (rtc_include_ilbc) {
|
|
audio_codec_defines += [ "WEBRTC_CODEC_ILBC" ]
|
|
}
|
|
if (rtc_include_opus) {
|
|
audio_codec_defines += [ "WEBRTC_CODEC_OPUS" ]
|
|
}
|
|
if (rtc_opus_support_120ms_ptime) {
|
|
audio_codec_defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=1" ]
|
|
} else {
|
|
audio_codec_defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=0" ]
|
|
}
|
|
|
|
audio_coding_defines = audio_codec_defines
|
|
neteq_defines = audio_codec_defines
|