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fbeb76ab51
This reverts commit b46c4bf27b.
Reason for revert: breaks a downstream project
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: Ice138004e84e8c5f896684e8d01133d4b2a77bb7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283800
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#38655}
30 lines
993 B
Text
30 lines
993 B
Text
# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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#
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# Use of this source code is governed by a BSD-style license
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# that can be found in the LICENSE file in the root of the source
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# tree. An additional intellectual property rights grant can be found
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# in the file PATENTS. All contributing project authors may
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# be found in the AUTHORS file in the root of the source tree.
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import("../../webrtc.gni")
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audio_codec_defines = []
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if (rtc_include_ilbc) {
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audio_codec_defines += [ "WEBRTC_CODEC_ILBC" ]
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}
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if (rtc_include_opus) {
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audio_codec_defines += [ "WEBRTC_CODEC_OPUS" ]
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}
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if (rtc_opus_support_120ms_ptime) {
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audio_codec_defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=1" ]
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} else {
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audio_codec_defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=0" ]
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}
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if (current_cpu == "arm") {
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audio_codec_defines += [ "WEBRTC_CODEC_ISACFX" ]
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} else {
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audio_codec_defines += [ "WEBRTC_CODEC_ISAC" ]
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}
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audio_coding_defines = audio_codec_defines
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neteq_defines = audio_codec_defines
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