fbeb76ab51
This reverts commit b46c4bf27ba5c417fcba7f200d80fa4634e7e1a1. 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}
31 lines
993 B
Plaintext
31 lines
993 B
Plaintext
# 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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