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}
29 lines
1.1 KiB
C++
29 lines
1.1 KiB
C++
/*
|
|
* Copyright (c) 2015 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.
|
|
*/
|
|
|
|
#include "modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h"
|
|
#include "test/fuzzers/audio_decoder_fuzzer.h"
|
|
|
|
namespace webrtc {
|
|
void FuzzOneInput(const uint8_t* data, size_t size) {
|
|
if (size > 20000) {
|
|
return;
|
|
}
|
|
const int sample_rate_hz = size % 2 == 0 ? 16000 : 32000; // 16 or 32 kHz.
|
|
static const size_t kAllocatedOuputSizeSamples = 32000 / 10; // 100 ms.
|
|
int16_t output[kAllocatedOuputSizeSamples];
|
|
AudioDecoderIsacFloatImpl::Config c;
|
|
c.sample_rate_hz = sample_rate_hz;
|
|
AudioDecoderIsacFloatImpl dec(c);
|
|
FuzzAudioDecoder(DecoderFunctionType::kNormalDecode, data, size, &dec,
|
|
sample_rate_hz, sizeof(output), output);
|
|
}
|
|
} // namespace webrtc
|