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-rw-r--r--third_party/libwebrtc/test/fuzzers/audio_decoder_g722_fuzzer.cc39
1 files changed, 39 insertions, 0 deletions
diff --git a/third_party/libwebrtc/test/fuzzers/audio_decoder_g722_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/audio_decoder_g722_fuzzer.cc
new file mode 100644
index 0000000000..08599aa333
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/audio_decoder_g722_fuzzer.cc
@@ -0,0 +1,39 @@
+/*
+ * Copyright (c) 2022 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/g722/audio_decoder_g722.h"
+#include "test/fuzzers/audio_decoder_fuzzer.h"
+
+namespace webrtc {
+void FuzzOneInput(const uint8_t* data, size_t size) {
+ if (size > 10000 || size < 1) {
+ return;
+ }
+
+ std::unique_ptr<AudioDecoder> dec;
+ size_t num_channels;
+ if (data[0] % 2) {
+ dec = std::make_unique<AudioDecoderG722Impl>();
+ num_channels = 1;
+ } else {
+ dec = std::make_unique<AudioDecoderG722StereoImpl>();
+ num_channels = 2;
+ }
+ // Allocate a maximum output size of 100 ms.
+ const int sample_rate_hz = dec->SampleRateHz();
+ const size_t allocated_ouput_size_samples =
+ sample_rate_hz / 10 * num_channels;
+ std::unique_ptr<int16_t[]> output =
+ std::make_unique<int16_t[]>(allocated_ouput_size_samples);
+ FuzzAudioDecoder(
+ DecoderFunctionType::kNormalDecode, data, size, dec.get(), sample_rate_hz,
+ allocated_ouput_size_samples * sizeof(int16_t), output.get());
+}
+} // namespace webrtc