From ba2677061aa0872f742a8f4b7327e2878f461df6 Mon Sep 17 00:00:00 2001 From: Henrik Lundin Date: Wed, 16 Feb 2022 15:15:12 +0000 Subject: [PATCH] Add fuzzer test for G722 and fix a fuzzer problem The problem was fixed by implementing the methid PacketDuration() in AudioDecoderG722StereoImpl, which catches the issue in AudioDecoder::Decode(). Bug: chromium:1280851 Change-Id: I31f974b9999f3c1c62b0e5dc39bb3e56a9a9388d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251842 Reviewed-by: Minyue Li Commit-Queue: Henrik Lundin Cr-Commit-Position: refs/heads/main@{#36034} --- .../codecs/g722/audio_decoder_g722.cc | 6 +++ .../codecs/g722/audio_decoder_g722.h | 1 + test/fuzzers/BUILD.gn | 8 ++++ test/fuzzers/audio_decoder_g722_fuzzer.cc | 39 +++++++++++++++++++ 4 files changed, 54 insertions(+) create mode 100644 test/fuzzers/audio_decoder_g722_fuzzer.cc diff --git a/modules/audio_coding/codecs/g722/audio_decoder_g722.cc b/modules/audio_coding/codecs/g722/audio_decoder_g722.cc index f02ca7f896..c21ab9341f 100644 --- a/modules/audio_coding/codecs/g722/audio_decoder_g722.cc +++ b/modules/audio_coding/codecs/g722/audio_decoder_g722.cc @@ -114,6 +114,12 @@ int AudioDecoderG722StereoImpl::DecodeInternal(const uint8_t* encoded, return static_cast(ret); } +int AudioDecoderG722StereoImpl::PacketDuration(const uint8_t* encoded, + size_t encoded_len) const { + // 1/2 encoded byte per sample per channel. + return static_cast(2 * encoded_len / Channels()); +} + int AudioDecoderG722StereoImpl::SampleRateHz() const { return 16000; } diff --git a/modules/audio_coding/codecs/g722/audio_decoder_g722.h b/modules/audio_coding/codecs/g722/audio_decoder_g722.h index 39e9e630be..5872fad5de 100644 --- a/modules/audio_coding/codecs/g722/audio_decoder_g722.h +++ b/modules/audio_coding/codecs/g722/audio_decoder_g722.h @@ -57,6 +57,7 @@ class AudioDecoderG722StereoImpl final : public AudioDecoder { std::vector ParsePayload(rtc::Buffer&& payload, uint32_t timestamp) override; int SampleRateHz() const override; + int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override; size_t Channels() const override; protected: diff --git a/test/fuzzers/BUILD.gn b/test/fuzzers/BUILD.gn index a5e2e44ff1..cc8b312ab3 100644 --- a/test/fuzzers/BUILD.gn +++ b/test/fuzzers/BUILD.gn @@ -259,6 +259,14 @@ rtc_library("audio_decoder_fuzzer") { absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ] } +webrtc_fuzzer_test("audio_decoder_g722_fuzzer") { + sources = [ "audio_decoder_g722_fuzzer.cc" ] + deps = [ + ":audio_decoder_fuzzer", + "../../modules/audio_coding:g722", + ] +} + webrtc_fuzzer_test("audio_decoder_ilbc_fuzzer") { sources = [ "audio_decoder_ilbc_fuzzer.cc" ] deps = [ diff --git a/test/fuzzers/audio_decoder_g722_fuzzer.cc b/test/fuzzers/audio_decoder_g722_fuzzer.cc new file mode 100644 index 0000000000..08599aa333 --- /dev/null +++ b/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 dec; + size_t num_channels; + if (data[0] % 2) { + dec = std::make_unique(); + num_channels = 1; + } else { + dec = std::make_unique(); + 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 output = + std::make_unique(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