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chore: cherry-pick ac341df436 from webrtc (#33671)

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Co-authored-by: Electron Bot <electron@github.com>
This commit is contained in:
Pedro Pontes
2022-04-28 16:26:57 +02:00
committed by GitHub
parent af7d3b7171
commit 5b4acf406c
2 changed files with 112 additions and 0 deletions
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1
patches/webrtc/.patches
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@@ -1,2 +1,3 @@
add_thread_local_to_x_error_trap_cc.patch
adding_fuzzer_for_pcm16b_decoder_and_fixing_a_fuzzer_problem.patch
cherry-pick-a18fddcb53e6.patch

111
patches/webrtc/adding_fuzzer_for_pcm16b_decoder_and_fixing_a_fuzzer_problem.patch Normal file
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@@ -0,0 +1,111 @@
From 0000000000000000000000000000000000000000 Mon Sep 17 00:00:00 2001
From: Henrik Lundin <henrik.lundin@webrtc.org>
Date: Tue, 15 Feb 2022 15:13:34 +0000
Subject: Adding fuzzer for PCM16b decoder and fixing a fuzzer problem
Bug: chromium:1280852
Change-Id: I7f6c5de86ceee01156743c0389c59f875e53bb5f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251580
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36005}
diff --git a/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc b/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc
index 1dd2ff289ee10280963c3db5de58545886c60d49..7761efe8b3b2d1627021e9f19dfb25ca871be049 100644
--- a/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc
+++ b/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc
@@ -42,7 +42,12 @@ int AudioDecoderPcm16B::DecodeInternal(const uint8_t* encoded,
int16_t* decoded,
SpeechType* speech_type) {
RTC_DCHECK_EQ(sample_rate_hz_, sample_rate_hz);
- size_t ret = WebRtcPcm16b_Decode(encoded, encoded_len, decoded);
+ // Adjust the encoded length down to ensure the same number of samples in each
+ // channel.
+ const size_t encoded_len_adjusted =
+ PacketDuration(encoded, encoded_len) * 2 *
+ Channels(); // 2 bytes per sample per channel
+ size_t ret = WebRtcPcm16b_Decode(encoded, encoded_len_adjusted, decoded);
*speech_type = ConvertSpeechType(1);
return static_cast<int>(ret);
}
diff --git a/test/fuzzers/BUILD.gn b/test/fuzzers/BUILD.gn
index 2e94f40fa2f9deefc867428e1a80aaec3a2d822b..e32da276c91ea67e02eeadf826947e2903090c61 100644
--- a/test/fuzzers/BUILD.gn
+++ b/test/fuzzers/BUILD.gn
@@ -308,6 +308,14 @@ webrtc_fuzzer_test("audio_decoder_multiopus_fuzzer") {
]
}
+webrtc_fuzzer_test("audio_decoder_pcm16b_fuzzer") {
+ sources = [ "audio_decoder_pcm16b_fuzzer.cc" ]
+ deps = [
+ ":audio_decoder_fuzzer",
+ "../../modules/audio_coding:pcm16b",
+ ]
+}
+
rtc_library("audio_encoder_fuzzer") {
testonly = true
sources = [
diff --git a/test/fuzzers/audio_decoder_pcm16b_fuzzer.cc b/test/fuzzers/audio_decoder_pcm16b_fuzzer.cc
new file mode 100644
index 0000000000000000000000000000000000000000..6e5d6e2190638185578ac71b400913f46f6b67cf
--- /dev/null
+++ b/test/fuzzers/audio_decoder_pcm16b_fuzzer.cc
@@ -0,0 +1,56 @@
+/*
+ * 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 <memory>
+
+#include "modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h"
+#include "test/fuzzers/audio_decoder_fuzzer.h"
+
+namespace webrtc {
+void FuzzOneInput(const uint8_t* data, size_t size) {
+ if (size > 10000 || size < 2) {
+ return;
+ }
+
+ int sample_rate_hz;
+ switch (data[0] % 4) {
+ case 0:
+ sample_rate_hz = 8000;
+ break;
+ case 1:
+ sample_rate_hz = 16000;
+ break;
+ case 2:
+ sample_rate_hz = 32000;
+ break;
+ case 3:
+ sample_rate_hz = 48000;
+ break;
+ default:
+ RTC_DCHECK_NOTREACHED();
+ return;
+ }
+ const size_t num_channels = data[1] % 16 + 1;
+
+ // Two first bytes of the data are used. Move forward.
+ data += 2;
+ size -= 2;
+
+ AudioDecoderPcm16B dec(sample_rate_hz, num_channels);
+ // Allocate a maximum output size of 100 ms.
+ const size_t allocated_ouput_size_samples =
+ sample_rate_hz * num_channels / 10;
+ std::unique_ptr<int16_t[]> output =
+ std::make_unique<int16_t[]>(allocated_ouput_size_samples);
+ FuzzAudioDecoder(
+ DecoderFunctionType::kNormalDecode, data, size, &dec, sample_rate_hz,
+ allocated_ouput_size_samples * sizeof(int16_t), output.get());
+}
+} // namespace webrtc