From ac341df436895ff901aeb1310f150241499c29d7 Mon Sep 17 00:00:00 2001
From: Henrik Lundin <henrik.lundin@webrtc.org>
Date: Tue, 15 Feb 2022 15:13:34 +0000
Subject: [PATCH] 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}
---
 .../codecs/pcm16b/audio_decoder_pcm16b.cc     |  7 ++-
 test/fuzzers/BUILD.gn                         |  8 +++
 test/fuzzers/audio_decoder_pcm16b_fuzzer.cc   | 56 +++++++++++++++++++
 3 files changed, 70 insertions(+), 1 deletion(-)
 create mode 100644 test/fuzzers/audio_decoder_pcm16b_fuzzer.cc

diff --git a/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc b/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc
index 1dd2ff289e..7761efe8b3 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 69c4d3d29f..a5e2e44ff1 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 0000000000..6e5d6e2190
--- /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