Adding fuzzer for G711/PCM u/A decoders and fixing a fuzzer problem

Bug: chromium:1279775
Change-Id: I8cc3f5fe25b9e707e9d171251026bd5a8bad5da5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251844
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36036}

modules/audio_coding/codecs/g711/audio_decoder_pcm.cc

@@ -40,8 +40,14 @@ int AudioDecoderPcmU::DecodeInternal(const uint8_t* encoded,
                                      int16_t* decoded,
                                      SpeechType* speech_type) {
   RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
+  // 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) *
+      Channels();         // 1 byte per sample per channel
   int16_t temp_type = 1;  // Default is speech.
-  size_t ret = WebRtcG711_DecodeU(encoded, encoded_len, decoded, &temp_type);
+  size_t ret =
+      WebRtcG711_DecodeU(encoded, encoded_len_adjusted, decoded, &temp_type);
   *speech_type = ConvertSpeechType(temp_type);
   return static_cast<int>(ret);
 }
@@ -75,8 +81,14 @@ int AudioDecoderPcmA::DecodeInternal(const uint8_t* encoded,
                                      int16_t* decoded,
                                      SpeechType* speech_type) {
   RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
+  // 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) *
+      Channels();         // 1 byte per sample per channel
   int16_t temp_type = 1;  // Default is speech.
-  size_t ret = WebRtcG711_DecodeA(encoded, encoded_len, decoded, &temp_type);
+  size_t ret =
+      WebRtcG711_DecodeA(encoded, encoded_len_adjusted, decoded, &temp_type);
   *speech_type = ConvertSpeechType(temp_type);
   return static_cast<int>(ret);
 }

test/fuzzers/BUILD.gn

@@ -316,6 +316,14 @@ webrtc_fuzzer_test("audio_decoder_multiopus_fuzzer") {
   ]
 }
 
+webrtc_fuzzer_test("audio_decoder_pcm_fuzzer") {
+  sources = [ "audio_decoder_pcm_fuzzer.cc" ]
+  deps = [
+    ":audio_decoder_fuzzer",
+    "../../modules/audio_coding:g711",
+  ]
+}
+
 webrtc_fuzzer_test("audio_decoder_pcm16b_fuzzer") {
   sources = [ "audio_decoder_pcm16b_fuzzer.cc" ]
   deps = [

test/fuzzers/audio_decoder_pcm_fuzzer.cc

@@ -0,0 +1,45 @@
+/*
+ *  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/g711/audio_decoder_pcm.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;
+  }
+
+  const size_t num_channels = data[0] % 16 + 1;
+
+  std::unique_ptr<AudioDecoder> dec;
+  if (data[1] % 2) {
+    dec = std::make_unique<AudioDecoderPcmU>(num_channels);
+  } else {
+    dec = std::make_unique<AudioDecoderPcmA>(num_channels);
+  }
+
+  // Two first bytes of the data are used. Move forward.
+  data += 2;
+  size -= 2;
+
+  // Allocate a maximum output size of 100 ms.
+  const size_t allocated_ouput_size_samples =
+      dec->SampleRateHz() * num_channels / 10;
+  std::unique_ptr<int16_t[]> output =
+      std::make_unique<int16_t[]>(allocated_ouput_size_samples);
+  FuzzAudioDecoder(DecoderFunctionType::kNormalDecode, data, size, dec.get(),
+                   dec->SampleRateHz(),
+                   allocated_ouput_size_samples * sizeof(int16_t),
+                   output.get());
+}
+}  // namespace webrtc