Opus Encoder fuzzer: separate target for FuzzAudioEncoder

Move FuzzAudioEncoder to a separate target to make it available for
other encoders.

Bug: webrtc:11388
Change-Id: I8b9a0f810791880eedb129b55eb33f154790e48f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169364
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30634}

test/fuzzers/BUILD.gn

@@ -28,6 +28,7 @@ rtc_library("webrtc_fuzzer_main") {
 }
 
 rtc_library("fuzz_data_helper") {
+  testonly = true
   sources = [
     "fuzz_data_helper.cc",
     "fuzz_data_helper.h",
@@ -228,6 +229,7 @@ webrtc_fuzzer_test("congestion_controller_feedback_fuzzer") {
 }
 
 rtc_library("audio_decoder_fuzzer") {
+  testonly = true
   sources = [
     "audio_decoder_fuzzer.cc",
     "audio_decoder_fuzzer.h",
@@ -290,13 +292,27 @@ webrtc_fuzzer_test("audio_decoder_multiopus_fuzzer") {
   ]
 }
 
+rtc_library("audio_encoder_fuzzer") {
+  testonly = true
+  sources = [
+    "audio_encoder_fuzzer.cc",
+    "audio_encoder_fuzzer.h",
+  ]
+  deps = [
+    ":fuzz_data_helper",
+    "../../api:array_view",
+    "../../api/audio_codecs:audio_codecs_api",
+    "../../rtc_base:checks",
+    "../../rtc_base:rtc_base_approved",
+  ]
+}
+
 webrtc_fuzzer_test("audio_encoder_opus_fuzzer") {
   sources = [ "audio_encoder_opus_fuzzer.cc" ]
   deps = [
-    "../../api:array_view",
+    ":audio_encoder_fuzzer",
     "../../api/audio_codecs/opus:audio_encoder_opus",
     "../../rtc_base:checks",
-    "../../rtc_base:rtc_base_approved",
   ]
 }
 
@@ -391,6 +407,7 @@ webrtc_fuzzer_test("pseudotcp_parser_fuzzer") {
 }
 
 rtc_library("audio_processing_fuzzer_helper") {
+  testonly = true
   sources = [
     "audio_processing_fuzzer_helper.cc",
     "audio_processing_fuzzer_helper.h",

test/fuzzers/audio_encoder_fuzzer.cc

@@ -0,0 +1,53 @@
+/*
+ *  Copyright (c) 2020 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 "test/fuzzers/audio_encoder_fuzzer.h"
+
+#include <cstring>
+
+#include "rtc_base/buffer.h"
+#include "rtc_base/checks.h"
+#include "test/fuzzers/fuzz_data_helper.h"
+
+namespace webrtc {
+
+// This function reads bytes from |data_view|, interprets them as RTP timestamp
+// and input samples, and sends them for encoding. The process continues until
+// no more data is available.
+void FuzzAudioEncoder(rtc::ArrayView<const uint8_t> data_view,
+                      std::unique_ptr<AudioEncoder> encoder) {
+  test::FuzzDataHelper data(data_view);
+  const size_t block_size_samples =
+      encoder->SampleRateHz() / 100 * encoder->NumChannels();
+  const size_t block_size_bytes = block_size_samples * sizeof(int16_t);
+  if (data_view.size() / block_size_bytes > 1000) {
+    // If the size of the fuzzer data is more than 1000 input blocks (i.e., more
+    // than 10 seconds), then don't fuzz at all for the fear of timing out.
+    return;
+  }
+
+  rtc::BufferT<int16_t> input_aligned(block_size_samples);
+  rtc::Buffer encoded;
+
+  // Each round in the loop below will need one block of samples + a 32-bit
+  // timestamp from the fuzzer input.
+  const size_t bytes_to_read = block_size_bytes + sizeof(uint32_t);
+  while (data.CanReadBytes(bytes_to_read)) {
+    const uint32_t timestamp = data.Read<uint32_t>();
+    auto byte_array = data.ReadByteArray(block_size_bytes);
+    // Align the data by copying to another array.
+    RTC_DCHECK_EQ(input_aligned.size() * sizeof(int16_t),
+                  byte_array.size() * sizeof(uint8_t));
+    memcpy(input_aligned.data(), byte_array.data(), byte_array.size());
+    auto info = encoder->Encode(timestamp, input_aligned, &encoded);
+  }
+}
+
+}  // namespace webrtc

test/fuzzers/audio_encoder_fuzzer.h

@@ -0,0 +1,26 @@
+/*
+ *  Copyright (c) 2020 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.
+ */
+
+#ifndef TEST_FUZZERS_AUDIO_ENCODER_FUZZER_H_
+#define TEST_FUZZERS_AUDIO_ENCODER_FUZZER_H_
+
+#include <memory>
+
+#include "api/array_view.h"
+#include "api/audio_codecs/audio_encoder.h"
+
+namespace webrtc {
+
+void FuzzAudioEncoder(rtc::ArrayView<const uint8_t> data_view,
+                      std::unique_ptr<AudioEncoder> encoder);
+
+}  // namespace webrtc
+
+#endif  // TEST_FUZZERS_AUDIO_ENCODER_FUZZER_H_

test/fuzzers/audio_encoder_opus_fuzzer.cc

@@ -8,57 +8,20 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "api/array_view.h"
 #include "api/audio_codecs/opus/audio_encoder_opus.h"
-#include "rtc_base/buffer.h"
 #include "rtc_base/checks.h"
-#include "test/fuzzers/fuzz_data_helper.h"
+#include "test/fuzzers/audio_encoder_fuzzer.h"
 
 namespace webrtc {
-namespace {
-
-// This function reads bytes from |data_view|, interprets them
-// as RTP timestamp and input samples, and sends them for encoding. The process
-// continues until no more data is available.
-void FuzzAudioEncoder(rtc::ArrayView<const uint8_t> data_view,
-                      AudioEncoder* encoder) {
-  test::FuzzDataHelper data(data_view);
-  const size_t block_size_samples =
-      encoder->SampleRateHz() / 100 * encoder->NumChannels();
-  const size_t block_size_bytes = block_size_samples * sizeof(int16_t);
-  if (data_view.size() / block_size_bytes > 1000) {
-    // If the size of the fuzzer data is more than 1000 input blocks (i.e., more
-    // than 10 seconds), then don't fuzz at all for the fear of timing out.
-    return;
-  }
-
-  rtc::BufferT<int16_t> input_aligned(block_size_samples);
-  rtc::Buffer encoded;
-
-  // Each round in the loop below will need one block of samples + a 32-bit
-  // timestamp from the fuzzer input.
-  const size_t bytes_to_read = block_size_bytes + sizeof(uint32_t);
-  while (data.CanReadBytes(bytes_to_read)) {
-    const uint32_t timestamp = data.Read<uint32_t>();
-    auto byte_array = data.ReadByteArray(block_size_bytes);
-    // Align the data by copying to another array.
-    RTC_DCHECK_EQ(input_aligned.size() * sizeof(int16_t),
-                  byte_array.size() * sizeof(uint8_t));
-    memcpy(input_aligned.data(), byte_array.data(), byte_array.size());
-    auto info = encoder->Encode(timestamp, input_aligned, &encoded);
-  }
-}
-
-}  // namespace
 
 void FuzzOneInput(const uint8_t* data, size_t size) {
   AudioEncoderOpus::Config config;
   config.frame_size_ms = 20;
   RTC_CHECK(config.IsOk());
   constexpr int kPayloadType = 100;
-  std::unique_ptr<AudioEncoder> enc =
-      AudioEncoderOpus::MakeAudioEncoder(config, kPayloadType);
-  FuzzAudioEncoder(rtc::ArrayView<const uint8_t>(data, size), enc.get());
+  FuzzAudioEncoder(
+      /*data_view=*/{data, size},
+      /*encoder=*/AudioEncoderOpus::MakeAudioEncoder(config, kPayloadType));
 }
 
 }  // namespace webrtc