1 files changed, 64 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/audio_checksum.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/audio_checksum.h
new file mode 100644
index 0000000000..42e3a3a3a0
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+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/audio_checksum.h
@@ -0,0 +1,64 @@
+/*
+ *  Copyright (c) 2014 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 MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_CHECKSUM_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_CHECKSUM_H_
+
+#include <memory>
+#include <string>
+
+#include "modules/audio_coding/neteq/tools/audio_sink.h"
+#include "rtc_base/buffer.h"
+#include "rtc_base/message_digest.h"
+#include "rtc_base/string_encode.h"
+#include "rtc_base/system/arch.h"
+
+namespace webrtc {
+namespace test {
+
+class AudioChecksum : public AudioSink {
+ public:
+  AudioChecksum()
+      : checksum_(rtc::MessageDigestFactory::Create(rtc::DIGEST_MD5)),
+        checksum_result_(checksum_->Size()),
+        finished_(false) {}
+
+  AudioChecksum(const AudioChecksum&) = delete;
+  AudioChecksum& operator=(const AudioChecksum&) = delete;
+
+  bool WriteArray(const int16_t* audio, size_t num_samples) override {
+    if (finished_)
+      return false;
+
+#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
+#error "Big-endian gives a different checksum"
+#endif
+    checksum_->Update(audio, num_samples * sizeof(*audio));
+    return true;
+  }
+
+  // Finalizes the computations, and returns the checksum.
+  std::string Finish() {
+    if (!finished_) {
+      finished_ = true;
+      checksum_->Finish(checksum_result_.data(), checksum_result_.size());
+    }
+    return rtc::hex_encode(checksum_result_);
+  }
+
+ private:
+  std::unique_ptr<rtc::MessageDigest> checksum_;
+  rtc::Buffer checksum_result_;
+  bool finished_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_CHECKSUM_H_