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/*
* 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_
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