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In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
58 lines
2.2 KiB
C++
58 lines
2.2 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_INPUT_AUDIO_FILE_H_
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#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_INPUT_AUDIO_FILE_H_
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#include <stdio.h>
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#include <string>
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#include "webrtc/rtc_base/constructormagic.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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namespace test {
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// Class for handling a looping input audio file.
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class InputAudioFile {
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public:
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explicit InputAudioFile(const std::string file_name);
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virtual ~InputAudioFile();
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// Reads |samples| elements from source file to |destination|. Returns true
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// if the read was successful, otherwise false. If the file end is reached,
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// the file is rewound and reading continues from the beginning.
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// The output |destination| must have the capacity to hold |samples| elements.
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virtual bool Read(size_t samples, int16_t* destination);
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// Fast-forwards (|samples| > 0) or -backwards (|samples| < 0) the file by the
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// indicated number of samples. Just like Read(), Seek() starts over at the
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// beginning of the file if the end is reached. However, seeking backwards
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// past the beginning of the file is not possible.
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virtual bool Seek(int samples);
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// Creates a multi-channel signal from a mono signal. Each sample is repeated
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// |channels| times to create an interleaved multi-channel signal where all
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// channels are identical. The output |destination| must have the capacity to
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// hold samples * channels elements. Note that |source| and |destination| can
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// be the same array (i.e., point to the same address).
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static void DuplicateInterleaved(const int16_t* source, size_t samples,
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size_t channels, int16_t* destination);
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private:
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FILE* fp_;
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RTC_DISALLOW_COPY_AND_ASSIGN(InputAudioFile);
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};
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} // namespace test
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_INPUT_AUDIO_FILE_H_
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