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Diffstat (limited to 'chromium/third_party/webrtc/modules/audio_coding/neteq/tools/input_audio_file.h')
-rw-r--r-- | chromium/third_party/webrtc/modules/audio_coding/neteq/tools/input_audio_file.h | 51 |
1 files changed, 51 insertions, 0 deletions
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/input_audio_file.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/input_audio_file.h new file mode 100644 index 00000000000..274f8ea07e5 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/input_audio_file.h @@ -0,0 +1,51 @@ +/* + * Copyright (c) 2013 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 WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_INPUT_AUDIO_FILE_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_INPUT_AUDIO_FILE_H_ + +#include <stdio.h> + +#include <string> + +#include "webrtc/base/constructormagic.h" +#include "webrtc/typedefs.h" + +namespace webrtc { +namespace test { + +// Class for handling a looping input audio file. +class InputAudioFile { + public: + explicit InputAudioFile(const std::string file_name); + + virtual ~InputAudioFile(); + + // Reads |samples| elements from source file to |destination|. Returns true + // if the read was successful, otherwise false. If the file end is reached, + // the file is rewound and reading continues from the beginning. + // The output |destination| must have the capacity to hold |samples| elements. + bool Read(size_t samples, int16_t* destination); + + // Creates a multi-channel signal from a mono signal. Each sample is repeated + // |channels| times to create an interleaved multi-channel signal where all + // channels are identical. The output |destination| must have the capacity to + // hold samples * channels elements. + static void DuplicateInterleaved(const int16_t* source, size_t samples, + size_t channels, int16_t* destination); + + private: + FILE* fp_; + DISALLOW_COPY_AND_ASSIGN(InputAudioFile); +}; + +} // namespace test +} // namespace webrtc +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_INPUT_AUDIO_FILE_H_ |