mirror of
https://github.com/danog/libtgvoip.git
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5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
123 lines
3.6 KiB
C++
123 lines
3.6 KiB
C++
/*
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* Copyright (c) 2014 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 COMMON_AUDIO_WAV_FILE_H_
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#define COMMON_AUDIO_WAV_FILE_H_
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#ifdef __cplusplus
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#include <stdint.h>
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#include <cstddef>
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#include <string>
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#include "rtc_base/constructormagic.h"
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#include "rtc_base/platform_file.h"
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namespace webrtc {
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// Interface to provide access to WAV file parameters.
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class WavFile {
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public:
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virtual ~WavFile() {}
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virtual int sample_rate() const = 0;
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virtual size_t num_channels() const = 0;
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virtual size_t num_samples() const = 0;
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};
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// Simple C++ class for writing 16-bit PCM WAV files. All error handling is
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// by calls to RTC_CHECK(), making it unsuitable for anything but debug code.
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class WavWriter final : public WavFile {
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public:
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// Open a new WAV file for writing.
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WavWriter(const std::string& filename, int sample_rate, size_t num_channels);
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// Open a new WAV file for writing.
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WavWriter(rtc::PlatformFile file, int sample_rate, size_t num_channels);
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// Close the WAV file, after writing its header.
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~WavWriter() override;
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// Write additional samples to the file. Each sample is in the range
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// [-32768,32767], and there must be the previously specified number of
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// interleaved channels.
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void WriteSamples(const float* samples, size_t num_samples);
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void WriteSamples(const int16_t* samples, size_t num_samples);
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int sample_rate() const override;
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size_t num_channels() const override;
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size_t num_samples() const override;
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private:
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void Close();
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const int sample_rate_;
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const size_t num_channels_;
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size_t num_samples_; // Total number of samples written to file.
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FILE* file_handle_; // Output file, owned by this class
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RTC_DISALLOW_COPY_AND_ASSIGN(WavWriter);
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};
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// Follows the conventions of WavWriter.
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class WavReader final : public WavFile {
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public:
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// Opens an existing WAV file for reading.
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explicit WavReader(const std::string& filename);
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// Opens an existing WAV file for reading.
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explicit WavReader(rtc::PlatformFile file);
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// Close the WAV file.
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~WavReader() override;
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// Returns the number of samples read. If this is less than requested,
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// verifies that the end of the file was reached.
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size_t ReadSamples(size_t num_samples, float* samples);
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size_t ReadSamples(size_t num_samples, int16_t* samples);
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int sample_rate() const override;
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size_t num_channels() const override;
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size_t num_samples() const override;
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private:
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void Close();
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int sample_rate_;
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size_t num_channels_;
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size_t num_samples_; // Total number of samples in the file.
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size_t num_samples_remaining_;
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FILE* file_handle_; // Input file, owned by this class.
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RTC_DISALLOW_COPY_AND_ASSIGN(WavReader);
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};
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} // namespace webrtc
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extern "C" {
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#endif // __cplusplus
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// C wrappers for the WavWriter class.
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typedef struct rtc_WavWriter rtc_WavWriter;
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rtc_WavWriter* rtc_WavOpen(const char* filename,
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int sample_rate,
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size_t num_channels);
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void rtc_WavClose(rtc_WavWriter* wf);
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void rtc_WavWriteSamples(rtc_WavWriter* wf,
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const float* samples,
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size_t num_samples);
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int rtc_WavSampleRate(const rtc_WavWriter* wf);
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size_t rtc_WavNumChannels(const rtc_WavWriter* wf);
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size_t rtc_WavNumSamples(const rtc_WavWriter* wf);
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#ifdef __cplusplus
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} // extern "C"
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#endif
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#endif // COMMON_AUDIO_WAV_FILE_H_
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