mirror of
https://github.com/danog/libtgvoip.git
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333c4a1101
Added simple audio resampler Replaced prebuilt static libs with their sources & added that to all project files (closes #5)
65 lines
2.2 KiB
C++
65 lines
2.2 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 WEBRTC_COMMON_AUDIO_WAV_HEADER_H_
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#define WEBRTC_COMMON_AUDIO_WAV_HEADER_H_
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#include <stddef.h>
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#include <stdint.h>
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namespace webrtc {
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static const size_t kWavHeaderSize = 44;
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class ReadableWav {
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public:
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// Returns the number of bytes read.
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size_t virtual Read(void* buf, size_t num_bytes) = 0;
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virtual ~ReadableWav() {}
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};
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enum WavFormat {
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kWavFormatPcm = 1, // PCM, each sample of size bytes_per_sample
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kWavFormatALaw = 6, // 8-bit ITU-T G.711 A-law
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kWavFormatMuLaw = 7, // 8-bit ITU-T G.711 mu-law
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};
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// Return true if the given parameters will make a well-formed WAV header.
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bool CheckWavParameters(size_t num_channels,
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int sample_rate,
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WavFormat format,
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size_t bytes_per_sample,
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size_t num_samples);
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// Write a kWavHeaderSize bytes long WAV header to buf. The payload that
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// follows the header is supposed to have the specified number of interleaved
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// channels and contain the specified total number of samples of the specified
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// type. CHECKs the input parameters for validity.
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void WriteWavHeader(uint8_t* buf,
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size_t num_channels,
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int sample_rate,
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WavFormat format,
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size_t bytes_per_sample,
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size_t num_samples);
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// Read a WAV header from an implemented ReadableWav and parse the values into
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// the provided output parameters. ReadableWav is used because the header can
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// be variably sized. Returns false if the header is invalid.
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bool ReadWavHeader(ReadableWav* readable,
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size_t* num_channels,
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int* sample_rate,
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WavFormat* format,
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size_t* bytes_per_sample,
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size_t* num_samples);
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} // namespace webrtc
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#endif // WEBRTC_COMMON_AUDIO_WAV_HEADER_H_
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