src/audio/SDL_audio_c.h
author Ryan C. Gordon <icculus@icculus.org>
Sun, 08 Jan 2017 14:18:03 -0500
changeset 10790 92726dd20acd
parent 10764 f9bf759e9dd1
child 10791 64c46e09df9d
permissions -rw-r--r--
audio: libsamplerate loading now happens once at init time.
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/*
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  Simple DirectMedia Layer
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  Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
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  This software is provided 'as-is', without any express or implied
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  warranty.  In no event will the authors be held liable for any damages
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  arising from the use of this software.
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  Permission is granted to anyone to use this software for any purpose,
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  including commercial applications, and to alter it and redistribute it
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  freely, subject to the following restrictions:
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  1. The origin of this software must not be misrepresented; you must not
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     claim that you wrote the original software. If you use this software
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     in a product, an acknowledgment in the product documentation would be
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     appreciated but is not required.
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  2. Altered source versions must be plainly marked as such, and must not be
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     misrepresented as being the original software.
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  3. This notice may not be removed or altered from any source distribution.
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*/
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#ifndef SDL_audio_c_h_
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#define SDL_audio_c_h_
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#include "../SDL_internal.h"
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#ifndef DEBUG_CONVERT
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#define DEBUG_CONVERT 0
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#endif
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#if DEBUG_CONVERT
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#define LOG_DEBUG_CONVERT(from, to) fprintf(stderr, "Converting %s to %s.\n", from, to);
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#else
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#define LOG_DEBUG_CONVERT(from, to)
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#endif
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/* Functions and variables exported from SDL_audio.c for SDL_sysaudio.c */
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#ifdef HAVE_LIBSAMPLERATE_H
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extern SDL_bool SRC_available;
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typedef struct SRC_STATE SRC_STATE;
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extern SRC_STATE* (*SRC_src_new)(int converter_type, int channels, int *error);
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extern int (*SRC_src_process)(SRC_STATE *state, SRC_DATA *data);
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extern int (*SRC_src_reset)(SRC_STATE *state);
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extern SRC_STATE* (*SRC_src_delete)(SRC_STATE *state);
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extern const char* (*SRC_src_strerror)(int error);
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#endif
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/* Functions to get a list of "close" audio formats */
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extern SDL_AudioFormat SDL_FirstAudioFormat(SDL_AudioFormat format);
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extern SDL_AudioFormat SDL_NextAudioFormat(void);
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/* Function to calculate the size and silence for a SDL_AudioSpec */
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extern void SDL_CalculateAudioSpec(SDL_AudioSpec * spec);
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void SDLCALL SDL_Convert_S8_to_F32(SDL_AudioCVT *cvt, SDL_AudioFormat format);
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void SDLCALL SDL_Convert_U8_to_F32(SDL_AudioCVT *cvt, SDL_AudioFormat format);
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void SDLCALL SDL_Convert_S16_to_F32(SDL_AudioCVT *cvt, SDL_AudioFormat format);
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void SDLCALL SDL_Convert_U16_to_F32(SDL_AudioCVT *cvt, SDL_AudioFormat format);
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void SDLCALL SDL_Convert_S32_to_F32(SDL_AudioCVT *cvt, SDL_AudioFormat format);
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void SDLCALL SDL_Convert_F32_to_S8(SDL_AudioCVT *cvt, SDL_AudioFormat format);
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void SDLCALL SDL_Convert_F32_to_U8(SDL_AudioCVT *cvt, SDL_AudioFormat format);
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void SDLCALL SDL_Convert_F32_to_S16(SDL_AudioCVT *cvt, SDL_AudioFormat format);
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void SDLCALL SDL_Convert_F32_to_U16(SDL_AudioCVT *cvt, SDL_AudioFormat format);
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void SDLCALL SDL_Convert_F32_to_S32(SDL_AudioCVT *cvt, SDL_AudioFormat format);
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void SDL_Upsample_Arbitrary(SDL_AudioCVT *cvt, const int channels);
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void SDL_Upsample_Multiple(SDL_AudioCVT *cvt, const int channels);
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void SDL_Downsample_Arbitrary(SDL_AudioCVT *cvt, const int channels);
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void SDL_Downsample_Multiple(SDL_AudioCVT *cvt, const int channels);
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/* SDL_AudioStream is a new audio conversion interface. It
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    might eventually become a public API.
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   The benefits vs SDL_AudioCVT:
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    - it can handle resampling data in chunks without generating
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      artifacts, when it doesn't have the complete buffer available.
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    - it can handle incoming data in any variable size.
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    - You push data as you have it, and pull it when you need it
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    (Note that currently this converts as data is put into the stream, so
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    you need to push more than a handful of bytes if you want decent
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    resampling. This can be changed later.)
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 */
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/* this is opaque to the outside world. */
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typedef struct SDL_AudioStream SDL_AudioStream;
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/* create a new stream */
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SDL_AudioStream *SDL_NewAudioStream(const SDL_AudioFormat src_format,
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                                    const Uint8 src_channels,
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                                    const int src_rate,
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                                    const SDL_AudioFormat dst_format,
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                                    const Uint8 dst_channels,
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                                    const int dst_rate);
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/* add data to be converted/resampled to the stream */
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int SDL_AudioStreamPut(SDL_AudioStream *stream, const void *buf, const Uint32 len);
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/* get converted/resampled data from the stream */
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int SDL_AudioStreamGet(SDL_AudioStream *stream, void *buf, const Uint32 len);
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/* clear any pending data in the stream without converting it. */
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void SDL_AudioStreamClear(SDL_AudioStream *stream);
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/* number of converted/resampled bytes available */
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int SDL_AudioStreamAvailable(SDL_AudioStream *stream);
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/* dispose of a stream */
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void SDL_FreeAudioStream(SDL_AudioStream *stream);
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#endif
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