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https://github.com/shadps4-emu/ext-SDL.git
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Added audio stream conversion functions:
SDL_NewAudioStream
SDL_AudioStreamPut
SDL_AudioStreamGet
SDL_AudioStreamAvailable
SDL_AudioStreamClear
SDL_FreeAudioStream
This commit is contained in:
Sam Lantinga
parent
9fd0d6191c
commit
80f8464d97
@ -6,6 +6,13 @@ This is a list of major changes in SDL's version history.
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---------------------------------------------------------------------------
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General:
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* Added audio stream conversion functions:
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SDL_NewAudioStream
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SDL_AudioStreamPut
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SDL_AudioStreamGet
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SDL_AudioStreamAvailable
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SDL_AudioStreamClear
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SDL_FreeAudioStream
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* Added functions to query and set the SDL memory allocation functions:
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SDL_GetMemoryFunctions()
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SDL_SetMemoryFunctions()
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@ -477,6 +477,106 @@ extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT * cvt,
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*/
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extern DECLSPEC int SDLCALL SDL_ConvertAudio(SDL_AudioCVT * cvt);
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/* SDL_AudioStream is a new audio conversion interface.
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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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*/
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/* this is opaque to the outside world. */
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struct _SDL_AudioStream;
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typedef struct _SDL_AudioStream SDL_AudioStream;
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/**
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* Create a new audio stream
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*
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* \param src_format The format of the source audio
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* \param src_channels The number of channels of the source audio
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* \param src_rate The sampling rate of the source audio
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* \param dst_format The format of the desired audio output
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* \param dst_channels The number of channels of the desired audio output
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* \param dst_rate The sampling rate of the desired audio output
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* \return 0 on success, or -1 on error.
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*
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* \sa SDL_AudioStreamPut
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* \sa SDL_AudioStreamGet
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* \sa SDL_AudioStreamAvailable
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* \sa SDL_AudioStreamClear
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* \sa SDL_FreeAudioStream
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*/
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extern DECLSPEC SDL_AudioStream * SDLCALL 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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/**
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* Add data to be converted/resampled to the stream
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*
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* \param stream The stream the audio data is being added to
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* \param buf A pointer to the audio data to add
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* \param int The number of bytes to write to the stream
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* \return 0 on success, or -1 on error.
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*
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* \sa SDL_NewAudioStream
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* \sa SDL_AudioStreamGet
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* \sa SDL_AudioStreamAvailable
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* \sa SDL_AudioStreamClear
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* \sa SDL_FreeAudioStream
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*/
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extern DECLSPEC int SDLCALL SDL_AudioStreamPut(SDL_AudioStream *stream, const void *buf, int len);
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/**
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* Get converted/resampled data from the stream
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*
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* \param stream The stream the audio is being requested from
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* \param buf A buffer to fill with audio data
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* \param len The maximum number of bytes to fill
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* \return The number of bytes read from the stream, or -1 on error
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*
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* \sa SDL_NewAudioStream
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* \sa SDL_AudioStreamPut
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* \sa SDL_AudioStreamAvailable
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* \sa SDL_AudioStreamClear
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* \sa SDL_FreeAudioStream
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*/
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extern DECLSPEC int SDLCALL SDL_AudioStreamGet(SDL_AudioStream *stream, void *buf, int len);
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/**
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* Get the number of converted/resampled bytes available
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*
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* \sa SDL_NewAudioStream
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* \sa SDL_AudioStreamPut
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* \sa SDL_AudioStreamGet
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* \sa SDL_AudioStreamClear
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* \sa SDL_FreeAudioStream
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*/
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extern DECLSPEC int SDLCALL SDL_AudioStreamAvailable(SDL_AudioStream *stream);
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/**
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* Clear any pending data in the stream without converting it
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*
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* \sa SDL_NewAudioStream
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* \sa SDL_AudioStreamPut
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* \sa SDL_AudioStreamGet
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* \sa SDL_AudioStreamAvailable
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* \sa SDL_FreeAudioStream
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*/
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extern DECLSPEC void SDLCALL SDL_AudioStreamClear(SDL_AudioStream *stream);
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/**
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* Free an audio stream
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*
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* \sa SDL_NewAudioStream
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* \sa SDL_AudioStreamPut
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* \sa SDL_AudioStreamGet
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* \sa SDL_AudioStreamAvailable
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* \sa SDL_AudioStreamClear
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*/
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extern DECLSPEC void SDLCALL SDL_FreeAudioStream(SDL_AudioStream *stream);
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#define SDL_MIX_MAXVOLUME 128
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/**
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* This takes two audio buffers of the playing audio format and mixes
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@ -532,7 +632,7 @@ extern DECLSPEC void SDLCALL SDL_MixAudioFormat(Uint8 * dst,
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* \param dev The device ID to which we will queue audio.
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* \param data The data to queue to the device for later playback.
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* \param len The number of bytes (not samples!) to which (data) points.
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* \return zero on success, -1 on error.
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* \return 0 on success, or -1 on error.
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*
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* \sa SDL_GetQueuedAudioSize
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* \sa SDL_ClearQueuedAudio
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@ -74,46 +74,6 @@ extern SDL_AudioFilter SDL_Convert_F32_to_S32;
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extern int SDL_PrepareResampleFilter(void);
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extern void SDL_FreeResampleFilter(void);
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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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extern 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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extern 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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extern 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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extern void SDL_AudioStreamClear(SDL_AudioStream *stream);
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/* number of converted/resampled bytes available */
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extern int SDL_AudioStreamAvailable(SDL_AudioStream *stream);
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/* dispose of a stream */
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extern void SDL_FreeAudioStream(SDL_AudioStream *stream);
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#endif /* SDL_audio_c_h_ */
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/* vi: set ts=4 sw=4 expandtab: */
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@ -1077,7 +1077,7 @@ typedef int (*SDL_ResampleAudioStreamFunc)(SDL_AudioStream *stream, const void *
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typedef void (*SDL_ResetAudioStreamResamplerFunc)(SDL_AudioStream *stream);
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typedef void (*SDL_CleanupAudioStreamResamplerFunc)(SDL_AudioStream *stream);
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struct SDL_AudioStream
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struct _SDL_AudioStream
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{
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SDL_AudioCVT cvt_before_resampling;
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SDL_AudioCVT cvt_after_resampling;
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@ -1349,9 +1349,9 @@ SDL_NewAudioStream(const SDL_AudioFormat src_format,
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}
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int
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SDL_AudioStreamPut(SDL_AudioStream *stream, const void *buf, const Uint32 _buflen)
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SDL_AudioStreamPut(SDL_AudioStream *stream, const void *buf, int len)
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{
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int buflen = (int) _buflen;
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int buflen = len;
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int workbuflen;
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Uint8 *workbuf;
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Uint8 *resamplebuf = NULL;
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@ -1495,34 +1495,19 @@ SDL_AudioStreamPut(SDL_AudioStream *stream, const void *buf, const Uint32 _bufle
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return buflen ? SDL_WriteToDataQueue(stream->queue, resamplebuf, buflen) : 0;
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}
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void
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SDL_AudioStreamClear(SDL_AudioStream *stream)
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{
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if (!stream) {
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SDL_InvalidParamError("stream");
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} else {
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SDL_ClearDataQueue(stream->queue, stream->packetlen * 2);
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if (stream->reset_resampler_func) {
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stream->reset_resampler_func(stream);
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}
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stream->first_run = SDL_TRUE;
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}
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}
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/* get converted/resampled data from the stream */
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int
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SDL_AudioStreamGet(SDL_AudioStream *stream, void *buf, const Uint32 len)
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SDL_AudioStreamGet(SDL_AudioStream *stream, void *buf, int len)
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{
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#if DEBUG_AUDIOSTREAM
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printf("AUDIOSTREAM: want to get %u converted bytes\n", (unsigned int) len);
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printf("AUDIOSTREAM: want to get %d converted bytes\n", len);
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#endif
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if (!stream) {
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return SDL_InvalidParamError("stream");
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} else if (!buf) {
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return SDL_InvalidParamError("buf");
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} else if (len == 0) {
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} else if (len <= 0) {
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return 0; /* nothing to do. */
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} else if ((len % stream->dst_sample_frame_size) != 0) {
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return SDL_SetError("Can't request partial sample frames");
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@ -1538,6 +1523,20 @@ SDL_AudioStreamAvailable(SDL_AudioStream *stream)
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return stream ? (int) SDL_CountDataQueue(stream->queue) : 0;
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}
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void
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SDL_AudioStreamClear(SDL_AudioStream *stream)
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{
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if (!stream) {
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SDL_InvalidParamError("stream");
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} else {
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SDL_ClearDataQueue(stream->queue, stream->packetlen * 2);
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if (stream->reset_resampler_func) {
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stream->reset_resampler_func(stream);
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}
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stream->first_run = SDL_TRUE;
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}
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}
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/* dispose of a stream */
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void
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SDL_FreeAudioStream(SDL_AudioStream *stream)
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@ -640,3 +640,9 @@
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#define SDL_GetMemoryFunctions SDL_GetMemoryFunctions_REAL
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#define SDL_SetMemoryFunctions SDL_SetMemoryFunctions_REAL
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#define SDL_GetNumAllocations SDL_GetNumAllocations_REAL
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#define SDL_NewAudioStream SDL_NewAudioStream_REAL
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#define SDL_AudioStreamPut SDL_AudioStreamPut_REAL
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#define SDL_AudioStreamGet SDL_AudioStreamGet_REAL
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#define SDL_AudioStreamClear SDL_AudioStreamClear_REAL
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#define SDL_AudioStreamAvailable SDL_AudioStreamAvailable_REAL
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#define SDL_FreeAudioStream SDL_FreeAudioStream_REAL
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@ -674,3 +674,9 @@ SDL_DYNAPI_PROC(void,SDL_UnlockJoysticks,(void),(),)
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SDL_DYNAPI_PROC(void,SDL_GetMemoryFunctions,(SDL_malloc_func *a, SDL_calloc_func *b, SDL_realloc_func *c, SDL_free_func *d),(a,b,c,d),)
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SDL_DYNAPI_PROC(int,SDL_SetMemoryFunctions,(SDL_malloc_func a, SDL_calloc_func b, SDL_realloc_func c, SDL_free_func d),(a,b,c,d),return)
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SDL_DYNAPI_PROC(int,SDL_GetNumAllocations,(void),(),return)
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SDL_DYNAPI_PROC(SDL_AudioStream*,SDL_NewAudioStream,(const SDL_AudioFormat a, const Uint8 b, const int c, const SDL_AudioFormat d, const Uint8 e, const int f),(a,b,c,d,e,f),return)
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SDL_DYNAPI_PROC(int,SDL_AudioStreamPut,(SDL_AudioStream *a, const void *b, int c),(a,b,c),return)
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SDL_DYNAPI_PROC(int,SDL_AudioStreamGet,(SDL_AudioStream *a, void *b, int c),(a,b,c),return)
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SDL_DYNAPI_PROC(void,SDL_AudioStreamClear,(SDL_AudioStream *a),(a),)
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SDL_DYNAPI_PROC(int,SDL_AudioStreamAvailable,(SDL_AudioStream *a),(a),return)
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SDL_DYNAPI_PROC(void,SDL_FreeAudioStream,(SDL_AudioStream *a),(a),)
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