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	audio: SDL_AudioStream now has callbacks for Get and Put operations.
This allows code to feed a stream (or feed from a stream) on-demand, which is to say: it can efficiently simulate the SDL2 audio callback.
This commit is contained in:
		| @@ -53,11 +53,11 @@ The following structures have been renamed: | ||||
|  | ||||
| ## SDL_audio.h | ||||
|  | ||||
| The audio subsystem in SDL3 is dramatically different than SDL2. There is no longer an audio callback; instead you bind SDL_AudioStreams to devices. | ||||
| The audio subsystem in SDL3 is dramatically different than SDL2. The primary way to play audio is no longer an audio callback; instead you bind SDL_AudioStreams to devices. | ||||
|  | ||||
| The SDL 1.2 audio compatibility API has also been removed, as it was a simplified version of the audio callback interface. | ||||
|  | ||||
| If your app depends on the callback method, you can use the single-header library at https://github.com/libsdl-org/SDL3_audio_callback (to be written!) to simulate it on top of SDL3's new API. | ||||
| If your app depends on the callback method, there is a similar approach you can take. But first, this is the new approach: | ||||
|  | ||||
| In SDL2, you might have done something like this to play audio: | ||||
|  | ||||
| @@ -84,7 +84,6 @@ in SDL3: | ||||
|  | ||||
| ```c | ||||
|     /* ...somewhere near startup... */ | ||||
|     my_desired_audio_format.callback = MyAudioCallback;  /* etc */ | ||||
|     SDL_AudioDeviceID my_audio_device = SDL_OpenAudioDevice(0, SDL_AUDIO_S16, 2, 44100); | ||||
|     SDL_AudioSteam *stream = SDL_CreateAndBindAudioStream(my_audio_device, SDL_AUDIO_S16, 2, 44100); | ||||
|  | ||||
| @@ -93,6 +92,23 @@ in SDL3: | ||||
|     SDL_PutAudioStreamData(stream, buf, buflen); | ||||
| ``` | ||||
|  | ||||
| If you absolutely require the callback method, SDL_AudioStreams can use a callback whenever more data is to be read from them, which can be used to simulate SDL2 semantics: | ||||
|  | ||||
| ```c | ||||
|     void SDLCALL MyAudioCallback(SDL_AudioStream *stream, int len, void *userdata) | ||||
|     { | ||||
|         /* calculate a little more audio here, maybe using `userdata`, write it to `stream` */ | ||||
|         SDL_PutAudioStreamData(stream, newdata, len); | ||||
|     } | ||||
|  | ||||
|     /* ...somewhere near startup... */ | ||||
|     SDL_AudioDeviceID my_audio_device = SDL_OpenAudioDevice(0, SDL_AUDIO_S16, 2, 44100); | ||||
|     SDL_AudioSteam *stream = SDL_CreateAndBindAudioStream(my_audio_device, SDL_AUDIO_S16, 2, 44100); | ||||
|     SDL_SetAudioStreamGetCallback(stream, MyAudioCallback); | ||||
|  | ||||
|     /* MyAudioCallback will be called whenever the device requests more audio data. */ | ||||
| ``` | ||||
|  | ||||
| SDL_AudioInit() and SDL_AudioQuit() have been removed. Instead you can call SDL_InitSubSystem() and SDL_QuitSubSystem() with SDL_INIT_AUDIO, which will properly refcount the subsystems. You can choose a specific audio driver using SDL_AUDIO_DRIVER hint. | ||||
|  | ||||
| The `SDL_AUDIO_ALLOW_*` symbols have been removed; now one may request the format they desire from the audio device, but ultimately SDL_AudioStream will manage the difference. One can use SDL_GetAudioDeviceFormat() to see what the final format is, if any "allowed" changes should be accomodated by the app. | ||||
|   | ||||
| @@ -713,6 +713,171 @@ extern DECLSPEC int SDLCALL SDL_FlushAudioStream(SDL_AudioStream *stream); | ||||
|  */ | ||||
| extern DECLSPEC int SDLCALL SDL_ClearAudioStream(SDL_AudioStream *stream); | ||||
|  | ||||
| /** | ||||
|  * Lock an audio stream for serialized access. | ||||
|  * | ||||
|  * Each SDL_AudioStream has an internal mutex it uses to | ||||
|  * protect its data structures from threading conflicts. This function | ||||
|  * allows an app to lock that mutex, which could be useful if | ||||
|  * registering callbacks on this stream. | ||||
|  * | ||||
|  * One does not need to lock a stream to use in it most cases, | ||||
|  * as the stream manages this lock internally. However, this lock | ||||
|  * is held during callbacks, which may run from arbitrary threads | ||||
|  * at any time, so if an app needs to protect shared data during | ||||
|  * those callbacks, locking the stream guarantees that the | ||||
|  * callback is not running while the lock is held. | ||||
|  * | ||||
|  * As this is just a wrapper over SDL_LockMutex for an internal | ||||
|  * lock, it has all the same attributes (recursive locks are | ||||
|  * allowed, etc). | ||||
|  * | ||||
|  * \param stream The audio stream to lock. | ||||
|  * \returns 0 on success or a negative error code on failure; call | ||||
|  *          SDL_GetError() for more information. | ||||
|  * | ||||
|  * \since This function is available since SDL 3.0.0. | ||||
|  * | ||||
|  * \threadsafety It is safe to call this function from any thread. | ||||
|  * | ||||
|  * \sa SDL_UnlockAudioStream | ||||
|  * \sa SDL_SetAudioStreamPutCallback | ||||
|  * \sa SDL_SetAudioStreamGetCallback | ||||
|  */ | ||||
| extern DECLSPEC int SDLCALL SDL_LockAudioStream(SDL_AudioStream *stream); | ||||
|  | ||||
|  | ||||
| /** | ||||
|  * Unlock an audio stream for serialized access. | ||||
|  * | ||||
|  * This unlocks an audio stream after a call to SDL_LockAudioStream. | ||||
|  * | ||||
|  * \param stream The audio stream to unlock. | ||||
|  * \returns 0 on success or a negative error code on failure; call | ||||
|  *          SDL_GetError() for more information. | ||||
|  * | ||||
|  * \since This function is available since SDL 3.0.0. | ||||
|  * | ||||
|  * \threadsafety You should only call this from the same thread that | ||||
|  *               previously called SDL_LockAudioStream. | ||||
|  * | ||||
|  * \sa SDL_LockAudioStream | ||||
|  * \sa SDL_SetAudioStreamPutCallback | ||||
|  * \sa SDL_SetAudioStreamGetCallback | ||||
|  */ | ||||
| extern DECLSPEC int SDLCALL SDL_UnlockAudioStream(SDL_AudioStream *stream); | ||||
|  | ||||
| /** | ||||
|  * A callback that fires when data passes through an SDL_AudioStream. | ||||
|  * | ||||
|  * Apps can (optionally) register a callback with an audio stream that | ||||
|  * is called when data is added with SDL_PutAudioStreamData, or requested | ||||
|  * with SDL_GetAudioStreamData. These callbacks may run from any | ||||
|  * thread, so if you need to protect shared data, you should use | ||||
|  * SDL_LockAudioStream to serialize access; this lock will be held by | ||||
|  * before your callback is called, so your callback does not need to | ||||
|  * manage the lock explicitly. | ||||
|  * | ||||
|  * \param stream The SDL audio stream associated with this callback. | ||||
|  * \param approx_request The _approximate_ amout of data, in bytes, that is requested. | ||||
|  *                       This might be slightly overestimated due to buffering or | ||||
|  *                       resampling, and may change from call to call anyhow. | ||||
|  * \param userdata An opaque pointer provided by the app for their personal use. | ||||
|  */ | ||||
| typedef void (SDLCALL *SDL_AudioStreamRequestCallback)(SDL_AudioStream *stream, int approx_request, void *userdata); | ||||
|  | ||||
| /** | ||||
|  * Set a callback that runs when data is requested from an audio stream. | ||||
|  * | ||||
|  * This callback is called _before_ data is obtained from the stream, | ||||
|  * giving the callback the chance to add more on-demand. | ||||
|  * | ||||
|  * The callback can (optionally) call SDL_PutAudioStreamData() to add | ||||
|  * more audio to the stream during this call; if needed, the request | ||||
|  * that triggered this callback will obtain the new data immediately. | ||||
|  * | ||||
|  * The callback's `approx_request` argument is roughly how many bytes | ||||
|  * of _unconverted_ data (in the stream's input format) is needed by | ||||
|  * the caller, although this may overestimate a little for safety. | ||||
|  * This takes into account how much is already in the stream and only | ||||
|  * asks for any extra necessary to resolve the request, which means | ||||
|  * the callback may be asked for zero bytes, and a different amount | ||||
|  * on each call. | ||||
|  * | ||||
|  * The callback is not required to supply exact amounts; it is allowed | ||||
|  * to supply too much or too little or none at all. The caller will | ||||
|  * get what's available, up to the amount they requested, regardless | ||||
|  * of this callback's outcome. | ||||
|  * | ||||
|  * Clearing or flushing an audio stream does not call this callback. | ||||
|  * | ||||
|  * This function obtains the stream's lock, which means any existing | ||||
|  * callback (get or put) in progress will finish running before setting | ||||
|  * the new callback. | ||||
|  * | ||||
|  * Setting a NULL function turns off the callback. | ||||
|  * | ||||
|  * \param stream the audio stream to set the new callback on. | ||||
|  * \param callback the new callback function to call when data is added to the stream. | ||||
|  * \param userdata an opaque pointer provided to the callback for its own personal use. | ||||
|  * \returns 0 on success, -1 on error. This only fails if `stream` is NULL. | ||||
|  * | ||||
|  * \since This function is available since SDL 3.0.0. | ||||
|  * | ||||
|  * \threadsafety It is safe to call this function from any thread. | ||||
|  * | ||||
|  * \sa SDL_SetAudioStreamPutCallback | ||||
|  */ | ||||
| extern DECLSPEC int SDLCALL SDL_SetAudioStreamGetCallback(SDL_AudioStream *stream, SDL_AudioStreamRequestCallback callback, void *userdata); | ||||
|  | ||||
| /** | ||||
|  * Set a callback that runs when data is added to an audio stream. | ||||
|  * | ||||
|  * This callback is called _after_ the data is added to the stream, | ||||
|  * giving the callback the chance to obtain it immediately. | ||||
|  * | ||||
|  * The callback can (optionally) call SDL_GetAudioStreamData() to | ||||
|  * obtain audio from the stream during this call. | ||||
|  * | ||||
|  * The callback's `approx_request` argument is how many bytes | ||||
|  * of _converted_ data (in the stream's output format) was provided | ||||
|  * by the caller, although this may underestimate a little for safety. | ||||
|  * This value might be less than what is currently available in the | ||||
|  * stream, if data was already there, and might be less than the | ||||
|  * caller provided if the stream needs to keep a buffer to aid in | ||||
|  * resampling. Which means the callback may be provided with zero | ||||
|  * bytes, and a different amount on each call. | ||||
|  * | ||||
|  * The callback may call SDL_GetAudioStreamAvailable to see the | ||||
|  * total amount currently available to read from the stream, instead | ||||
|  * of the total provided by the current call. | ||||
|  * | ||||
|  * The callback is not required to obtain all data. It is allowed | ||||
|  * to read less or none at all. Anything not read now simply remains | ||||
|  * in the stream for later access. | ||||
|  * | ||||
|  * Clearing or flushing an audio stream does not call this callback. | ||||
|  * | ||||
|  * This function obtains the stream's lock, which means any existing | ||||
|  * callback (get or put) in progress will finish running before setting | ||||
|  * the new callback. | ||||
|  * | ||||
|  * Setting a NULL function turns off the callback. | ||||
|  * | ||||
|  * \param stream the audio stream to set the new callback on. | ||||
|  * \param callback the new callback function to call when data is added to the stream. | ||||
|  * \param userdata an opaque pointer provided to the callback for its own personal use. | ||||
|  * \returns 0 on success, -1 on error. This only fails if `stream` is NULL. | ||||
|  * | ||||
|  * \since This function is available since SDL 3.0.0. | ||||
|  * | ||||
|  * \threadsafety It is safe to call this function from any thread. | ||||
|  * | ||||
|  * \sa SDL_SetAudioStreamGetCallback | ||||
|  */ | ||||
| extern DECLSPEC int SDLCALL SDL_SetAudioStreamPutCallback(SDL_AudioStream *stream, SDL_AudioStreamRequestCallback callback, void *userdata); | ||||
|  | ||||
|  | ||||
| /** | ||||
|  * Free an audio stream | ||||
|  * | ||||
|   | ||||
| @@ -629,6 +629,40 @@ SDL_AudioStream *SDL_CreateAudioStream(SDL_AudioFormat src_format, | ||||
|     return retval; | ||||
| } | ||||
|  | ||||
| int SDL_SetAudioStreamGetCallback(SDL_AudioStream *stream, SDL_AudioStreamRequestCallback callback, void *userdata) | ||||
| { | ||||
|     if (!stream) { | ||||
|         return SDL_InvalidParamError("stream"); | ||||
|     } | ||||
|     SDL_LockMutex(stream->lock); | ||||
|     stream->get_callback = callback; | ||||
|     stream->get_callback_userdata = userdata; | ||||
|     SDL_UnlockMutex(stream->lock); | ||||
|     return 0; | ||||
| } | ||||
|  | ||||
| int SDL_SetAudioStreamPutCallback(SDL_AudioStream *stream, SDL_AudioStreamRequestCallback callback, void *userdata) | ||||
| { | ||||
|     if (!stream) { | ||||
|         return SDL_InvalidParamError("stream"); | ||||
|     } | ||||
|     SDL_LockMutex(stream->lock); | ||||
|     stream->put_callback = callback; | ||||
|     stream->put_callback_userdata = userdata; | ||||
|     SDL_UnlockMutex(stream->lock); | ||||
|     return 0; | ||||
| } | ||||
|  | ||||
| int SDL_LockAudioStream(SDL_AudioStream *stream) | ||||
| { | ||||
|     return stream ? SDL_LockMutex(stream->lock) : SDL_InvalidParamError("stream"); | ||||
| } | ||||
|  | ||||
| int SDL_UnlockAudioStream(SDL_AudioStream *stream) | ||||
| { | ||||
|     return stream ? SDL_UnlockMutex(stream->lock) : SDL_InvalidParamError("stream"); | ||||
| } | ||||
|  | ||||
| int SDL_GetAudioStreamFormat(SDL_AudioStream *stream, SDL_AudioFormat *src_format, int *src_channels, int *src_rate, SDL_AudioFormat *dst_format, int *dst_channels, int *dst_rate) | ||||
| { | ||||
|     if (!stream) { | ||||
| @@ -696,6 +730,8 @@ int SDL_PutAudioStreamData(SDL_AudioStream *stream, const void *buf, int len) | ||||
|  | ||||
|     SDL_LockMutex(stream->lock); | ||||
|  | ||||
|     const int prev_available = stream->put_callback ? SDL_GetAudioStreamAvailable(stream) : 0; | ||||
|  | ||||
|     if ((len % stream->src_sample_frame_size) != 0) { | ||||
|         SDL_UnlockMutex(stream->lock); | ||||
|         return SDL_SetError("Can't add partial sample frames"); | ||||
| @@ -704,6 +740,11 @@ int SDL_PutAudioStreamData(SDL_AudioStream *stream, const void *buf, int len) | ||||
|     /* just queue the data, we convert/resample when dequeueing. */ | ||||
|     retval = SDL_WriteToDataQueue(stream->queue, buf, len); | ||||
|     stream->flushed = SDL_FALSE; | ||||
|  | ||||
|     if (stream->put_callback) { | ||||
|         stream->put_callback(stream, SDL_GetAudioStreamAvailable(stream) - prev_available, stream->put_callback_userdata); | ||||
|     } | ||||
|  | ||||
|     SDL_UnlockMutex(stream->lock); | ||||
|  | ||||
|     return retval; | ||||
| @@ -978,6 +1019,23 @@ int SDL_GetAudioStreamData(SDL_AudioStream *stream, void *voidbuf, int len) | ||||
|  | ||||
|     len -= len % stream->dst_sample_frame_size;  /* chop off any fractional sample frame. */ | ||||
|  | ||||
|     // give the callback a chance to fill in more stream data if it wants. | ||||
|     if (stream->get_callback) { | ||||
|         int approx_request = len / stream->dst_sample_frame_size;  /* start with sample frames desired */ | ||||
|         if (stream->src_rate != stream->dst_rate) { | ||||
|             /* calculate difference in dataset size after resampling. Use a Uint64 so the multiplication doesn't overflow. */ | ||||
|             approx_request = (size_t) ((((Uint64) approx_request) * stream->src_rate) / stream->dst_rate); | ||||
|             if (!stream->flushed) {  /* do we need to fill the future buffer to accomodate this, too? */ | ||||
|                 approx_request += stream->future_buffer_filled_frames - stream->resampler_padding_frames; | ||||
|             } | ||||
|         } | ||||
|  | ||||
|         approx_request *= stream->src_sample_frame_size;  // convert sample frames to bytes. | ||||
|         const int already_have = SDL_GetAudioStreamAvailable(stream); | ||||
|         approx_request -= SDL_min(approx_request, already_have);  // we definitely have this much output already packed in. | ||||
|         stream->get_callback(stream, approx_request, stream->get_callback_userdata); | ||||
|     } | ||||
|  | ||||
|     /* we convert in chunks, so we don't end up allocating a massive work buffer, etc. */ | ||||
|     while (len > 0) { /* didn't ask for a whole sample frame, nothing to do */ | ||||
|         const int chunk_size = 1024 * 1024;  /* !!! FIXME: a megabyte might be overly-aggressive. */ | ||||
|   | ||||
| @@ -137,6 +137,11 @@ struct SDL_AudioStream | ||||
|     SDL_DataQueue *queue; | ||||
|     SDL_Mutex *lock;  /* this is just a copy of `queue`'s mutex. We share a lock. */ | ||||
|  | ||||
|     SDL_AudioStreamRequestCallback get_callback; | ||||
|     void *get_callback_userdata; | ||||
|     SDL_AudioStreamRequestCallback put_callback; | ||||
|     void *put_callback_userdata; | ||||
|  | ||||
|     Uint8 *work_buffer;    /* used for scratch space during data conversion/resampling. */ | ||||
|     Uint8 *history_buffer;  /* history for left padding and future sample rate changes. */ | ||||
|     Uint8 *future_buffer;  /* stuff that left the queue for the right padding and will be next read's data. */ | ||||
|   | ||||
| @@ -879,6 +879,10 @@ SDL3_0.0.0 { | ||||
|     SDL_MixAudioFormat; | ||||
|     SDL_ConvertAudioSamples; | ||||
|     SDL_GetSilenceValueForFormat; | ||||
|     SDL_LockAudioStream; | ||||
|     SDL_UnlockAudioStream; | ||||
|     SDL_SetAudioStreamGetCallback; | ||||
|     SDL_SetAudioStreamPutCallback; | ||||
|     # extra symbols go here (don't modify this line) | ||||
|   local: *; | ||||
| }; | ||||
|   | ||||
| @@ -905,3 +905,7 @@ | ||||
| #define SDL_MixAudioFormat SDL_MixAudioFormat_REAL | ||||
| #define SDL_ConvertAudioSamples SDL_ConvertAudioSamples_REAL | ||||
| #define SDL_GetSilenceValueForFormat SDL_GetSilenceValueForFormat_REAL | ||||
| #define SDL_LockAudioStream SDL_LockAudioStream_REAL | ||||
| #define SDL_UnlockAudioStream SDL_UnlockAudioStream_REAL | ||||
| #define SDL_SetAudioStreamGetCallback SDL_SetAudioStreamGetCallback_REAL | ||||
| #define SDL_SetAudioStreamPutCallback SDL_SetAudioStreamPutCallback_REAL | ||||
|   | ||||
| @@ -950,3 +950,7 @@ SDL_DYNAPI_PROC(int,SDL_LoadWAV_RW,(SDL_RWops *a, int b, SDL_AudioFormat *c, int | ||||
| SDL_DYNAPI_PROC(int,SDL_MixAudioFormat,(Uint8 *a, const Uint8 *b, SDL_AudioFormat c, Uint32 d, int e),(a,b,c,d,e),return) | ||||
| SDL_DYNAPI_PROC(int,SDL_ConvertAudioSamples,(SDL_AudioFormat a, int b, int c, const Uint8 *d, int e, SDL_AudioFormat f, int g, int h, Uint8 **i, int *j),(a,b,c,d,e,f,g,h,i,j),return) | ||||
| SDL_DYNAPI_PROC(int,SDL_GetSilenceValueForFormat,(SDL_AudioFormat a),(a),return) | ||||
| SDL_DYNAPI_PROC(int,SDL_LockAudioStream,(SDL_AudioStream *a),(a),return) | ||||
| SDL_DYNAPI_PROC(int,SDL_UnlockAudioStream,(SDL_AudioStream *a),(a),return) | ||||
| SDL_DYNAPI_PROC(int,SDL_SetAudioStreamGetCallback,(SDL_AudioStream *a, SDL_AudioStreamRequestCallback b, void *c),(a,b,c),return) | ||||
| SDL_DYNAPI_PROC(int,SDL_SetAudioStreamPutCallback,(SDL_AudioStream *a, SDL_AudioStreamRequestCallback b, void *c),(a,b,c),return) | ||||
|   | ||||
| @@ -33,11 +33,36 @@ static struct | ||||
|     int freq; | ||||
|     Uint8 *sound;    /* Pointer to wave data */ | ||||
|     Uint32 soundlen; /* Length of wave data */ | ||||
|     Uint32 soundpos; | ||||
| } wave; | ||||
|  | ||||
| static SDL_AudioDeviceID device; | ||||
| static SDL_AudioStream *stream; | ||||
|  | ||||
| static void SDLCALL | ||||
| fillerup(SDL_AudioStream *stream, int len, void *unused) | ||||
| { | ||||
|     Uint8 *waveptr; | ||||
|     int waveleft; | ||||
|  | ||||
|     /*SDL_Log("CALLBACK WANTS %d MORE BYTES!", len);*/ | ||||
|  | ||||
|     /* Set up the pointers */ | ||||
|     waveptr = wave.sound + wave.soundpos; | ||||
|     waveleft = wave.soundlen - wave.soundpos; | ||||
|  | ||||
|     /* Go! */ | ||||
|     while (waveleft <= len) { | ||||
|         SDL_PutAudioStreamData(stream, waveptr, waveleft); | ||||
|         len -= waveleft; | ||||
|         waveptr = wave.sound; | ||||
|         waveleft = wave.soundlen; | ||||
|         wave.soundpos = 0; | ||||
|     } | ||||
|     SDL_PutAudioStreamData(stream, waveptr, len); | ||||
|     wave.soundpos += len; | ||||
| } | ||||
|  | ||||
| /* Call this instead of exit(), so we can clean up SDL: atexit() is evil. */ | ||||
| static void | ||||
| quit(int rc) | ||||
| @@ -78,6 +103,8 @@ open_audio(void) | ||||
|         SDL_free(wave.sound); | ||||
|         quit(2); | ||||
|     } | ||||
|  | ||||
|     SDL_SetAudioStreamGetCallback(stream, fillerup, NULL); | ||||
| } | ||||
|  | ||||
| #ifndef __EMSCRIPTEN__ | ||||
| @@ -91,12 +118,6 @@ static void reopen_audio(void) | ||||
|  | ||||
| static int done = 0; | ||||
|  | ||||
| static void fillerup(void) | ||||
| { | ||||
|     if (SDL_GetAudioStreamAvailable(stream) < (wave.soundlen / 2)) { | ||||
|         SDL_PutAudioStreamData(stream, wave.sound, wave.soundlen); | ||||
|     } | ||||
| } | ||||
|  | ||||
|  | ||||
| #ifdef __EMSCRIPTEN__ | ||||
| @@ -192,7 +213,6 @@ int main(int argc, char *argv[]) | ||||
|             } | ||||
|         } | ||||
|  | ||||
|         fillerup(); | ||||
|         SDL_Delay(100); | ||||
|     } | ||||
| #endif | ||||
|   | ||||
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