SDL
2.0
|
#include "SDL_stdinc.h"
#include "SDL_error.h"
#include "SDL_endian.h"
#include "SDL_mutex.h"
#include "SDL_thread.h"
#include "SDL_rwops.h"
#include "begin_code.h"
#include "close_code.h"
Go to the source code of this file.
Data Structures | |
struct | SDL_AudioSpec |
struct | SDL_AudioCVT |
A structure to hold a set of audio conversion filters and buffers. More... | |
Typedefs | |
typedef Uint16 | SDL_AudioFormat |
Audio format flags. More... | |
Functions | |
Driver discovery functions | |
These functions return the list of built in audio drivers, in the order that they are normally initialized by default. | |
int | SDL_GetNumAudioDrivers (void) |
const char * | SDL_GetAudioDriver (int index) |
Audio lock functions | |
The lock manipulated by these functions protects the callback function. During a SDL_LockAudio()/SDL_UnlockAudio() pair, you can be guaranteed that the callback function is not running. Do not call these from the callback function or you will cause deadlock. | |
void | SDL_LockAudio (void) |
void | SDL_LockAudioDevice (SDL_AudioDeviceID dev) |
void | SDL_UnlockAudio (void) |
void | SDL_UnlockAudioDevice (SDL_AudioDeviceID dev) |
void | SDL_CloseAudio (void) |
void | SDL_CloseAudioDevice (SDL_AudioDeviceID dev) |
Allow change flags | |
Which audio format changes are allowed when opening a device. | |
#define | SDL_AUDIO_ALLOW_FREQUENCY_CHANGE 0x00000001 |
#define | SDL_AUDIO_ALLOW_FORMAT_CHANGE 0x00000002 |
#define | SDL_AUDIO_ALLOW_CHANNELS_CHANGE 0x00000004 |
#define | SDL_AUDIO_ALLOW_SAMPLES_CHANGE 0x00000008 |
#define | SDL_AUDIO_ALLOW_ANY_CHANGE (SDL_AUDIO_ALLOW_FREQUENCY_CHANGE|SDL_AUDIO_ALLOW_FORMAT_CHANGE|SDL_AUDIO_ALLOW_CHANNELS_CHANGE|SDL_AUDIO_ALLOW_SAMPLES_CHANGE) |
#define | SDL_AUDIOCVT_MAX_FILTERS 9 |
Upper limit of filters in SDL_AudioCVT. More... | |
#define | SDL_AUDIOCVT_PACKED |
typedef void(* | SDL_AudioCallback) (void *userdata, Uint8 *stream, int len) |
typedef void(* | SDL_AudioFilter) (struct SDL_AudioCVT *cvt, SDL_AudioFormat format) |
Pause audio functions | |
These functions pause and unpause the audio callback processing. They should be called with a parameter of 0 after opening the audio device to start playing sound. This is so you can safely initialize data for your callback function after opening the audio device. Silence will be written to the audio device during the pause. | |
#define | SDL_LoadWAV(file, spec, audio_buf, audio_len) SDL_LoadWAV_RW(SDL_RWFromFile(file, "rb"),1, spec,audio_buf,audio_len) |
#define | SDL_MIX_MAXVOLUME 128 |
typedef struct _SDL_AudioStream | SDL_AudioStream |
void | SDL_PauseAudio (int pause_on) |
void | SDL_PauseAudioDevice (SDL_AudioDeviceID dev, int pause_on) |
SDL_AudioSpec * | SDL_LoadWAV_RW (SDL_RWops *src, int freesrc, SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len) |
void | SDL_FreeWAV (Uint8 *audio_buf) |
int | SDL_BuildAudioCVT (SDL_AudioCVT *cvt, SDL_AudioFormat src_format, Uint8 src_channels, int src_rate, SDL_AudioFormat dst_format, Uint8 dst_channels, int dst_rate) |
int | SDL_ConvertAudio (SDL_AudioCVT *cvt) |
SDL_AudioStream * | SDL_NewAudioStream (const SDL_AudioFormat src_format, const Uint8 src_channels, const int src_rate, const SDL_AudioFormat dst_format, const Uint8 dst_channels, const int dst_rate) |
int | SDL_AudioStreamPut (SDL_AudioStream *stream, const void *buf, int len) |
int | SDL_AudioStreamGet (SDL_AudioStream *stream, void *buf, int len) |
int | SDL_AudioStreamAvailable (SDL_AudioStream *stream) |
int | SDL_AudioStreamFlush (SDL_AudioStream *stream) |
void | SDL_AudioStreamClear (SDL_AudioStream *stream) |
void | SDL_FreeAudioStream (SDL_AudioStream *stream) |
void | SDL_MixAudio (Uint8 *dst, const Uint8 *src, Uint32 len, int volume) |
void | SDL_MixAudioFormat (Uint8 *dst, const Uint8 *src, SDL_AudioFormat format, Uint32 len, int volume) |
int | SDL_QueueAudio (SDL_AudioDeviceID dev, const void *data, Uint32 len) |
Uint32 | SDL_DequeueAudio (SDL_AudioDeviceID dev, void *data, Uint32 len) |
Uint32 | SDL_GetQueuedAudioSize (SDL_AudioDeviceID dev) |
void | SDL_ClearQueuedAudio (SDL_AudioDeviceID dev) |
Audio state | |
enum | SDL_AudioStatus { SDL_AUDIO_STOPPED = 0 , SDL_AUDIO_PLAYING , SDL_AUDIO_PAUSED } |
SDL_AudioStatus | SDL_GetAudioStatus (void) |
SDL_AudioStatus | SDL_GetAudioDeviceStatus (SDL_AudioDeviceID dev) |
Initialization and cleanup | |
These functions are used internally, and should not be used unless you have a specific need to specify the audio driver you want to use. You should normally use SDL_Init() or SDL_InitSubSystem(). | |
typedef Uint32 | SDL_AudioDeviceID |
int | SDL_AudioInit (const char *driver_name) |
void | SDL_AudioQuit (void) |
const char * | SDL_GetCurrentAudioDriver (void) |
int | SDL_OpenAudio (SDL_AudioSpec *desired, SDL_AudioSpec *obtained) |
int | SDL_GetNumAudioDevices (int iscapture) |
const char * | SDL_GetAudioDeviceName (int index, int iscapture) |
int | SDL_GetAudioDeviceSpec (int index, int iscapture, SDL_AudioSpec *spec) |
SDL_AudioDeviceID | SDL_OpenAudioDevice (const char *device, int iscapture, const SDL_AudioSpec *desired, SDL_AudioSpec *obtained, int allowed_changes) |
Access to the raw audio mixing buffer for the SDL library.
Definition in file SDL_audio.h.
#define AUDIO_F32 AUDIO_F32LSB |
Definition at line 116 of file SDL_audio.h.
#define AUDIO_F32LSB 0x8120 |
32-bit floating point samples
Definition at line 114 of file SDL_audio.h.
#define AUDIO_F32MSB 0x9120 |
As above, but big-endian byte order
Definition at line 115 of file SDL_audio.h.
#define AUDIO_F32SYS AUDIO_F32LSB |
Definition at line 127 of file SDL_audio.h.
#define AUDIO_S16 AUDIO_S16LSB |
Definition at line 98 of file SDL_audio.h.
#define AUDIO_S16LSB 0x8010 |
Signed 16-bit samples
Definition at line 94 of file SDL_audio.h.
#define AUDIO_S16MSB 0x9010 |
As above, but big-endian byte order
Definition at line 96 of file SDL_audio.h.
#define AUDIO_S16SYS AUDIO_S16LSB |
Definition at line 125 of file SDL_audio.h.
#define AUDIO_S32 AUDIO_S32LSB |
Definition at line 107 of file SDL_audio.h.
#define AUDIO_S32LSB 0x8020 |
32-bit integer samples
Definition at line 105 of file SDL_audio.h.
#define AUDIO_S32MSB 0x9020 |
As above, but big-endian byte order
Definition at line 106 of file SDL_audio.h.
#define AUDIO_S32SYS AUDIO_S32LSB |
Definition at line 126 of file SDL_audio.h.
#define AUDIO_S8 0x8008 |
Signed 8-bit samples
Definition at line 92 of file SDL_audio.h.
#define AUDIO_U16 AUDIO_U16LSB |
Definition at line 97 of file SDL_audio.h.
#define AUDIO_U16LSB 0x0010 |
Unsigned 16-bit samples
Definition at line 93 of file SDL_audio.h.
#define AUDIO_U16MSB 0x1010 |
As above, but big-endian byte order
Definition at line 95 of file SDL_audio.h.
#define AUDIO_U16SYS AUDIO_U16LSB |
Definition at line 124 of file SDL_audio.h.
#define AUDIO_U8 0x0008 |
Unsigned 8-bit samples
Definition at line 91 of file SDL_audio.h.
#define SDL_AUDIO_ALLOW_ANY_CHANGE (SDL_AUDIO_ALLOW_FREQUENCY_CHANGE|SDL_AUDIO_ALLOW_FORMAT_CHANGE|SDL_AUDIO_ALLOW_CHANNELS_CHANGE|SDL_AUDIO_ALLOW_SAMPLES_CHANGE) |
Definition at line 146 of file SDL_audio.h.
#define SDL_AUDIO_ALLOW_CHANNELS_CHANGE 0x00000004 |
Definition at line 144 of file SDL_audio.h.
#define SDL_AUDIO_ALLOW_FORMAT_CHANGE 0x00000002 |
Definition at line 143 of file SDL_audio.h.
#define SDL_AUDIO_ALLOW_FREQUENCY_CHANGE 0x00000001 |
Definition at line 142 of file SDL_audio.h.
#define SDL_AUDIO_ALLOW_SAMPLES_CHANGE 0x00000008 |
Definition at line 145 of file SDL_audio.h.
#define SDL_AUDIO_BITSIZE | ( | x | ) | (x & SDL_AUDIO_MASK_BITSIZE) |
Definition at line 77 of file SDL_audio.h.
#define SDL_AUDIO_ISBIGENDIAN | ( | x | ) | (x & SDL_AUDIO_MASK_ENDIAN) |
Definition at line 79 of file SDL_audio.h.
#define SDL_AUDIO_ISFLOAT | ( | x | ) | (x & SDL_AUDIO_MASK_DATATYPE) |
Definition at line 78 of file SDL_audio.h.
#define SDL_AUDIO_ISINT | ( | x | ) | (!SDL_AUDIO_ISFLOAT(x)) |
Definition at line 81 of file SDL_audio.h.
#define SDL_AUDIO_ISLITTLEENDIAN | ( | x | ) | (!SDL_AUDIO_ISBIGENDIAN(x)) |
Definition at line 82 of file SDL_audio.h.
#define SDL_AUDIO_ISSIGNED | ( | x | ) | (x & SDL_AUDIO_MASK_SIGNED) |
Definition at line 80 of file SDL_audio.h.
#define SDL_AUDIO_ISUNSIGNED | ( | x | ) | (!SDL_AUDIO_ISSIGNED(x)) |
Definition at line 83 of file SDL_audio.h.
#define SDL_AUDIO_MASK_BITSIZE (0xFF) |
Definition at line 73 of file SDL_audio.h.
#define SDL_AUDIO_MASK_DATATYPE (1<<8) |
Definition at line 74 of file SDL_audio.h.
#define SDL_AUDIO_MASK_ENDIAN (1<<12) |
Definition at line 75 of file SDL_audio.h.
#define SDL_AUDIO_MASK_SIGNED (1<<15) |
Definition at line 76 of file SDL_audio.h.
#define SDL_AUDIOCVT_MAX_FILTERS 9 |
Upper limit of filters in SDL_AudioCVT.
The maximum number of SDL_AudioFilter functions in SDL_AudioCVT is currently limited to 9. The SDL_AudioCVT.filters array has 10 pointers, one of which is the terminating NULL pointer.
Definition at line 205 of file SDL_audio.h.
#define SDL_AUDIOCVT_PACKED |
Definition at line 228 of file SDL_audio.h.
#define SDL_LoadWAV | ( | file, | |
spec, | |||
audio_buf, | |||
audio_len | |||
) | SDL_LoadWAV_RW(SDL_RWFromFile(file, "rb"),1, spec,audio_buf,audio_len) |
Loads a WAV from a file. Compatibility convenience function.
Definition at line 844 of file SDL_audio.h.
#define SDL_MIX_MAXVOLUME 128 |
Definition at line 1085 of file SDL_audio.h.
typedef void( * SDL_AudioCallback) (void *userdata, Uint8 *stream, int len) |
This function is called when the audio device needs more data.
userdata | An application-specific parameter saved in the SDL_AudioSpec structure |
stream | A pointer to the audio data buffer. |
len | The length of that buffer in bytes. |
Once the callback returns, the buffer will no longer be valid. Stereo samples are stored in a LRLRLR ordering.
You can choose to avoid callbacks and use SDL_QueueAudio() instead, if you like. Just open your audio device with a NULL callback.
Definition at line 165 of file SDL_audio.h.
typedef Uint32 SDL_AudioDeviceID |
SDL Audio Device IDs.
A successful call to SDL_OpenAudio() is always device id 1, and legacy SDL audio APIs assume you want this device ID. SDL_OpenAudioDevice() calls always returns devices >= 2 on success. The legacy calls are good both for backwards compatibility and when you don't care about multiple, specific, or capture devices.
Definition at line 419 of file SDL_audio.h.
typedef void( * SDL_AudioFilter) (struct SDL_AudioCVT *cvt, SDL_AudioFormat format) |
Definition at line 195 of file SDL_audio.h.
typedef Uint16 SDL_AudioFormat |
Audio format flags.
These are what the 16 bits in SDL_AudioFormat currently mean... (Unspecified bits are always zero).
++-----------------------sample is signed if set || || ++-----------sample is bigendian if set || || || || ++---sample is float if set || || || || || || +---sample bit size---+ || || || | | 15 14 13 12 11 10 09 08 07 06 05 04 03 02 01 00
There are macros in SDL 2.0 and later to query these bits.
Definition at line 66 of file SDL_audio.h.
typedef struct _SDL_AudioStream SDL_AudioStream |
Definition at line 942 of file SDL_audio.h.
enum SDL_AudioStatus |
Enumerator | |
---|---|
SDL_AUDIO_STOPPED | |
SDL_AUDIO_PLAYING | |
SDL_AUDIO_PAUSED |
Definition at line 648 of file SDL_audio.h.
int SDL_AudioInit | ( | const char * | driver_name | ) |
Use this function to initialize a particular audio driver.
This function is used internally, and should not be used unless you have a specific need to designate the audio driver you want to use. You should normally use SDL_Init() or SDL_InitSubSystem().
driver_name | the name of the desired audio driver |
void SDL_AudioQuit | ( | void | ) |
Use this function to shut down audio if you initialized it with SDL_AudioInit().
This function is used internally, and should not be used unless you have a specific need to specify the audio driver you want to use. You should normally use SDL_Quit() or SDL_QuitSubSystem().
int SDL_AudioStreamAvailable | ( | SDL_AudioStream * | stream | ) |
Get the number of converted/resampled bytes available.
The stream may be buffering data behind the scenes until it has enough to resample correctly, so this number might be lower than what you expect, or even be zero. Add more data or flush the stream if you need the data now.
void SDL_AudioStreamClear | ( | SDL_AudioStream * | stream | ) |
Clear any pending data in the stream without converting it
int SDL_AudioStreamFlush | ( | SDL_AudioStream * | stream | ) |
Tell the stream that you're done sending data, and anything being buffered should be converted/resampled and made available immediately.
It is legal to add more data to a stream after flushing, but there will be audio gaps in the output. Generally this is intended to signal the end of input, so the complete output becomes available.
int SDL_AudioStreamGet | ( | SDL_AudioStream * | stream, |
void * | buf, | ||
int | len | ||
) |
Get converted/resampled data from the stream
stream | The stream the audio is being requested from |
buf | A buffer to fill with audio data |
len | The maximum number of bytes to fill |
int SDL_AudioStreamPut | ( | SDL_AudioStream * | stream, |
const void * | buf, | ||
int | len | ||
) |
Add data to be converted/resampled to the stream.
stream | The stream the audio data is being added to |
buf | A pointer to the audio data to add |
len | The number of bytes to write to the stream |
int SDL_BuildAudioCVT | ( | SDL_AudioCVT * | cvt, |
SDL_AudioFormat | src_format, | ||
Uint8 | src_channels, | ||
int | src_rate, | ||
SDL_AudioFormat | dst_format, | ||
Uint8 | dst_channels, | ||
int | dst_rate | ||
) |
Initialize an SDL_AudioCVT structure for conversion.
Before an SDL_AudioCVT structure can be used to convert audio data it must be initialized with source and destination information.
This function will zero out every field of the SDL_AudioCVT, so it must be called before the application fills in the final buffer information.
Once this function has returned successfully, and reported that a conversion is necessary, the application fills in the rest of the fields in SDL_AudioCVT, now that it knows how large a buffer it needs to allocate, and then can call SDL_ConvertAudio() to complete the conversion.
cvt | an SDL_AudioCVT structure filled in with audio conversion information |
src_format | the source format of the audio data; for more info see SDL_AudioFormat |
src_channels | the number of channels in the source |
src_rate | the frequency (sample-frames-per-second) of the source |
dst_format | the destination format of the audio data; for more info see SDL_AudioFormat |
dst_channels | the number of channels in the destination |
dst_rate | the frequency (sample-frames-per-second) of the destination |
void SDL_ClearQueuedAudio | ( | SDL_AudioDeviceID | dev | ) |
Drop any queued audio data waiting to be sent to the hardware.
Immediately after this call, SDL_GetQueuedAudioSize() will return 0. For output devices, the hardware will start playing silence if more audio isn't queued. For capture devices, the hardware will start filling the empty queue with new data if the capture device isn't paused.
This will not prevent playback of queued audio that's already been sent to the hardware, as we can not undo that, so expect there to be some fraction of a second of audio that might still be heard. This can be useful if you want to, say, drop any pending music or any unprocessed microphone input during a level change in your game.
You may not queue or dequeue audio on a device that is using an application-supplied callback; calling this function on such a device always returns 0. You have to use the audio callback or queue audio, but not both.
You should not call SDL_LockAudio() on the device before clearing the queue; SDL handles locking internally for this function.
This function always succeeds and thus returns void.
dev | the device ID of which to clear the audio queue |
void SDL_CloseAudio | ( | void | ) |
This function is a legacy means of closing the audio device.
This function is equivalent to calling...
...and is only useful if you used the legacy SDL_OpenAudio() function.
void SDL_CloseAudioDevice | ( | SDL_AudioDeviceID | dev | ) |
Use this function to shut down audio processing and close the audio device.
The application should close open audio devices once they are no longer needed. Calling this function will wait until the device's audio callback is not running, release the audio hardware and then clean up internal state. No further audio will play from this device once this function returns.
This function may block briefly while pending audio data is played by the hardware, so that applications don't drop the last buffer of data they supplied.
The device ID is invalid as soon as the device is closed, and is eligible for reuse in a new SDL_OpenAudioDevice() call immediately.
dev | an audio device previously opened with SDL_OpenAudioDevice() |
int SDL_ConvertAudio | ( | SDL_AudioCVT * | cvt | ) |
Convert audio data to a desired audio format.
This function does the actual audio data conversion, after the application has called SDL_BuildAudioCVT() to prepare the conversion information and then filled in the buffer details.
Once the application has initialized the cvt
structure using SDL_BuildAudioCVT(), allocated an audio buffer and filled it with audio data in the source format, this function will convert the buffer, in-place, to the desired format.
The data conversion may go through several passes; any given pass may possibly temporarily increase the size of the data. For example, SDL might expand 16-bit data to 32 bits before resampling to a lower frequency, shrinking the data size after having grown it briefly. Since the supplied buffer will be both the source and destination, converting as necessary in-place, the application must allocate a buffer that will fully contain the data during its largest conversion pass. After SDL_BuildAudioCVT() returns, the application should set the cvt->len
field to the size, in bytes, of the source data, and allocate a buffer that is cvt->len * cvt->len_mult
bytes long for the buf
field.
The source data should be copied into this buffer before the call to SDL_ConvertAudio(). Upon successful return, this buffer will contain the converted audio, and cvt->len_cvt
will be the size of the converted data, in bytes. Any bytes in the buffer past cvt->len_cvt
are undefined once this function returns.
cvt | an SDL_AudioCVT structure that was previously set up by SDL_BuildAudioCVT(). |
Uint32 SDL_DequeueAudio | ( | SDL_AudioDeviceID | dev, |
void * | data, | ||
Uint32 | len | ||
) |
Dequeue more audio on non-callback devices.
If you are looking to queue audio for output on a non-callback playback device, you want SDL_QueueAudio() instead. SDL_DequeueAudio() will always return 0 if you use it with playback devices.
SDL offers two ways to retrieve audio from a capture device: you can either supply a callback that SDL triggers with some frequency as the device records more audio data, (push method), or you can supply no callback, and then SDL will expect you to retrieve data at regular intervals (pull method) with this function.
There are no limits on the amount of data you can queue, short of exhaustion of address space. Data from the device will keep queuing as necessary without further intervention from you. This means you will eventually run out of memory if you aren't routinely dequeueing data.
Capture devices will not queue data when paused; if you are expecting to not need captured audio for some length of time, use SDL_PauseAudioDevice() to stop the capture device from queueing more data. This can be useful during, say, level loading times. When unpaused, capture devices will start queueing data from that point, having flushed any capturable data available while paused.
This function is thread-safe, but dequeueing from the same device from two threads at once does not promise which thread will dequeue data first.
You may not dequeue audio from a device that is using an application-supplied callback; doing so returns an error. You have to use the audio callback, or dequeue audio with this function, but not both.
You should not call SDL_LockAudio() on the device before dequeueing; SDL handles locking internally for this function.
dev | the device ID from which we will dequeue audio |
data | a pointer into where audio data should be copied |
len | the number of bytes (not samples!) to which (data) points |
void SDL_FreeAudioStream | ( | SDL_AudioStream * | stream | ) |
Free an audio stream
void SDL_FreeWAV | ( | Uint8 * | audio_buf | ) |
Free data previously allocated with SDL_LoadWAV() or SDL_LoadWAV_RW().
After a WAVE file has been opened with SDL_LoadWAV() or SDL_LoadWAV_RW() its data can eventually be freed with SDL_FreeWAV(). It is safe to call this function with a NULL pointer.
audio_buf | a pointer to the buffer created by SDL_LoadWAV() or SDL_LoadWAV_RW() |
const char* SDL_GetAudioDeviceName | ( | int | index, |
int | iscapture | ||
) |
Get the human-readable name of a specific audio device.
This function is only valid after successfully initializing the audio subsystem. The values returned by this function reflect the latest call to SDL_GetNumAudioDevices(); re-call that function to redetect available hardware.
The string returned by this function is UTF-8 encoded, read-only, and managed internally. You are not to free it. If you need to keep the string for any length of time, you should make your own copy of it, as it will be invalid next time any of several other SDL functions are called.
index | the index of the audio device; valid values range from 0 to SDL_GetNumAudioDevices() - 1 |
iscapture | non-zero to query the list of recording devices, zero to query the list of output devices. |
int SDL_GetAudioDeviceSpec | ( | int | index, |
int | iscapture, | ||
SDL_AudioSpec * | spec | ||
) |
Get the preferred audio format of a specific audio device.
This function is only valid after a successfully initializing the audio subsystem. The values returned by this function reflect the latest call to SDL_GetNumAudioDevices(); re-call that function to redetect available hardware.
spec
will be filled with the sample rate, sample format, and channel count. All other values in the structure are filled with 0. When the supported struct members are 0, SDL was unable to get the property from the backend.
index | the index of the audio device; valid values range from 0 to SDL_GetNumAudioDevices() - 1 |
iscapture | non-zero to query the list of recording devices, zero to query the list of output devices. |
spec | The SDL_AudioSpec to be initialized by this function. |
SDL_AudioStatus SDL_GetAudioDeviceStatus | ( | SDL_AudioDeviceID | dev | ) |
Use this function to get the current audio state of an audio device.
dev | the ID of an audio device previously opened with SDL_OpenAudioDevice() |
const char* SDL_GetAudioDriver | ( | int | index | ) |
Use this function to get the name of a built in audio driver.
The list of audio drivers is given in the order that they are normally initialized by default; the drivers that seem more reasonable to choose first (as far as the SDL developers believe) are earlier in the list.
The names of drivers are all simple, low-ASCII identifiers, like "alsa", "coreaudio" or "xaudio2". These never have Unicode characters, and are not meant to be proper names.
index | the index of the audio driver; the value ranges from 0 to SDL_GetNumAudioDrivers() - 1 |
SDL_AudioStatus SDL_GetAudioStatus | ( | void | ) |
This function is a legacy means of querying the audio device.
New programs might want to use SDL_GetAudioDeviceStatus() instead. This function is equivalent to calling...
...and is only useful if you used the legacy SDL_OpenAudio() function.
const char* SDL_GetCurrentAudioDriver | ( | void | ) |
Get the name of the current audio driver.
The returned string points to internal static memory and thus never becomes invalid, even if you quit the audio subsystem and initialize a new driver (although such a case would return a different static string from another call to this function, of course). As such, you should not modify or free the returned string.
int SDL_GetNumAudioDevices | ( | int | iscapture | ) |
Get the number of built-in audio devices.
This function is only valid after successfully initializing the audio subsystem.
Note that audio capture support is not implemented as of SDL 2.0.4, so the iscapture
parameter is for future expansion and should always be zero for now.
This function will return -1 if an explicit list of devices can't be determined. Returning -1 is not an error. For example, if SDL is set up to talk to a remote audio server, it can't list every one available on the Internet, but it will still allow a specific host to be specified in SDL_OpenAudioDevice().
In many common cases, when this function returns a value <= 0, it can still successfully open the default device (NULL for first argument of SDL_OpenAudioDevice()).
This function may trigger a complete redetect of available hardware. It should not be called for each iteration of a loop, but rather once at the start of a loop:
iscapture | zero to request playback devices, non-zero to request recording devices |
int SDL_GetNumAudioDrivers | ( | void | ) |
Use this function to get the number of built-in audio drivers.
This function returns a hardcoded number. This never returns a negative value; if there are no drivers compiled into this build of SDL, this function returns zero. The presence of a driver in this list does not mean it will function, it just means SDL is capable of interacting with that interface. For example, a build of SDL might have esound support, but if there's no esound server available, SDL's esound driver would fail if used.
By default, SDL tries all drivers, in its preferred order, until one is found to be usable.
Uint32 SDL_GetQueuedAudioSize | ( | SDL_AudioDeviceID | dev | ) |
Get the number of bytes of still-queued audio.
For playback devices: this is the number of bytes that have been queued for playback with SDL_QueueAudio(), but have not yet been sent to the hardware.
Once we've sent it to the hardware, this function can not decide the exact byte boundary of what has been played. It's possible that we just gave the hardware several kilobytes right before you called this function, but it hasn't played any of it yet, or maybe half of it, etc.
For capture devices, this is the number of bytes that have been captured by the device and are waiting for you to dequeue. This number may grow at any time, so this only informs of the lower-bound of available data.
You may not queue or dequeue audio on a device that is using an application-supplied callback; calling this function on such a device always returns 0. You have to use the audio callback or queue audio, but not both.
You should not call SDL_LockAudio() on the device before querying; SDL handles locking internally for this function.
dev | the device ID of which we will query queued audio size |
SDL_AudioSpec* SDL_LoadWAV_RW | ( | SDL_RWops * | src, |
int | freesrc, | ||
SDL_AudioSpec * | spec, | ||
Uint8 ** | audio_buf, | ||
Uint32 * | audio_len | ||
) |
Load the audio data of a WAVE file into memory.
Loading a WAVE file requires src
, spec
, audio_buf
and audio_len
to be valid pointers. The entire data portion of the file is then loaded into memory and decoded if necessary.
If freesrc
is non-zero, the data source gets automatically closed and freed before the function returns.
Supported formats are RIFF WAVE files with the formats PCM (8, 16, 24, and 32 bits), IEEE Float (32 bits), Microsoft ADPCM and IMA ADPCM (4 bits), and A-law and mu-law (8 bits). Other formats are currently unsupported and cause an error.
If this function succeeds, the pointer returned by it is equal to spec
and the pointer to the audio data allocated by the function is written to audio_buf
and its length in bytes to audio_len
. The SDL_AudioSpec members freq
, channels
, and format
are set to the values of the audio data in the buffer. The samples
member is set to a sane default and all others are set to zero.
It's necessary to use SDL_FreeWAV() to free the audio data returned in audio_buf
when it is no longer used.
Because of the underspecification of the .WAV format, there are many problematic files in the wild that cause issues with strict decoders. To provide compatibility with these files, this decoder is lenient in regards to the truncation of the file, the fact chunk, and the size of the RIFF chunk. The hints SDL_HINT_WAVE_RIFF_CHUNK_SIZE
, SDL_HINT_WAVE_TRUNCATION
, and SDL_HINT_WAVE_FACT_CHUNK
can be used to tune the behavior of the loading process.
Any file that is invalid (due to truncation, corruption, or wrong values in the headers), too big, or unsupported causes an error. Additionally, any critical I/O error from the data source will terminate the loading process with an error. The function returns NULL on error and in all cases (with the exception of src
being NULL), an appropriate error message will be set.
It is required that the data source supports seeking.
Example:
Note that the SDL_LoadWAV macro does this same thing for you, but in a less messy way:
src | The data source for the WAVE data |
freesrc | If non-zero, SDL will always free the data source |
spec | An SDL_AudioSpec that will be filled in with the wave file's format details |
audio_buf | A pointer filled with the audio data, allocated by the function. |
audio_len | A pointer filled with the length of the audio data buffer in bytes |
spec
, which will be filled with the audio data format of the wave source data. audio_buf
will be filled with a pointer to an allocated buffer containing the audio data, and audio_len
is filled with the length of that audio buffer in bytes.This function returns NULL if the .WAV file cannot be opened, uses an unknown data format, or is corrupt; call SDL_GetError() for more information.
When the application is done with the data returned in audio_buf
, it should call SDL_FreeWAV() to dispose of it.
void SDL_LockAudio | ( | void | ) |
This function is a legacy means of locking the audio device.
New programs might want to use SDL_LockAudioDevice() instead. This function is equivalent to calling...
...and is only useful if you used the legacy SDL_OpenAudio() function.
void SDL_LockAudioDevice | ( | SDL_AudioDeviceID | dev | ) |
Use this function to lock out the audio callback function for a specified device.
The lock manipulated by these functions protects the audio callback function specified in SDL_OpenAudioDevice(). During a SDL_LockAudioDevice()/SDL_UnlockAudioDevice() pair, you can be guaranteed that the callback function for that device is not running, even if the device is not paused. While a device is locked, any other unpaused, unlocked devices may still run their callbacks.
Calling this function from inside your audio callback is unnecessary. SDL obtains this lock before calling your function, and releases it when the function returns.
You should not hold the lock longer than absolutely necessary. If you hold it too long, you'll experience dropouts in your audio playback. Ideally, your application locks the device, sets a few variables and unlocks again. Do not do heavy work while holding the lock for a device.
It is safe to lock the audio device multiple times, as long as you unlock it an equivalent number of times. The callback will not run until the device has been unlocked completely in this way. If your application fails to unlock the device appropriately, your callback will never run, you might hear repeating bursts of audio, and SDL_CloseAudioDevice() will probably deadlock.
Internally, the audio device lock is a mutex; if you lock from two threads at once, not only will you block the audio callback, you'll block the other thread.
dev | the ID of the device to be locked |
This function is a legacy means of mixing audio.
This function is equivalent to calling...
...where format
is the obtained format of the audio device from the legacy SDL_OpenAudio() function.
dst | the destination for the mixed audio |
src | the source audio buffer to be mixed |
len | the length of the audio buffer in bytes |
volume | ranges from 0 - 128, and should be set to SDL_MIX_MAXVOLUME for full audio volume |
void SDL_MixAudioFormat | ( | Uint8 * | dst, |
const Uint8 * | src, | ||
SDL_AudioFormat | format, | ||
Uint32 | len, | ||
int | volume | ||
) |
Mix audio data in a specified format.
This takes an audio buffer src
of len
bytes of format
data and mixes it into dst
, performing addition, volume adjustment, and overflow clipping. The buffer pointed to by dst
must also be len
bytes of format
data.
This is provided for convenience – you can mix your own audio data.
Do not use this function for mixing together more than two streams of sample data. The output from repeated application of this function may be distorted by clipping, because there is no accumulator with greater range than the input (not to mention this being an inefficient way of doing it).
It is a common misconception that this function is required to write audio data to an output stream in an audio callback. While you can do that, SDL_MixAudioFormat() is really only needed when you're mixing a single audio stream with a volume adjustment.
dst | the destination for the mixed audio |
src | the source audio buffer to be mixed |
format | the SDL_AudioFormat structure representing the desired audio format |
len | the length of the audio buffer in bytes |
volume | ranges from 0 - 128, and should be set to SDL_MIX_MAXVOLUME for full audio volume |
SDL_AudioStream* SDL_NewAudioStream | ( | const SDL_AudioFormat | src_format, |
const Uint8 | src_channels, | ||
const int | src_rate, | ||
const SDL_AudioFormat | dst_format, | ||
const Uint8 | dst_channels, | ||
const int | dst_rate | ||
) |
Create a new audio stream.
src_format | The format of the source audio |
src_channels | The number of channels of the source audio |
src_rate | The sampling rate of the source audio |
dst_format | The format of the desired audio output |
dst_channels | The number of channels of the desired audio output |
dst_rate | The sampling rate of the desired audio output |
int SDL_OpenAudio | ( | SDL_AudioSpec * | desired, |
SDL_AudioSpec * | obtained | ||
) |
This function is a legacy means of opening the audio device.
This function remains for compatibility with SDL 1.2, but also because it's slightly easier to use than the new functions in SDL 2.0. The new, more powerful, and preferred way to do this is SDL_OpenAudioDevice().
This function is roughly equivalent to:
With two notable exceptions:
obtained
is NULL, we use desired
(and allow no changes), which means desired will be modified to have the correct values for silence, etc, and SDL will convert any differences between your app's specific request and the hardware behind the scenes.desired | an SDL_AudioSpec structure representing the desired output format. Please refer to the SDL_OpenAudioDevice documentation for details on how to prepare this structure. |
obtained | an SDL_AudioSpec structure filled in with the actual parameters, or NULL. |
obtained
.If obtained
is NULL, the audio data passed to the callback function will be guaranteed to be in the requested format, and will be automatically converted to the actual hardware audio format if necessary. If obtained
is NULL, desired
will have fields modified.
This function returns a negative error code on failure to open the audio device or failure to set up the audio thread; call SDL_GetError() for more information.
SDL_AudioDeviceID SDL_OpenAudioDevice | ( | const char * | device, |
int | iscapture, | ||
const SDL_AudioSpec * | desired, | ||
SDL_AudioSpec * | obtained, | ||
int | allowed_changes | ||
) |
Open a specific audio device.
SDL_OpenAudio(), unlike this function, always acts on device ID 1. As such, this function will never return a 1 so as not to conflict with the legacy function.
Please note that SDL 2.0 before 2.0.5 did not support recording; as such, this function would fail if iscapture
was not zero. Starting with SDL 2.0.5, recording is implemented and this value can be non-zero.
Passing in a device
name of NULL requests the most reasonable default (and is equivalent to what SDL_OpenAudio() does to choose a device). The device
name is a UTF-8 string reported by SDL_GetAudioDeviceName(), but some drivers allow arbitrary and driver-specific strings, such as a hostname/IP address for a remote audio server, or a filename in the diskaudio driver.
An opened audio device starts out paused, and should be enabled for playing by calling SDL_PauseAudioDevice(devid, 0) when you are ready for your audio callback function to be called. Since the audio driver may modify the requested size of the audio buffer, you should allocate any local mixing buffers after you open the audio device.
The audio callback runs in a separate thread in most cases; you can prevent race conditions between your callback and other threads without fully pausing playback with SDL_LockAudioDevice(). For more information about the callback, see SDL_AudioSpec.
Managing the audio spec via 'desired' and 'obtained':
When filling in the desired audio spec structure:
desired->freq
should be the frequency in sample-frames-per-second (Hz).desired->format
should be the audio format (AUDIO_S16SYS
, etc).desired->samples
is the desired size of the audio buffer, in sample frames (with stereo output, two samples–left and right–would make a single sample frame). This number should be a power of two, and may be adjusted by the audio driver to a value more suitable for the hardware. Good values seem to range between 512 and 8096 inclusive, depending on the application and CPU speed. Smaller values reduce latency, but can lead to underflow if the application is doing heavy processing and cannot fill the audio buffer in time. Note that the number of sample frames is directly related to time by the following formula: ms = (sampleframes*1000)/freq
desired->size
is the size in bytes of the audio buffer, and is calculated by SDL_OpenAudioDevice(). You don't initialize this.desired->silence
is the value used to set the buffer to silence, and is calculated by SDL_OpenAudioDevice(). You don't initialize this.desired->callback
should be set to a function that will be called when the audio device is ready for more data. It is passed a pointer to the audio buffer, and the length in bytes of the audio buffer. This function usually runs in a separate thread, and so you should protect data structures that it accesses by calling SDL_LockAudioDevice() and SDL_UnlockAudioDevice() in your code. Alternately, you may pass a NULL pointer here, and call SDL_QueueAudio() with some frequency, to queue more audio samples to be played (or for capture devices, call SDL_DequeueAudio() with some frequency, to obtain audio samples).desired->userdata
is passed as the first parameter to your callback function. If you passed a NULL callback, this value is ignored.allowed_changes
can have the following flags OR'd together:
SDL_AUDIO_ALLOW_FREQUENCY_CHANGE
SDL_AUDIO_ALLOW_FORMAT_CHANGE
SDL_AUDIO_ALLOW_CHANNELS_CHANGE
SDL_AUDIO_ALLOW_ANY_CHANGE
These flags specify how SDL should behave when a device cannot offer a specific feature. If the application requests a feature that the hardware doesn't offer, SDL will always try to get the closest equivalent.
For example, if you ask for float32 audio format, but the sound card only supports int16, SDL will set the hardware to int16. If you had set SDL_AUDIO_ALLOW_FORMAT_CHANGE, SDL will change the format in the obtained
structure. If that flag was not set, SDL will prepare to convert your callback's float32 audio to int16 before feeding it to the hardware and will keep the originally requested format in the obtained
structure.
The resulting audio specs, varying depending on hardware and on what changes were allowed, will then be written back to obtained
.
If your application can only handle one specific data format, pass a zero for allowed_changes
and let SDL transparently handle any differences.
device | a UTF-8 string reported by SDL_GetAudioDeviceName() or a driver-specific name as appropriate. NULL requests the most reasonable default device. |
iscapture | non-zero to specify a device should be opened for recording, not playback |
desired | an SDL_AudioSpec structure representing the desired output format; see SDL_OpenAudio() for more information |
obtained | an SDL_AudioSpec structure filled in with the actual output format; see SDL_OpenAudio() for more information |
allowed_changes | 0, or one or more flags OR'd together |
For compatibility with SDL 1.2, this will never return 1, since SDL reserves that ID for the legacy SDL_OpenAudio() function.
void SDL_PauseAudio | ( | int | pause_on | ) |
This function is a legacy means of pausing the audio device.
New programs might want to use SDL_PauseAudioDevice() instead. This function is equivalent to calling...
...and is only useful if you used the legacy SDL_OpenAudio() function.
pause_on | non-zero to pause, 0 to unpause |
void SDL_PauseAudioDevice | ( | SDL_AudioDeviceID | dev, |
int | pause_on | ||
) |
Use this function to pause and unpause audio playback on a specified device.
This function pauses and unpauses the audio callback processing for a given device. Newly-opened audio devices start in the paused state, so you must call this function with pause_on=0 after opening the specified audio device to start playing sound. This allows you to safely initialize data for your callback function after opening the audio device. Silence will be written to the audio device while paused, and the audio callback is guaranteed to not be called. Pausing one device does not prevent other unpaused devices from running their callbacks.
Pausing state does not stack; even if you pause a device several times, a single unpause will start the device playing again, and vice versa. This is different from how SDL_LockAudioDevice() works.
If you just need to protect a few variables from race conditions vs your callback, you shouldn't pause the audio device, as it will lead to dropouts in the audio playback. Instead, you should use SDL_LockAudioDevice().
dev | a device opened by SDL_OpenAudioDevice() |
pause_on | non-zero to pause, 0 to unpause |
int SDL_QueueAudio | ( | SDL_AudioDeviceID | dev, |
const void * | data, | ||
Uint32 | len | ||
) |
Queue more audio on non-callback devices.
If you are looking to retrieve queued audio from a non-callback capture device, you want SDL_DequeueAudio() instead. SDL_QueueAudio() will return -1 to signify an error if you use it with capture devices.
SDL offers two ways to feed audio to the device: you can either supply a callback that SDL triggers with some frequency to obtain more audio (pull method), or you can supply no callback, and then SDL will expect you to supply data at regular intervals (push method) with this function.
There are no limits on the amount of data you can queue, short of exhaustion of address space. Queued data will drain to the device as necessary without further intervention from you. If the device needs audio but there is not enough queued, it will play silence to make up the difference. This means you will have skips in your audio playback if you aren't routinely queueing sufficient data.
This function copies the supplied data, so you are safe to free it when the function returns. This function is thread-safe, but queueing to the same device from two threads at once does not promise which buffer will be queued first.
You may not queue audio on a device that is using an application-supplied callback; doing so returns an error. You have to use the audio callback or queue audio with this function, but not both.
You should not call SDL_LockAudio() on the device before queueing; SDL handles locking internally for this function.
Note that SDL2 does not support planar audio. You will need to resample from planar audio formats into a non-planar one (see SDL_AudioFormat) before queuing audio.
dev | the device ID to which we will queue audio |
data | the data to queue to the device for later playback |
len | the number of bytes (not samples!) to which data points |
void SDL_UnlockAudio | ( | void | ) |
This function is a legacy means of unlocking the audio device.
New programs might want to use SDL_UnlockAudioDevice() instead. This function is equivalent to calling...
...and is only useful if you used the legacy SDL_OpenAudio() function.
void SDL_UnlockAudioDevice | ( | SDL_AudioDeviceID | dev | ) |
Use this function to unlock the audio callback function for a specified device.
This function should be paired with a previous SDL_LockAudioDevice() call.
dev | the ID of the device to be unlocked |