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arsa
2.7
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#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.
Classes | |
| 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... | |
| typedef void(SDLCALL * | SDL_AudioCallback) (void *userdata, Uint8 *stream, int len) |
| typedef struct SDL_AudioSpec | SDL_AudioSpec |
| typedef SDL_AudioFormat | format |
| typedef struct SDL_AudioCVT | SDL_AudioCVT |
| typedef Uint32 | SDL_AudioDeviceID |
| typedef struct _SDL_AudioStream | SDL_AudioStream |
Audio state | |
| enum | SDL_AudioStatus { SDL_AUDIO_STOPPED = 0, SDL_AUDIO_PLAYING, SDL_AUDIO_PAUSED } |
| DECLSPEC SDL_AudioStatus SDLCALL | SDL_GetAudioStatus (void) |
| DECLSPEC SDL_AudioStatus SDLCALL | SDL_GetAudioDeviceStatus (SDL_AudioDeviceID dev) |
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 114 of file SDL_audio.h.
| #define AUDIO_F32LSB 0x8120 |
32-bit floating point samples
Definition at line 112 of file SDL_audio.h.
| #define AUDIO_F32MSB 0x9120 |
As above, but big-endian byte order
Definition at line 113 of file SDL_audio.h.
| #define AUDIO_F32SYS AUDIO_F32LSB |
Definition at line 125 of file SDL_audio.h.
| #define AUDIO_S16 AUDIO_S16LSB |
Definition at line 96 of file SDL_audio.h.
| #define AUDIO_S16LSB 0x8010 |
Signed 16-bit samples
Definition at line 92 of file SDL_audio.h.
| #define AUDIO_S16MSB 0x9010 |
As above, but big-endian byte order
Definition at line 94 of file SDL_audio.h.
| #define AUDIO_S16SYS AUDIO_S16LSB |
Definition at line 123 of file SDL_audio.h.
| #define AUDIO_S32 AUDIO_S32LSB |
Definition at line 105 of file SDL_audio.h.
| #define AUDIO_S32LSB 0x8020 |
32-bit integer samples
Definition at line 103 of file SDL_audio.h.
| #define AUDIO_S32MSB 0x9020 |
As above, but big-endian byte order
Definition at line 104 of file SDL_audio.h.
| #define AUDIO_S32SYS AUDIO_S32LSB |
Definition at line 124 of file SDL_audio.h.
| #define AUDIO_S8 0x8008 |
Signed 8-bit samples
Definition at line 90 of file SDL_audio.h.
| #define AUDIO_U16 AUDIO_U16LSB |
Definition at line 95 of file SDL_audio.h.
| #define AUDIO_U16LSB 0x0010 |
Unsigned 16-bit samples
Definition at line 91 of file SDL_audio.h.
| #define AUDIO_U16MSB 0x1010 |
As above, but big-endian byte order
Definition at line 93 of file SDL_audio.h.
| #define AUDIO_U16SYS AUDIO_U16LSB |
Definition at line 122 of file SDL_audio.h.
| #define AUDIO_U8 0x0008 |
Unsigned 8-bit samples
Definition at line 89 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 144 of file SDL_audio.h.
| #define SDL_AUDIO_ALLOW_CHANNELS_CHANGE 0x00000004 |
Definition at line 142 of file SDL_audio.h.
| #define SDL_AUDIO_ALLOW_FORMAT_CHANGE 0x00000002 |
Definition at line 141 of file SDL_audio.h.
| #define SDL_AUDIO_ALLOW_FREQUENCY_CHANGE 0x00000001 |
Definition at line 140 of file SDL_audio.h.
| #define SDL_AUDIO_ALLOW_SAMPLES_CHANGE 0x00000008 |
Definition at line 143 of file SDL_audio.h.
| #define SDL_AUDIO_BITSIZE | ( | x | ) | (x & SDL_AUDIO_MASK_BITSIZE) |
Definition at line 75 of file SDL_audio.h.
| #define SDL_AUDIO_ISBIGENDIAN | ( | x | ) | (x & SDL_AUDIO_MASK_ENDIAN) |
Definition at line 77 of file SDL_audio.h.
| #define SDL_AUDIO_ISFLOAT | ( | x | ) | (x & SDL_AUDIO_MASK_DATATYPE) |
Definition at line 76 of file SDL_audio.h.
| #define SDL_AUDIO_ISINT | ( | x | ) | (!SDL_AUDIO_ISFLOAT(x)) |
Definition at line 79 of file SDL_audio.h.
| #define SDL_AUDIO_ISLITTLEENDIAN | ( | x | ) | (!SDL_AUDIO_ISBIGENDIAN(x)) |
Definition at line 80 of file SDL_audio.h.
| #define SDL_AUDIO_ISSIGNED | ( | x | ) | (x & SDL_AUDIO_MASK_SIGNED) |
Definition at line 78 of file SDL_audio.h.
| #define SDL_AUDIO_ISUNSIGNED | ( | x | ) | (!SDL_AUDIO_ISSIGNED(x)) |
Definition at line 81 of file SDL_audio.h.
| #define SDL_AUDIO_MASK_BITSIZE (0xFF) |
Definition at line 71 of file SDL_audio.h.
| #define SDL_AUDIO_MASK_DATATYPE (1<<8) |
Definition at line 72 of file SDL_audio.h.
| #define SDL_AUDIO_MASK_ENDIAN (1<<12) |
Definition at line 73 of file SDL_audio.h.
| #define SDL_AUDIO_MASK_SIGNED (1<<15) |
Definition at line 74 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 203 of file SDL_audio.h.
| #define SDL_AUDIOCVT_PACKED |
Definition at line 223 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 484 of file SDL_audio.h.
| #define SDL_MIX_MAXVOLUME 128 |
Definition at line 649 of file SDL_audio.h.
Definition at line 194 of file SDL_audio.h.
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 163 of file SDL_audio.h.
| typedef struct SDL_AudioCVT SDL_AudioCVT |
| 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 330 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 64 of file SDL_audio.h.
| typedef struct SDL_AudioSpec SDL_AudioSpec |
The calculated values in this structure are calculated by SDL_OpenAudio().
For multi-channel audio, the default SDL channel mapping is: 2: FL FR (stereo) 3: FL FR LFE (2.1 surround) 4: FL FR BL BR (quad) 5: FL FR FC BL BR (quad + center) 6: FL FR FC LFE SL SR (5.1 surround - last two can also be BL BR) 7: FL FR FC LFE BC SL SR (6.1 surround) 8: FL FR FC LFE BL BR SL SR (7.1 surround)
| typedef struct _SDL_AudioStream SDL_AudioStream |
Definition at line 532 of file SDL_audio.h.
| enum SDL_AudioStatus |
| Enumerator | |
|---|---|
| SDL_AUDIO_STOPPED | |
| SDL_AUDIO_PLAYING | |
| SDL_AUDIO_PAUSED | |
Definition at line 395 of file SDL_audio.h.
| DECLSPEC int SDLCALL 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.
| DECLSPEC void SDLCALL SDL_AudioStreamClear | ( | SDL_AudioStream * | stream | ) |
Clear any pending data in the stream without converting it
| DECLSPEC int SDLCALL 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.
| DECLSPEC int SDLCALL 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 |
| DECLSPEC int SDLCALL 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 |
| DECLSPEC int SDLCALL 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 | ||
| ) |
This function takes a source format and rate and a destination format and rate, and initializes the cvt structure with information needed by SDL_ConvertAudio() to convert a buffer of audio data from one format to the other. An unsupported format causes an error and -1 will be returned.
| DECLSPEC void SDLCALL SDL_ClearQueuedAudio | ( | SDL_AudioDeviceID | dev | ) |
Drop any queued audio data. For playback devices, this is any queued data still waiting to be submitted to the hardware. For capture devices, this is any data that was queued by the device that hasn't yet been dequeued by the application.
Immediately after this call, SDL_GetQueuedAudioSize() will return 0. For playback devices, the hardware will start playing silence if more audio isn't queued. Unpaused capture devices will start filling the queue again as soon as they have more data available (which, depending on the state of the hardware and the thread, could be before this function call returns!).
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 during a level change in your game.
You may not queue audio on a device that is using an application-supplied callback; calling this function on such a device is always a no-op. You have to queue audio with SDL_QueueAudio()/SDL_DequeueAudio(), or use the audio callback, 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. |
This function shuts down audio processing and closes the audio device.
| DECLSPEC void SDLCALL SDL_CloseAudioDevice | ( | SDL_AudioDeviceID | dev | ) |
| DECLSPEC int SDLCALL SDL_ConvertAudio | ( | SDL_AudioCVT * | cvt | ) |
Once you have initialized the cvt structure using SDL_BuildAudioCVT(), created an audio buffer cvt->buf, and filled it with cvt->len bytes of audio data in the source format, this function will convert it in-place to the desired format.
The data conversion may expand the size of the audio data, so the buffer cvt->buf should be allocated after the cvt structure is initialized by SDL_BuildAudioCVT(), and should be cvt->len*cvt->len_mult bytes long.
cvt->buf is NULL. 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. This 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 dequeued 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 queueing; 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. |
| DECLSPEC void SDLCALL SDL_FreeAudioStream | ( | SDL_AudioStream * | stream | ) |
Free an audio stream
This function frees data previously allocated with SDL_LoadWAV_RW()
Get the human-readable name of a specific audio device. Must be a value between 0 and (number of audio devices-1). Only valid after a successfully initializing the audio subsystem. The values returned by this function reflect the latest call to SDL_GetNumAudioDevices(); recall 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 is called.
| DECLSPEC SDL_AudioStatus SDLCALL SDL_GetAudioDeviceStatus | ( | SDL_AudioDeviceID | dev | ) |
| DECLSPEC SDL_AudioStatus SDLCALL SDL_GetAudioStatus | ( | void | ) |
This function returns the name of the current audio driver, or NULL if no driver has been initialized.
Get the number of available devices exposed by the current driver. Only valid after a successfully initializing the audio subsystem. Returns -1 if an explicit list of devices can't be determined; this 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 to 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()).
| DECLSPEC Uint32 SDLCALL SDL_GetQueuedAudioSize | ( | SDL_AudioDeviceID | dev | ) |
Get the number of bytes of still-queued audio.
For playback device:
This is the number of bytes that have been queued for playback with SDL_QueueAudio(), but have not yet been sent to the hardware. This number may shrink at any time, so this only informs of pending data.
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 audio on a device that is using an application-supplied callback; calling this function on such a device always returns 0. You have to queue audio with SDL_QueueAudio()/SDL_DequeueAudio(), or use the audio callback, 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. |
| DECLSPEC SDL_AudioSpec* SDLCALL 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 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 ยต-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 Waveform 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:
| src | The data source with the WAVE data |
| freesrc | A integer value that makes the function close the data source if non-zero |
| spec | A pointer filled with the audio format of the audio data |
| 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 |
| DECLSPEC void SDLCALL SDL_LockAudioDevice | ( | SDL_AudioDeviceID | dev | ) |
This takes two audio buffers of the playing audio format and mixes them, performing addition, volume adjustment, and overflow clipping. The volume ranges from 0 - 128, and should be set to SDL_MIX_MAXVOLUME for full audio volume. Note this does not change hardware volume. This is provided for convenience – you can mix your own audio data.
| DECLSPEC void SDLCALL SDL_MixAudioFormat | ( | Uint8 * | dst, |
| const Uint8 * | src, | ||
| SDL_AudioFormat | format, | ||
| Uint32 | len, | ||
| int | volume | ||
| ) |
This works like SDL_MixAudio(), but you specify the audio format instead of using the format of audio device 1. Thus it can be used when no audio device is open at all.
| DECLSPEC SDL_AudioStream* SDLCALL 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 |
| DECLSPEC int SDLCALL SDL_OpenAudio | ( | SDL_AudioSpec * | desired, |
| SDL_AudioSpec * | obtained | ||
| ) |
This function opens the audio device with the desired parameters, and returns 0 if successful, placing the actual hardware parameters in the structure pointed to by 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 hardware audio format if necessary. This function returns -1 if it failed to open the audio device, or couldn't set up the audio thread.
When filling in the desired audio spec structure,
desired->freq should be the desired audio frequency in samples-per- second.desired->format should be the desired audio format.desired->samples is the desired size of the audio buffer, in samples. 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 yield faster response time, but can lead to underflow if the application is doing heavy processing and cannot fill the audio buffer in time. A stereo sample consists of both right and left channels in LR ordering. Note that the number of samples is directly related to time by the following formula:desired->size is the size in bytes of the audio buffer, and is calculated by SDL_OpenAudio().desired->silence is the value used to set the buffer to silence, and is calculated by SDL_OpenAudio().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_LockAudio() and SDL_UnlockAudio() 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.The audio device starts out playing silence when it's opened, and should be enabled for playing by calling SDL_PauseAudio(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.
| DECLSPEC SDL_AudioDeviceID SDLCALL SDL_OpenAudioDevice | ( | const char * | device, |
| int | iscapture, | ||
| const SDL_AudioSpec * | desired, | ||
| SDL_AudioSpec * | obtained, | ||
| int | allowed_changes | ||
| ) |
Open a specific audio device. Passing in a device name of NULL requests the most reasonable default (and is equivalent to calling SDL_OpenAudio()).
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.
SDL_OpenAudio(), unlike this function, always acts on device ID 1.
| DECLSPEC void SDLCALL SDL_PauseAudioDevice | ( | SDL_AudioDeviceID | dev, |
| int | pause_on | ||
| ) |
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. This 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.
| 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. |
| DECLSPEC void SDLCALL SDL_UnlockAudioDevice | ( | SDL_AudioDeviceID | dev | ) |
| typedef void | ( | SDLCALL * | SDL_AudioFilter | ) |
1.8.15