Open a specific audio device.
SDL_AudioDeviceID SDL_OpenAudioDevice(
const char *device,
int iscapture,
const SDL_AudioSpec *desired,
SDL_AudioSpec *obtained,
int allowed_changes);| 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 |
| obtained | an SDL_AudioSpec structure filled in with the actual output format |
| allowed_changes | 0, or one or more flags OR'd together |
Returns a valid device ID that is > 0 on success or 0 on failure; call SDL_GetError() for more information.
For compatibility with SDL 1.2, this will never return 1, since SDL reserves that ID for the legacy SDL_OpenAudio() function.
Passing in a device name of NULL requests the most reasonable default. 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_PlayAudioDevice(devid) 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)/freqdesired->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_CHANGESDL_AUDIO_ALLOW_FORMAT_CHANGESDL_AUDIO_ALLOW_CHANNELS_CHANGESDL_AUDIO_ALLOW_SAMPLES_CHANGESDL_AUDIO_ALLOW_ANY_CHANGEThese 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.
This function is available since SDL 3.0.0.
SDL_AudioSpec want, have;
SDL_AudioDeviceID dev;
SDL_memset(&want, 0, sizeof(want)); /* or SDL_zero(want) */
want.freq = 48000;
want.format = AUDIO_F32;
want.channels = 2;
want.samples = 4096;
want.callback = MyAudioCallback; /* you wrote this function elsewhere -- see SDL_AudioSpec for details */
dev = SDL_OpenAudioDevice(NULL, 0, &want, &have, SDL_AUDIO_ALLOW_FORMAT_CHANGE);
if (dev == 0) {
SDL_Log("Failed to open audio: %s", SDL_GetError());
} else {
if (have.format != want.format) { /* we let this one thing change. */
SDL_Log("We didn't get Float32 audio format.");
}
SDL_PauseAudioDevice(dev, 0); /* start audio playing. */
SDL_Delay(5000); /* let the audio callback play some sound for 5 seconds. */
SDL_CloseAudioDevice(dev);
}