static ALenum
openal_audio_format(struct audio_format *audio_format)
{
switch (audio_format->format) {
case SAMPLE_FORMAT_S16:
if (audio_format->channels == 2)
return AL_FORMAT_STEREO16;
if (audio_format->channels == 1)
return AL_FORMAT_MONO16;
/* fall back to mono */
audio_format->channels = 1;
return openal_audio_format(audio_format);
case SAMPLE_FORMAT_S8:
if (audio_format->channels == 2)
return AL_FORMAT_STEREO8;
if (audio_format->channels == 1)
return AL_FORMAT_MONO8;
/* fall back to mono */
audio_format->channels = 1;
return openal_audio_format(audio_format);
default:
/* fall back to 16 bit */
audio_format->format = SAMPLE_FORMAT_S16;
return openal_audio_format(audio_format);
}
}
static ALenum
openal_audio_format(struct audio_format *audio_format)
{
/* note: cannot map SAMPLE_FORMAT_S8 to AL_FORMAT_STEREO8 or
AL_FORMAT_MONO8 since OpenAL expects unsigned 8 bit
samples, while MPD uses signed samples */
switch (audio_format->format) {
case SAMPLE_FORMAT_S16:
if (audio_format->channels == 2)
return AL_FORMAT_STEREO16;
if (audio_format->channels == 1)
return AL_FORMAT_MONO16;
/* fall back to mono */
audio_format->channels = 1;
return openal_audio_format(audio_format);
default:
/* fall back to 16 bit */
audio_format->format = SAMPLE_FORMAT_S16;
return openal_audio_format(audio_format);
}
}
static bool
openal_open(void *data, struct audio_format *audio_format,
GError **error)
{
struct openal_data *od = data;
od->format = openal_audio_format(audio_format);
if (!od->format) {
struct audio_format_string s;
g_set_error(error, openal_output_quark(), 0,
"Unsupported audio format: %s",
audio_format_to_string(audio_format, &s));
return false;
}
if (!openal_setup_context(od, error)) {
return false;
}
alcMakeContextCurrent(od->context);
alGenBuffers(NUM_BUFFERS, od->buffers);
if (alGetError() != AL_NO_ERROR) {
g_set_error(error, openal_output_quark(), 0,
"Failed to generate buffers");
return false;
}
alGenSources(1, &od->source);
if (alGetError() != AL_NO_ERROR) {
g_set_error(error, openal_output_quark(), 0,
"Failed to generate source");
alDeleteBuffers(NUM_BUFFERS, od->buffers);
return false;
}
od->filled = 0;
od->timer = timer_new(audio_format);
od->frequency = audio_format->sample_rate;
return true;
}
