static int hardware_mixer = 0;
static int has_softvol = 0;
+static snd_pcm_sframes_t (*alsa_pcm_write)(snd_pcm_t *, const void *, snd_pcm_uframes_t) = snd_pcm_writei;
+
static int play_number;
static int64_t accumulated_delay, accumulated_da_delay;
int alsa_characteristics_already_listed = 0;
unsigned int my_sample_rate = desired_sample_rate;
// snd_pcm_uframes_t frames = 441 * 10;
snd_pcm_uframes_t buffer_size, actual_buffer_length;
+ snd_pcm_access_t access;
ret = snd_pcm_open(&alsa_handle, alsa_out_dev, SND_PCM_STREAM_PLAYBACK, 0);
if (ret < 0)
alsa_out_dev);
}
- ret = snd_pcm_hw_params_set_access(alsa_handle, alsa_params,
- SND_PCM_ACCESS_RW_INTERLEAVED);
+ if (snd_pcm_hw_params_set_access(alsa_handle, alsa_params, SND_PCM_ACCESS_MMAP_INTERLEAVED) >= 0) {
+ access = SND_PCM_ACCESS_MMAP_INTERLEAVED;
+ alsa_pcm_write = snd_pcm_mmap_writei;
+ } else {
+ access = SND_PCM_ACCESS_RW_INTERLEAVED;
+ alsa_pcm_write = snd_pcm_writei;
+ }
+
+ ret = snd_pcm_hw_params_set_access(alsa_handle, alsa_params, access);
if (ret < 0) {
die("audio_alsa: Access type not available for device \"%s\": %s",
alsa_out_dev, snd_strerror(ret));
if (samples==0)
debug(1,"empty buffer being passed to pcm_writei -- skipping it");
if ((samples!=0) && (buf!=NULL)) {
- err = snd_pcm_writei(alsa_handle, (char *)buf, samples);
+ err = alsa_pcm_write(alsa_handle, (char *)buf, samples);
if (err < 0) {
debug(1, "Error %d writing %d samples in play(): \"%s\".", err, samples,
snd_strerror(err));