526 lines
12 KiB
C
526 lines
12 KiB
C
/* public domain */
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#include "qemu-common.h"
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#include "audio.h"
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#include <pulse/simple.h>
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#include <pulse/error.h>
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#define AUDIO_CAP "pulseaudio"
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#include "audio_int.h"
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#include "audio_pt_int.h"
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typedef struct {
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HWVoiceOut hw;
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int done;
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int live;
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int decr;
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int rpos;
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pa_simple *s;
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void *pcm_buf;
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struct audio_pt pt;
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} PAVoiceOut;
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typedef struct {
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HWVoiceIn hw;
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int done;
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int dead;
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int incr;
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int wpos;
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pa_simple *s;
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void *pcm_buf;
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struct audio_pt pt;
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} PAVoiceIn;
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static struct {
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int samples;
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char *server;
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char *sink;
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char *source;
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} conf = {
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.samples = 4096,
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};
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static void GCC_FMT_ATTR (2, 3) qpa_logerr (int err, const char *fmt, ...)
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{
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va_list ap;
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va_start (ap, fmt);
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AUD_vlog (AUDIO_CAP, fmt, ap);
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va_end (ap);
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AUD_log (AUDIO_CAP, "Reason: %s\n", pa_strerror (err));
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}
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static void *qpa_thread_out (void *arg)
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{
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PAVoiceOut *pa = arg;
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HWVoiceOut *hw = &pa->hw;
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if (audio_pt_lock (&pa->pt, AUDIO_FUNC)) {
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return NULL;
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}
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for (;;) {
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int decr, to_mix, rpos;
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for (;;) {
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if (pa->done) {
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goto exit;
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}
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if (pa->live > 0) {
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break;
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}
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if (audio_pt_wait (&pa->pt, AUDIO_FUNC)) {
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goto exit;
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}
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}
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decr = to_mix = audio_MIN (pa->live, conf.samples >> 2);
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rpos = pa->rpos;
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if (audio_pt_unlock (&pa->pt, AUDIO_FUNC)) {
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return NULL;
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}
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while (to_mix) {
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int error;
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int chunk = audio_MIN (to_mix, hw->samples - rpos);
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struct st_sample *src = hw->mix_buf + rpos;
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hw->clip (pa->pcm_buf, src, chunk);
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if (pa_simple_write (pa->s, pa->pcm_buf,
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chunk << hw->info.shift, &error) < 0) {
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qpa_logerr (error, "pa_simple_write failed\n");
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return NULL;
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}
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rpos = (rpos + chunk) % hw->samples;
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to_mix -= chunk;
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}
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if (audio_pt_lock (&pa->pt, AUDIO_FUNC)) {
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return NULL;
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}
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pa->rpos = rpos;
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pa->live -= decr;
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pa->decr += decr;
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}
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exit:
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audio_pt_unlock (&pa->pt, AUDIO_FUNC);
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return NULL;
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}
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static int qpa_run_out (HWVoiceOut *hw, int live)
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{
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int decr;
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PAVoiceOut *pa = (PAVoiceOut *) hw;
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if (audio_pt_lock (&pa->pt, AUDIO_FUNC)) {
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return 0;
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}
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decr = audio_MIN (live, pa->decr);
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pa->decr -= decr;
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pa->live = live - decr;
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hw->rpos = pa->rpos;
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if (pa->live > 0) {
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audio_pt_unlock_and_signal (&pa->pt, AUDIO_FUNC);
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}
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else {
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audio_pt_unlock (&pa->pt, AUDIO_FUNC);
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}
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return decr;
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}
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static int qpa_write (SWVoiceOut *sw, void *buf, int len)
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{
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return audio_pcm_sw_write (sw, buf, len);
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}
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/* capture */
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static void *qpa_thread_in (void *arg)
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{
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PAVoiceIn *pa = arg;
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HWVoiceIn *hw = &pa->hw;
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if (audio_pt_lock (&pa->pt, AUDIO_FUNC)) {
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return NULL;
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}
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for (;;) {
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int incr, to_grab, wpos;
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for (;;) {
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if (pa->done) {
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goto exit;
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}
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if (pa->dead > 0) {
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break;
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}
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if (audio_pt_wait (&pa->pt, AUDIO_FUNC)) {
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goto exit;
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}
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}
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incr = to_grab = audio_MIN (pa->dead, conf.samples >> 2);
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wpos = pa->wpos;
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if (audio_pt_unlock (&pa->pt, AUDIO_FUNC)) {
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return NULL;
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}
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while (to_grab) {
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int error;
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int chunk = audio_MIN (to_grab, hw->samples - wpos);
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void *buf = advance (pa->pcm_buf, wpos);
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if (pa_simple_read (pa->s, buf,
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chunk << hw->info.shift, &error) < 0) {
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qpa_logerr (error, "pa_simple_read failed\n");
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return NULL;
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}
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hw->conv (hw->conv_buf + wpos, buf, chunk);
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wpos = (wpos + chunk) % hw->samples;
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to_grab -= chunk;
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}
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if (audio_pt_lock (&pa->pt, AUDIO_FUNC)) {
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return NULL;
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}
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pa->wpos = wpos;
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pa->dead -= incr;
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pa->incr += incr;
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}
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exit:
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audio_pt_unlock (&pa->pt, AUDIO_FUNC);
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return NULL;
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}
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static int qpa_run_in (HWVoiceIn *hw)
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{
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int live, incr, dead;
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PAVoiceIn *pa = (PAVoiceIn *) hw;
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if (audio_pt_lock (&pa->pt, AUDIO_FUNC)) {
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return 0;
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}
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live = audio_pcm_hw_get_live_in (hw);
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dead = hw->samples - live;
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incr = audio_MIN (dead, pa->incr);
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pa->incr -= incr;
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pa->dead = dead - incr;
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hw->wpos = pa->wpos;
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if (pa->dead > 0) {
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audio_pt_unlock_and_signal (&pa->pt, AUDIO_FUNC);
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}
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else {
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audio_pt_unlock (&pa->pt, AUDIO_FUNC);
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}
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return incr;
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}
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static int qpa_read (SWVoiceIn *sw, void *buf, int len)
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{
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return audio_pcm_sw_read (sw, buf, len);
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}
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static pa_sample_format_t audfmt_to_pa (audfmt_e afmt, int endianness)
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{
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int format;
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switch (afmt) {
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case AUD_FMT_S8:
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case AUD_FMT_U8:
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format = PA_SAMPLE_U8;
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break;
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case AUD_FMT_S16:
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case AUD_FMT_U16:
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format = endianness ? PA_SAMPLE_S16BE : PA_SAMPLE_S16LE;
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break;
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case AUD_FMT_S32:
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case AUD_FMT_U32:
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format = endianness ? PA_SAMPLE_S32BE : PA_SAMPLE_S32LE;
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break;
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default:
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dolog ("Internal logic error: Bad audio format %d\n", afmt);
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format = PA_SAMPLE_U8;
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break;
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}
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return format;
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}
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static audfmt_e pa_to_audfmt (pa_sample_format_t fmt, int *endianness)
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{
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switch (fmt) {
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case PA_SAMPLE_U8:
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return AUD_FMT_U8;
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case PA_SAMPLE_S16BE:
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*endianness = 1;
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return AUD_FMT_S16;
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case PA_SAMPLE_S16LE:
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*endianness = 0;
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return AUD_FMT_S16;
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case PA_SAMPLE_S32BE:
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*endianness = 1;
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return AUD_FMT_S32;
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case PA_SAMPLE_S32LE:
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*endianness = 0;
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return AUD_FMT_S32;
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default:
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dolog ("Internal logic error: Bad pa_sample_format %d\n", fmt);
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return AUD_FMT_U8;
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}
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}
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static int qpa_init_out (HWVoiceOut *hw, struct audsettings *as)
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{
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int error;
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static pa_sample_spec ss;
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static pa_buffer_attr ba;
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struct audsettings obt_as = *as;
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PAVoiceOut *pa = (PAVoiceOut *) hw;
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ss.format = audfmt_to_pa (as->fmt, as->endianness);
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ss.channels = as->nchannels;
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ss.rate = as->freq;
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/*
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* qemu audio tick runs at 250 Hz (by default), so processing
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* data chunks worth 4 ms of sound should be a good fit.
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*/
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ba.tlength = pa_usec_to_bytes (4 * 1000, &ss);
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ba.minreq = pa_usec_to_bytes (2 * 1000, &ss);
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ba.maxlength = -1;
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ba.prebuf = -1;
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obt_as.fmt = pa_to_audfmt (ss.format, &obt_as.endianness);
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pa->s = pa_simple_new (
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conf.server,
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"qemu",
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PA_STREAM_PLAYBACK,
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conf.sink,
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"pcm.playback",
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&ss,
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NULL, /* channel map */
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&ba, /* buffering attributes */
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&error
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);
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if (!pa->s) {
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qpa_logerr (error, "pa_simple_new for playback failed\n");
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goto fail1;
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}
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audio_pcm_init_info (&hw->info, &obt_as);
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hw->samples = conf.samples;
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pa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
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pa->rpos = hw->rpos;
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if (!pa->pcm_buf) {
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dolog ("Could not allocate buffer (%d bytes)\n",
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hw->samples << hw->info.shift);
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goto fail2;
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}
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if (audio_pt_init (&pa->pt, qpa_thread_out, hw, AUDIO_CAP, AUDIO_FUNC)) {
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goto fail3;
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}
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return 0;
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fail3:
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g_free (pa->pcm_buf);
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pa->pcm_buf = NULL;
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fail2:
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pa_simple_free (pa->s);
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pa->s = NULL;
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fail1:
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return -1;
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}
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static int qpa_init_in (HWVoiceIn *hw, struct audsettings *as)
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{
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int error;
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static pa_sample_spec ss;
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struct audsettings obt_as = *as;
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PAVoiceIn *pa = (PAVoiceIn *) hw;
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ss.format = audfmt_to_pa (as->fmt, as->endianness);
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ss.channels = as->nchannels;
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ss.rate = as->freq;
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obt_as.fmt = pa_to_audfmt (ss.format, &obt_as.endianness);
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pa->s = pa_simple_new (
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conf.server,
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"qemu",
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PA_STREAM_RECORD,
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conf.source,
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"pcm.capture",
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&ss,
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NULL, /* channel map */
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NULL, /* buffering attributes */
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&error
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);
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if (!pa->s) {
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qpa_logerr (error, "pa_simple_new for capture failed\n");
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goto fail1;
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}
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audio_pcm_init_info (&hw->info, &obt_as);
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hw->samples = conf.samples;
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pa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
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pa->wpos = hw->wpos;
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if (!pa->pcm_buf) {
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dolog ("Could not allocate buffer (%d bytes)\n",
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hw->samples << hw->info.shift);
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goto fail2;
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}
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if (audio_pt_init (&pa->pt, qpa_thread_in, hw, AUDIO_CAP, AUDIO_FUNC)) {
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goto fail3;
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}
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return 0;
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fail3:
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g_free (pa->pcm_buf);
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pa->pcm_buf = NULL;
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fail2:
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pa_simple_free (pa->s);
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pa->s = NULL;
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fail1:
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return -1;
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}
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static void qpa_fini_out (HWVoiceOut *hw)
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{
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void *ret;
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PAVoiceOut *pa = (PAVoiceOut *) hw;
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audio_pt_lock (&pa->pt, AUDIO_FUNC);
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pa->done = 1;
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audio_pt_unlock_and_signal (&pa->pt, AUDIO_FUNC);
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audio_pt_join (&pa->pt, &ret, AUDIO_FUNC);
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if (pa->s) {
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pa_simple_free (pa->s);
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pa->s = NULL;
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}
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audio_pt_fini (&pa->pt, AUDIO_FUNC);
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g_free (pa->pcm_buf);
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pa->pcm_buf = NULL;
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}
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static void qpa_fini_in (HWVoiceIn *hw)
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{
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void *ret;
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PAVoiceIn *pa = (PAVoiceIn *) hw;
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audio_pt_lock (&pa->pt, AUDIO_FUNC);
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pa->done = 1;
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audio_pt_unlock_and_signal (&pa->pt, AUDIO_FUNC);
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audio_pt_join (&pa->pt, &ret, AUDIO_FUNC);
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if (pa->s) {
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pa_simple_free (pa->s);
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pa->s = NULL;
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}
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audio_pt_fini (&pa->pt, AUDIO_FUNC);
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g_free (pa->pcm_buf);
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pa->pcm_buf = NULL;
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}
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static int qpa_ctl_out (HWVoiceOut *hw, int cmd, ...)
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{
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(void) hw;
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(void) cmd;
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return 0;
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}
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static int qpa_ctl_in (HWVoiceIn *hw, int cmd, ...)
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{
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(void) hw;
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(void) cmd;
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return 0;
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}
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/* common */
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static void *qpa_audio_init (void)
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{
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return &conf;
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}
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static void qpa_audio_fini (void *opaque)
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{
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(void) opaque;
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}
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struct audio_option qpa_options[] = {
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{
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.name = "SAMPLES",
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.tag = AUD_OPT_INT,
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.valp = &conf.samples,
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.descr = "buffer size in samples"
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},
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{
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.name = "SERVER",
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.tag = AUD_OPT_STR,
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.valp = &conf.server,
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.descr = "server address"
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},
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{
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.name = "SINK",
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.tag = AUD_OPT_STR,
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.valp = &conf.sink,
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.descr = "sink device name"
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},
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{
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.name = "SOURCE",
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.tag = AUD_OPT_STR,
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.valp = &conf.source,
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.descr = "source device name"
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},
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{ /* End of list */ }
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};
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static struct audio_pcm_ops qpa_pcm_ops = {
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.init_out = qpa_init_out,
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.fini_out = qpa_fini_out,
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.run_out = qpa_run_out,
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.write = qpa_write,
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.ctl_out = qpa_ctl_out,
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.init_in = qpa_init_in,
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.fini_in = qpa_fini_in,
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.run_in = qpa_run_in,
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.read = qpa_read,
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.ctl_in = qpa_ctl_in
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};
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struct audio_driver pa_audio_driver = {
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.name = "pa",
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.descr = "http://www.pulseaudio.org/",
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.options = qpa_options,
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.init = qpa_audio_init,
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.fini = qpa_audio_fini,
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.pcm_ops = &qpa_pcm_ops,
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.can_be_default = 1,
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.max_voices_out = INT_MAX,
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.max_voices_in = INT_MAX,
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.voice_size_out = sizeof (PAVoiceOut),
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.voice_size_in = sizeof (PAVoiceIn)
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};
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