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6cdc2d189c
The rate of pulseaudio absorbing the audio stream is used to control the the rate of the guests audio stream. When the emulated hardware uses small chunks (like intel-hda does) we need small chunks on the audio backend side too, otherwise that feedback loop doesn't work very well. Cc: Max Ehrlich <maxehr@umiacs.umd.edu> Cc: Martin Schrodt <martin@schrodt.org> Buglink: https://bugs.launchpad.net/bugs/1795527 Signed-off-by: Gerd Hoffmann <kraxel@redhat.com> Reviewed-by: Philippe Mathieu-Daudé <philmd@redhat.com> Message-id: 20181109142032.1628-1-kraxel@redhat.com
959 lines
24 KiB
C
959 lines
24 KiB
C
/* public domain */
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#include "qemu/osdep.h"
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#include "qemu-common.h"
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#include "audio.h"
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#include <pulse/pulseaudio.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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int samples;
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char *server;
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char *sink;
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char *source;
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} PAConf;
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typedef struct {
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PAConf conf;
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pa_threaded_mainloop *mainloop;
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pa_context *context;
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} paaudio;
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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_stream *stream;
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void *pcm_buf;
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struct audio_pt pt;
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paaudio *g;
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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_stream *stream;
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void *pcm_buf;
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struct audio_pt pt;
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const void *read_data;
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size_t read_index, read_length;
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paaudio *g;
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} PAVoiceIn;
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static void qpa_audio_fini(void *opaque);
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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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#ifndef PA_CONTEXT_IS_GOOD
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static inline int PA_CONTEXT_IS_GOOD(pa_context_state_t x)
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{
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return
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x == PA_CONTEXT_CONNECTING ||
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x == PA_CONTEXT_AUTHORIZING ||
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x == PA_CONTEXT_SETTING_NAME ||
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x == PA_CONTEXT_READY;
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}
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#endif
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#ifndef PA_STREAM_IS_GOOD
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static inline int PA_STREAM_IS_GOOD(pa_stream_state_t x)
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{
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return
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x == PA_STREAM_CREATING ||
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x == PA_STREAM_READY;
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}
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#endif
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#define CHECK_SUCCESS_GOTO(c, rerror, expression, label) \
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do { \
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if (!(expression)) { \
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if (rerror) { \
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*(rerror) = pa_context_errno ((c)->context); \
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} \
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goto label; \
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} \
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} while (0)
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#define CHECK_DEAD_GOTO(c, stream, rerror, label) \
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do { \
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if (!(c)->context || !PA_CONTEXT_IS_GOOD (pa_context_get_state((c)->context)) || \
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!(stream) || !PA_STREAM_IS_GOOD (pa_stream_get_state ((stream)))) { \
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if (((c)->context && pa_context_get_state ((c)->context) == PA_CONTEXT_FAILED) || \
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((stream) && pa_stream_get_state ((stream)) == PA_STREAM_FAILED)) { \
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if (rerror) { \
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*(rerror) = pa_context_errno ((c)->context); \
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} \
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} else { \
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if (rerror) { \
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*(rerror) = PA_ERR_BADSTATE; \
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} \
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} \
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goto label; \
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} \
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} while (0)
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static int qpa_simple_read (PAVoiceIn *p, void *data, size_t length, int *rerror)
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{
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paaudio *g = p->g;
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pa_threaded_mainloop_lock (g->mainloop);
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CHECK_DEAD_GOTO (g, p->stream, rerror, unlock_and_fail);
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while (length > 0) {
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size_t l;
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while (!p->read_data) {
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int r;
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r = pa_stream_peek (p->stream, &p->read_data, &p->read_length);
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CHECK_SUCCESS_GOTO (g, rerror, r == 0, unlock_and_fail);
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if (!p->read_data) {
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pa_threaded_mainloop_wait (g->mainloop);
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CHECK_DEAD_GOTO (g, p->stream, rerror, unlock_and_fail);
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} else {
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p->read_index = 0;
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}
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}
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l = p->read_length < length ? p->read_length : length;
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memcpy (data, (const uint8_t *) p->read_data+p->read_index, l);
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data = (uint8_t *) data + l;
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length -= l;
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p->read_index += l;
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p->read_length -= l;
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if (!p->read_length) {
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int r;
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r = pa_stream_drop (p->stream);
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p->read_data = NULL;
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p->read_length = 0;
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p->read_index = 0;
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CHECK_SUCCESS_GOTO (g, rerror, r == 0, unlock_and_fail);
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}
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}
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pa_threaded_mainloop_unlock (g->mainloop);
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return 0;
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unlock_and_fail:
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pa_threaded_mainloop_unlock (g->mainloop);
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return -1;
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}
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static int qpa_simple_write (PAVoiceOut *p, const void *data, size_t length, int *rerror)
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{
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paaudio *g = p->g;
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pa_threaded_mainloop_lock (g->mainloop);
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CHECK_DEAD_GOTO (g, p->stream, rerror, unlock_and_fail);
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while (length > 0) {
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size_t l;
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int r;
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while (!(l = pa_stream_writable_size (p->stream))) {
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pa_threaded_mainloop_wait (g->mainloop);
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CHECK_DEAD_GOTO (g, p->stream, rerror, unlock_and_fail);
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}
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CHECK_SUCCESS_GOTO (g, rerror, l != (size_t) -1, unlock_and_fail);
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if (l > length) {
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l = length;
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}
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r = pa_stream_write (p->stream, data, l, NULL, 0LL, PA_SEEK_RELATIVE);
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CHECK_SUCCESS_GOTO (g, rerror, r >= 0, unlock_and_fail);
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data = (const uint8_t *) data + l;
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length -= l;
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}
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pa_threaded_mainloop_unlock (g->mainloop);
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return 0;
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unlock_and_fail:
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pa_threaded_mainloop_unlock (g->mainloop);
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return -1;
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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, __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, __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, pa->g->conf.samples >> 5);
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rpos = pa->rpos;
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if (audio_pt_unlock(&pa->pt, __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 (qpa_simple_write (pa, 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, __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, __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, __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, __func__);
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}
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else {
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audio_pt_unlock(&pa->pt, __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, __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, __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, pa->g->conf.samples >> 5);
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wpos = pa->wpos;
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if (audio_pt_unlock(&pa->pt, __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 (qpa_simple_read (pa, 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, __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, __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, __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, __func__);
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}
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else {
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audio_pt_unlock(&pa->pt, __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 void context_state_cb (pa_context *c, void *userdata)
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{
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paaudio *g = userdata;
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switch (pa_context_get_state(c)) {
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case PA_CONTEXT_READY:
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case PA_CONTEXT_TERMINATED:
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case PA_CONTEXT_FAILED:
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pa_threaded_mainloop_signal (g->mainloop, 0);
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break;
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case PA_CONTEXT_UNCONNECTED:
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case PA_CONTEXT_CONNECTING:
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case PA_CONTEXT_AUTHORIZING:
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case PA_CONTEXT_SETTING_NAME:
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break;
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}
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}
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static void stream_state_cb (pa_stream *s, void * userdata)
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{
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paaudio *g = userdata;
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switch (pa_stream_get_state (s)) {
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case PA_STREAM_READY:
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case PA_STREAM_FAILED:
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case PA_STREAM_TERMINATED:
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pa_threaded_mainloop_signal (g->mainloop, 0);
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break;
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case PA_STREAM_UNCONNECTED:
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case PA_STREAM_CREATING:
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break;
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}
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}
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static void stream_request_cb (pa_stream *s, size_t length, void *userdata)
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{
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paaudio *g = userdata;
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pa_threaded_mainloop_signal (g->mainloop, 0);
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}
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static pa_stream *qpa_simple_new (
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paaudio *g,
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const char *name,
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pa_stream_direction_t dir,
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const char *dev,
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const pa_sample_spec *ss,
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const pa_channel_map *map,
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const pa_buffer_attr *attr,
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int *rerror)
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{
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int r;
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pa_stream *stream;
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pa_threaded_mainloop_lock (g->mainloop);
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stream = pa_stream_new (g->context, name, ss, map);
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if (!stream) {
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goto fail;
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}
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pa_stream_set_state_callback (stream, stream_state_cb, g);
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pa_stream_set_read_callback (stream, stream_request_cb, g);
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pa_stream_set_write_callback (stream, stream_request_cb, g);
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if (dir == PA_STREAM_PLAYBACK) {
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r = pa_stream_connect_playback (stream, dev, attr,
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PA_STREAM_INTERPOLATE_TIMING
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#ifdef PA_STREAM_ADJUST_LATENCY
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|PA_STREAM_ADJUST_LATENCY
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#endif
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|PA_STREAM_AUTO_TIMING_UPDATE, NULL, NULL);
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} else {
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r = pa_stream_connect_record (stream, dev, attr,
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PA_STREAM_INTERPOLATE_TIMING
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#ifdef PA_STREAM_ADJUST_LATENCY
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|PA_STREAM_ADJUST_LATENCY
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#endif
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|PA_STREAM_AUTO_TIMING_UPDATE);
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}
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if (r < 0) {
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goto fail;
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}
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pa_threaded_mainloop_unlock (g->mainloop);
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return stream;
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fail:
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pa_threaded_mainloop_unlock (g->mainloop);
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if (stream) {
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pa_stream_unref (stream);
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}
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*rerror = pa_context_errno (g->context);
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return NULL;
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}
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static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
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void *drv_opaque)
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{
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int error;
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pa_sample_spec ss;
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pa_buffer_attr ba;
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struct audsettings obt_as = *as;
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PAVoiceOut *pa = (PAVoiceOut *) hw;
|
|
paaudio *g = pa->g = drv_opaque;
|
|
|
|
ss.format = audfmt_to_pa (as->fmt, as->endianness);
|
|
ss.channels = as->nchannels;
|
|
ss.rate = as->freq;
|
|
|
|
/*
|
|
* qemu audio tick runs at 100 Hz (by default), so processing
|
|
* data chunks worth 10 ms of sound should be a good fit.
|
|
*/
|
|
ba.tlength = pa_usec_to_bytes (10 * 1000, &ss);
|
|
ba.minreq = pa_usec_to_bytes (5 * 1000, &ss);
|
|
ba.maxlength = -1;
|
|
ba.prebuf = -1;
|
|
|
|
obt_as.fmt = pa_to_audfmt (ss.format, &obt_as.endianness);
|
|
|
|
pa->stream = qpa_simple_new (
|
|
g,
|
|
"qemu",
|
|
PA_STREAM_PLAYBACK,
|
|
g->conf.sink,
|
|
&ss,
|
|
NULL, /* channel map */
|
|
&ba, /* buffering attributes */
|
|
&error
|
|
);
|
|
if (!pa->stream) {
|
|
qpa_logerr (error, "pa_simple_new for playback failed\n");
|
|
goto fail1;
|
|
}
|
|
|
|
audio_pcm_init_info (&hw->info, &obt_as);
|
|
hw->samples = g->conf.samples;
|
|
pa->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
|
|
pa->rpos = hw->rpos;
|
|
if (!pa->pcm_buf) {
|
|
dolog ("Could not allocate buffer (%d bytes)\n",
|
|
hw->samples << hw->info.shift);
|
|
goto fail2;
|
|
}
|
|
|
|
if (audio_pt_init(&pa->pt, qpa_thread_out, hw, AUDIO_CAP, __func__)) {
|
|
goto fail3;
|
|
}
|
|
|
|
return 0;
|
|
|
|
fail3:
|
|
g_free (pa->pcm_buf);
|
|
pa->pcm_buf = NULL;
|
|
fail2:
|
|
if (pa->stream) {
|
|
pa_stream_unref (pa->stream);
|
|
pa->stream = NULL;
|
|
}
|
|
fail1:
|
|
return -1;
|
|
}
|
|
|
|
static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
|
|
{
|
|
int error;
|
|
pa_sample_spec ss;
|
|
struct audsettings obt_as = *as;
|
|
PAVoiceIn *pa = (PAVoiceIn *) hw;
|
|
paaudio *g = pa->g = drv_opaque;
|
|
|
|
ss.format = audfmt_to_pa (as->fmt, as->endianness);
|
|
ss.channels = as->nchannels;
|
|
ss.rate = as->freq;
|
|
|
|
obt_as.fmt = pa_to_audfmt (ss.format, &obt_as.endianness);
|
|
|
|
pa->stream = qpa_simple_new (
|
|
g,
|
|
"qemu",
|
|
PA_STREAM_RECORD,
|
|
g->conf.source,
|
|
&ss,
|
|
NULL, /* channel map */
|
|
NULL, /* buffering attributes */
|
|
&error
|
|
);
|
|
if (!pa->stream) {
|
|
qpa_logerr (error, "pa_simple_new for capture failed\n");
|
|
goto fail1;
|
|
}
|
|
|
|
audio_pcm_init_info (&hw->info, &obt_as);
|
|
hw->samples = g->conf.samples;
|
|
pa->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
|
|
pa->wpos = hw->wpos;
|
|
if (!pa->pcm_buf) {
|
|
dolog ("Could not allocate buffer (%d bytes)\n",
|
|
hw->samples << hw->info.shift);
|
|
goto fail2;
|
|
}
|
|
|
|
if (audio_pt_init(&pa->pt, qpa_thread_in, hw, AUDIO_CAP, __func__)) {
|
|
goto fail3;
|
|
}
|
|
|
|
return 0;
|
|
|
|
fail3:
|
|
g_free (pa->pcm_buf);
|
|
pa->pcm_buf = NULL;
|
|
fail2:
|
|
if (pa->stream) {
|
|
pa_stream_unref (pa->stream);
|
|
pa->stream = NULL;
|
|
}
|
|
fail1:
|
|
return -1;
|
|
}
|
|
|
|
static void qpa_fini_out (HWVoiceOut *hw)
|
|
{
|
|
void *ret;
|
|
PAVoiceOut *pa = (PAVoiceOut *) hw;
|
|
|
|
audio_pt_lock(&pa->pt, __func__);
|
|
pa->done = 1;
|
|
audio_pt_unlock_and_signal(&pa->pt, __func__);
|
|
audio_pt_join(&pa->pt, &ret, __func__);
|
|
|
|
if (pa->stream) {
|
|
pa_stream_unref (pa->stream);
|
|
pa->stream = NULL;
|
|
}
|
|
|
|
audio_pt_fini(&pa->pt, __func__);
|
|
g_free (pa->pcm_buf);
|
|
pa->pcm_buf = NULL;
|
|
}
|
|
|
|
static void qpa_fini_in (HWVoiceIn *hw)
|
|
{
|
|
void *ret;
|
|
PAVoiceIn *pa = (PAVoiceIn *) hw;
|
|
|
|
audio_pt_lock(&pa->pt, __func__);
|
|
pa->done = 1;
|
|
audio_pt_unlock_and_signal(&pa->pt, __func__);
|
|
audio_pt_join(&pa->pt, &ret, __func__);
|
|
|
|
if (pa->stream) {
|
|
pa_stream_unref (pa->stream);
|
|
pa->stream = NULL;
|
|
}
|
|
|
|
audio_pt_fini(&pa->pt, __func__);
|
|
g_free (pa->pcm_buf);
|
|
pa->pcm_buf = NULL;
|
|
}
|
|
|
|
static int qpa_ctl_out (HWVoiceOut *hw, int cmd, ...)
|
|
{
|
|
PAVoiceOut *pa = (PAVoiceOut *) hw;
|
|
pa_operation *op;
|
|
pa_cvolume v;
|
|
paaudio *g = pa->g;
|
|
|
|
#ifdef PA_CHECK_VERSION /* macro is present in 0.9.16+ */
|
|
pa_cvolume_init (&v); /* function is present in 0.9.13+ */
|
|
#endif
|
|
|
|
switch (cmd) {
|
|
case VOICE_VOLUME:
|
|
{
|
|
SWVoiceOut *sw;
|
|
va_list ap;
|
|
|
|
va_start (ap, cmd);
|
|
sw = va_arg (ap, SWVoiceOut *);
|
|
va_end (ap);
|
|
|
|
v.channels = 2;
|
|
v.values[0] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * sw->vol.l) / UINT32_MAX;
|
|
v.values[1] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * sw->vol.r) / UINT32_MAX;
|
|
|
|
pa_threaded_mainloop_lock (g->mainloop);
|
|
|
|
op = pa_context_set_sink_input_volume (g->context,
|
|
pa_stream_get_index (pa->stream),
|
|
&v, NULL, NULL);
|
|
if (!op)
|
|
qpa_logerr (pa_context_errno (g->context),
|
|
"set_sink_input_volume() failed\n");
|
|
else
|
|
pa_operation_unref (op);
|
|
|
|
op = pa_context_set_sink_input_mute (g->context,
|
|
pa_stream_get_index (pa->stream),
|
|
sw->vol.mute, NULL, NULL);
|
|
if (!op) {
|
|
qpa_logerr (pa_context_errno (g->context),
|
|
"set_sink_input_mute() failed\n");
|
|
} else {
|
|
pa_operation_unref (op);
|
|
}
|
|
|
|
pa_threaded_mainloop_unlock (g->mainloop);
|
|
}
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
static int qpa_ctl_in (HWVoiceIn *hw, int cmd, ...)
|
|
{
|
|
PAVoiceIn *pa = (PAVoiceIn *) hw;
|
|
pa_operation *op;
|
|
pa_cvolume v;
|
|
paaudio *g = pa->g;
|
|
|
|
#ifdef PA_CHECK_VERSION
|
|
pa_cvolume_init (&v);
|
|
#endif
|
|
|
|
switch (cmd) {
|
|
case VOICE_VOLUME:
|
|
{
|
|
SWVoiceIn *sw;
|
|
va_list ap;
|
|
|
|
va_start (ap, cmd);
|
|
sw = va_arg (ap, SWVoiceIn *);
|
|
va_end (ap);
|
|
|
|
v.channels = 2;
|
|
v.values[0] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * sw->vol.l) / UINT32_MAX;
|
|
v.values[1] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * sw->vol.r) / UINT32_MAX;
|
|
|
|
pa_threaded_mainloop_lock (g->mainloop);
|
|
|
|
op = pa_context_set_source_output_volume (g->context,
|
|
pa_stream_get_index (pa->stream),
|
|
&v, NULL, NULL);
|
|
if (!op) {
|
|
qpa_logerr (pa_context_errno (g->context),
|
|
"set_source_output_volume() failed\n");
|
|
} else {
|
|
pa_operation_unref(op);
|
|
}
|
|
|
|
op = pa_context_set_source_output_mute (g->context,
|
|
pa_stream_get_index (pa->stream),
|
|
sw->vol.mute, NULL, NULL);
|
|
if (!op) {
|
|
qpa_logerr (pa_context_errno (g->context),
|
|
"set_source_output_mute() failed\n");
|
|
} else {
|
|
pa_operation_unref (op);
|
|
}
|
|
|
|
pa_threaded_mainloop_unlock (g->mainloop);
|
|
}
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
/* common */
|
|
static PAConf glob_conf = {
|
|
.samples = 4096,
|
|
};
|
|
|
|
static void *qpa_audio_init (void)
|
|
{
|
|
paaudio *g = g_malloc(sizeof(paaudio));
|
|
g->conf = glob_conf;
|
|
g->mainloop = NULL;
|
|
g->context = NULL;
|
|
|
|
g->mainloop = pa_threaded_mainloop_new ();
|
|
if (!g->mainloop) {
|
|
goto fail;
|
|
}
|
|
|
|
g->context = pa_context_new (pa_threaded_mainloop_get_api (g->mainloop),
|
|
g->conf.server);
|
|
if (!g->context) {
|
|
goto fail;
|
|
}
|
|
|
|
pa_context_set_state_callback (g->context, context_state_cb, g);
|
|
|
|
if (pa_context_connect (g->context, g->conf.server, 0, NULL) < 0) {
|
|
qpa_logerr (pa_context_errno (g->context),
|
|
"pa_context_connect() failed\n");
|
|
goto fail;
|
|
}
|
|
|
|
pa_threaded_mainloop_lock (g->mainloop);
|
|
|
|
if (pa_threaded_mainloop_start (g->mainloop) < 0) {
|
|
goto unlock_and_fail;
|
|
}
|
|
|
|
for (;;) {
|
|
pa_context_state_t state;
|
|
|
|
state = pa_context_get_state (g->context);
|
|
|
|
if (state == PA_CONTEXT_READY) {
|
|
break;
|
|
}
|
|
|
|
if (!PA_CONTEXT_IS_GOOD (state)) {
|
|
qpa_logerr (pa_context_errno (g->context),
|
|
"Wrong context state\n");
|
|
goto unlock_and_fail;
|
|
}
|
|
|
|
/* Wait until the context is ready */
|
|
pa_threaded_mainloop_wait (g->mainloop);
|
|
}
|
|
|
|
pa_threaded_mainloop_unlock (g->mainloop);
|
|
|
|
return g;
|
|
|
|
unlock_and_fail:
|
|
pa_threaded_mainloop_unlock (g->mainloop);
|
|
fail:
|
|
AUD_log (AUDIO_CAP, "Failed to initialize PA context");
|
|
qpa_audio_fini(g);
|
|
return NULL;
|
|
}
|
|
|
|
static void qpa_audio_fini (void *opaque)
|
|
{
|
|
paaudio *g = opaque;
|
|
|
|
if (g->mainloop) {
|
|
pa_threaded_mainloop_stop (g->mainloop);
|
|
}
|
|
|
|
if (g->context) {
|
|
pa_context_disconnect (g->context);
|
|
pa_context_unref (g->context);
|
|
}
|
|
|
|
if (g->mainloop) {
|
|
pa_threaded_mainloop_free (g->mainloop);
|
|
}
|
|
|
|
g_free(g);
|
|
}
|
|
|
|
struct audio_option qpa_options[] = {
|
|
{
|
|
.name = "SAMPLES",
|
|
.tag = AUD_OPT_INT,
|
|
.valp = &glob_conf.samples,
|
|
.descr = "buffer size in samples"
|
|
},
|
|
{
|
|
.name = "SERVER",
|
|
.tag = AUD_OPT_STR,
|
|
.valp = &glob_conf.server,
|
|
.descr = "server address"
|
|
},
|
|
{
|
|
.name = "SINK",
|
|
.tag = AUD_OPT_STR,
|
|
.valp = &glob_conf.sink,
|
|
.descr = "sink device name"
|
|
},
|
|
{
|
|
.name = "SOURCE",
|
|
.tag = AUD_OPT_STR,
|
|
.valp = &glob_conf.source,
|
|
.descr = "source device name"
|
|
},
|
|
{ /* End of list */ }
|
|
};
|
|
|
|
static struct audio_pcm_ops qpa_pcm_ops = {
|
|
.init_out = qpa_init_out,
|
|
.fini_out = qpa_fini_out,
|
|
.run_out = qpa_run_out,
|
|
.write = qpa_write,
|
|
.ctl_out = qpa_ctl_out,
|
|
|
|
.init_in = qpa_init_in,
|
|
.fini_in = qpa_fini_in,
|
|
.run_in = qpa_run_in,
|
|
.read = qpa_read,
|
|
.ctl_in = qpa_ctl_in
|
|
};
|
|
|
|
static struct audio_driver pa_audio_driver = {
|
|
.name = "pa",
|
|
.descr = "http://www.pulseaudio.org/",
|
|
.options = qpa_options,
|
|
.init = qpa_audio_init,
|
|
.fini = qpa_audio_fini,
|
|
.pcm_ops = &qpa_pcm_ops,
|
|
.can_be_default = 1,
|
|
.max_voices_out = INT_MAX,
|
|
.max_voices_in = INT_MAX,
|
|
.voice_size_out = sizeof (PAVoiceOut),
|
|
.voice_size_in = sizeof (PAVoiceIn),
|
|
.ctl_caps = VOICE_VOLUME_CAP
|
|
};
|
|
|
|
static void register_audio_pa(void)
|
|
{
|
|
audio_driver_register(&pa_audio_driver);
|
|
}
|
|
type_init(register_audio_pa);
|