diff options
Diffstat (limited to 'audio')
| -rw-r--r-- | audio/alsaaudio.c | 49 | ||||
| -rw-r--r-- | audio/audio.c | 347 | ||||
| -rw-r--r-- | audio/audio.h | 37 | ||||
| -rw-r--r-- | audio/audio_int.h | 43 | ||||
| -rw-r--r-- | audio/audio_template.h | 62 | ||||
| -rw-r--r-- | audio/coreaudio.c | 18 | ||||
| -rw-r--r-- | audio/dsoundaudio.c | 31 | ||||
| -rw-r--r-- | audio/mixeng.h | 9 | ||||
| -rw-r--r-- | audio/noaudio.c | 39 | ||||
| -rw-r--r-- | audio/ossaudio.c | 75 | ||||
| -rw-r--r-- | audio/paaudio.c | 413 | ||||
| -rw-r--r-- | audio/rate_template.h | 2 | ||||
| -rw-r--r-- | audio/sdlaudio.c | 30 | ||||
| -rw-r--r-- | audio/spiceaudio.c | 34 | ||||
| -rw-r--r-- | audio/wavaudio.c | 18 | ||||
| -rw-r--r-- | audio/wavcapture.c | 6 |
16 files changed, 622 insertions, 591 deletions
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c index 3745c823ad..591344dccd 100644 --- a/audio/alsaaudio.c +++ b/audio/alsaaudio.c @@ -39,6 +39,7 @@ struct pollhlp { struct pollfd *pfds; int count; int mask; + AudioState *s; }; typedef struct ALSAVoiceOut { @@ -199,11 +200,11 @@ static void alsa_poll_handler (void *opaque) break; case SND_PCM_STATE_PREPARED: - audio_run ("alsa run (prepared)"); + audio_run(hlp->s, "alsa run (prepared)"); break; case SND_PCM_STATE_RUNNING: - audio_run ("alsa run (running)"); + audio_run(hlp->s, "alsa run (running)"); break; default: @@ -269,11 +270,6 @@ static int alsa_poll_in (HWVoiceIn *hw) return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLIN); } -static int alsa_write (SWVoiceOut *sw, void *buf, int len) -{ - return audio_pcm_sw_write (sw, buf, len); -} - static snd_pcm_format_t aud_to_alsafmt (AudioFormat fmt, int endianness) { switch (fmt) { @@ -634,7 +630,7 @@ static void alsa_write_pending (ALSAVoiceOut *alsa) while (alsa->pending) { int left_till_end_samples = hw->samples - alsa->wpos; - int len = audio_MIN (alsa->pending, left_till_end_samples); + int len = MIN (alsa->pending, left_till_end_samples); char *src = advance (alsa->pcm_buf, alsa->wpos << hw->info.shift); while (len) { @@ -685,10 +681,10 @@ static void alsa_write_pending (ALSAVoiceOut *alsa) } } -static int alsa_run_out (HWVoiceOut *hw, int live) +static size_t alsa_run_out(HWVoiceOut *hw, size_t live) { ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; - int decr; + size_t decr; snd_pcm_sframes_t avail; avail = alsa_get_avail (alsa->handle); @@ -697,7 +693,7 @@ static int alsa_run_out (HWVoiceOut *hw, int live) return 0; } - decr = audio_MIN (live, avail); + decr = MIN (live, avail); decr = audio_pcm_hw_clip_out (hw, alsa->pcm_buf, decr, alsa->pending); alsa->pending += decr; alsa_write_pending (alsa); @@ -743,12 +739,13 @@ static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as, alsa->pcm_buf = audio_calloc(__func__, obt.samples, 1 << hw->info.shift); if (!alsa->pcm_buf) { - dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n", - hw->samples, 1 << hw->info.shift); + dolog("Could not allocate DAC buffer (%zu samples, each %d bytes)\n", + hw->samples, 1 << hw->info.shift); alsa_anal_close1 (&handle); return -1; } + alsa->pollhlp.s = hw->s; alsa->handle = handle; alsa->dev = dev; return 0; @@ -844,12 +841,13 @@ static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque) alsa->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift); if (!alsa->pcm_buf) { - dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n", - hw->samples, 1 << hw->info.shift); + dolog("Could not allocate ADC buffer (%zu samples, each %d bytes)\n", + hw->samples, 1 << hw->info.shift); alsa_anal_close1 (&handle); return -1; } + alsa->pollhlp.s = hw->s; alsa->handle = handle; alsa->dev = dev; return 0; @@ -865,17 +863,17 @@ static void alsa_fini_in (HWVoiceIn *hw) alsa->pcm_buf = NULL; } -static int alsa_run_in (HWVoiceIn *hw) +static size_t alsa_run_in(HWVoiceIn *hw) { ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; int hwshift = hw->info.shift; int i; - int live = audio_pcm_hw_get_live_in (hw); - int dead = hw->samples - live; - int decr; + size_t live = audio_pcm_hw_get_live_in (hw); + size_t dead = hw->samples - live; + size_t decr; struct { - int add; - int len; + size_t add; + size_t len; } bufs[2] = { { .add = hw->wpos, .len = 0 }, { .add = 0, .len = 0 } @@ -915,7 +913,7 @@ static int alsa_run_in (HWVoiceIn *hw) } } - decr = audio_MIN (dead, avail); + decr = MIN(dead, avail); if (!decr) { return 0; } @@ -985,11 +983,6 @@ static int alsa_run_in (HWVoiceIn *hw) return read_samples; } -static int alsa_read (SWVoiceIn *sw, void *buf, int size) -{ - return audio_pcm_sw_read (sw, buf, size); -} - static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...) { ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; @@ -1073,13 +1066,11 @@ static struct audio_pcm_ops alsa_pcm_ops = { .init_out = alsa_init_out, .fini_out = alsa_fini_out, .run_out = alsa_run_out, - .write = alsa_write, .ctl_out = alsa_ctl_out, .init_in = alsa_init_in, .fini_in = alsa_fini_in, .run_in = alsa_run_in, - .read = alsa_read, .ctl_in = alsa_ctl_in, }; diff --git a/audio/audio.c b/audio/audio.c index c8b88d892d..7d715332c9 100644 --- a/audio/audio.c +++ b/audio/audio.c @@ -87,7 +87,8 @@ audio_driver *audio_driver_lookup(const char *name) return NULL; } -static AudioState glob_audio_state; +static QTAILQ_HEAD(AudioStateHead, AudioState) audio_states = + QTAILQ_HEAD_INITIALIZER(audio_states); const struct mixeng_volume nominal_volume = { .mute = 0, @@ -100,6 +101,8 @@ const struct mixeng_volume nominal_volume = { #endif }; +static bool legacy_config = true; + #ifdef AUDIO_IS_FLAWLESS_AND_NO_CHECKS_ARE_REQURIED #error No its not #else @@ -306,6 +309,7 @@ void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as) case AUDIO_FORMAT_S16: sign = 1; + /* fall through */ case AUDIO_FORMAT_U16: bits = 16; shift = 1; @@ -313,6 +317,7 @@ void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as) case AUDIO_FORMAT_S32: sign = 1; + /* fall through */ case AUDIO_FORMAT_U32: bits = 32; shift = 2; @@ -399,12 +404,10 @@ static void noop_conv (struct st_sample *dst, const void *src, int samples) (void) samples; } -static CaptureVoiceOut *audio_pcm_capture_find_specific ( - struct audsettings *as - ) +static CaptureVoiceOut *audio_pcm_capture_find_specific(AudioState *s, + struct audsettings *as) { CaptureVoiceOut *cap; - AudioState *s = &glob_audio_state; for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) { if (audio_pcm_info_eq (&cap->hw.info, as)) { @@ -481,7 +484,7 @@ static void audio_detach_capture (HWVoiceOut *hw) static int audio_attach_capture (HWVoiceOut *hw) { - AudioState *s = &glob_audio_state; + AudioState *s = hw->s; CaptureVoiceOut *cap; audio_detach_capture (hw); @@ -525,41 +528,41 @@ static int audio_attach_capture (HWVoiceOut *hw) /* * Hard voice (capture) */ -static int audio_pcm_hw_find_min_in (HWVoiceIn *hw) +static size_t audio_pcm_hw_find_min_in (HWVoiceIn *hw) { SWVoiceIn *sw; - int m = hw->total_samples_captured; + size_t m = hw->total_samples_captured; for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) { if (sw->active) { - m = audio_MIN (m, sw->total_hw_samples_acquired); + m = MIN (m, sw->total_hw_samples_acquired); } } return m; } -int audio_pcm_hw_get_live_in (HWVoiceIn *hw) +size_t audio_pcm_hw_get_live_in(HWVoiceIn *hw) { - int live = hw->total_samples_captured - audio_pcm_hw_find_min_in (hw); - if (audio_bug(__func__, live < 0 || live > hw->samples)) { - dolog ("live=%d hw->samples=%d\n", live, hw->samples); + size_t live = hw->total_samples_captured - audio_pcm_hw_find_min_in (hw); + if (audio_bug(__func__, live > hw->samples)) { + dolog("live=%zu hw->samples=%zu\n", live, hw->samples); return 0; } return live; } -int audio_pcm_hw_clip_out (HWVoiceOut *hw, void *pcm_buf, - int live, int pending) +size_t audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf, + size_t live, size_t pending) { - int left = hw->samples - pending; - int len = audio_MIN (left, live); - int clipped = 0; + size_t left = hw->samples - pending; + size_t len = MIN (left, live); + size_t clipped = 0; while (len) { struct st_sample *src = hw->mix_buf + hw->rpos; uint8_t *dst = advance (pcm_buf, hw->rpos << hw->info.shift); - int samples_till_end_of_buf = hw->samples - hw->rpos; - int samples_to_clip = audio_MIN (len, samples_till_end_of_buf); + size_t samples_till_end_of_buf = hw->samples - hw->rpos; + size_t samples_to_clip = MIN (len, samples_till_end_of_buf); hw->clip (dst, src, samples_to_clip); @@ -573,14 +576,14 @@ int audio_pcm_hw_clip_out (HWVoiceOut *hw, void *pcm_buf, /* * Soft voice (capture) */ -static int audio_pcm_sw_get_rpos_in (SWVoiceIn *sw) +static size_t audio_pcm_sw_get_rpos_in(SWVoiceIn *sw) { HWVoiceIn *hw = sw->hw; - int live = hw->total_samples_captured - sw->total_hw_samples_acquired; - int rpos; + ssize_t live = hw->total_samples_captured - sw->total_hw_samples_acquired; + ssize_t rpos; if (audio_bug(__func__, live < 0 || live > hw->samples)) { - dolog ("live=%d hw->samples=%d\n", live, hw->samples); + dolog("live=%zu hw->samples=%zu\n", live, hw->samples); return 0; } @@ -593,17 +596,17 @@ static int audio_pcm_sw_get_rpos_in (SWVoiceIn *sw) } } -int audio_pcm_sw_read (SWVoiceIn *sw, void *buf, int size) +static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size) { HWVoiceIn *hw = sw->hw; - int samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0; + size_t samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0; struct st_sample *src, *dst = sw->buf; rpos = audio_pcm_sw_get_rpos_in (sw) % hw->samples; live = hw->total_samples_captured - sw->total_hw_samples_acquired; - if (audio_bug(__func__, live < 0 || live > hw->samples)) { - dolog ("live_in=%d hw->samples=%d\n", live, hw->samples); + if (audio_bug(__func__, live > hw->samples)) { + dolog("live_in=%zu hw->samples=%zu\n", live, hw->samples); return 0; } @@ -613,13 +616,13 @@ int audio_pcm_sw_read (SWVoiceIn *sw, void *buf, int size) } swlim = (live * sw->ratio) >> 32; - swlim = audio_MIN (swlim, samples); + swlim = MIN (swlim, samples); while (swlim) { src = hw->conv_buf + rpos; - isamp = hw->wpos - rpos; - /* XXX: <= ? */ - if (isamp <= 0) { + if (hw->wpos > rpos) { + isamp = hw->wpos - rpos; + } else { isamp = hw->samples - rpos; } @@ -628,11 +631,6 @@ int audio_pcm_sw_read (SWVoiceIn *sw, void *buf, int size) } osamp = swlim; - if (audio_bug(__func__, osamp < 0)) { - dolog ("osamp=%d\n", osamp); - return 0; - } - st_rate_flow (sw->rate, src, dst, &isamp, &osamp); swlim -= osamp; rpos = (rpos + isamp) % hw->samples; @@ -653,15 +651,15 @@ int audio_pcm_sw_read (SWVoiceIn *sw, void *buf, int size) /* * Hard voice (playback) */ -static int audio_pcm_hw_find_min_out (HWVoiceOut *hw, int *nb_livep) +static size_t audio_pcm_hw_find_min_out (HWVoiceOut *hw, int *nb_livep) { SWVoiceOut *sw; - int m = INT_MAX; + size_t m = SIZE_MAX; int nb_live = 0; for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) { if (sw->active || !sw->empty) { - m = audio_MIN (m, sw->total_hw_samples_mixed); + m = MIN (m, sw->total_hw_samples_mixed); nb_live += 1; } } @@ -670,9 +668,9 @@ static int audio_pcm_hw_find_min_out (HWVoiceOut *hw, int *nb_livep) return m; } -static int audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live) +static size_t audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live) { - int smin; + size_t smin; int nb_live1; smin = audio_pcm_hw_find_min_out (hw, &nb_live1); @@ -681,10 +679,10 @@ static int audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live) } if (nb_live1) { - int live = smin; + size_t live = smin; - if (audio_bug(__func__, live < 0 || live > hw->samples)) { - dolog ("live=%d hw->samples=%d\n", live, hw->samples); + if (audio_bug(__func__, live > hw->samples)) { + dolog("live=%zu hw->samples=%zu\n", live, hw->samples); return 0; } return live; @@ -695,10 +693,10 @@ static int audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live) /* * Soft voice (playback) */ -int audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int size) +static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size) { - int hwsamples, samples, isamp, osamp, wpos, live, dead, left, swlim, blck; - int ret = 0, pos = 0, total = 0; + size_t hwsamples, samples, isamp, osamp, wpos, live, dead, left, swlim, blck; + size_t ret = 0, pos = 0, total = 0; if (!sw) { return size; @@ -707,8 +705,8 @@ int audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int size) hwsamples = sw->hw->samples; live = sw->total_hw_samples_mixed; - if (audio_bug(__func__, live < 0 || live > hwsamples)) { - dolog ("live=%d hw->samples=%d\n", live, hwsamples); + if (audio_bug(__func__, live > hwsamples)) { + dolog("live=%zu hw->samples=%zu\n", live, hwsamples); return 0; } @@ -724,7 +722,7 @@ int audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int size) dead = hwsamples - live; swlim = ((int64_t) dead << 32) / sw->ratio; - swlim = audio_MIN (swlim, samples); + swlim = MIN (swlim, samples); if (swlim) { sw->conv (sw->buf, buf, swlim); @@ -736,7 +734,7 @@ int audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int size) while (swlim) { dead = hwsamples - live; left = hwsamples - wpos; - blck = audio_MIN (dead, left); + blck = MIN (dead, left); if (!blck) { break; } @@ -762,7 +760,7 @@ int audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int size) #ifdef DEBUG_OUT dolog ( - "%s: write size %d ret %d total sw %d\n", + "%s: write size %zu ret %zu total sw %zu\n", SW_NAME (sw), size >> sw->info.shift, ret, @@ -789,19 +787,15 @@ static void audio_pcm_print_info (const char *cap, struct audio_pcm_info *info) /* * Timer */ - -static bool audio_timer_running; -static uint64_t audio_timer_last; - -static int audio_is_timer_needed (void) +static int audio_is_timer_needed(AudioState *s) { HWVoiceIn *hwi = NULL; HWVoiceOut *hwo = NULL; - while ((hwo = audio_pcm_hw_find_any_enabled_out (hwo))) { + while ((hwo = audio_pcm_hw_find_any_enabled_out(s, hwo))) { if (!hwo->poll_mode) return 1; } - while ((hwi = audio_pcm_hw_find_any_enabled_in (hwi))) { + while ((hwi = audio_pcm_hw_find_any_enabled_in(s, hwi))) { if (!hwi->poll_mode) return 1; } return 0; @@ -809,18 +803,18 @@ static int audio_is_timer_needed (void) static void audio_reset_timer (AudioState *s) { - if (audio_is_timer_needed ()) { + if (audio_is_timer_needed(s)) { timer_mod_anticipate_ns(s->ts, qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL) + s->period_ticks); - if (!audio_timer_running) { - audio_timer_running = true; - audio_timer_last = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL); + if (!s->timer_running) { + s->timer_running = true; + s->timer_last = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL); trace_audio_timer_start(s->period_ticks / SCALE_MS); } } else { timer_del(s->ts); - if (audio_timer_running) { - audio_timer_running = false; + if (s->timer_running) { + s->timer_running = false; trace_audio_timer_stop(); } } @@ -832,20 +826,20 @@ static void audio_timer (void *opaque) AudioState *s = opaque; now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL); - diff = now - audio_timer_last; + diff = now - s->timer_last; if (diff > s->period_ticks * 3 / 2) { trace_audio_timer_delayed(diff / SCALE_MS); } - audio_timer_last = now; + s->timer_last = now; - audio_run("timer"); + audio_run(s, "timer"); audio_reset_timer(s); } /* * Public API */ -int AUD_write (SWVoiceOut *sw, void *buf, int size) +size_t AUD_write(SWVoiceOut *sw, void *buf, size_t size) { if (!sw) { /* XXX: Consider options */ @@ -857,10 +851,10 @@ int AUD_write (SWVoiceOut *sw, void *buf, int size) return 0; } - return sw->hw->pcm_ops->write(sw, buf, size); + return audio_pcm_sw_write(sw, buf, size); } -int AUD_read (SWVoiceIn *sw, void *buf, int size) +size_t AUD_read(SWVoiceIn *sw, void *buf, size_t size) { if (!sw) { /* XXX: Consider options */ @@ -872,7 +866,7 @@ int AUD_read (SWVoiceIn *sw, void *buf, int size) return 0; } - return sw->hw->pcm_ops->read(sw, buf, size); + return audio_pcm_sw_read(sw, buf, size); } int AUD_get_buffer_size_out (SWVoiceOut *sw) @@ -890,7 +884,7 @@ void AUD_set_active_out (SWVoiceOut *sw, int on) hw = sw->hw; if (sw->active != on) { - AudioState *s = &glob_audio_state; + AudioState *s = sw->s; SWVoiceOut *temp_sw; SWVoiceCap *sc; @@ -937,7 +931,7 @@ void AUD_set_active_in (SWVoiceIn *sw, int on) hw = sw->hw; if (sw->active != on) { - AudioState *s = &glob_audio_state; + AudioState *s = sw->s; SWVoiceIn *temp_sw; if (on) { @@ -969,17 +963,17 @@ void AUD_set_active_in (SWVoiceIn *sw, int on) } } -static int audio_get_avail (SWVoiceIn *sw) +static size_t audio_get_avail (SWVoiceIn *sw) { - int live; + size_t live; if (!sw) { return 0; } live = sw->hw->total_samples_captured - sw->total_hw_samples_acquired; - if (audio_bug(__func__, live < 0 || live > sw->hw->samples)) { - dolog ("live=%d sw->hw->samples=%d\n", live, sw->hw->samples); + if (audio_bug(__func__, live > sw->hw->samples)) { + dolog("live=%zu sw->hw->samples=%zu\n", live, sw->hw->samples); return 0; } @@ -992,9 +986,9 @@ static int audio_get_avail (SWVoiceIn *sw) return (((int64_t) live << 32) / sw->ratio) << sw->info.shift; } -static int audio_get_free (SWVoiceOut *sw) +static size_t audio_get_free(SWVoiceOut *sw) { - int live, dead; + size_t live, dead; if (!sw) { return 0; @@ -1002,8 +996,8 @@ static int audio_get_free (SWVoiceOut *sw) live = sw->total_hw_samples_mixed; - if (audio_bug(__func__, live < 0 || live > sw->hw->samples)) { - dolog ("live=%d sw->hw->samples=%d\n", live, sw->hw->samples); + if (audio_bug(__func__, live > sw->hw->samples)) { + dolog("live=%zu sw->hw->samples=%zu\n", live, sw->hw->samples); return 0; } @@ -1018,9 +1012,10 @@ static int audio_get_free (SWVoiceOut *sw) return (((int64_t) dead << 32) / sw->ratio) << sw->info.shift; } -static void audio_capture_mix_and_clear (HWVoiceOut *hw, int rpos, int samples) +static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos, + size_t samples) { - int n; + size_t n; if (hw->enabled) { SWVoiceCap *sc; @@ -1031,17 +1026,17 @@ static void audio_capture_mix_and_clear (HWVoiceOut *hw, int rpos, int samples) n = samples; while (n) { - int till_end_of_hw = hw->samples - rpos2; - int to_write = audio_MIN (till_end_of_hw, n); - int bytes = to_write << hw->info.shift; - int written; + size_t till_end_of_hw = hw->samples - rpos2; + size_t to_write = MIN(till_end_of_hw, n); + size_t bytes = to_write << hw->info.shift; + size_t written; sw->buf = hw->mix_buf + rpos2; written = audio_pcm_sw_write (sw, NULL, bytes); if (written - bytes) { - dolog ("Could not mix %d bytes into a capture " - "buffer, mixed %d\n", - bytes, written); + dolog("Could not mix %zu bytes into a capture " + "buffer, mixed %zu\n", + bytes, written); break; } n -= to_write; @@ -1050,9 +1045,9 @@ static void audio_capture_mix_and_clear (HWVoiceOut *hw, int rpos, int samples) } } - n = audio_MIN (samples, hw->samples - rpos); - mixeng_clear (hw->mix_buf + rpos, n); - mixeng_clear (hw->mix_buf, samples - n); + n = MIN(samples, hw->samples - rpos); + mixeng_clear(hw->mix_buf + rpos, n); + mixeng_clear(hw->mix_buf, samples - n); } static void audio_run_out (AudioState *s) @@ -1060,17 +1055,17 @@ static void audio_run_out (AudioState *s) HWVoiceOut *hw = NULL; SWVoiceOut *sw; - while ((hw = audio_pcm_hw_find_any_enabled_out (hw))) { - int played; - int live, free, nb_live, cleanup_required, prev_rpos; + while ((hw = audio_pcm_hw_find_any_enabled_out(s, hw))) { + size_t played, live, prev_rpos, free; + int nb_live, cleanup_required; live = audio_pcm_hw_get_live_out (hw, &nb_live); if (!nb_live) { live = 0; } - if (audio_bug(__func__, live < 0 || live > hw->samples)) { - dolog ("live=%d hw->samples=%d\n", live, hw->samples); + if (audio_bug(__func__, live > hw->samples)) { + dolog ("live=%zu hw->samples=%zu\n", live, hw->samples); continue; } @@ -1105,13 +1100,13 @@ static void audio_run_out (AudioState *s) played = hw->pcm_ops->run_out (hw, live); replay_audio_out(&played); if (audio_bug(__func__, hw->rpos >= hw->samples)) { - dolog ("hw->rpos=%d hw->samples=%d played=%d\n", - hw->rpos, hw->samples, played); + dolog("hw->rpos=%zu hw->samples=%zu played=%zu\n", + hw->rpos, hw->samples, played); hw->rpos = 0; } #ifdef DEBUG_OUT - dolog ("played=%d\n", played); + dolog("played=%zu\n", played); #endif if (played) { @@ -1126,8 +1121,8 @@ static void audio_run_out (AudioState *s) } if (audio_bug(__func__, played > sw->total_hw_samples_mixed)) { - dolog ("played=%d sw->total_hw_samples_mixed=%d\n", - played, sw->total_hw_samples_mixed); + dolog("played=%zu sw->total_hw_samples_mixed=%zu\n", + played, sw->total_hw_samples_mixed); played = sw->total_hw_samples_mixed; } @@ -1165,9 +1160,9 @@ static void audio_run_in (AudioState *s) { HWVoiceIn *hw = NULL; - while ((hw = audio_pcm_hw_find_any_enabled_in (hw))) { + while ((hw = audio_pcm_hw_find_any_enabled_in(s, hw))) { SWVoiceIn *sw; - int captured = 0, min; + size_t captured = 0, min; if (replay_mode != REPLAY_MODE_PLAY) { captured = hw->pcm_ops->run_in(hw); @@ -1182,7 +1177,7 @@ static void audio_run_in (AudioState *s) sw->total_hw_samples_acquired -= min; if (sw->active) { - int avail; + size_t avail; avail = audio_get_avail (sw); if (avail > 0) { @@ -1198,15 +1193,15 @@ static void audio_run_capture (AudioState *s) CaptureVoiceOut *cap; for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) { - int live, rpos, captured; + size_t live, rpos, captured; HWVoiceOut *hw = &cap->hw; SWVoiceOut *sw; captured = live = audio_pcm_hw_get_live_out (hw, NULL); rpos = hw->rpos; while (live) { - int left = hw->samples - rpos; - int to_capture = audio_MIN (live, left); + size_t left = hw->samples - rpos; + size_t to_capture = MIN(live, left); struct st_sample *src; struct capture_callback *cb; @@ -1229,8 +1224,8 @@ static void audio_run_capture (AudioState *s) } if (audio_bug(__func__, captured > sw->total_hw_samples_mixed)) { - dolog ("captured=%d sw->total_hw_samples_mixed=%d\n", - captured, sw->total_hw_samples_mixed); + dolog("captured=%zu sw->total_hw_samples_mixed=%zu\n", + captured, sw->total_hw_samples_mixed); captured = sw->total_hw_samples_mixed; } @@ -1240,13 +1235,12 @@ static void audio_run_capture (AudioState *s) } } -void audio_run (const char *msg) +void audio_run(AudioState *s, const char *msg) { - AudioState *s = &glob_audio_state; + audio_run_out(s); + audio_run_in(s); + audio_run_capture(s); - audio_run_out (s); - audio_run_in (s); - audio_run_capture (s); #ifdef DEBUG_POLL { static double prevtime; @@ -1271,8 +1265,8 @@ static int audio_driver_init(AudioState *s, struct audio_driver *drv, s->drv_opaque = drv->init(dev); if (s->drv_opaque) { - audio_init_nb_voices_out (drv); - audio_init_nb_voices_in (drv); + audio_init_nb_voices_out(s, drv); + audio_init_nb_voices_in(s, drv); s->drv = drv; return 0; } @@ -1293,11 +1287,11 @@ static void audio_vm_change_state_handler (void *opaque, int running, int op = running ? VOICE_ENABLE : VOICE_DISABLE; s->vm_running = running; - while ((hwo = audio_pcm_hw_find_any_enabled_out (hwo))) { + while ((hwo = audio_pcm_hw_find_any_enabled_out(s, hwo))) { hwo->pcm_ops->ctl_out(hwo, op); } - while ((hwi = audio_pcm_hw_find_any_enabled_in (hwi))) { + while ((hwi = audio_pcm_hw_find_any_enabled_in(s, hwi))) { hwi->pcm_ops->ctl_in(hwi, op); } audio_reset_timer (s); @@ -1310,14 +1304,12 @@ bool audio_is_cleaning_up(void) return is_cleaning_up; } -void audio_cleanup(void) +static void free_audio_state(AudioState *s) { - AudioState *s = &glob_audio_state; HWVoiceOut *hwo, *hwon; HWVoiceIn *hwi, *hwin; - is_cleaning_up = true; - QLIST_FOREACH_SAFE(hwo, &glob_audio_state.hw_head_out, entries, hwon) { + QLIST_FOREACH_SAFE(hwo, &s->hw_head_out, entries, hwon) { SWVoiceCap *sc; if (hwo->enabled) { @@ -1336,7 +1328,7 @@ void audio_cleanup(void) QLIST_REMOVE(hwo, entries); } - QLIST_FOREACH_SAFE(hwi, &glob_audio_state.hw_head_in, entries, hwin) { + QLIST_FOREACH_SAFE(hwi, &s->hw_head_in, entries, hwin) { if (hwi->enabled) { hwi->pcm_ops->ctl_in (hwi, VOICE_DISABLE); } @@ -1353,6 +1345,23 @@ void audio_cleanup(void) qapi_free_Audiodev(s->dev); s->dev = NULL; } + + if (s->ts) { + timer_free(s->ts); + s->ts = NULL; + } + + g_free(s); +} + +void audio_cleanup(void) +{ + is_cleaning_up = true; + while (!QTAILQ_EMPTY(&audio_states)) { + AudioState *s = QTAILQ_FIRST(&audio_states); + QTAILQ_REMOVE(&audio_states, s, list); + free_audio_state(s); + } } static const VMStateDescription vmstate_audio = { @@ -1379,28 +1388,34 @@ static AudiodevListEntry *audiodev_find( return NULL; } -static int audio_init(Audiodev *dev) +/* + * if we have dev, this function was called because of an -audiodev argument => + * initialize a new state with it + * if dev == NULL => legacy implicit initialization, return the already created + * state or create a new one + */ +static AudioState *audio_init(Audiodev *dev, const char *name) { + static bool atexit_registered; size_t i; int done = 0; const char *drvname = NULL; VMChangeStateEntry *e; - AudioState *s = &glob_audio_state; + AudioState *s; struct audio_driver *driver; /* silence gcc warning about uninitialized variable */ AudiodevListHead head = QSIMPLEQ_HEAD_INITIALIZER(head); - if (s->drv) { - if (dev) { - dolog("Cannot create more than one audio backend, sorry\n"); - qapi_free_Audiodev(dev); - } - return -1; - } - if (dev) { /* -audiodev option */ + legacy_config = false; drvname = AudiodevDriver_str(dev->driver); + } else if (!QTAILQ_EMPTY(&audio_states)) { + if (!legacy_config) { + dolog("You must specify an audiodev= for the device %s\n", name); + exit(1); + } + return QTAILQ_FIRST(&audio_states); } else { /* legacy implicit initialization */ head = audio_handle_legacy_opts(); @@ -1414,12 +1429,18 @@ static int audio_init(Audiodev *dev) dev = QSIMPLEQ_FIRST(&head)->dev; audio_validate_opts(dev, &error_abort); } + + s = g_malloc0(sizeof(AudioState)); s->dev = dev; QLIST_INIT (&s->hw_head_out); QLIST_INIT (&s->hw_head_in); QLIST_INIT (&s->cap_head); - atexit(audio_cleanup); + if (!atexit_registered) { + atexit(audio_cleanup); + atexit_registered = true; + } + QTAILQ_INSERT_TAIL(&audio_states, s, list); s->ts = timer_new_ns(QEMU_CLOCK_VIRTUAL, audio_timer, s); @@ -1484,7 +1505,7 @@ static int audio_init(Audiodev *dev) QLIST_INIT (&s->card_head); vmstate_register (NULL, 0, &vmstate_audio, s); - return 0; + return s; } void audio_free_audiodev_list(AudiodevListHead *head) @@ -1499,10 +1520,13 @@ void audio_free_audiodev_list(AudiodevListHead *head) void AUD_register_card (const char *name, QEMUSoundCard *card) { - audio_init(NULL); + if (!card->state) { + card->state = audio_init(NULL, name); + } + card->name = g_strdup (name); memset (&card->entries, 0, sizeof (card->entries)); - QLIST_INSERT_HEAD (&glob_audio_state.card_head, card, entries); + QLIST_INSERT_HEAD(&card->state->card_head, card, entries); } void AUD_remove_card (QEMUSoundCard *card) @@ -1512,16 +1536,24 @@ void AUD_remove_card (QEMUSoundCard *card) } -CaptureVoiceOut *AUD_add_capture ( +CaptureVoiceOut *AUD_add_capture( + AudioState *s, struct audsettings *as, struct audio_capture_ops *ops, void *cb_opaque ) { - AudioState *s = &glob_audio_state; CaptureVoiceOut *cap; struct capture_callback *cb; + if (!s) { + if (!legacy_config) { + dolog("You must specify audiodev when trying to capture\n"); + return NULL; + } + s = audio_init(NULL, NULL); + } + if (audio_validate_settings (as)) { dolog ("Invalid settings were passed when trying to add capture\n"); audio_print_settings (as); @@ -1532,7 +1564,7 @@ CaptureVoiceOut *AUD_add_capture ( cb->ops = *ops; cb->opaque = cb_opaque; - cap = audio_pcm_capture_find_specific (as); + cap = audio_pcm_capture_find_specific(s, as); if (cap) { QLIST_INSERT_HEAD (&cap->cb_head, cb, entries); return cap; @@ -1544,6 +1576,7 @@ CaptureVoiceOut *AUD_add_capture ( cap = g_malloc0(sizeof(*cap)); hw = &cap->hw; + hw->s = s; QLIST_INIT (&hw->sw_head); QLIST_INIT (&cap->cb_head); @@ -1564,7 +1597,7 @@ CaptureVoiceOut *AUD_add_capture ( QLIST_INSERT_HEAD (&s->cap_head, cap, entries); QLIST_INSERT_HEAD (&cap->cb_head, cb, entries); - QLIST_FOREACH(hw, &glob_audio_state.hw_head_out, entries) { + QLIST_FOREACH(hw, &s->hw_head_out, entries) { audio_attach_capture (hw); } return cap; @@ -1749,7 +1782,7 @@ void audio_init_audiodevs(void) AudiodevListEntry *e; QSIMPLEQ_FOREACH(e, &audiodevs, next) { - audio_init(e->dev); + audio_init(e->dev, NULL); } } @@ -1810,3 +1843,25 @@ int audio_buffer_bytes(AudiodevPerDirectionOptions *pdo, return audio_buffer_samples(pdo, as, def_usecs) * audioformat_bytes_per_sample(as->fmt); } + +AudioState *audio_state_by_name(const char *name) +{ + AudioState *s; + QTAILQ_FOREACH(s, &audio_states, list) { + assert(s->dev); + if (strcmp(name, s->dev->id) == 0) { + return s; + } + } + return NULL; +} + +const char *audio_get_id(QEMUSoundCard *card) +{ + if (card->state) { + assert(card->state->dev); + return card->state->dev->id; + } else { + return ""; + } +} diff --git a/audio/audio.h b/audio/audio.h index 64b0f761bc..c74abb8c47 100644 --- a/audio/audio.h +++ b/audio/audio.h @@ -27,6 +27,7 @@ #include "qemu/queue.h" #include "qapi/qapi-types-audio.h" +#include "hw/qdev-properties.h" typedef void (*audio_callback_fn) (void *opaque, int avail); @@ -78,8 +79,10 @@ typedef struct SWVoiceOut SWVoiceOut; typedef struct CaptureVoiceOut CaptureVoiceOut; typedef struct SWVoiceIn SWVoiceIn; +typedef struct AudioState AudioState; typedef struct QEMUSoundCard { char *name; + AudioState *state; QLIST_ENTRY (QEMUSoundCard) entries; } QEMUSoundCard; @@ -92,7 +95,8 @@ void AUD_log (const char *cap, const char *fmt, ...) GCC_FMT_ATTR(2, 3); void AUD_register_card (const char *name, QEMUSoundCard *card); void AUD_remove_card (QEMUSoundCard *card); -CaptureVoiceOut *AUD_add_capture ( +CaptureVoiceOut *AUD_add_capture( + AudioState *s, struct audsettings *as, struct audio_capture_ops *ops, void *opaque @@ -109,7 +113,7 @@ SWVoiceOut *AUD_open_out ( ); void AUD_close_out (QEMUSoundCard *card, SWVoiceOut *sw); -int AUD_write (SWVoiceOut *sw, void *pcm_buf, int size); +size_t AUD_write (SWVoiceOut *sw, void *pcm_buf, size_t size); int AUD_get_buffer_size_out (SWVoiceOut *sw); void AUD_set_active_out (SWVoiceOut *sw, int on); int AUD_is_active_out (SWVoiceOut *sw); @@ -130,7 +134,7 @@ SWVoiceIn *AUD_open_in ( ); void AUD_close_in (QEMUSoundCard *card, SWVoiceIn *sw); -int AUD_read (SWVoiceIn *sw, void *pcm_buf, int size); +size_t AUD_read (SWVoiceIn *sw, void *pcm_buf, size_t size); void AUD_set_active_in (SWVoiceIn *sw, int on); int AUD_is_active_in (SWVoiceIn *sw); @@ -143,25 +147,8 @@ static inline void *advance (void *p, int incr) return (d + incr); } -#ifdef __GNUC__ -#define audio_MIN(a, b) ( __extension__ ({ \ - __typeof (a) ta = a; \ - __typeof (b) tb = b; \ - ((ta)>(tb)?(tb):(ta)); \ -})) - -#define audio_MAX(a, b) ( __extension__ ({ \ - __typeof (a) ta = a; \ - __typeof (b) tb = b; \ - ((ta)<(tb)?(tb):(ta)); \ -})) -#else -#define audio_MIN(a, b) ((a)>(b)?(b):(a)) -#define audio_MAX(a, b) ((a)<(b)?(b):(a)) -#endif - -int wav_start_capture (CaptureState *s, const char *path, int freq, - int bits, int nchannels); +int wav_start_capture(AudioState *state, CaptureState *s, const char *path, + int freq, int bits, int nchannels); bool audio_is_cleaning_up(void); void audio_cleanup(void); @@ -175,4 +162,10 @@ void audio_parse_option(const char *opt); void audio_init_audiodevs(void); void audio_legacy_help(void); +AudioState *audio_state_by_name(const char *name); +const char *audio_get_id(QEMUSoundCard *card); + +#define DEFINE_AUDIO_PROPERTIES(_s, _f) \ + DEFINE_PROP_AUDIODEV("audiodev", _s, _f) + #endif /* QEMU_AUDIO_H */ diff --git a/audio/audio_int.h b/audio/audio_int.h index 3f14842709..a674c5374a 100644 --- a/audio/audio_int.h +++ b/audio/audio_int.h @@ -49,9 +49,11 @@ struct audio_pcm_info { int swap_endianness; }; +typedef struct AudioState AudioState; typedef struct SWVoiceCap SWVoiceCap; typedef struct HWVoiceOut { + AudioState *s; int enabled; int poll_mode; int pending_disable; @@ -59,12 +61,12 @@ typedef struct HWVoiceOut { f_sample *clip; - int rpos; + size_t rpos; uint64_t ts_helper; struct st_sample *mix_buf; - int samples; + size_t samples; QLIST_HEAD (sw_out_listhead, SWVoiceOut) sw_head; QLIST_HEAD (sw_cap_listhead, SWVoiceCap) cap_head; int ctl_caps; @@ -73,19 +75,20 @@ typedef struct HWVoiceOut { } HWVoiceOut; typedef struct HWVoiceIn { + AudioState *s; int enabled; int poll_mode; struct audio_pcm_info info; t_sample *conv; - int wpos; - int total_samples_captured; + size_t wpos; + size_t total_samples_captured; uint64_t ts_helper; struct st_sample *conv_buf; - int samples; + size_t samples; QLIST_HEAD (sw_in_listhead, SWVoiceIn) sw_head; int ctl_caps; struct audio_pcm_ops *pcm_ops; @@ -94,12 +97,13 @@ typedef struct HWVoiceIn { struct SWVoiceOut { QEMUSoundCard *card; + AudioState *s; struct audio_pcm_info info; t_sample *conv; int64_t ratio; struct st_sample *buf; void *rate; - int total_hw_samples_mixed; + size_t total_hw_samples_mixed; int active; int empty; HWVoiceOut *hw; @@ -111,11 +115,12 @@ struct SWVoiceOut { struct SWVoiceIn { QEMUSoundCard *card; + AudioState *s; int active; struct audio_pcm_info info; int64_t ratio; void *rate; - int total_hw_samples_acquired; + size_t total_hw_samples_acquired; struct st_sample *buf; f_sample *clip; HWVoiceIn *hw; @@ -144,14 +149,12 @@ struct audio_driver { struct audio_pcm_ops { int (*init_out)(HWVoiceOut *hw, struct audsettings *as, void *drv_opaque); void (*fini_out)(HWVoiceOut *hw); - int (*run_out) (HWVoiceOut *hw, int live); - int (*write) (SWVoiceOut *sw, void *buf, int size); + size_t (*run_out)(HWVoiceOut *hw, size_t live); int (*ctl_out) (HWVoiceOut *hw, int cmd, ...); int (*init_in) (HWVoiceIn *hw, struct audsettings *as, void *drv_opaque); void (*fini_in) (HWVoiceIn *hw); - int (*run_in) (HWVoiceIn *hw); - int (*read) (SWVoiceIn *sw, void *buf, int size); + size_t (*run_in)(HWVoiceIn *hw); int (*ctl_in) (HWVoiceIn *hw, int cmd, ...); }; @@ -188,6 +191,11 @@ typedef struct AudioState { int nb_hw_voices_in; int vm_running; int64_t period_ticks; + + bool timer_running; + uint64_t timer_last; + + QTAILQ_ENTRY(AudioState) list; } AudioState; extern const struct mixeng_volume nominal_volume; @@ -200,18 +208,15 @@ audio_driver *audio_driver_lookup(const char *name); void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as); void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len); -int audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int len); -int audio_pcm_hw_get_live_in (HWVoiceIn *hw); - -int audio_pcm_sw_read (SWVoiceIn *sw, void *buf, int len); +size_t audio_pcm_hw_get_live_in(HWVoiceIn *hw); -int audio_pcm_hw_clip_out (HWVoiceOut *hw, void *pcm_buf, - int live, int pending); +size_t audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf, + size_t live, size_t pending); int audio_bug (const char *funcname, int cond); void *audio_calloc (const char *funcname, int nmemb, size_t size); -void audio_run (const char *msg); +void audio_run(AudioState *s, const char *msg); #define VOICE_ENABLE 1 #define VOICE_DISABLE 2 @@ -219,7 +224,7 @@ void audio_run (const char *msg); #define VOICE_VOLUME_CAP (1 << VOICE_VOLUME) -static inline int audio_ring_dist (int dst, int src, int len) +static inline size_t audio_ring_dist(size_t dst, size_t src, size_t len) { return (dst >= src) ? (dst - src) : (len - src + dst); } diff --git a/audio/audio_template.h b/audio/audio_template.h index 1232bb54db..2562bf5f00 100644 --- a/audio/audio_template.h +++ b/audio/audio_template.h @@ -36,9 +36,9 @@ #define HWBUF hw->conv_buf #endif -static void glue (audio_init_nb_voices_, TYPE) (struct audio_driver *drv) +static void glue(audio_init_nb_voices_, TYPE)(AudioState *s, + struct audio_driver *drv) { - AudioState *s = &glob_audio_state; int max_voices = glue (drv->max_voices_, TYPE); int voice_size = glue (drv->voice_size_, TYPE); @@ -75,16 +75,16 @@ static void glue (audio_pcm_hw_free_resources_, TYPE) (HW *hw) HWBUF = NULL; } -static int glue (audio_pcm_hw_alloc_resources_, TYPE) (HW *hw) +static bool glue(audio_pcm_hw_alloc_resources_, TYPE)(HW *hw) { HWBUF = audio_calloc(__func__, hw->samples, sizeof(struct st_sample)); if (!HWBUF) { - dolog ("Could not allocate " NAME " buffer (%d samples)\n", - hw->samples); - return -1; + dolog("Could not allocate " NAME " buffer (%zu samples)\n", + hw->samples); + return false; } - return 0; + return true; } static void glue (audio_pcm_sw_free_resources_, TYPE) (SW *sw) @@ -183,8 +183,8 @@ static void glue (audio_pcm_hw_del_sw_, TYPE) (SW *sw) static void glue (audio_pcm_hw_gc_, TYPE) (HW **hwp) { - AudioState *s = &glob_audio_state; HW *hw = *hwp; + AudioState *s = hw->s; if (!hw->sw_head.lh_first) { #ifdef DAC @@ -199,15 +199,14 @@ static void glue (audio_pcm_hw_gc_, TYPE) (HW **hwp) } } -static HW *glue (audio_pcm_hw_find_any_, TYPE) (HW *hw) +static HW *glue(audio_pcm_hw_find_any_, TYPE)(AudioState *s, HW *hw) { - AudioState *s = &glob_audio_state; return hw ? hw->entries.le_next : glue (s->hw_head_, TYPE).lh_first; } -static HW *glue (audio_pcm_hw_find_any_enabled_, TYPE) (HW *hw) +static HW *glue(audio_pcm_hw_find_any_enabled_, TYPE)(AudioState *s, HW *hw) { - while ((hw = glue (audio_pcm_hw_find_any_, TYPE) (hw))) { + while ((hw = glue(audio_pcm_hw_find_any_, TYPE)(s, hw))) { if (hw->enabled) { return hw; } @@ -215,12 +214,10 @@ static HW *glue (audio_pcm_hw_find_any_enabled_, TYPE) (HW *hw) return NULL; } -static HW *glue (audio_pcm_hw_find_specific_, TYPE) ( - HW *hw, - struct audsettings *as - ) +static HW *glue(audio_pcm_hw_find_specific_, TYPE)(AudioState *s, HW *hw, + struct audsettings *as) { - while ((hw = glue (audio_pcm_hw_find_any_, TYPE) (hw))) { + while ((hw = glue(audio_pcm_hw_find_any_, TYPE)(s, hw))) { if (audio_pcm_info_eq (&hw->info, as)) { return hw; } @@ -228,10 +225,10 @@ static HW *glue (audio_pcm_hw_find_specific_, TYPE) ( return NULL; } -static HW *glue (audio_pcm_hw_add_new_, TYPE) (struct audsettings *as) +static HW *glue(audio_pcm_hw_add_new_, TYPE)(AudioState *s, + struct audsettings *as) { HW *hw; - AudioState *s = &glob_audio_state; struct audio_driver *drv = s->drv; if (!glue (s->nb_hw_voices_, TYPE)) { @@ -255,6 +252,7 @@ static HW *glue (audio_pcm_hw_add_new_, TYPE) (struct audsettings *as) return NULL; } + hw->s = s; hw->pcm_ops = drv->pcm_ops; hw->ctl_caps = drv->ctl_caps; @@ -267,7 +265,7 @@ static HW *glue (audio_pcm_hw_add_new_, TYPE) (struct audsettings *as) } if (audio_bug(__func__, hw->samples <= 0)) { - dolog ("hw->samples=%d\n", hw->samples); + dolog("hw->samples=%zd\n", hw->samples); goto err1; } @@ -281,7 +279,7 @@ static HW *glue (audio_pcm_hw_add_new_, TYPE) (struct audsettings *as) [hw->info.swap_endianness] [audio_bits_to_index (hw->info.bits)]; - if (glue (audio_pcm_hw_alloc_resources_, TYPE) (hw)) { + if (!glue(audio_pcm_hw_alloc_resources_, TYPE)(hw)) { goto err1; } @@ -328,33 +326,33 @@ AudiodevPerDirectionOptions *glue(audio_get_pdo_, TYPE)(Audiodev *dev) abort(); } -static HW *glue (audio_pcm_hw_add_, TYPE) (struct audsettings *as) +static HW *glue(audio_pcm_hw_add_, TYPE)(AudioState *s, struct audsettings *as) { HW *hw; - AudioState *s = &glob_audio_state; AudiodevPerDirectionOptions *pdo = glue(audio_get_pdo_, TYPE)(s->dev); if (pdo->fixed_settings) { - hw = glue (audio_pcm_hw_add_new_, TYPE) (as); + hw = glue(audio_pcm_hw_add_new_, TYPE)(s, as); if (hw) { return hw; } } - hw = glue (audio_pcm_hw_find_specific_, TYPE) (NULL, as); + hw = glue(audio_pcm_hw_find_specific_, TYPE)(s, NULL, as); if (hw) { return hw; } - hw = glue (audio_pcm_hw_add_new_, TYPE) (as); + hw = glue(audio_pcm_hw_add_new_, TYPE)(s, as); if (hw) { return hw; } - return glue (audio_pcm_hw_find_any_, TYPE) (NULL); + return glue(audio_pcm_hw_find_any_, TYPE)(s, NULL); } -static SW *glue (audio_pcm_create_voice_pair_, TYPE) ( +static SW *glue(audio_pcm_create_voice_pair_, TYPE)( + AudioState *s, const char *sw_name, struct audsettings *as ) @@ -362,7 +360,6 @@ static SW *glue (audio_pcm_create_voice_pair_, TYPE) ( SW *sw; HW *hw; struct audsettings hw_as; - AudioState *s = &glob_audio_state; AudiodevPerDirectionOptions *pdo = glue(audio_get_pdo_, TYPE)(s->dev); if (pdo->fixed_settings) { @@ -378,8 +375,9 @@ static SW *glue (audio_pcm_create_voice_pair_, TYPE) ( sw_name ? sw_name : "unknown", sizeof (*sw)); goto err1; } + sw->s = s; - hw = glue (audio_pcm_hw_add_, TYPE) (&hw_as); + hw = glue(audio_pcm_hw_add_, TYPE)(s, &hw_as); if (!hw) { goto err2; } @@ -430,7 +428,7 @@ SW *glue (AUD_open_, TYPE) ( struct audsettings *as ) { - AudioState *s = &glob_audio_state; + AudioState *s = card->state; AudiodevPerDirectionOptions *pdo = glue(audio_get_pdo_, TYPE)(s->dev); if (audio_bug(__func__, !card || !name || !callback_fn || !as)) { @@ -476,7 +474,7 @@ SW *glue (AUD_open_, TYPE) ( } } else { - sw = glue (audio_pcm_create_voice_pair_, TYPE) (name, as); + sw = glue(audio_pcm_create_voice_pair_, TYPE)(s, name, as); if (!sw) { dolog ("Failed to create voice `%s'\n", name); return NULL; diff --git a/audio/coreaudio.c b/audio/coreaudio.c index 4bec6c8c5c..d1be58b40a 100644 --- a/audio/coreaudio.c +++ b/audio/coreaudio.c @@ -43,9 +43,9 @@ typedef struct coreaudioVoiceOut { UInt32 audioDevicePropertyBufferFrameSize; AudioStreamBasicDescription outputStreamBasicDescription; AudioDeviceIOProcID ioprocid; - int live; - int decr; - int rpos; + size_t live; + size_t decr; + size_t rpos; } coreaudioVoiceOut; #if MAC_OS_X_VERSION_MAX_ALLOWED >= MAC_OS_X_VERSION_10_6 @@ -397,9 +397,9 @@ static int coreaudio_unlock (coreaudioVoiceOut *core, const char *fn_name) return 0; } -static int coreaudio_run_out (HWVoiceOut *hw, int live) +static size_t coreaudio_run_out(HWVoiceOut *hw, size_t live) { - int decr; + size_t decr; coreaudioVoiceOut *core = (coreaudioVoiceOut *) hw; if (coreaudio_lock (core, "coreaudio_run_out")) { @@ -413,7 +413,7 @@ static int coreaudio_run_out (HWVoiceOut *hw, int live) core->live); } - decr = audio_MIN (core->decr, live); + decr = MIN (core->decr, live); core->decr -= decr; core->live = live - decr; @@ -489,11 +489,6 @@ static OSStatus audioDeviceIOProc( return 0; } -static int coreaudio_write (SWVoiceOut *sw, void *buf, int len) -{ - return audio_pcm_sw_write (sw, buf, len); -} - static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as, void *drv_opaque) { @@ -692,7 +687,6 @@ static struct audio_pcm_ops coreaudio_pcm_ops = { .init_out = coreaudio_init_out, .fini_out = coreaudio_fini_out, .run_out = coreaudio_run_out, - .write = coreaudio_write, .ctl_out = coreaudio_ctl_out }; diff --git a/audio/dsoundaudio.c b/audio/dsoundaudio.c index 5da4c864c3..2fc118b795 100644 --- a/audio/dsoundaudio.c +++ b/audio/dsoundaudio.c @@ -454,24 +454,20 @@ static int dsound_ctl_out (HWVoiceOut *hw, int cmd, ...) return 0; } -static int dsound_write (SWVoiceOut *sw, void *buf, int len) -{ - return audio_pcm_sw_write (sw, buf, len); -} - -static int dsound_run_out (HWVoiceOut *hw, int live) +static size_t dsound_run_out(HWVoiceOut *hw, size_t live) { int err; HRESULT hr; DSoundVoiceOut *ds = (DSoundVoiceOut *) hw; LPDIRECTSOUNDBUFFER dsb = ds->dsound_buffer; - int len, hwshift; + size_t len; + int hwshift; DWORD blen1, blen2; DWORD len1, len2; DWORD decr; DWORD wpos, ppos, old_pos; LPVOID p1, p2; - int bufsize; + size_t bufsize; dsound *s = ds->s; AudiodevDsoundOptions *dso = &s->dev->u.dsound; @@ -538,9 +534,9 @@ static int dsound_run_out (HWVoiceOut *hw, int live) } } - if (audio_bug(__func__, len < 0 || len > bufsize)) { - dolog ("len=%d bufsize=%d old_pos=%ld ppos=%ld\n", - len, bufsize, old_pos, ppos); + if (audio_bug(__func__, len > bufsize)) { + dolog("len=%zu bufsize=%zu old_pos=%ld ppos=%ld\n", + len, bufsize, old_pos, ppos); return 0; } @@ -645,18 +641,13 @@ static int dsound_ctl_in (HWVoiceIn *hw, int cmd, ...) return 0; } -static int dsound_read (SWVoiceIn *sw, void *buf, int len) -{ - return audio_pcm_sw_read (sw, buf, len); -} - -static int dsound_run_in (HWVoiceIn *hw) +static size_t dsound_run_in(HWVoiceIn *hw) { int err; HRESULT hr; DSoundVoiceIn *ds = (DSoundVoiceIn *) hw; LPDIRECTSOUNDCAPTUREBUFFER dscb = ds->dsound_capture_buffer; - int live, len, dead; + size_t live, len, dead; DWORD blen1, blen2; DWORD len1, len2; DWORD decr; @@ -707,7 +698,7 @@ static int dsound_run_in (HWVoiceIn *hw) if (!len) { return 0; } - len = audio_MIN (len, dead); + len = MIN (len, dead); err = dsound_lock_in ( dscb, @@ -856,13 +847,11 @@ static struct audio_pcm_ops dsound_pcm_ops = { .init_out = dsound_init_out, .fini_out = dsound_fini_out, .run_out = dsound_run_out, - .write = dsound_write, .ctl_out = dsound_ctl_out, .init_in = dsound_init_in, .fini_in = dsound_fini_in, .run_in = dsound_run_in, - .read = dsound_read, .ctl_in = dsound_ctl_in }; diff --git a/audio/mixeng.h b/audio/mixeng.h index b53a5ef99a..18e62c7c49 100644 --- a/audio/mixeng.h +++ b/audio/mixeng.h @@ -33,6 +33,7 @@ struct st_sample { mixeng_real l; mixeng_real r; }; struct mixeng_volume { int mute; int64_t r; int64_t l; }; struct st_sample { int64_t l; int64_t r; }; #endif +typedef struct st_sample st_sample; typedef void (t_sample) (struct st_sample *dst, const void *src, int samples); typedef void (f_sample) (void *dst, const struct st_sample *src, int samples); @@ -41,10 +42,10 @@ extern t_sample *mixeng_conv[2][2][2][3]; extern f_sample *mixeng_clip[2][2][2][3]; void *st_rate_start (int inrate, int outrate); -void st_rate_flow (void *opaque, struct st_sample *ibuf, struct st_sample *obuf, - int *isamp, int *osamp); -void st_rate_flow_mix (void *opaque, struct st_sample *ibuf, struct st_sample *obuf, - int *isamp, int *osamp); +void st_rate_flow(void *opaque, st_sample *ibuf, st_sample *obuf, + size_t *isamp, size_t *osamp); +void st_rate_flow_mix(void *opaque, st_sample *ibuf, st_sample *obuf, + size_t *isamp, size_t *osamp); void st_rate_stop (void *opaque); void mixeng_clear (struct st_sample *buf, int len); void mixeng_volume (struct st_sample *buf, int len, struct mixeng_volume *vol); diff --git a/audio/noaudio.c b/audio/noaudio.c index 9b195dc52c..0fb2629cf2 100644 --- a/audio/noaudio.c +++ b/audio/noaudio.c @@ -41,10 +41,10 @@ typedef struct NoVoiceIn { int64_t old_ticks; } NoVoiceIn; -static int no_run_out (HWVoiceOut *hw, int live) +static size_t no_run_out(HWVoiceOut *hw, size_t live) { NoVoiceOut *no = (NoVoiceOut *) hw; - int decr, samples; + size_t decr, samples; int64_t now; int64_t ticks; int64_t bytes; @@ -52,20 +52,15 @@ static int no_run_out (HWVoiceOut *hw, int live) now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL); ticks = now - no->old_ticks; bytes = muldiv64(ticks, hw->info.bytes_per_second, NANOSECONDS_PER_SECOND); - bytes = audio_MIN(bytes, INT_MAX); + bytes = MIN(bytes, SIZE_MAX); samples = bytes >> hw->info.shift; no->old_ticks = now; - decr = audio_MIN (live, samples); + decr = MIN (live, samples); hw->rpos = (hw->rpos + decr) % hw->samples; return decr; } -static int no_write (SWVoiceOut *sw, void *buf, int len) -{ - return audio_pcm_sw_write(sw, buf, len); -} - static int no_init_out(HWVoiceOut *hw, struct audsettings *as, void *drv_opaque) { audio_pcm_init_info (&hw->info, as); @@ -97,12 +92,12 @@ static void no_fini_in (HWVoiceIn *hw) (void) hw; } -static int no_run_in (HWVoiceIn *hw) +static size_t no_run_in(HWVoiceIn *hw) { NoVoiceIn *no = (NoVoiceIn *) hw; - int live = audio_pcm_hw_get_live_in (hw); - int dead = hw->samples - live; - int samples = 0; + size_t live = audio_pcm_hw_get_live_in(hw); + size_t dead = hw->samples - live; + size_t samples = 0; if (dead) { int64_t now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL); @@ -111,25 +106,13 @@ static int no_run_in (HWVoiceIn *hw) muldiv64(ticks, hw->info.bytes_per_second, NANOSECONDS_PER_SECOND); no->old_ticks = now; - bytes = audio_MIN (bytes, INT_MAX); + bytes = MIN (bytes, SIZE_MAX); samples = bytes >> hw->info.shift; - samples = audio_MIN (samples, dead); + samples = MIN (samples, dead); } return samples; } -static int no_read (SWVoiceIn *sw, void *buf, int size) -{ - /* use custom code here instead of audio_pcm_sw_read() to avoid - * useless resampling/mixing */ - int samples = size >> sw->info.shift; - int total = sw->hw->total_samples_captured - sw->total_hw_samples_acquired; - int to_clear = audio_MIN (samples, total); - sw->total_hw_samples_acquired += total; - audio_pcm_info_clear_buf (&sw->info, buf, to_clear); - return to_clear << sw->info.shift; -} - static int no_ctl_in (HWVoiceIn *hw, int cmd, ...) { (void) hw; @@ -151,13 +134,11 @@ static struct audio_pcm_ops no_pcm_ops = { .init_out = no_init_out, .fini_out = no_fini_out, .run_out = no_run_out, - .write = no_write, .ctl_out = no_ctl_out, .init_in = no_init_in, .fini_in = no_fini_in, .run_in = no_run_in, - .read = no_read, .ctl_in = no_ctl_in }; diff --git a/audio/ossaudio.c b/audio/ossaudio.c index c0af065b6f..1696933688 100644 --- a/audio/ossaudio.c +++ b/audio/ossaudio.c @@ -110,33 +110,28 @@ static void oss_anal_close (int *fdp) static void oss_helper_poll_out (void *opaque) { - (void) opaque; - audio_run ("oss_poll_out"); + AudioState *s = opaque; + audio_run(s, "oss_poll_out"); } static void oss_helper_poll_in (void *opaque) { - (void) opaque; - audio_run ("oss_poll_in"); + AudioState *s = opaque; + audio_run(s, "oss_poll_in"); } static void oss_poll_out (HWVoiceOut *hw) { OSSVoiceOut *oss = (OSSVoiceOut *) hw; - qemu_set_fd_handler (oss->fd, NULL, oss_helper_poll_out, NULL); + qemu_set_fd_handler(oss->fd, NULL, oss_helper_poll_out, hw->s); } static void oss_poll_in (HWVoiceIn *hw) { OSSVoiceIn *oss = (OSSVoiceIn *) hw; - qemu_set_fd_handler (oss->fd, oss_helper_poll_in, NULL, NULL); -} - -static int oss_write (SWVoiceOut *sw, void *buf, int len) -{ - return audio_pcm_sw_write (sw, buf, len); + qemu_set_fd_handler(oss->fd, oss_helper_poll_in, NULL, hw->s); } static int aud_to_ossfmt (AudioFormat fmt, int endianness) @@ -388,7 +383,7 @@ static void oss_write_pending (OSSVoiceOut *oss) int samples_written; ssize_t bytes_written; int samples_till_end = hw->samples - oss->wpos; - int samples_to_write = audio_MIN (oss->pending, samples_till_end); + int samples_to_write = MIN (oss->pending, samples_till_end); int bytes_to_write = samples_to_write << hw->info.shift; void *pcm = advance (oss->pcm_buf, oss->wpos << hw->info.shift); @@ -416,13 +411,14 @@ static void oss_write_pending (OSSVoiceOut *oss) } } -static int oss_run_out (HWVoiceOut *hw, int live) +static size_t oss_run_out(HWVoiceOut *hw, size_t live) { OSSVoiceOut *oss = (OSSVoiceOut *) hw; - int err, decr; + int err; + size_t decr; struct audio_buf_info abinfo; struct count_info cntinfo; - int bufsize; + size_t bufsize; bufsize = hw->samples << hw->info.shift; @@ -437,7 +433,7 @@ static int oss_run_out (HWVoiceOut *hw, int live) pos = hw->rpos << hw->info.shift; bytes = audio_ring_dist (cntinfo.ptr, pos, bufsize); - decr = audio_MIN (bytes >> hw->info.shift, live); + decr = MIN (bytes >> hw->info.shift, live); } else { err = ioctl (oss->fd, SNDCTL_DSP_GETOSPACE, &abinfo); @@ -456,7 +452,7 @@ static int oss_run_out (HWVoiceOut *hw, int live) return 0; } - decr = audio_MIN (abinfo.bytes >> hw->info.shift, live); + decr = MIN (abinfo.bytes >> hw->info.shift, live); if (!decr) { return 0; } @@ -481,8 +477,8 @@ static void oss_fini_out (HWVoiceOut *hw) if (oss->mmapped) { err = munmap (oss->pcm_buf, hw->samples << hw->info.shift); if (err) { - oss_logerr (errno, "Failed to unmap buffer %p, size %d\n", - oss->pcm_buf, hw->samples << hw->info.shift); + oss_logerr(errno, "Failed to unmap buffer %p, size %zu\n", + oss->pcm_buf, hw->samples << hw->info.shift); } } else { @@ -548,8 +544,8 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as, 0 ); if (oss->pcm_buf == MAP_FAILED) { - oss_logerr (errno, "Failed to map %d bytes of DAC\n", - hw->samples << hw->info.shift); + oss_logerr(errno, "Failed to map %zu bytes of DAC\n", + hw->samples << hw->info.shift); } else { int err; @@ -573,8 +569,8 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as, if (!oss->mmapped) { err = munmap (oss->pcm_buf, hw->samples << hw->info.shift); if (err) { - oss_logerr (errno, "Failed to unmap buffer %p size %d\n", - oss->pcm_buf, hw->samples << hw->info.shift); + oss_logerr(errno, "Failed to unmap buffer %p size %zu\n", + oss->pcm_buf, hw->samples << hw->info.shift); } } } @@ -586,7 +582,7 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as, 1 << hw->info.shift); if (!oss->pcm_buf) { dolog ( - "Could not allocate DAC buffer (%d samples, each %d bytes)\n", + "Could not allocate DAC buffer (%zu samples, each %d bytes)\n", hw->samples, 1 << hw->info.shift ); @@ -698,8 +694,8 @@ static int oss_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque) hw->samples = (obt.nfrags * obt.fragsize) >> hw->info.shift; oss->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift); if (!oss->pcm_buf) { - dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n", - hw->samples, 1 << hw->info.shift); + dolog("Could not allocate ADC buffer (%zu samples, each %d bytes)\n", + hw->samples, 1 << hw->info.shift); oss_anal_close (&fd); return -1; } @@ -719,17 +715,17 @@ static void oss_fini_in (HWVoiceIn *hw) oss->pcm_buf = NULL; } -static int oss_run_in (HWVoiceIn *hw) +static size_t oss_run_in(HWVoiceIn *hw) { OSSVoiceIn *oss = (OSSVoiceIn *) hw; int hwshift = hw->info.shift; int i; - int live = audio_pcm_hw_get_live_in (hw); - int dead = hw->samples - live; + size_t live = audio_pcm_hw_get_live_in (hw); + size_t dead = hw->samples - live; size_t read_samples = 0; struct { - int add; - int len; + size_t add; + size_t len; } bufs[2] = { { .add = hw->wpos, .len = 0 }, { .add = 0, .len = 0 } @@ -756,9 +752,9 @@ static int oss_run_in (HWVoiceIn *hw) if (nread > 0) { if (nread & hw->info.align) { - dolog ("warning: Misaligned read %zd (requested %d), " - "alignment %d\n", nread, bufs[i].add << hwshift, - hw->info.align + 1); + dolog("warning: Misaligned read %zd (requested %zu), " + "alignment %d\n", nread, bufs[i].add << hwshift, + hw->info.align + 1); } read_samples += nread >> hwshift; hw->conv (hw->conv_buf + bufs[i].add, p, nread >> hwshift); @@ -771,9 +767,9 @@ static int oss_run_in (HWVoiceIn *hw) case EAGAIN: break; default: - oss_logerr ( + oss_logerr( errno, - "Failed to read %d bytes of audio (to %p)\n", + "Failed to read %zu bytes of audio (to %p)\n", bufs[i].len, p ); break; @@ -788,11 +784,6 @@ static int oss_run_in (HWVoiceIn *hw) return read_samples; } -static int oss_read (SWVoiceIn *sw, void *buf, int size) -{ - return audio_pcm_sw_read (sw, buf, size); -} - static int oss_ctl_in (HWVoiceIn *hw, int cmd, ...) { OSSVoiceIn *oss = (OSSVoiceIn *) hw; @@ -855,13 +846,11 @@ static struct audio_pcm_ops oss_pcm_ops = { .init_out = oss_init_out, .fini_out = oss_fini_out, .run_out = oss_run_out, - .write = oss_write, .ctl_out = oss_ctl_out, .init_in = oss_init_in, .fini_in = oss_fini_in, .run_in = oss_run_in, - .read = oss_read, .ctl_in = oss_ctl_in }; diff --git a/audio/paaudio.c b/audio/paaudio.c index 5fc886bb33..bfef9acaad 100644 --- a/audio/paaudio.c +++ b/audio/paaudio.c @@ -11,41 +11,52 @@ #include "audio_int.h" #include "audio_pt_int.h" -typedef struct { - Audiodev *dev; +typedef struct PAConnection { + char *server; + int refcount; + QTAILQ_ENTRY(PAConnection) list; + pa_threaded_mainloop *mainloop; pa_context *context; +} PAConnection; + +static QTAILQ_HEAD(PAConnectionHead, PAConnection) pa_conns = + QTAILQ_HEAD_INITIALIZER(pa_conns); + +typedef struct { + Audiodev *dev; + PAConnection *conn; } paaudio; typedef struct { HWVoiceOut hw; - int done; - int live; - int decr; - int rpos; + size_t done; + size_t live; + size_t decr; + size_t rpos; pa_stream *stream; void *pcm_buf; struct audio_pt pt; paaudio *g; - int samples; + size_t samples; } PAVoiceOut; typedef struct { HWVoiceIn hw; - int done; - int dead; - int incr; - int wpos; + size_t done; + size_t dead; + size_t incr; + size_t wpos; pa_stream *stream; void *pcm_buf; struct audio_pt pt; const void *read_data; size_t read_index, read_length; paaudio *g; - int samples; + size_t samples; } PAVoiceIn; -static void qpa_audio_fini(void *opaque); +static void qpa_conn_fini(PAConnection *c); static void GCC_FMT_ATTR (2, 3) qpa_logerr (int err, const char *fmt, ...) { @@ -108,11 +119,11 @@ static inline int PA_STREAM_IS_GOOD(pa_stream_state_t x) static int qpa_simple_read (PAVoiceIn *p, void *data, size_t length, int *rerror) { - paaudio *g = p->g; + PAConnection *c = p->g->conn; - pa_threaded_mainloop_lock (g->mainloop); + pa_threaded_mainloop_lock(c->mainloop); - CHECK_DEAD_GOTO (g, p->stream, rerror, unlock_and_fail); + CHECK_DEAD_GOTO(c, p->stream, rerror, unlock_and_fail); while (length > 0) { size_t l; @@ -121,11 +132,11 @@ static int qpa_simple_read (PAVoiceIn *p, void *data, size_t length, int *rerror int r; r = pa_stream_peek (p->stream, &p->read_data, &p->read_length); - CHECK_SUCCESS_GOTO (g, rerror, r == 0, unlock_and_fail); + CHECK_SUCCESS_GOTO(c, rerror, r == 0, unlock_and_fail); if (!p->read_data) { - pa_threaded_mainloop_wait (g->mainloop); - CHECK_DEAD_GOTO (g, p->stream, rerror, unlock_and_fail); + pa_threaded_mainloop_wait(c->mainloop); + CHECK_DEAD_GOTO(c, p->stream, rerror, unlock_and_fail); } else { p->read_index = 0; } @@ -148,53 +159,53 @@ static int qpa_simple_read (PAVoiceIn *p, void *data, size_t length, int *rerror p->read_length = 0; p->read_index = 0; - CHECK_SUCCESS_GOTO (g, rerror, r == 0, unlock_and_fail); + CHECK_SUCCESS_GOTO(c, rerror, r == 0, unlock_and_fail); } } - pa_threaded_mainloop_unlock (g->mainloop); + pa_threaded_mainloop_unlock(c->mainloop); return 0; unlock_and_fail: - pa_threaded_mainloop_unlock (g->mainloop); + pa_threaded_mainloop_unlock(c->mainloop); return -1; } static int qpa_simple_write (PAVoiceOut *p, const void *data, size_t length, int *rerror) { - paaudio *g = p->g; + PAConnection *c = p->g->conn; - pa_threaded_mainloop_lock (g->mainloop); + pa_threaded_mainloop_lock(c->mainloop); - CHECK_DEAD_GOTO (g, p->stream, rerror, unlock_and_fail); + CHECK_DEAD_GOTO(c, p->stream, rerror, unlock_and_fail); while (length > 0) { size_t l; int r; while (!(l = pa_stream_writable_size (p->stream))) { - pa_threaded_mainloop_wait (g->mainloop); - CHECK_DEAD_GOTO (g, p->stream, rerror, unlock_and_fail); + pa_threaded_mainloop_wait(c->mainloop); + CHECK_DEAD_GOTO(c, p->stream, rerror, unlock_and_fail); } - CHECK_SUCCESS_GOTO (g, rerror, l != (size_t) -1, unlock_and_fail); + CHECK_SUCCESS_GOTO(c, rerror, l != (size_t) -1, unlock_and_fail); if (l > length) { l = length; } r = pa_stream_write (p->stream, data, l, NULL, 0LL, PA_SEEK_RELATIVE); - CHECK_SUCCESS_GOTO (g, rerror, r >= 0, unlock_and_fail); + CHECK_SUCCESS_GOTO(c, rerror, r >= 0, unlock_and_fail); data = (const uint8_t *) data + l; length -= l; } - pa_threaded_mainloop_unlock (g->mainloop); + pa_threaded_mainloop_unlock(c->mainloop); return 0; unlock_and_fail: - pa_threaded_mainloop_unlock (g->mainloop); + pa_threaded_mainloop_unlock(c->mainloop); return -1; } @@ -208,7 +219,7 @@ static void *qpa_thread_out (void *arg) } for (;;) { - int decr, to_mix, rpos; + size_t decr, to_mix, rpos; for (;;) { if (pa->done) { @@ -224,7 +235,7 @@ static void *qpa_thread_out (void *arg) } } - decr = to_mix = audio_MIN(pa->live, pa->samples >> 5); + decr = to_mix = MIN(pa->live, pa->samples >> 5); rpos = pa->rpos; if (audio_pt_unlock(&pa->pt, __func__)) { @@ -233,7 +244,7 @@ static void *qpa_thread_out (void *arg) while (to_mix) { int error; - int chunk = audio_MIN (to_mix, hw->samples - rpos); + size_t chunk = MIN (to_mix, hw->samples - rpos); struct st_sample *src = hw->mix_buf + rpos; hw->clip (pa->pcm_buf, src, chunk); @@ -262,16 +273,16 @@ static void *qpa_thread_out (void *arg) return NULL; } -static int qpa_run_out (HWVoiceOut *hw, int live) +static size_t qpa_run_out(HWVoiceOut *hw, size_t live) { - int decr; + size_t decr; PAVoiceOut *pa = (PAVoiceOut *) hw; if (audio_pt_lock(&pa->pt, __func__)) { return 0; } - decr = audio_MIN (live, pa->decr); + decr = MIN (live, pa->decr); pa->decr -= decr; pa->live = live - decr; hw->rpos = pa->rpos; @@ -284,11 +295,6 @@ static int qpa_run_out (HWVoiceOut *hw, int live) return decr; } -static int qpa_write (SWVoiceOut *sw, void *buf, int len) -{ - return audio_pcm_sw_write (sw, buf, len); -} - /* capture */ static void *qpa_thread_in (void *arg) { @@ -300,7 +306,7 @@ static void *qpa_thread_in (void *arg) } for (;;) { - int incr, to_grab, wpos; + size_t incr, to_grab, wpos; for (;;) { if (pa->done) { @@ -316,7 +322,7 @@ static void *qpa_thread_in (void *arg) } } - incr = to_grab = audio_MIN(pa->dead, pa->samples >> 5); + incr = to_grab = MIN(pa->dead, pa->samples >> 5); wpos = pa->wpos; if (audio_pt_unlock(&pa->pt, __func__)) { @@ -325,7 +331,7 @@ static void *qpa_thread_in (void *arg) while (to_grab) { int error; - int chunk = audio_MIN (to_grab, hw->samples - wpos); + size_t chunk = MIN (to_grab, hw->samples - wpos); void *buf = advance (pa->pcm_buf, wpos); if (qpa_simple_read (pa, buf, @@ -353,9 +359,9 @@ static void *qpa_thread_in (void *arg) return NULL; } -static int qpa_run_in (HWVoiceIn *hw) +static size_t qpa_run_in(HWVoiceIn *hw) { - int live, incr, dead; + size_t live, incr, dead; PAVoiceIn *pa = (PAVoiceIn *) hw; if (audio_pt_lock(&pa->pt, __func__)) { @@ -364,7 +370,7 @@ static int qpa_run_in (HWVoiceIn *hw) live = audio_pcm_hw_get_live_in (hw); dead = hw->samples - live; - incr = audio_MIN (dead, pa->incr); + incr = MIN (dead, pa->incr); pa->incr -= incr; pa->dead = dead - incr; hw->wpos = pa->wpos; @@ -377,11 +383,6 @@ static int qpa_run_in (HWVoiceIn *hw) return incr; } -static int qpa_read (SWVoiceIn *sw, void *buf, int len) -{ - return audio_pcm_sw_read (sw, buf, len); -} - static pa_sample_format_t audfmt_to_pa (AudioFormat afmt, int endianness) { int format; @@ -432,13 +433,13 @@ static AudioFormat pa_to_audfmt (pa_sample_format_t fmt, int *endianness) static void context_state_cb (pa_context *c, void *userdata) { - paaudio *g = userdata; + PAConnection *conn = userdata; switch (pa_context_get_state(c)) { case PA_CONTEXT_READY: case PA_CONTEXT_TERMINATED: case PA_CONTEXT_FAILED: - pa_threaded_mainloop_signal (g->mainloop, 0); + pa_threaded_mainloop_signal(conn->mainloop, 0); break; case PA_CONTEXT_UNCONNECTED: @@ -451,14 +452,14 @@ static void context_state_cb (pa_context *c, void *userdata) static void stream_state_cb (pa_stream *s, void * userdata) { - paaudio *g = userdata; + PAConnection *c = userdata; switch (pa_stream_get_state (s)) { case PA_STREAM_READY: case PA_STREAM_FAILED: case PA_STREAM_TERMINATED: - pa_threaded_mainloop_signal (g->mainloop, 0); + pa_threaded_mainloop_signal(c->mainloop, 0); break; case PA_STREAM_UNCONNECTED: @@ -469,13 +470,13 @@ static void stream_state_cb (pa_stream *s, void * userdata) static void stream_request_cb (pa_stream *s, size_t length, void *userdata) { - paaudio *g = userdata; + PAConnection *c = userdata; - pa_threaded_mainloop_signal (g->mainloop, 0); + pa_threaded_mainloop_signal(c->mainloop, 0); } static pa_stream *qpa_simple_new ( - paaudio *g, + PAConnection *c, const char *name, pa_stream_direction_t dir, const char *dev, @@ -486,50 +487,51 @@ static pa_stream *qpa_simple_new ( { int r; pa_stream *stream; + pa_stream_flags_t flags; - pa_threaded_mainloop_lock (g->mainloop); + pa_threaded_mainloop_lock(c->mainloop); - stream = pa_stream_new (g->context, name, ss, map); + stream = pa_stream_new(c->context, name, ss, map); if (!stream) { goto fail; } - pa_stream_set_state_callback (stream, stream_state_cb, g); - pa_stream_set_read_callback (stream, stream_request_cb, g); - pa_stream_set_write_callback (stream, stream_request_cb, g); + pa_stream_set_state_callback(stream, stream_state_cb, c); + pa_stream_set_read_callback(stream, stream_request_cb, c); + pa_stream_set_write_callback(stream, stream_request_cb, c); + + flags = + PA_STREAM_INTERPOLATE_TIMING + | PA_STREAM_AUTO_TIMING_UPDATE + | PA_STREAM_EARLY_REQUESTS; + + if (dev) { + /* don't move the stream if the user specified a sink/source */ + flags |= PA_STREAM_DONT_MOVE; + } if (dir == PA_STREAM_PLAYBACK) { - r = pa_stream_connect_playback (stream, dev, attr, - PA_STREAM_INTERPOLATE_TIMING -#ifdef PA_STREAM_ADJUST_LATENCY - |PA_STREAM_ADJUST_LATENCY -#endif - |PA_STREAM_AUTO_TIMING_UPDATE, NULL, NULL); + r = pa_stream_connect_playback(stream, dev, attr, flags, NULL, NULL); } else { - r = pa_stream_connect_record (stream, dev, attr, - PA_STREAM_INTERPOLATE_TIMING -#ifdef PA_STREAM_ADJUST_LATENCY - |PA_STREAM_ADJUST_LATENCY -#endif - |PA_STREAM_AUTO_TIMING_UPDATE); + r = pa_stream_connect_record(stream, dev, attr, flags); } if (r < 0) { goto fail; } - pa_threaded_mainloop_unlock (g->mainloop); + pa_threaded_mainloop_unlock(c->mainloop); return stream; fail: - pa_threaded_mainloop_unlock (g->mainloop); + pa_threaded_mainloop_unlock(c->mainloop); if (stream) { pa_stream_unref (stream); } - *rerror = pa_context_errno (g->context); + *rerror = pa_context_errno(c->context); return NULL; } @@ -545,6 +547,7 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as, paaudio *g = pa->g = drv_opaque; AudiodevPaOptions *popts = &g->dev->u.pa; AudiodevPaPerDirectionOptions *ppdo = popts->out; + PAConnection *c = g->conn; ss.format = audfmt_to_pa (as->fmt, as->endianness); ss.channels = as->nchannels; @@ -558,7 +561,7 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as, obt_as.fmt = pa_to_audfmt (ss.format, &obt_as.endianness); pa->stream = qpa_simple_new ( - g, + c, "qemu", PA_STREAM_PLAYBACK, ppdo->has_name ? ppdo->name : NULL, @@ -579,8 +582,8 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as, 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); + dolog("Could not allocate buffer (%zu bytes)\n", + hw->samples << hw->info.shift); goto fail2; } @@ -612,6 +615,7 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque) paaudio *g = pa->g = drv_opaque; AudiodevPaOptions *popts = &g->dev->u.pa; AudiodevPaPerDirectionOptions *ppdo = popts->in; + PAConnection *c = g->conn; ss.format = audfmt_to_pa (as->fmt, as->endianness); ss.channels = as->nchannels; @@ -625,7 +629,7 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque) obt_as.fmt = pa_to_audfmt (ss.format, &obt_as.endianness); pa->stream = qpa_simple_new ( - g, + c, "qemu", PA_STREAM_RECORD, ppdo->has_name ? ppdo->name : NULL, @@ -646,8 +650,8 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque) 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); + dolog("Could not allocate buffer (%zu bytes)\n", + hw->samples << hw->info.shift); goto fail2; } @@ -669,6 +673,27 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque) return -1; } +static void qpa_simple_disconnect(PAConnection *c, pa_stream *stream) +{ + int err; + + pa_threaded_mainloop_lock(c->mainloop); + /* + * wait until actually connects. workaround pa bug #247 + * https://gitlab.freedesktop.org/pulseaudio/pulseaudio/issues/247 + */ + while (pa_stream_get_state(stream) == PA_STREAM_CREATING) { + pa_threaded_mainloop_wait(c->mainloop); + } + + err = pa_stream_disconnect(stream); + if (err != 0) { + dolog("Failed to disconnect! err=%d\n", err); + } + pa_stream_unref(stream); + pa_threaded_mainloop_unlock(c->mainloop); +} + static void qpa_fini_out (HWVoiceOut *hw) { void *ret; @@ -680,7 +705,7 @@ static void qpa_fini_out (HWVoiceOut *hw) audio_pt_join(&pa->pt, &ret, __func__); if (pa->stream) { - pa_stream_unref (pa->stream); + qpa_simple_disconnect(pa->g->conn, pa->stream); pa->stream = NULL; } @@ -700,7 +725,7 @@ static void qpa_fini_in (HWVoiceIn *hw) audio_pt_join(&pa->pt, &ret, __func__); if (pa->stream) { - pa_stream_unref (pa->stream); + qpa_simple_disconnect(pa->g->conn, pa->stream); pa->stream = NULL; } @@ -714,7 +739,7 @@ static int qpa_ctl_out (HWVoiceOut *hw, int cmd, ...) PAVoiceOut *pa = (PAVoiceOut *) hw; pa_operation *op; pa_cvolume v; - paaudio *g = pa->g; + PAConnection *c = pa->g->conn; #ifdef PA_CHECK_VERSION /* macro is present in 0.9.16+ */ pa_cvolume_init (&v); /* function is present in 0.9.13+ */ @@ -734,28 +759,29 @@ static int qpa_ctl_out (HWVoiceOut *hw, int cmd, ...) 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); + pa_threaded_mainloop_lock(c->mainloop); - op = pa_context_set_sink_input_volume (g->context, + op = pa_context_set_sink_input_volume(c->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); + if (!op) { + qpa_logerr(pa_context_errno(c->context), + "set_sink_input_volume() failed\n"); + } else { + pa_operation_unref(op); + } - op = pa_context_set_sink_input_mute (g->context, + op = pa_context_set_sink_input_mute(c->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"); + qpa_logerr(pa_context_errno(c->context), + "set_sink_input_mute() failed\n"); } else { - pa_operation_unref (op); + pa_operation_unref(op); } - pa_threaded_mainloop_unlock (g->mainloop); + pa_threaded_mainloop_unlock(c->mainloop); } } return 0; @@ -766,7 +792,7 @@ static int qpa_ctl_in (HWVoiceIn *hw, int cmd, ...) PAVoiceIn *pa = (PAVoiceIn *) hw; pa_operation *op; pa_cvolume v; - paaudio *g = pa->g; + PAConnection *c = pa->g->conn; #ifdef PA_CHECK_VERSION pa_cvolume_init (&v); @@ -786,29 +812,29 @@ static int qpa_ctl_in (HWVoiceIn *hw, int cmd, ...) 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); + pa_threaded_mainloop_lock(c->mainloop); - op = pa_context_set_source_output_volume (g->context, - pa_stream_get_index (pa->stream), + op = pa_context_set_source_output_volume(c->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"); + qpa_logerr(pa_context_errno(c->context), + "set_source_output_volume() failed\n"); } else { pa_operation_unref(op); } - op = pa_context_set_source_output_mute (g->context, + op = pa_context_set_source_output_mute(c->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"); + qpa_logerr(pa_context_errno(c->context), + "set_source_output_mute() failed\n"); } else { pa_operation_unref (op); } - pa_threaded_mainloop_unlock (g->mainloop); + pa_threaded_mainloop_unlock(c->mainloop); } } return 0; @@ -828,11 +854,75 @@ static int qpa_validate_per_direction_opts(Audiodev *dev, return 1; } +/* common */ +static void *qpa_conn_init(const char *server) +{ + PAConnection *c = g_malloc0(sizeof(PAConnection)); + QTAILQ_INSERT_TAIL(&pa_conns, c, list); + + c->mainloop = pa_threaded_mainloop_new(); + if (!c->mainloop) { + goto fail; + } + + c->context = pa_context_new(pa_threaded_mainloop_get_api(c->mainloop), + server); + if (!c->context) { + goto fail; + } + + pa_context_set_state_callback(c->context, context_state_cb, c); + + if (pa_context_connect(c->context, server, 0, NULL) < 0) { + qpa_logerr(pa_context_errno(c->context), + "pa_context_connect() failed\n"); + goto fail; + } + + pa_threaded_mainloop_lock(c->mainloop); + + if (pa_threaded_mainloop_start(c->mainloop) < 0) { + goto unlock_and_fail; + } + + for (;;) { + pa_context_state_t state; + + state = pa_context_get_state(c->context); + + if (state == PA_CONTEXT_READY) { + break; + } + + if (!PA_CONTEXT_IS_GOOD(state)) { + qpa_logerr(pa_context_errno(c->context), + "Wrong context state\n"); + goto unlock_and_fail; + } + + /* Wait until the context is ready */ + pa_threaded_mainloop_wait(c->mainloop); + } + + pa_threaded_mainloop_unlock(c->mainloop); + return c; + +unlock_and_fail: + pa_threaded_mainloop_unlock(c->mainloop); +fail: + AUD_log (AUDIO_CAP, "Failed to initialize PA context"); + qpa_conn_fini(c); + return NULL; +} + static void *qpa_audio_init(Audiodev *dev) { paaudio *g; AudiodevPaOptions *popts = &dev->u.pa; const char *server; + PAConnection *c; + + assert(dev->driver == AUDIODEV_DRIVER_PA); if (!popts->has_server) { char pidfile[64]; @@ -849,93 +939,64 @@ static void *qpa_audio_init(Audiodev *dev) } } - assert(dev->driver == AUDIODEV_DRIVER_PA); - - g = g_malloc(sizeof(paaudio)); - server = popts->has_server ? popts->server : NULL; - if (!qpa_validate_per_direction_opts(dev, popts->in)) { - goto fail; + return NULL; } if (!qpa_validate_per_direction_opts(dev, popts->out)) { - goto fail; + return NULL; } + g = g_malloc0(sizeof(paaudio)); + server = popts->has_server ? popts->server : NULL; + g->dev = dev; - g->mainloop = NULL; - g->context = NULL; - g->mainloop = pa_threaded_mainloop_new (); - if (!g->mainloop) { - goto fail; + QTAILQ_FOREACH(c, &pa_conns, list) { + if (server == NULL || c->server == NULL ? + server == c->server : + strcmp(server, c->server) == 0) { + g->conn = c; + break; + } } - - g->context = pa_context_new (pa_threaded_mainloop_get_api (g->mainloop), - server); - if (!g->context) { - goto fail; + if (!g->conn) { + g->conn = qpa_conn_init(server); } - - pa_context_set_state_callback (g->context, context_state_cb, g); - - if (pa_context_connect(g->context, server, 0, NULL) < 0) { - qpa_logerr (pa_context_errno (g->context), - "pa_context_connect() failed\n"); - goto fail; + if (!g->conn) { + g_free(g); + return NULL; } - pa_threaded_mainloop_lock (g->mainloop); + ++g->conn->refcount; + return g; +} - if (pa_threaded_mainloop_start (g->mainloop) < 0) { - goto unlock_and_fail; +static void qpa_conn_fini(PAConnection *c) +{ + if (c->mainloop) { + pa_threaded_mainloop_stop(c->mainloop); } - 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); + if (c->context) { + pa_context_disconnect(c->context); + pa_context_unref(c->context); } - pa_threaded_mainloop_unlock (g->mainloop); - - return g; + if (c->mainloop) { + pa_threaded_mainloop_free(c->mainloop); + } -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; + QTAILQ_REMOVE(&pa_conns, c, list); + g_free(c); } static void qpa_audio_fini (void *opaque) { paaudio *g = opaque; + PAConnection *c = g->conn; - 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); + if (--c->refcount == 0) { + qpa_conn_fini(c); } g_free(g); @@ -945,13 +1006,11 @@ 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 }; diff --git a/audio/rate_template.h b/audio/rate_template.h index 6e93588877..f94c940c61 100644 --- a/audio/rate_template.h +++ b/audio/rate_template.h @@ -28,7 +28,7 @@ * Return number of samples processed. */ void NAME (void *opaque, struct st_sample *ibuf, struct st_sample *obuf, - int *isamp, int *osamp) + size_t *isamp, size_t *osamp) { struct rate *rate = opaque; struct st_sample *istart, *iend; diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c index e7179ff1d4..14b11f0335 100644 --- a/audio/sdlaudio.c +++ b/audio/sdlaudio.c @@ -41,8 +41,8 @@ typedef struct SDLVoiceOut { HWVoiceOut hw; - int live; - int decr; + size_t live; + size_t decr; } SDLVoiceOut; static struct SDLAudioState { @@ -184,22 +184,22 @@ static void sdl_callback (void *opaque, Uint8 *buf, int len) SDLVoiceOut *sdl = opaque; SDLAudioState *s = &glob_sdl; HWVoiceOut *hw = &sdl->hw; - int samples = len >> hw->info.shift; - int to_mix, decr; + size_t samples = len >> hw->info.shift; + size_t to_mix, decr; if (s->exit || !sdl->live) { return; } - /* dolog ("in callback samples=%d live=%d\n", samples, sdl->live); */ + /* dolog ("in callback samples=%zu live=%zu\n", samples, sdl->live); */ - to_mix = audio_MIN(samples, sdl->live); + to_mix = MIN(samples, sdl->live); decr = to_mix; while (to_mix) { - int chunk = audio_MIN(to_mix, hw->samples - hw->rpos); + size_t chunk = MIN(to_mix, hw->samples - hw->rpos); struct st_sample *src = hw->mix_buf + hw->rpos; - /* dolog ("in callback to_mix %d, chunk %d\n", to_mix, chunk); */ + /* dolog ("in callback to_mix %zu, chunk %zu\n", to_mix, chunk); */ hw->clip(buf, src, chunk); hw->rpos = (hw->rpos + chunk) % hw->samples; to_mix -= chunk; @@ -209,7 +209,7 @@ static void sdl_callback (void *opaque, Uint8 *buf, int len) sdl->live -= decr; sdl->decr += decr; - /* dolog ("done len=%d\n", len); */ + /* dolog ("done len=%zu\n", len); */ /* SDL2 does not clear the remaining buffer for us, so do it on our own */ if (samples) { @@ -217,14 +217,9 @@ static void sdl_callback (void *opaque, Uint8 *buf, int len) } } -static int sdl_write_out (SWVoiceOut *sw, void *buf, int len) +static size_t sdl_run_out(HWVoiceOut *hw, size_t live) { - return audio_pcm_sw_write (sw, buf, len); -} - -static int sdl_run_out (HWVoiceOut *hw, int live) -{ - int decr; + size_t decr; SDLVoiceOut *sdl = (SDLVoiceOut *) hw; SDL_LockAudio(); @@ -236,7 +231,7 @@ static int sdl_run_out (HWVoiceOut *hw, int live) sdl->live); } - decr = audio_MIN (sdl->decr, live); + decr = MIN (sdl->decr, live); sdl->decr -= decr; sdl->live = live; @@ -342,7 +337,6 @@ static struct audio_pcm_ops sdl_pcm_ops = { .init_out = sdl_init_out, .fini_out = sdl_fini_out, .run_out = sdl_run_out, - .write = sdl_write_out, .ctl_out = sdl_ctl_out, }; diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c index ec1c8fe936..26873c7f22 100644 --- a/audio/spiceaudio.c +++ b/audio/spiceaudio.c @@ -152,31 +152,31 @@ static void line_out_fini (HWVoiceOut *hw) spice_server_remove_interface (&out->sin.base); } -static int line_out_run (HWVoiceOut *hw, int live) +static size_t line_out_run (HWVoiceOut *hw, size_t live) { SpiceVoiceOut *out = container_of (hw, SpiceVoiceOut, hw); - int rpos, decr; - int samples; + size_t rpos, decr; + size_t samples; if (!live) { return 0; } decr = rate_get_samples (&hw->info, &out->rate); - decr = audio_MIN (live, decr); + decr = MIN (live, decr); samples = decr; rpos = hw->rpos; while (samples) { int left_till_end_samples = hw->samples - rpos; - int len = audio_MIN (samples, left_till_end_samples); + int len = MIN (samples, left_till_end_samples); if (!out->frame) { spice_server_playback_get_buffer (&out->sin, &out->frame, &out->fsize); out->fpos = out->frame; } if (out->frame) { - len = audio_MIN (len, out->fsize); + len = MIN (len, out->fsize); hw->clip (out->fpos, hw->mix_buf + rpos, len); out->fsize -= len; out->fpos += len; @@ -192,11 +192,6 @@ static int line_out_run (HWVoiceOut *hw, int live) return decr; } -static int line_out_write (SWVoiceOut *sw, void *buf, int len) -{ - return audio_pcm_sw_write (sw, buf, len); -} - static int line_out_ctl (HWVoiceOut *hw, int cmd, ...) { SpiceVoiceOut *out = container_of (hw, SpiceVoiceOut, hw); @@ -280,12 +275,12 @@ static void line_in_fini (HWVoiceIn *hw) spice_server_remove_interface (&in->sin.base); } -static int line_in_run (HWVoiceIn *hw) +static size_t line_in_run(HWVoiceIn *hw) { SpiceVoiceIn *in = container_of (hw, SpiceVoiceIn, hw); - int num_samples; + size_t num_samples; int ready; - int len[2]; + size_t len[2]; uint64_t delta_samp; const uint32_t *samples; @@ -294,7 +289,7 @@ static int line_in_run (HWVoiceIn *hw) } delta_samp = rate_get_samples (&hw->info, &in->rate); - num_samples = audio_MIN (num_samples, delta_samp); + num_samples = MIN (num_samples, delta_samp); ready = spice_server_record_get_samples (&in->sin, in->samples, num_samples); samples = in->samples; @@ -304,7 +299,7 @@ static int line_in_run (HWVoiceIn *hw) ready = LINE_IN_SAMPLES; } - num_samples = audio_MIN (ready, num_samples); + num_samples = MIN (ready, num_samples); if (hw->wpos + num_samples > hw->samples) { len[0] = hw->samples - hw->wpos; @@ -325,11 +320,6 @@ static int line_in_run (HWVoiceIn *hw) return num_samples; } -static int line_in_read (SWVoiceIn *sw, void *buf, int size) -{ - return audio_pcm_sw_read (sw, buf, size); -} - static int line_in_ctl (HWVoiceIn *hw, int cmd, ...) { SpiceVoiceIn *in = container_of (hw, SpiceVoiceIn, hw); @@ -377,13 +367,11 @@ static struct audio_pcm_ops audio_callbacks = { .init_out = line_out_init, .fini_out = line_out_fini, .run_out = line_out_run, - .write = line_out_write, .ctl_out = line_out_ctl, .init_in = line_in_init, .fini_in = line_in_fini, .run_in = line_in_run, - .read = line_in_read, .ctl_in = line_in_ctl, }; diff --git a/audio/wavaudio.c b/audio/wavaudio.c index 803b6cb1f3..b6eeeb4e26 100644 --- a/audio/wavaudio.c +++ b/audio/wavaudio.c @@ -40,10 +40,10 @@ typedef struct WAVVoiceOut { int total_samples; } WAVVoiceOut; -static int wav_run_out (HWVoiceOut *hw, int live) +static size_t wav_run_out(HWVoiceOut *hw, size_t live) { WAVVoiceOut *wav = (WAVVoiceOut *) hw; - int rpos, decr, samples; + size_t rpos, decr, samples; uint8_t *dst; struct st_sample *src; int64_t now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL); @@ -59,12 +59,12 @@ static int wav_run_out (HWVoiceOut *hw, int live) } wav->old_ticks = now; - decr = audio_MIN (live, samples); + decr = MIN (live, samples); samples = decr; rpos = hw->rpos; while (samples) { int left_till_end_samples = hw->samples - rpos; - int convert_samples = audio_MIN (samples, left_till_end_samples); + int convert_samples = MIN (samples, left_till_end_samples); src = hw->mix_buf + rpos; dst = advance (wav->pcm_buf, rpos << hw->info.shift); @@ -84,11 +84,6 @@ static int wav_run_out (HWVoiceOut *hw, int live) return decr; } -static int wav_write_out (SWVoiceOut *sw, void *buf, int len) -{ - return audio_pcm_sw_write (sw, buf, len); -} - /* VICE code: Store number as little endian. */ static void le_store (uint8_t *buf, uint32_t val, int len) { @@ -144,8 +139,8 @@ static int wav_init_out(HWVoiceOut *hw, struct audsettings *as, hw->samples = 1024; wav->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift); if (!wav->pcm_buf) { - dolog ("Could not allocate buffer (%d bytes)\n", - hw->samples << hw->info.shift); + dolog("Could not allocate buffer (%zu bytes)\n", + hw->samples << hw->info.shift); return -1; } @@ -240,7 +235,6 @@ static struct audio_pcm_ops wav_pcm_ops = { .init_out = wav_init_out, .fini_out = wav_fini_out, .run_out = wav_run_out, - .write = wav_write_out, .ctl_out = wav_ctl_out, }; diff --git a/audio/wavcapture.c b/audio/wavcapture.c index 493edc60e4..8d7ce2eda1 100644 --- a/audio/wavcapture.c +++ b/audio/wavcapture.c @@ -104,8 +104,8 @@ static struct capture_ops wav_capture_ops = { .info = wav_capture_info }; -int wav_start_capture (CaptureState *s, const char *path, int freq, - int bits, int nchannels) +int wav_start_capture(AudioState *state, CaptureState *s, const char *path, + int freq, int bits, int nchannels) { WAVState *wav; uint8_t hdr[] = { @@ -170,7 +170,7 @@ int wav_start_capture (CaptureState *s, const char *path, int freq, goto error_free; } - cap = AUD_add_capture (&as, &ops, wav); + cap = AUD_add_capture(state, &as, &ops, wav); if (!cap) { error_report("Failed to add audio capture"); goto error_free; |