From: "Kővágó, Zoltán" <dirty.ice.hu@gmail.com>
To: qemu-devel@nongnu.org
Cc: Gerd Hoffmann <kraxel@redhat.com>
Subject: [Qemu-devel] [PATCH v3 37/50] audio: unify input and output mixeng buffer management
Date: Thu, 17 Jan 2019 00:37:10 +0100 [thread overview]
Message-ID: <4d2457aead94211b435d153732e987cc75b0eabd.1547681517.git.DirtY.iCE.hu@gmail.com> (raw)
In-Reply-To: <cover.1547681517.git.DirtY.iCE.hu@gmail.com>
Usage notes: hw->samples became hw->{mix,conv}_buf->size, except before
initialization (audio_pcm_hw_alloc_resources_*), hw->samples gives the
initial size of the STSampleBuffer. The next commit tries to fix this
inconsistency.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio_int.h | 12 ++--
audio/audio_template.h | 19 +++---
audio/audio.c | 130 ++++++++++++++++++++---------------------
audio/ossaudio.c | 2 +-
4 files changed, 80 insertions(+), 83 deletions(-)
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 1f6ec15e18..97d159f28e 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -51,6 +51,11 @@ struct audio_pcm_info {
typedef struct SWVoiceCap SWVoiceCap;
+typedef struct STSampleBuffer {
+ size_t pos, size;
+ st_sample samples[];
+} STSampleBuffer;
+
typedef struct HWVoiceOut {
AudioState *s;
int enabled;
@@ -59,11 +64,9 @@ typedef struct HWVoiceOut {
struct audio_pcm_info info;
f_sample *clip;
-
- size_t rpos;
uint64_t ts_helper;
- struct st_sample *mix_buf;
+ STSampleBuffer *mix_buf;
void *buf_emul;
size_t pos_emul, pending_emul, size_emul;
@@ -83,11 +86,10 @@ typedef struct HWVoiceIn {
t_sample *conv;
- size_t wpos;
size_t total_samples_captured;
uint64_t ts_helper;
- struct st_sample *conv_buf;
+ STSampleBuffer *conv_buf;
void *buf_emul;
size_t pos_emul, pending_emul, size_emul;
diff --git a/audio/audio_template.h b/audio/audio_template.h
index fcab583cfc..83ffc62183 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -76,16 +76,15 @@ static void glue (audio_pcm_hw_free_resources_, TYPE) (HW *hw)
HWBUF = NULL;
}
-static bool glue(audio_pcm_hw_alloc_resources_, TYPE)(HW *hw)
+static void 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 (%zu samples)\n",
- hw->samples);
- return false;
+ size_t samples = hw->samples;
+ if (audio_bug(__func__, samples == 0)) {
+ dolog("Attempted to allocate empty buffer\n");
}
- return true;
+ HWBUF = g_malloc0(sizeof(STSampleBuffer) + sizeof(st_sample) * samples);
+ HWBUF->size = samples;
}
static void glue (audio_pcm_sw_free_resources_, TYPE) (SW *sw)
@@ -104,7 +103,7 @@ static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw)
{
int samples;
- samples = ((int64_t) sw->hw->samples << 32) / sw->ratio;
+ samples = ((int64_t) sw->HWBUF->size << 32) / sw->ratio;
sw->buf = audio_calloc(__func__, samples, sizeof(struct st_sample));
if (!sw->buf) {
@@ -280,9 +279,7 @@ static HW *glue(audio_pcm_hw_add_new_, TYPE)(AudioState *s,
[hw->info.swap_endianness]
[audio_bits_to_index (hw->info.bits)];
- if (!glue(audio_pcm_hw_alloc_resources_, TYPE)(hw)) {
- goto err1;
- }
+ glue(audio_pcm_hw_alloc_resources_, TYPE)(hw);
QLIST_INSERT_HEAD (&s->glue (hw_head_, TYPE), hw, entries);
glue (s->nb_hw_voices_, TYPE) -= 1;
diff --git a/audio/audio.c b/audio/audio.c
index 8bfc122e60..b5dbf5228b 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -545,8 +545,8 @@ static size_t audio_pcm_hw_find_min_in (HWVoiceIn *hw)
static size_t audio_pcm_hw_get_live_in(HWVoiceIn *hw)
{
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);
+ if (audio_bug(__func__, live > hw->conv_buf->size)) {
+ dolog("live=%zu hw->conv_buf->size=%zu\n", live, hw->conv_buf->size);
return 0;
}
return live;
@@ -555,17 +555,17 @@ static size_t audio_pcm_hw_get_live_in(HWVoiceIn *hw)
static void audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf, size_t len)
{
size_t clipped = 0;
- size_t pos = hw->rpos;
+ size_t pos = hw->mix_buf->pos;
while (len) {
- st_sample *src = hw->mix_buf + pos;
+ st_sample *src = hw->mix_buf->samples + pos;
uint8_t *dst = advance (pcm_buf, clipped << hw->info.shift);
- size_t samples_till_end_of_buf = hw->samples - pos;
+ size_t samples_till_end_of_buf = hw->mix_buf->size - pos;
size_t samples_to_clip = MIN(len, samples_till_end_of_buf);
hw->clip (dst, src, samples_to_clip);
- pos = (pos + samples_to_clip) % hw->samples;
+ pos = (pos + samples_to_clip) % hw->mix_buf->size;
len -= samples_to_clip;
clipped += samples_to_clip;
}
@@ -580,17 +580,17 @@ static size_t audio_pcm_sw_get_rpos_in(SWVoiceIn *sw)
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=%zu hw->samples=%zu\n", live, hw->samples);
+ if (audio_bug(__func__, live < 0 || live > hw->conv_buf->size)) {
+ dolog("live=%zu hw->conv_buf->size=%zu\n", live, hw->conv_buf->size);
return 0;
}
- rpos = hw->wpos - live;
+ rpos = hw->conv_buf->pos - live;
if (rpos >= 0) {
return rpos;
}
else {
- return hw->samples + rpos;
+ return hw->conv_buf->size + rpos;
}
}
@@ -600,11 +600,11 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
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;
+ rpos = audio_pcm_sw_get_rpos_in(sw) % hw->conv_buf->size;
live = hw->total_samples_captured - sw->total_hw_samples_acquired;
- if (audio_bug(__func__, live > hw->samples)) {
- dolog("live_in=%zu hw->samples=%zu\n", live, hw->samples);
+ if (audio_bug(__func__, live > hw->conv_buf->size)) {
+ dolog("live_in=%zu hw->conv_buf->size=%zu\n", live, hw->conv_buf->size);
return 0;
}
@@ -617,11 +617,11 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
swlim = MIN (swlim, samples);
while (swlim) {
- src = hw->conv_buf + rpos;
- if (hw->wpos > rpos) {
- isamp = hw->wpos - rpos;
+ src = hw->conv_buf->samples + rpos;
+ if (hw->conv_buf->pos > rpos) {
+ isamp = hw->conv_buf->pos - rpos;
} else {
- isamp = hw->samples - rpos;
+ isamp = hw->conv_buf->size - rpos;
}
if (!isamp) {
@@ -631,7 +631,7 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
st_rate_flow (sw->rate, src, dst, &isamp, &osamp);
swlim -= osamp;
- rpos = (rpos + isamp) % hw->samples;
+ rpos = (rpos + isamp) % hw->conv_buf->size;
dst += osamp;
ret += osamp;
total += isamp;
@@ -679,8 +679,8 @@ static size_t audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live)
if (nb_live1) {
size_t live = smin;
- if (audio_bug(__func__, live > hw->samples)) {
- dolog("live=%zu hw->samples=%zu\n", live, hw->samples);
+ if (audio_bug(__func__, live > hw->mix_buf->size)) {
+ dolog("live=%zu hw->mix_buf->size=%zu\n", live, hw->mix_buf->size);
return 0;
}
return live;
@@ -700,11 +700,11 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
return size;
}
- hwsamples = sw->hw->samples;
+ hwsamples = sw->hw->mix_buf->size;
live = sw->total_hw_samples_mixed;
if (audio_bug(__func__, live > hwsamples)) {
- dolog("live=%zu hw->samples=%zu\n", live, hwsamples);
+ dolog("live=%zu hw->mix_buf->size=%zu\n", live, hwsamples);
return 0;
}
@@ -715,7 +715,7 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
return 0;
}
- wpos = (sw->hw->rpos + live) % hwsamples;
+ wpos = (sw->hw->mix_buf->pos + live) % hwsamples;
samples = size >> sw->info.shift;
dead = hwsamples - live;
@@ -741,7 +741,7 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
st_rate_flow_mix (
sw->rate,
sw->buf + pos,
- sw->hw->mix_buf + wpos,
+ sw->hw->mix_buf->samples + wpos,
&isamp,
&osamp
);
@@ -869,7 +869,7 @@ size_t AUD_read(SWVoiceIn *sw, void *buf, size_t size)
int AUD_get_buffer_size_out (SWVoiceOut *sw)
{
- return sw->hw->samples << sw->hw->info.shift;
+ return sw->hw->mix_buf->size << sw->hw->info.shift;
}
void AUD_set_active_out (SWVoiceOut *sw, int on)
@@ -970,8 +970,9 @@ static size_t audio_get_avail (SWVoiceIn *sw)
}
live = sw->hw->total_samples_captured - sw->total_hw_samples_acquired;
- if (audio_bug(__func__, live > sw->hw->samples)) {
- dolog("live=%zu sw->hw->samples=%zu\n", live, sw->hw->samples);
+ if (audio_bug(__func__, live > sw->hw->conv_buf->size)) {
+ dolog("live=%zu sw->hw->conv_buf->size=%zu\n", live,
+ sw->hw->conv_buf->size);
return 0;
}
@@ -994,12 +995,13 @@ static size_t audio_get_free(SWVoiceOut *sw)
live = sw->total_hw_samples_mixed;
- if (audio_bug(__func__, live > sw->hw->samples)) {
- dolog("live=%zu sw->hw->samples=%zu\n", live, sw->hw->samples);
+ if (audio_bug(__func__, live > sw->hw->mix_buf->size)) {
+ dolog("live=%zu sw->hw->mix_buf->size=%zu\n", live,
+ sw->hw->mix_buf->size);
return 0;
}
- dead = sw->hw->samples - live;
+ dead = sw->hw->mix_buf->size - live;
#ifdef DEBUG_OUT
dolog ("%s: get_free live %d dead %d ret %" PRId64 "\n",
@@ -1024,12 +1026,12 @@ static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos,
n = samples;
while (n) {
- size_t till_end_of_hw = hw->samples - rpos2;
+ size_t till_end_of_hw = hw->mix_buf->size - 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;
+ sw->buf = hw->mix_buf->samples + rpos2;
written = audio_pcm_sw_write (sw, NULL, bytes);
if (written - bytes) {
dolog("Could not mix %zu bytes into a capture "
@@ -1038,14 +1040,14 @@ static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos,
break;
}
n -= to_write;
- rpos2 = (rpos2 + to_write) % hw->samples;
+ rpos2 = (rpos2 + to_write) % hw->mix_buf->size;
}
}
}
- n = MIN(samples, hw->samples - rpos);
- mixeng_clear(hw->mix_buf + rpos, n);
- mixeng_clear(hw->mix_buf, samples - n);
+ n = MIN(samples, hw->mix_buf->size - rpos);
+ mixeng_clear(hw->mix_buf->samples + rpos, n);
+ mixeng_clear(hw->mix_buf->samples, samples - n);
}
static size_t audio_pcm_hw_run_out(HWVoiceOut *hw, size_t live)
@@ -1063,7 +1065,7 @@ static size_t audio_pcm_hw_run_out(HWVoiceOut *hw, size_t live)
live -= proc;
clipped += proc;
- hw->rpos = (hw->rpos + proc) % hw->samples;
+ hw->mix_buf->pos = (hw->mix_buf->pos + proc) % hw->mix_buf->size;
if (proc == 0 || proc < decr) {
break;
@@ -1087,8 +1089,8 @@ static void audio_run_out (AudioState *s)
live = 0;
}
- if (audio_bug(__func__, live > hw->samples)) {
- dolog ("live=%zu hw->samples=%zu\n", live, hw->samples);
+ if (audio_bug(__func__, live > hw->mix_buf->size)) {
+ dolog ("live=%zu hw->mix_buf->size=%zu\n", live, hw->mix_buf->size);
continue;
}
@@ -1119,13 +1121,13 @@ static void audio_run_out (AudioState *s)
continue;
}
- prev_rpos = hw->rpos;
+ prev_rpos = hw->mix_buf->pos;
played = audio_pcm_hw_run_out(hw, live);
replay_audio_out(&played);
- if (audio_bug(__func__, hw->rpos >= hw->samples)) {
- dolog("hw->rpos=%zu hw->samples=%zu played=%zu\n",
- hw->rpos, hw->samples, played);
- hw->rpos = 0;
+ if (audio_bug(__func__, hw->mix_buf->pos >= hw->mix_buf->size)) {
+ dolog("hw->mix_buf->pos=%zu hw->mix_buf->size=%zu played=%zu\n",
+ hw->mix_buf->pos, hw->mix_buf->size, played);
+ hw->mix_buf->pos = 0;
}
#ifdef DEBUG_OUT
@@ -1182,6 +1184,7 @@ static void audio_run_out (AudioState *s)
static size_t audio_pcm_hw_run_in(HWVoiceIn *hw, size_t samples)
{
size_t conv = 0;
+ STSampleBuffer *conv_buf = hw->conv_buf;
while (samples) {
size_t proc;
@@ -1195,10 +1198,10 @@ static size_t audio_pcm_hw_run_in(HWVoiceIn *hw, size_t samples)
}
proc = MIN(size >> hw->info.shift,
- hw->samples - hw->wpos);
+ conv_buf->size - conv_buf->pos);
- hw->conv(hw->conv_buf + hw->wpos, buf, proc);
- hw->wpos = (hw->wpos + proc) % hw->samples;
+ hw->conv(conv_buf->samples + conv_buf->pos, buf, proc);
+ conv_buf->pos = (conv_buf->pos + proc) % conv_buf->size;
samples -= proc;
conv += proc;
@@ -1218,9 +1221,10 @@ static void audio_run_in (AudioState *s)
if (replay_mode != REPLAY_MODE_PLAY) {
captured = audio_pcm_hw_run_in(
- hw, hw->samples - audio_pcm_hw_get_live_in(hw));
+ hw, hw->conv_buf->size - audio_pcm_hw_get_live_in(hw));
}
- replay_audio_in(&captured, hw->conv_buf, &hw->wpos, hw->samples);
+ replay_audio_in(&captured, hw->conv_buf->samples, &hw->conv_buf->pos,
+ hw->conv_buf->size);
min = audio_pcm_hw_find_min_in (hw);
hw->total_samples_captured += captured - min;
@@ -1251,14 +1255,14 @@ static void audio_run_capture (AudioState *s)
SWVoiceOut *sw;
captured = live = audio_pcm_hw_get_live_out (hw, NULL);
- rpos = hw->rpos;
+ rpos = hw->mix_buf->pos;
while (live) {
- size_t left = hw->samples - rpos;
+ size_t left = hw->mix_buf->size - rpos;
size_t to_capture = MIN(live, left);
struct st_sample *src;
struct capture_callback *cb;
- src = hw->mix_buf + rpos;
+ src = hw->mix_buf->samples + rpos;
hw->clip (cap->buf, src, to_capture);
mixeng_clear (src, to_capture);
@@ -1266,10 +1270,10 @@ static void audio_run_capture (AudioState *s)
cb->ops.capture (cb->opaque, cap->buf,
to_capture << hw->info.shift);
}
- rpos = (rpos + to_capture) % hw->samples;
+ rpos = (rpos + to_capture) % hw->mix_buf->size;
live -= to_capture;
}
- hw->rpos = rpos;
+ hw->mix_buf->pos = rpos;
for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
if (!sw->active && sw->empty) {
@@ -1317,7 +1321,7 @@ void *audio_generic_get_buffer_in(HWVoiceIn *hw, size_t *size)
ssize_t start;
if (unlikely(!hw->buf_emul)) {
- size_t calc_size = hw->samples << hw->info.shift;
+ size_t calc_size = hw->conv_buf->size << hw->info.shift;
hw->buf_emul = g_malloc(calc_size);
hw->size_emul = calc_size;
hw->pos_emul = hw->pending_emul = 0;
@@ -1353,7 +1357,7 @@ void audio_generic_put_buffer_in(HWVoiceIn *hw, void *buf, size_t size)
void *audio_generic_get_buffer_out(HWVoiceOut *hw, size_t *size)
{
if (unlikely(!hw->buf_emul)) {
- size_t calc_size = hw->samples << hw->info.shift;
+ size_t calc_size = hw->mix_buf->size << hw->info.shift;
hw->buf_emul = g_malloc(calc_size);
hw->size_emul = calc_size;
@@ -1761,21 +1765,16 @@ CaptureVoiceOut *AUD_add_capture(
/* XXX find a more elegant way */
hw->samples = 4096 * 4;
- hw->mix_buf = audio_calloc(__func__, hw->samples,
- sizeof(struct st_sample));
- if (!hw->mix_buf) {
- dolog("Could not allocate capture mix buffer (%zu samples)\n",
- hw->samples);
- goto err2;
- }
+ audio_pcm_hw_alloc_resources_out(hw);
audio_pcm_init_info (&hw->info, as);
- cap->buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
+ cap->buf = audio_calloc(__func__, hw->mix_buf->size,
+ 1 << hw->info.shift);
if (!cap->buf) {
dolog ("Could not allocate capture buffer "
"(%zu samples, each %d bytes)\n",
- hw->samples, 1 << hw->info.shift);
+ hw->mix_buf->size, 1 << hw->info.shift);
goto err3;
}
@@ -1795,7 +1794,6 @@ CaptureVoiceOut *AUD_add_capture(
err3:
g_free (cap->hw.mix_buf);
- err2:
g_free (cap);
err1:
g_free (cb);
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index 774a41fbf9..957d14eb8e 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -584,7 +584,7 @@ static int oss_ctl_out (HWVoiceOut *hw, int cmd, ...)
return 0;
}
- audio_pcm_info_clear_buf(&hw->info, hw->buf_emul, hw->samples);
+ audio_pcm_info_clear_buf(&hw->info, hw->buf_emul, hw->mix_buf->size);
trig = PCM_ENABLE_OUTPUT;
if (ioctl (oss->fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
oss_logerr (
--
2.20.1
next prev parent reply other threads:[~2019-01-16 23:38 UTC|newest]
Thread overview: 64+ messages / expand[flat|nested] mbox.gz Atom feed top
2019-01-16 23:36 [Qemu-devel] [PATCH v3 00/50] Audio 5.1 patches Kővágó, Zoltán
2019-01-16 23:36 ` [Qemu-devel] [PATCH v3 01/50] qapi: qapi for audio backends Kővágó, Zoltán
2019-01-17 8:54 ` Gerd Hoffmann
2019-01-16 23:36 ` [Qemu-devel] [PATCH v3 02/50] audio: use qapi AudioFormat instead of audfmt_e Kővágó, Zoltán
2019-01-16 23:36 ` [Qemu-devel] [PATCH v3 03/50] audio: -audiodev command line option: documentation Kővágó, Zoltán
2019-01-16 23:36 ` [Qemu-devel] [PATCH v3 04/50] audio: -audiodev command line option basic implementation Kővágó, Zoltán
2019-01-16 23:36 ` [Qemu-devel] [PATCH v3 05/50] alsaaudio: port to -audiodev config Kővágó, Zoltán
2019-01-16 23:36 ` [Qemu-devel] [PATCH v3 06/50] coreaudio: " Kővágó, Zoltán
2019-01-16 23:36 ` [Qemu-devel] [PATCH v3 07/50] dsoundaudio: " Kővágó, Zoltán
2019-01-16 23:36 ` [Qemu-devel] [PATCH v3 08/50] noaudio: " Kővágó, Zoltán
2019-01-16 23:36 ` [Qemu-devel] [PATCH v3 09/50] ossaudio: " Kővágó, Zoltán
2019-01-16 23:36 ` [Qemu-devel] [PATCH v3 10/50] paaudio: " Kővágó, Zoltán
2019-01-16 23:36 ` [Qemu-devel] [PATCH v3 11/50] sdlaudio: " Kővágó, Zoltán
2019-01-16 23:36 ` [Qemu-devel] [PATCH v3 12/50] spiceaudio: " Kővágó, Zoltán
2019-01-16 23:36 ` [Qemu-devel] [PATCH v3 13/50] wavaudio: " Kővágó, Zoltán
2019-01-16 23:36 ` [Qemu-devel] [PATCH v3 14/50] audio: -audiodev command line option: cleanup Kővágó, Zoltán
2019-01-16 23:36 ` [Qemu-devel] [PATCH v3 15/50] audio: reduce glob_audio_state usage Kővágó, Zoltán
2019-01-17 9:22 ` Gerd Hoffmann
2019-01-23 20:16 ` Zoltán Kővágó
2019-01-24 7:42 ` Gerd Hoffmann
2019-01-24 11:19 ` Gerd Hoffmann
2019-01-24 20:12 ` Zoltán Kővágó
2019-01-25 6:57 ` Gerd Hoffmann
2019-01-16 23:36 ` [Qemu-devel] [PATCH v3 16/50] audio: basic support for multi backend audio Kővágó, Zoltán
2019-01-16 23:36 ` [Qemu-devel] [PATCH v3 17/50] audio: add audiodev properties to frontends Kővágó, Zoltán
2019-01-16 23:36 ` [Qemu-devel] [PATCH v3 18/50] audio: audiodev= parameters no longer optional when -audiodev present Kővágó, Zoltán
2019-01-17 9:42 ` Gerd Hoffmann
2019-01-17 9:46 ` Gerd Hoffmann
2019-01-16 23:36 ` [Qemu-devel] [PATCH v3 19/50] paaudio: do not move stream when sink/source name is specified Kővágó, Zoltán
2019-01-16 23:36 ` [Qemu-devel] [PATCH v3 20/50] paaudio: properly disconnect streams in fini_* Kővágó, Zoltán
2019-01-17 5:53 ` Marc-André Lureau
2019-01-16 23:36 ` [Qemu-devel] [PATCH v3 21/50] audio: remove audio_MIN, audio_MAX Kővágó, Zoltán
2019-01-16 23:36 ` [Qemu-devel] [PATCH v3 22/50] audio: do not run each backend in audio_run Kővágó, Zoltán
2019-01-16 23:36 ` [Qemu-devel] [PATCH v3 23/50] paaudio: fix playback glitches Kővágó, Zoltán
2019-01-16 23:36 ` [Qemu-devel] [PATCH v3 24/50] audio: remove read and write pcm_ops Kővágó, Zoltán
2019-01-16 23:36 ` [Qemu-devel] [PATCH v3 25/50] audio: use size_t where makes sense Kővágó, Zoltán
2019-01-16 23:36 ` [Qemu-devel] [PATCH v3 26/50] audio: api for mixeng code free backends Kővágó, Zoltán
2019-01-17 9:52 ` Gerd Hoffmann
2019-01-16 23:37 ` [Qemu-devel] [PATCH v3 27/50] alsaaudio: port to the new audio backend api Kővágó, Zoltán
2019-01-16 23:37 ` [Qemu-devel] [PATCH v3 28/50] coreaudio: " Kővágó, Zoltán
2019-01-16 23:37 ` [Qemu-devel] [PATCH v3 29/50] dsoundaudio: " Kővágó, Zoltán
2019-01-16 23:37 ` [Qemu-devel] [PATCH v3 30/50] noaudio: " Kővágó, Zoltán
2019-01-16 23:37 ` [Qemu-devel] [PATCH v3 31/50] ossaudio: " Kővágó, Zoltán
2019-01-16 23:37 ` [Qemu-devel] [PATCH v3 32/50] paaudio: " Kővágó, Zoltán
2019-01-16 23:37 ` [Qemu-devel] [PATCH v3 33/50] sdlaudio: " Kővágó, Zoltán
2019-01-16 23:37 ` [Qemu-devel] [PATCH v3 34/50] spiceaudio: " Kővágó, Zoltán
2019-01-16 23:37 ` [Qemu-devel] [PATCH v3 35/50] wavaudio: " Kővágó, Zoltán
2019-01-16 23:37 ` [Qemu-devel] [PATCH v3 36/50] audio: remove remains of the old " Kővágó, Zoltán
2019-01-16 23:37 ` Kővágó, Zoltán [this message]
2019-01-16 23:37 ` [Qemu-devel] [PATCH v3 38/50] audio: remove hw->samples, buffer_size_in/out pcm_ops Kővágó, Zoltán
2019-01-16 23:37 ` [Qemu-devel] [PATCH v3 39/50] audio: common rate control code for timer based outputs Kővágó, Zoltán
2019-01-16 23:37 ` [Qemu-devel] [PATCH v3 40/50] audio: split ctl_* functions into enable_* and volume_* Kővágó, Zoltán
2019-01-16 23:37 ` [Qemu-devel] [PATCH v3 41/50] audio: add mixeng option (documentation) Kővágó, Zoltán
2019-01-16 23:37 ` [Qemu-devel] [PATCH v3 42/50] audio: make mixeng optional Kővágó, Zoltán
2019-01-16 23:37 ` [Qemu-devel] [PATCH v3 43/50] paaudio: get/put_buffer functions Kővágó, Zoltán
2019-01-16 23:37 ` [Qemu-devel] [PATCH v3 44/50] audio: support more than two channels in volume setting Kővágó, Zoltán
2019-01-16 23:37 ` [Qemu-devel] [PATCH v3 45/50] audio: replace shift in audio_pcm_info with bytes_per_frame Kővágó, Zoltán
2019-01-16 23:37 ` [Qemu-devel] [PATCH v3 46/50] audio: basic support for multichannel audio Kővágó, Zoltán
2019-01-16 23:37 ` [Qemu-devel] [PATCH v3 47/50] paaudio: channel-map option Kővágó, Zoltán
2019-01-17 10:03 ` Gerd Hoffmann
2019-01-23 20:13 ` Zoltán Kővágó
2019-01-23 20:33 ` Eric Blake
2019-01-16 23:37 ` [Qemu-devel] [PATCH v3 48/50] usb-audio: do not count on avail bytes actually available Kővágó, Zoltán
2019-01-16 23:37 ` [Qemu-devel] [PATCH v3 50/50] usbaudio: change playback counters to 64 bit Kővágó, Zoltán
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