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[Qemu-devel] [PATCH 12/25] audio: unify input and output mixeng buffer m
From: |
Kővágó, Zoltán |
Subject: |
[Qemu-devel] [PATCH 12/25] audio: unify input and output mixeng buffer management |
Date: |
Sun, 25 Aug 2019 20:46:14 +0200 |
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 <address@hidden>
---
audio/audio_int.h | 12 +++--
audio/audio_template.h | 19 +++----
audio/audio.c | 120 +++++++++++++++++++++--------------------
audio/ossaudio.c | 3 +-
4 files changed, 79 insertions(+), 75 deletions(-)
diff --git a/audio/audio_int.h b/audio/audio_int.h
index c76d7c39e8..20021df9e8 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -52,6 +52,11 @@ struct audio_pcm_info {
typedef struct AudioState AudioState;
typedef struct SWVoiceCap SWVoiceCap;
+typedef struct STSampleBuffer {
+ size_t pos, size;
+ st_sample samples[];
+} STSampleBuffer;
+
typedef struct HWVoiceOut {
AudioState *s;
int enabled;
@@ -60,11 +65,9 @@ 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;
@@ -84,11 +87,10 @@ 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 ff4a173f18..87c6d2d271 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 d1c9ce855f..e571eba6a1 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -544,8 +544,8 @@ 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;
@@ -554,17 +554,17 @@ 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;
}
@@ -579,17 +579,17 @@ 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;
}
}
@@ -599,11 +599,11 @@ 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;
}
@@ -616,11 +616,11 @@ 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) {
@@ -630,7 +630,7 @@ 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;
@@ -678,8 +678,8 @@ 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;
@@ -699,11 +699,11 @@ 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;
}
@@ -714,7 +714,7 @@ 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;
@@ -740,7 +740,7 @@ 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
);
@@ -868,7 +868,7 @@ 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)
@@ -969,8 +969,9 @@ 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;
}
@@ -993,12 +994,13 @@ 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",
@@ -1023,12 +1025,12 @@ 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 "
@@ -1037,14 +1039,14 @@ 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)
@@ -1062,7 +1064,7 @@ 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;
@@ -1086,8 +1088,8 @@ 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;
}
@@ -1118,13 +1120,13 @@ 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
@@ -1181,6 +1183,7 @@ 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;
@@ -1194,10 +1197,10 @@ 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;
@@ -1217,9 +1220,10 @@ 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;
@@ -1250,14 +1254,14 @@ 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);
@@ -1265,10 +1269,10 @@ 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) {
@@ -1316,7 +1320,7 @@ 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;
@@ -1352,7 +1356,7 @@ 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;
@@ -1759,11 +1763,11 @@ CaptureVoiceOut *AUD_add_capture(
/* XXX find a more elegant way */
hw->samples = 4096 * 4;
- hw->mix_buf = g_new0(struct st_sample, hw->samples);
+ audio_pcm_hw_alloc_resources_out(hw);
audio_pcm_init_info (&hw->info, as);
- cap->buf = g_malloc0_n(hw->samples, 1 << hw->info.shift);
+ cap->buf = g_malloc0_n(hw->mix_buf->size, 1 << hw->info.shift);
hw->clip = mixeng_clip
[hw->info.nchannels == 2]
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index 2782512706..76c082d5e2 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -585,7 +585,8 @@ 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 (
--
2.22.0
- [Qemu-devel] [PATCH 08/25] sdlaudio: port to the new audio backend api, (continued)
- [Qemu-devel] [PATCH 08/25] sdlaudio: port to the new audio backend api, Kővágó, Zoltán, 2019/08/25
- [Qemu-devel] [PATCH 10/25] wavaudio: port to the new audio backend api, Kővágó, Zoltán, 2019/08/25
- [Qemu-devel] [PATCH 07/25] paaudio: port to the new audio backend api, Kővágó, Zoltán, 2019/08/25
- [Qemu-devel] [PATCH 11/25] audio: remove remains of the old backend api, Kővágó, Zoltán, 2019/08/25
- [Qemu-devel] [PATCH 13/25] audio: remove hw->samples, buffer_size_in/out pcm_ops, Kővágó, Zoltán, 2019/08/25
- [Qemu-devel] [PATCH 16/25] audio: add mixeng option (documentation), Kővágó, Zoltán, 2019/08/25
- [Qemu-devel] [PATCH 17/25] audio: make mixeng optional, Kővágó, Zoltán, 2019/08/25
- [Qemu-devel] [PATCH 12/25] audio: unify input and output mixeng buffer management,
Kővágó, Zoltán <=
- [Qemu-devel] [PATCH 14/25] audio: common rate control code for timer based outputs, Kővágó, Zoltán, 2019/08/25
- [Qemu-devel] [PATCH 15/25] audio: split ctl_* functions into enable_* and volume_*, Kővágó, Zoltán, 2019/08/25
- [Qemu-devel] [PATCH 19/25] audio: support more than two channels in volume setting, Kővágó, Zoltán, 2019/08/25
- [Qemu-devel] [PATCH 21/25] audio: basic support for multichannel audio, Kővágó, Zoltán, 2019/08/25
- [Qemu-devel] [PATCH 23/25] usb-audio: do not count on avail bytes actually available, Kővágó, Zoltán, 2019/08/25
- [Qemu-devel] [PATCH 20/25] audio: replace shift in audio_pcm_info with bytes_per_frame, Kővágó, Zoltán, 2019/08/25
- [Qemu-devel] [PATCH 25/25] usbaudio: change playback counters to 64 bit, Kővágó, Zoltán, 2019/08/25
- [Qemu-devel] [PATCH 22/25] paaudio: channel-map option, Kővágó, Zoltán, 2019/08/25