OpenE2K
/
qemu-e2k
13
4
Fork 0
Code Issues 4 Pull Requests Projects 2 Releases Activity

Audio patches for QEMU 8.0 

Cleanups and improvements from Volker Rümelin.
 -----BEGIN PGP SIGNATURE-----
 
 iQJQBAABCAA6FiEEh6m9kz+HxgbSdvYt2ujhCXWWnOUFAmQFjUccHG1hcmNhbmRy
 ZS5sdXJlYXVAcmVkaGF0LmNvbQAKCRDa6OEJdZac5ZLMD/sFWmVYccRqvOfOJvuC
 dJ7EtABmC6b4MxLkMyP5WLMoCezSha0vgcYwAL3oZX1dIBoGfaUXXdxIM8IK8rAv
 J2Fe2p9D+6tqzqc8jkbx0DmY040A3Gb18P60jL3NeQfmqCt7KmzuQoCMn6htPyyy
 NMgSRG3wy+esglmeU2majhxQZh2Vmwsi6i37A5Fg8b27oWLYHxeS6Lgp3PNAwJsA
 k/RKJeZDMjDh0f0wSHZN+8w0VVFQa3BjqoNV4UAgqSjsPaRlnQxppG2FEXNNrtLb
 bJ4rbXgSO68wrcweHgC2L7qWhLnYwdcdQFAPotmt2V1jtH4xx6f+ovwCdf0vZ5Fo
 YhHixpP41gFvF7sJWSZO0wUFKXTyLSdWIPaUKP9iMTZfXpBxo5H6rkPsZ6v5yyG2
 tLFlL0Rv/h3liEkngmRbw81gB9te92ZGXOjkB2Q69tGQl/tXRkhrn5VT2mufb7BV
 WUsdtSFl6P231lDuzQkFp85PkoRtr8eDPkkZW8/xgRx5h8j5I2bc6LRMd63w2tE6
 TGdUfiaqH396r1T87ynBdBWSc4H6YuImTouy27rAv/50x8w4cgk66wY+UGB7D8FX
 Eg4rU5Dzco2D+1RY50zSqZgPxIWjnC4xtYQXLaJURe0Way1HQqt/Pvp/aVFKUOef
 snYV1FLDvHg2Cf9LCGqvpW8b/w==
 =d4LY
 -----END PGP SIGNATURE-----

Merge tag 'audio-pull-request' of https://gitlab.com/marcandre.lureau/qemu into staging

Audio patches for QEMU 8.0

Cleanups and improvements from Volker Rümelin.

# -----BEGIN PGP SIGNATURE-----
#
# iQJQBAABCAA6FiEEh6m9kz+HxgbSdvYt2ujhCXWWnOUFAmQFjUccHG1hcmNhbmRy
# ZS5sdXJlYXVAcmVkaGF0LmNvbQAKCRDa6OEJdZac5ZLMD/sFWmVYccRqvOfOJvuC
# dJ7EtABmC6b4MxLkMyP5WLMoCezSha0vgcYwAL3oZX1dIBoGfaUXXdxIM8IK8rAv
# J2Fe2p9D+6tqzqc8jkbx0DmY040A3Gb18P60jL3NeQfmqCt7KmzuQoCMn6htPyyy
# NMgSRG3wy+esglmeU2majhxQZh2Vmwsi6i37A5Fg8b27oWLYHxeS6Lgp3PNAwJsA
# k/RKJeZDMjDh0f0wSHZN+8w0VVFQa3BjqoNV4UAgqSjsPaRlnQxppG2FEXNNrtLb
# bJ4rbXgSO68wrcweHgC2L7qWhLnYwdcdQFAPotmt2V1jtH4xx6f+ovwCdf0vZ5Fo
# YhHixpP41gFvF7sJWSZO0wUFKXTyLSdWIPaUKP9iMTZfXpBxo5H6rkPsZ6v5yyG2
# tLFlL0Rv/h3liEkngmRbw81gB9te92ZGXOjkB2Q69tGQl/tXRkhrn5VT2mufb7BV
# WUsdtSFl6P231lDuzQkFp85PkoRtr8eDPkkZW8/xgRx5h8j5I2bc6LRMd63w2tE6
# TGdUfiaqH396r1T87ynBdBWSc4H6YuImTouy27rAv/50x8w4cgk66wY+UGB7D8FX
# Eg4rU5Dzco2D+1RY50zSqZgPxIWjnC4xtYQXLaJURe0Way1HQqt/Pvp/aVFKUOef
# snYV1FLDvHg2Cf9LCGqvpW8b/w==
# =d4LY
# -----END PGP SIGNATURE-----
# gpg: Signature made Mon 06 Mar 2023 06:50:47 GMT
# gpg:                using RSA key 87A9BD933F87C606D276F62DDAE8E10975969CE5
# gpg:                issuer "marcandre.lureau@redhat.com"
# gpg: Good signature from "Marc-André Lureau <marcandre.lureau@redhat.com>" [full]
# gpg:                 aka "Marc-André Lureau <marcandre.lureau@gmail.com>" [full]
# Primary key fingerprint: 87A9 BD93 3F87 C606 D276  F62D DAE8 E109 7596 9CE5

* tag 'audio-pull-request' of https://gitlab.com/marcandre.lureau/qemu: (27 commits)
  audio: remove sw->ratio
  audio/audio_template: substitute sw->hw with hw
  audio: handle leftover audio frame from upsampling
  audio: make recording packet length calculation exact
  audio: rename variables in audio_pcm_sw_read()
  audio: replace the resampling loop in audio_pcm_sw_read()
  audio: make playback packet length calculation exact
  audio: remove unused noop_conv() function
  audio: don't misuse audio_pcm_sw_write()
  audio: rename variables in audio_pcm_sw_write()
  audio: remove sw == NULL check
  audio: replace the resampling loop in audio_pcm_sw_write()
  audio: make the resampling code greedy
  audio: change type and name of the resample buffer
  audio: change type of mix_buf and conv_buf
  alsaaudio: reintroduce default recording settings
  alsaaudio: change default playback settings
  audio: remove audio_calloc() function
  audio/audio_template: use g_new0() to replace audio_calloc()
  audio/audio_template: use g_malloc0() to replace audio_calloc()
  ...

Signed-off-by: Peter Maydell <peter.maydell@linaro.org>
This commit is contained in:
Peter Maydell 2023-03-06 14:06:06 +00:00
parent f003dd8d81 2f886a34bb
commit 817fd33836
8 changed files with 360 additions and 298 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

2
MAINTAINERS
View File

@ -2490,6 +2490,7 @@ Subsystems
----------
Overall Audio backends
M: Gerd Hoffmann <kraxel@redhat.com>
M: Marc-André Lureau <marcandre.lureau@redhat.com>
S: Odd Fixes
F: audio/
X: audio/alsaaudio.c
@ -2785,6 +2786,7 @@ F: docs/spice-port-fqdn.txt
Graphics
M: Gerd Hoffmann <kraxel@redhat.com>
M: Marc-André Lureau <marcandre.lureau@redhat.com>
S: Odd Fixes
F: ui/
F: include/ui/

27
audio/alsaaudio.c
View File

@ -222,11 +222,7 @@ static int alsa_poll_helper (snd_pcm_t *handle, struct pollhlp *hlp, int mask)
return -1;
}
pfds = audio_calloc ("alsa_poll_helper", count, sizeof (*pfds));
if (!pfds) {
dolog ("Could not initialize poll mode\n");
return -1;
}
pfds = g_new0(struct pollfd, count);
err = snd_pcm_poll_descriptors (handle, pfds, count);
if (err < 0) {
@ -917,28 +913,23 @@ static void *alsa_audio_init(Audiodev *dev)
alsa_init_per_direction(aopts->in);
alsa_init_per_direction(aopts->out);
/*
* need to define them, as otherwise alsa produces no sound
* doesn't set has_* so alsa_open can identify it wasn't set by the user
*/
/* don't set has_* so alsa_open can identify it wasn't set by the user */
if (!dev->u.alsa.out->has_period_length) {
/* 1024 frames assuming 44100Hz */
dev->u.alsa.out->period_length = 1024 * 1000000 / 44100;
/* 256 frames assuming 44100Hz */
dev->u.alsa.out->period_length = 5805;
}
if (!dev->u.alsa.out->has_buffer_length) {
/* 4096 frames assuming 44100Hz */
dev->u.alsa.out->buffer_length = 4096ll * 1000000 / 44100;
dev->u.alsa.out->buffer_length = 92880;
}
/*
* OptsVisitor sets unspecified optional fields to zero, but do not depend
* on it...
*/
if (!dev->u.alsa.in->has_period_length) {
dev->u.alsa.in->period_length = 0;
/* 256 frames assuming 44100Hz */
dev->u.alsa.in->period_length = 5805;
}
if (!dev->u.alsa.in->has_buffer_length) {
dev->u.alsa.in->buffer_length = 0;
/* 4096 frames assuming 44100Hz */
dev->u.alsa.in->buffer_length = 92880;
}
return dev;

394
audio/audio.c
View File

@ -33,6 +33,7 @@
#include "qapi/qapi-visit-audio.h"
#include "qapi/qapi-commands-audio.h"
#include "qemu/cutils.h"
#include "qemu/log.h"
#include "qemu/module.h"
#include "qemu/help_option.h"
#include "sysemu/sysemu.h"
@ -148,26 +149,6 @@ static inline int audio_bits_to_index (int bits)
}
}
void *audio_calloc (const char *funcname, int nmemb, size_t size)
{
int cond;
size_t len;
len = nmemb * size;
cond = !nmemb || !size;
cond |= nmemb < 0;
cond |= len < size;
if (audio_bug ("audio_calloc", cond)) {
AUD_log (NULL, "%s passed invalid arguments to audio_calloc\n",
funcname);
AUD_log (NULL, "nmemb=%d size=%zu (len=%zu)\n", nmemb, size, len);
return NULL;
}
return g_malloc0 (len);
}
void AUD_vlog (const char *cap, const char *fmt, va_list ap)
{
if (cap) {
@ -400,13 +381,6 @@ void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len)
/*
* Capture
*/
static void noop_conv (struct st_sample *dst, const void *src, int samples)
{
(void) src;
(void) dst;
(void) samples;
}
static CaptureVoiceOut *audio_pcm_capture_find_specific(AudioState *s,
struct audsettings *as)
{
@ -504,15 +478,8 @@ static int audio_attach_capture (HWVoiceOut *hw)
sw->info = hw->info;
sw->empty = 1;
sw->active = hw->enabled;
sw->conv = noop_conv;
sw->ratio = ((int64_t) hw_cap->info.freq << 32) / sw->info.freq;
sw->vol = nominal_volume;
sw->rate = st_rate_start (sw->info.freq, hw_cap->info.freq);
if (!sw->rate) {
dolog ("Could not start rate conversion for `%s'\n", SW_NAME (sw));
g_free (sw);
return -1;
}
QLIST_INSERT_HEAD (&hw_cap->sw_head, sw, entries);
QLIST_INSERT_HEAD (&hw->cap_head, sc, entries);
#ifdef DEBUG_CAPTURE
@ -547,8 +514,8 @@ 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->conv_buf->size)) {
dolog("live=%zu hw->conv_buf->size=%zu\n", live, hw->conv_buf->size);
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;
@ -557,13 +524,13 @@ static size_t audio_pcm_hw_get_live_in(HWVoiceIn *hw)
static size_t audio_pcm_hw_conv_in(HWVoiceIn *hw, void *pcm_buf, size_t samples)
{
size_t conv = 0;
STSampleBuffer *conv_buf = hw->conv_buf;
STSampleBuffer *conv_buf = &hw->conv_buf;
while (samples) {
uint8_t *src = advance(pcm_buf, conv * hw->info.bytes_per_frame);
size_t proc = MIN(samples, conv_buf->size - conv_buf->pos);
hw->conv(conv_buf->samples + conv_buf->pos, src, proc);
hw->conv(conv_buf->buffer + conv_buf->pos, src, proc);
conv_buf->pos = (conv_buf->pos + proc) % conv_buf->size;
samples -= proc;
conv += proc;
@ -575,56 +542,65 @@ static size_t audio_pcm_hw_conv_in(HWVoiceIn *hw, void *pcm_buf, size_t samples)
/*
* Soft voice (capture)
*/
static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
static void audio_pcm_sw_resample_in(SWVoiceIn *sw,
size_t frames_in_max, size_t frames_out_max,
size_t *total_in, size_t *total_out)
{
HWVoiceIn *hw = sw->hw;
size_t samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0;
struct st_sample *src, *dst = sw->buf;
struct st_sample *src, *dst;
size_t live, rpos, frames_in, frames_out;
live = hw->total_samples_captured - sw->total_hw_samples_acquired;
rpos = audio_ring_posb(hw->conv_buf.pos, live, hw->conv_buf.size);
/* resample conv_buf from rpos to end of buffer */
src = hw->conv_buf.buffer + rpos;
frames_in = MIN(frames_in_max, hw->conv_buf.size - rpos);
dst = sw->resample_buf.buffer;
frames_out = frames_out_max;
st_rate_flow(sw->rate, src, dst, &frames_in, &frames_out);
rpos += frames_in;
*total_in = frames_in;
*total_out = frames_out;
/* resample conv_buf from start of buffer if there are input frames left */
if (frames_in_max - frames_in && rpos == hw->conv_buf.size) {
src = hw->conv_buf.buffer;
frames_in = frames_in_max - frames_in;
dst += frames_out;
frames_out = frames_out_max - frames_out;
st_rate_flow(sw->rate, src, dst, &frames_in, &frames_out);
*total_in += frames_in;
*total_out += frames_out;
}
}
static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t buf_len)
{
HWVoiceIn *hw = sw->hw;
size_t live, frames_out_max, total_in, total_out;
live = hw->total_samples_captured - sw->total_hw_samples_acquired;
if (!live) {
return 0;
}
if (audio_bug(__func__, live > hw->conv_buf->size)) {
dolog("live_in=%zu hw->conv_buf->size=%zu\n", live, hw->conv_buf->size);
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;
}
rpos = audio_ring_posb(hw->conv_buf->pos, live, hw->conv_buf->size);
frames_out_max = MIN(buf_len / sw->info.bytes_per_frame,
sw->resample_buf.size);
samples = size / sw->info.bytes_per_frame;
swlim = (live * sw->ratio) >> 32;
swlim = MIN (swlim, samples);
while (swlim) {
src = hw->conv_buf->samples + rpos;
if (hw->conv_buf->pos > rpos) {
isamp = hw->conv_buf->pos - rpos;
} else {
isamp = hw->conv_buf->size - rpos;
}
if (!isamp) {
break;
}
osamp = swlim;
st_rate_flow (sw->rate, src, dst, &isamp, &osamp);
swlim -= osamp;
rpos = (rpos + isamp) % hw->conv_buf->size;
dst += osamp;
ret += osamp;
total += isamp;
}
audio_pcm_sw_resample_in(sw, live, frames_out_max, &total_in, &total_out);
if (!hw->pcm_ops->volume_in) {
mixeng_volume (sw->buf, ret, &sw->vol);
mixeng_volume(sw->resample_buf.buffer, total_out, &sw->vol);
}
sw->clip(buf, sw->resample_buf.buffer, total_out);
sw->clip (buf, sw->buf, ret);
sw->total_hw_samples_acquired += total;
return ret * sw->info.bytes_per_frame;
sw->total_hw_samples_acquired += total_in;
return total_out * sw->info.bytes_per_frame;
}
/*
@ -660,8 +636,8 @@ 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->mix_buf->size)) {
dolog("live=%zu hw->mix_buf->size=%zu\n", live, hw->mix_buf->size);
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;
@ -678,17 +654,17 @@ static size_t audio_pcm_hw_get_free(HWVoiceOut *hw)
static void audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf, size_t len)
{
size_t clipped = 0;
size_t pos = hw->mix_buf->pos;
size_t pos = hw->mix_buf.pos;
while (len) {
st_sample *src = hw->mix_buf->samples + pos;
st_sample *src = hw->mix_buf.buffer + pos;
uint8_t *dst = advance(pcm_buf, clipped * hw->info.bytes_per_frame);
size_t samples_till_end_of_buf = hw->mix_buf->size - 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->mix_buf->size;
pos = (pos + samples_to_clip) % hw->mix_buf.size;
len -= samples_to_clip;
clipped += samples_to_clip;
}
@ -697,84 +673,113 @@ static void audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf, size_t len)
/*
* Soft voice (playback)
*/
static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
static void audio_pcm_sw_resample_out(SWVoiceOut *sw,
size_t frames_in_max, size_t frames_out_max,
size_t *total_in, size_t *total_out)
{
size_t hwsamples, samples, isamp, osamp, wpos, live, dead, left, blck;
size_t hw_free;
size_t ret = 0, pos = 0, total = 0;
if (!sw) {
return size;
}
hwsamples = sw->hw->mix_buf->size;
HWVoiceOut *hw = sw->hw;
struct st_sample *src, *dst;
size_t live, wpos, frames_in, frames_out;
live = sw->total_hw_samples_mixed;
if (audio_bug(__func__, live > hwsamples)) {
dolog("live=%zu hw->mix_buf->size=%zu\n", live, hwsamples);
wpos = (hw->mix_buf.pos + live) % hw->mix_buf.size;
/* write to mix_buf from wpos to end of buffer */
src = sw->resample_buf.buffer;
frames_in = frames_in_max;
dst = hw->mix_buf.buffer + wpos;
frames_out = MIN(frames_out_max, hw->mix_buf.size - wpos);
st_rate_flow_mix(sw->rate, src, dst, &frames_in, &frames_out);
wpos += frames_out;
*total_in = frames_in;
*total_out = frames_out;
/* write to mix_buf from start of buffer if there are input frames left */
if (frames_in_max - frames_in > 0 && wpos == hw->mix_buf.size) {
src += frames_in;
frames_in = frames_in_max - frames_in;
dst = hw->mix_buf.buffer;
frames_out = frames_out_max - frames_out;
st_rate_flow_mix(sw->rate, src, dst, &frames_in, &frames_out);
*total_in += frames_in;
*total_out += frames_out;
}
}
static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t buf_len)
{
HWVoiceOut *hw = sw->hw;
size_t live, dead, hw_free, sw_max, fe_max;
size_t frames_in_max, frames_out_max, total_in, total_out;
live = sw->total_hw_samples_mixed;
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;
}
if (live == hwsamples) {
if (live == hw->mix_buf.size) {
#ifdef DEBUG_OUT
dolog ("%s is full %zu\n", sw->name, live);
#endif
return 0;
}
wpos = (sw->hw->mix_buf->pos + live) % hwsamples;
dead = hwsamples - live;
hw_free = audio_pcm_hw_get_free(sw->hw);
dead = hw->mix_buf.size - live;
hw_free = audio_pcm_hw_get_free(hw);
hw_free = hw_free > live ? hw_free - live : 0;
samples = ((int64_t)MIN(dead, hw_free) << 32) / sw->ratio;
samples = MIN(samples, size / sw->info.bytes_per_frame);
if (samples) {
sw->conv(sw->buf, buf, samples);
frames_out_max = MIN(dead, hw_free);
sw_max = st_rate_frames_in(sw->rate, frames_out_max);
fe_max = MIN(buf_len / sw->info.bytes_per_frame + sw->resample_buf.pos,
sw->resample_buf.size);
frames_in_max = MIN(sw_max, fe_max);
if (!frames_in_max) {
return 0;
}
if (frames_in_max > sw->resample_buf.pos) {
sw->conv(sw->resample_buf.buffer + sw->resample_buf.pos,
buf, frames_in_max - sw->resample_buf.pos);
if (!sw->hw->pcm_ops->volume_out) {
mixeng_volume(sw->buf, samples, &sw->vol);
mixeng_volume(sw->resample_buf.buffer + sw->resample_buf.pos,
frames_in_max - sw->resample_buf.pos, &sw->vol);
}
}
while (samples) {
dead = hwsamples - live;
left = hwsamples - wpos;
blck = MIN (dead, left);
if (!blck) {
break;
}
isamp = samples;
osamp = blck;
st_rate_flow_mix (
sw->rate,
sw->buf + pos,
sw->hw->mix_buf->samples + wpos,
&isamp,
&osamp
);
ret += isamp;
samples -= isamp;
pos += isamp;
live += osamp;
wpos = (wpos + osamp) % hwsamples;
total += osamp;
}
audio_pcm_sw_resample_out(sw, frames_in_max, frames_out_max,
&total_in, &total_out);
sw->total_hw_samples_mixed += total;
sw->total_hw_samples_mixed += total_out;
sw->empty = sw->total_hw_samples_mixed == 0;
/*
* Upsampling may leave one audio frame in the resample buffer. Decrement
* total_in by one if there was a leftover frame from the previous resample
* pass in the resample buffer. Increment total_in by one if the current
* resample pass left one frame in the resample buffer.
*/
if (frames_in_max - total_in == 1) {
/* copy one leftover audio frame to the beginning of the buffer */
*sw->resample_buf.buffer = *(sw->resample_buf.buffer + total_in);
total_in += 1 - sw->resample_buf.pos;
sw->resample_buf.pos = 1;
} else if (total_in >= sw->resample_buf.pos) {
total_in -= sw->resample_buf.pos;
sw->resample_buf.pos = 0;
}
#ifdef DEBUG_OUT
dolog (
"%s: write size %zu ret %zu total sw %zu\n",
SW_NAME (sw),
size / sw->info.bytes_per_frame,
ret,
"%s: write size %zu written %zu total mixed %zu\n",
SW_NAME(sw),
buf_len / sw->info.bytes_per_frame,
total_in,
sw->total_hw_samples_mixed
);
#endif
return ret * sw->info.bytes_per_frame;
return total_in * sw->info.bytes_per_frame;
}
#ifdef DEBUG_AUDIO
@ -992,18 +997,6 @@ void AUD_set_active_in (SWVoiceIn *sw, int on)
}
}
/**
* audio_frontend_frames_in() - returns the number of frames the resampling
* code generates from frames_in frames
*
* @sw: audio recording frontend
* @frames_in: number of frames
*/
static size_t audio_frontend_frames_in(SWVoiceIn *sw, size_t frames_in)
{
return (int64_t)frames_in * sw->ratio >> 32;
}
static size_t audio_get_avail (SWVoiceIn *sw)
{
size_t live;
@ -1013,33 +1006,21 @@ 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->conv_buf->size)) {
dolog("live=%zu sw->hw->conv_buf->size=%zu\n", live,
sw->hw->conv_buf->size);
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;
}
ldebug (
"%s: get_avail live %zu frontend frames %zu\n",
"%s: get_avail live %zu frontend frames %u\n",
SW_NAME (sw),
live, audio_frontend_frames_in(sw, live)
live, st_rate_frames_out(sw->rate, live)
);
return live;
}
/**
* audio_frontend_frames_out() - returns the number of frames needed to
* get frames_out frames after resampling
*
* @sw: audio playback frontend
* @frames_out: number of frames
*/
static size_t audio_frontend_frames_out(SWVoiceOut *sw, size_t frames_out)
{
return ((int64_t)frames_out << 32) / sw->ratio;
}
static size_t audio_get_free(SWVoiceOut *sw)
{
size_t live, dead;
@ -1050,17 +1031,17 @@ static size_t audio_get_free(SWVoiceOut *sw)
live = sw->total_hw_samples_mixed;
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);
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->mix_buf->size - live;
dead = sw->hw->mix_buf.size - live;
#ifdef DEBUG_OUT
dolog("%s: get_free live %zu dead %zu frontend frames %zu\n",
SW_NAME(sw), live, dead, audio_frontend_frames_out(sw, dead));
dolog("%s: get_free live %zu dead %zu frontend frames %u\n",
SW_NAME(sw), live, dead, st_rate_frames_in(sw->rate, dead));
#endif
return dead;
@ -1076,32 +1057,40 @@ static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos,
for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
SWVoiceOut *sw = &sc->sw;
int rpos2 = rpos;
size_t rpos2 = rpos;
n = samples;
while (n) {
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.bytes_per_frame;
size_t written;
size_t till_end_of_hw = hw->mix_buf.size - rpos2;
size_t to_read = MIN(till_end_of_hw, n);
size_t live, frames_in, frames_out;
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 "
sw->resample_buf.buffer = hw->mix_buf.buffer + rpos2;
sw->resample_buf.size = to_read;
live = sw->total_hw_samples_mixed;
audio_pcm_sw_resample_out(sw,
to_read, sw->hw->mix_buf.size - live,
&frames_in, &frames_out);
sw->total_hw_samples_mixed += frames_out;
sw->empty = sw->total_hw_samples_mixed == 0;
if (to_read - frames_in) {
dolog("Could not mix %zu frames into a capture "
"buffer, mixed %zu\n",
bytes, written);
to_read, frames_in);
break;
}
n -= to_write;
rpos2 = (rpos2 + to_write) % hw->mix_buf->size;
n -= to_read;
rpos2 = (rpos2 + to_read) % hw->mix_buf.size;
}
}
}
n = MIN(samples, hw->mix_buf->size - rpos);
mixeng_clear(hw->mix_buf->samples + rpos, n);
mixeng_clear(hw->mix_buf->samples, samples - n);
n = MIN(samples, hw->mix_buf.size - rpos);
mixeng_clear(hw->mix_buf.buffer + rpos, n);
mixeng_clear(hw->mix_buf.buffer, samples - n);
}
static size_t audio_pcm_hw_run_out(HWVoiceOut *hw, size_t live)
@ -1127,7 +1116,7 @@ static size_t audio_pcm_hw_run_out(HWVoiceOut *hw, size_t live)
live -= proc;
clipped += proc;
hw->mix_buf->pos = (hw->mix_buf->pos + proc) % hw->mix_buf->size;
hw->mix_buf.pos = (hw->mix_buf.pos + proc) % hw->mix_buf.size;
if (proc == 0 || proc < decr) {
break;
@ -1181,12 +1170,14 @@ static void audio_run_out (AudioState *s)
size_t free;
if (hw_free > sw->total_hw_samples_mixed) {
free = audio_frontend_frames_out(sw,
free = st_rate_frames_in(sw->rate,
MIN(sw_free, hw_free - sw->total_hw_samples_mixed));
} else {
free = 0;
}
if (free > 0) {
if (free > sw->resample_buf.pos) {
free = MIN(free, sw->resample_buf.size)
- sw->resample_buf.pos;
sw->callback.fn(sw->callback.opaque,
free * sw->info.bytes_per_frame);
}
@ -1198,8 +1189,8 @@ static void audio_run_out (AudioState *s)
live = 0;
}
if (audio_bug(__func__, live > hw->mix_buf->size)) {
dolog("live=%zu hw->mix_buf->size=%zu\n", live, hw->mix_buf->size);
if (audio_bug(__func__, live > hw->mix_buf.size)) {
dolog("live=%zu hw->mix_buf.size=%zu\n", live, hw->mix_buf.size);
continue;
}
@ -1227,13 +1218,13 @@ static void audio_run_out (AudioState *s)
continue;
}
prev_rpos = hw->mix_buf->pos;
prev_rpos = hw->mix_buf.pos;
played = audio_pcm_hw_run_out(hw, live);
replay_audio_out(&played);
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;
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
@ -1314,10 +1305,10 @@ static void audio_run_in (AudioState *s)
if (replay_mode != REPLAY_MODE_PLAY) {
captured = audio_pcm_hw_run_in(
hw, hw->conv_buf->size - 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->samples, &hw->conv_buf->pos,
hw->conv_buf->size);
replay_audio_in(&captured, hw->conv_buf.buffer, &hw->conv_buf.pos,
hw->conv_buf.size);
min = audio_pcm_hw_find_min_in (hw);
hw->total_samples_captured += captured - min;
@ -1330,8 +1321,9 @@ static void audio_run_in (AudioState *s)
size_t sw_avail = audio_get_avail(sw);
size_t avail;
avail = audio_frontend_frames_in(sw, sw_avail);
avail = st_rate_frames_out(sw->rate, sw_avail);
if (avail > 0) {
avail = MIN(avail, sw->resample_buf.size);
sw->callback.fn(sw->callback.opaque,
avail * sw->info.bytes_per_frame);
}
@ -1350,14 +1342,14 @@ static void audio_run_capture (AudioState *s)
SWVoiceOut *sw;
captured = live = audio_pcm_hw_get_live_out (hw, NULL);
rpos = hw->mix_buf->pos;
rpos = hw->mix_buf.pos;
while (live) {
size_t left = hw->mix_buf->size - 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->samples + rpos;
src = hw->mix_buf.buffer + rpos;
hw->clip (cap->buf, src, to_capture);
mixeng_clear (src, to_capture);
@ -1365,10 +1357,10 @@ static void audio_run_capture (AudioState *s)
cb->ops.capture (cb->opaque, cap->buf,
to_capture * hw->info.bytes_per_frame);
}
rpos = (rpos + to_capture) % hw->mix_buf->size;
rpos = (rpos + to_capture) % hw->mix_buf.size;
live -= to_capture;
}
hw->mix_buf->pos = rpos;
hw->mix_buf.pos = rpos;
for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
if (!sw->active && sw->empty) {
@ -1927,7 +1919,7 @@ CaptureVoiceOut *AUD_add_capture(
audio_pcm_init_info (&hw->info, as);
cap->buf = g_malloc0_n(hw->mix_buf->size, hw->info.bytes_per_frame);
cap->buf = g_malloc0_n(hw->mix_buf.size, hw->info.bytes_per_frame);
if (hw->info.is_float) {
hw->clip = mixeng_clip_float[hw->info.nchannels == 2];
@ -1979,7 +1971,7 @@ void AUD_del_capture (CaptureVoiceOut *cap, void *cb_opaque)
sw = sw1;
}
QLIST_REMOVE (cap, entries);
g_free (cap->hw.mix_buf);
g_free(cap->hw.mix_buf.buffer);
g_free (cap->buf);
g_free (cap);
}

20
audio/audio_int.h
View File

@ -58,7 +58,7 @@ typedef struct SWVoiceCap SWVoiceCap;
typedef struct STSampleBuffer {
size_t pos, size;
st_sample samples[];
st_sample *buffer;
} STSampleBuffer;
typedef struct HWVoiceOut {
@ -71,7 +71,7 @@ typedef struct HWVoiceOut {
f_sample *clip;
uint64_t ts_helper;
STSampleBuffer *mix_buf;
STSampleBuffer mix_buf;
void *buf_emul;
size_t pos_emul, pending_emul, size_emul;
@ -93,7 +93,7 @@ typedef struct HWVoiceIn {
size_t total_samples_captured;
uint64_t ts_helper;
STSampleBuffer *conv_buf;
STSampleBuffer conv_buf;
void *buf_emul;
size_t pos_emul, pending_emul, size_emul;
@ -108,8 +108,7 @@ struct SWVoiceOut {
AudioState *s;
struct audio_pcm_info info;
t_sample *conv;
int64_t ratio;
struct st_sample *buf;
STSampleBuffer resample_buf;
void *rate;
size_t total_hw_samples_mixed;
int active;
@ -126,10 +125,9 @@ struct SWVoiceIn {
AudioState *s;
int active;
struct audio_pcm_info info;
int64_t ratio;
void *rate;
size_t total_hw_samples_acquired;
struct st_sample *buf;
STSampleBuffer resample_buf;
f_sample *clip;
HWVoiceIn *hw;
char *name;
@ -151,8 +149,8 @@ struct audio_driver {
int can_be_default;
int max_voices_out;
int max_voices_in;
int voice_size_out;
int voice_size_in;
size_t voice_size_out;
size_t voice_size_in;
QLIST_ENTRY(audio_driver) next;
};
@ -251,7 +249,6 @@ 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_bug (const char *funcname, int cond);
void *audio_calloc (const char *funcname, int nmemb, size_t size);
void audio_run(AudioState *s, const char *msg);
@ -294,9 +291,6 @@ static inline size_t audio_ring_posb(size_t pos, size_t dist, size_t len)
#define ldebug(fmt, ...) (void)0
#endif
#define AUDIO_STRINGIFY_(n) #n
#define AUDIO_STRINGIFY(n) AUDIO_STRINGIFY_(n)
typedef struct AudiodevListEntry {
Audiodev *dev;
QSIMPLEQ_ENTRY(AudiodevListEntry) next;

105
audio/audio_template.h
View File

@ -40,7 +40,7 @@ static void glue(audio_init_nb_voices_, TYPE)(AudioState *s,
struct audio_driver *drv)
{
int max_voices = glue (drv->max_voices_, TYPE);
int voice_size = glue (drv->voice_size_, TYPE);
size_t voice_size = glue(drv->voice_size_, TYPE);
if (glue (s->nb_hw_voices_, TYPE) > max_voices) {
if (!max_voices) {
@ -63,16 +63,17 @@ static void glue(audio_init_nb_voices_, TYPE)(AudioState *s,
}
if (audio_bug(__func__, voice_size && !max_voices)) {
dolog ("drv=`%s' voice_size=%d max_voices=0\n",
drv->name, voice_size);
dolog("drv=`%s' voice_size=%zu max_voices=0\n",
drv->name, voice_size);
}
}
static void glue (audio_pcm_hw_free_resources_, TYPE) (HW *hw)
{
g_free(hw->buf_emul);
g_free (HWBUF);
HWBUF = NULL;
g_free(HWBUF.buffer);
HWBUF.buffer = NULL;
HWBUF.size = 0;
}
static void glue(audio_pcm_hw_alloc_resources_, TYPE)(HW *hw)
@ -83,56 +84,67 @@ static void glue(audio_pcm_hw_alloc_resources_, TYPE)(HW *hw)
dolog("Attempted to allocate empty buffer\n");
}
HWBUF = g_malloc0(sizeof(STSampleBuffer) + sizeof(st_sample) * samples);
HWBUF->size = samples;
HWBUF.buffer = g_new0(st_sample, samples);
HWBUF.size = samples;
HWBUF.pos = 0;
} else {
HWBUF = NULL;
HWBUF.buffer = NULL;
HWBUF.size = 0;
}
}
static void glue (audio_pcm_sw_free_resources_, TYPE) (SW *sw)
{
g_free (sw->buf);
g_free(sw->resample_buf.buffer);
sw->resample_buf.buffer = NULL;
sw->resample_buf.size = 0;
if (sw->rate) {
st_rate_stop (sw->rate);
}
sw->buf = NULL;
sw->rate = NULL;
}
static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw)
{
int samples;
HW *hw = sw->hw;
uint64_t samples;
if (!glue(audio_get_pdo_, TYPE)(sw->s->dev)->mixing_engine) {
return 0;
}
#ifdef DAC
samples = ((int64_t) sw->HWBUF->size << 32) / sw->ratio;
#else
samples = (int64_t)sw->HWBUF->size * sw->ratio >> 32;
#endif
samples = muldiv64(HWBUF.size, sw->info.freq, hw->info.freq);
if (samples == 0) {
uint64_t f_fe_min;
uint64_t f_be = (uint32_t)hw->info.freq;
sw->buf = audio_calloc(__func__, samples, sizeof(struct st_sample));
if (!sw->buf) {
dolog ("Could not allocate buffer for `%s' (%d samples)\n",
SW_NAME (sw), samples);
/* f_fe_min = ceil(1 [frames] * f_be [Hz] / size_be [frames]) */
f_fe_min = (f_be + HWBUF.size - 1) / HWBUF.size;
qemu_log_mask(LOG_UNIMP,
AUDIO_CAP ": The guest selected a " NAME " sample rate"
" of %d Hz for %s. Only sample rates >= %" PRIu64 " Hz"
" are supported.\n",
sw->info.freq, sw->name, f_fe_min);
return -1;
}
/*
* Allocate one additional audio frame that is needed for upsampling
* if the resample buffer size is small. For large buffer sizes take
* care of overflows and truncation.
*/
samples = samples < SIZE_MAX ? samples + 1 : SIZE_MAX;
sw->resample_buf.buffer = g_new0(st_sample, samples);
sw->resample_buf.size = samples;
sw->resample_buf.pos = 0;
#ifdef DAC
sw->rate = st_rate_start (sw->info.freq, sw->hw->info.freq);
sw->rate = st_rate_start(sw->info.freq, hw->info.freq);
#else
sw->rate = st_rate_start (sw->hw->info.freq, sw->info.freq);
sw->rate = st_rate_start(hw->info.freq, sw->info.freq);
#endif
if (!sw->rate) {
g_free (sw->buf);
sw->buf = NULL;
return -1;
}
return 0;
}
@ -149,11 +161,8 @@ static int glue (audio_pcm_sw_init_, TYPE) (
sw->hw = hw;
sw->active = 0;
#ifdef DAC
sw->ratio = ((int64_t) sw->hw->info.freq << 32) / sw->info.freq;
sw->total_hw_samples_mixed = 0;
sw->empty = 1;
#else
sw->ratio = ((int64_t) sw->info.freq << 32) / sw->hw->info.freq;
#endif
if (sw->info.is_float) {
@ -264,13 +273,11 @@ static HW *glue(audio_pcm_hw_add_new_, TYPE)(AudioState *s,
return NULL;
}
hw = audio_calloc(__func__, 1, glue(drv->voice_size_, TYPE));
if (!hw) {
dolog ("Can not allocate voice `%s' size %d\n",
drv->name, glue (drv->voice_size_, TYPE));
return NULL;
}
/*
* Since glue(s->nb_hw_voices_, TYPE) is != 0, glue(drv->voice_size_, TYPE)
* is guaranteed to be != 0. See the audio_init_nb_voices_* functions.
*/
hw = g_malloc0(glue(drv->voice_size_, TYPE));
hw->s = s;
hw->pcm_ops = drv->pcm_ops;
@ -418,33 +425,28 @@ static SW *glue(audio_pcm_create_voice_pair_, TYPE)(
hw_as = *as;
}
sw = audio_calloc(__func__, 1, sizeof(*sw));
if (!sw) {
dolog ("Could not allocate soft voice `%s' (%zu bytes)\n",
sw_name ? sw_name : "unknown", sizeof (*sw));
goto err1;
}
sw = g_new0(SW, 1);
sw->s = s;
hw = glue(audio_pcm_hw_add_, TYPE)(s, &hw_as);
if (!hw) {
goto err2;
dolog("Could not create a backend for voice `%s'\n", sw_name);
goto err1;
}
glue (audio_pcm_hw_add_sw_, TYPE) (hw, sw);
if (glue (audio_pcm_sw_init_, TYPE) (sw, hw, sw_name, as)) {
goto err3;
goto err2;
}
return sw;
err3:
err2:
glue (audio_pcm_hw_del_sw_, TYPE) (sw);
glue (audio_pcm_hw_gc_, TYPE) (&hw);
err2:
g_free (sw);
err1:
g_free(sw);
return NULL;
}
@ -515,8 +517,8 @@ SW *glue (AUD_open_, TYPE) (
HW *hw = sw->hw;
if (!hw) {
dolog ("Internal logic error voice `%s' has no hardware store\n",
SW_NAME (sw));
dolog("Internal logic error: voice `%s' has no backend\n",
SW_NAME(sw));
goto fail;
}
@ -527,7 +529,6 @@ SW *glue (AUD_open_, TYPE) (
} else {
sw = glue(audio_pcm_create_voice_pair_, TYPE)(s, name, as);
if (!sw) {
dolog ("Failed to create voice `%s'\n", name);
return NULL;
}
}

87
audio/mixeng.c
View File

@ -414,12 +414,7 @@ struct rate {
*/
void *st_rate_start (int inrate, int outrate)
{
struct rate *rate = audio_calloc(__func__, 1, sizeof(*rate));
if (!rate) {
dolog ("Could not allocate resampler (%zu bytes)\n", sizeof (*rate));
return NULL;
}
struct rate *rate = g_new0(struct rate, 1);
rate->opos = 0;
@ -445,6 +440,86 @@ void st_rate_stop (void *opaque)
g_free (opaque);
}
/**
* st_rate_frames_out() - returns the number of frames the resampling code
* generates from frames_in frames
*
* @opaque: pointer to struct rate
* @frames_in: number of frames
*
* When upsampling, there may be more than one correct result. In this case,
* the function returns the maximum number of output frames the resampling
* code can generate.
*/
uint32_t st_rate_frames_out(void *opaque, uint32_t frames_in)
{
struct rate *rate = opaque;
uint64_t opos_end, opos_delta;
uint32_t ipos_end;
uint32_t frames_out;
if (rate->opos_inc == 1ULL << 32) {
return frames_in;
}
/* no output frame without at least one input frame */
if (!frames_in) {
return 0;
}
/* last frame read was at rate->ipos - 1 */
ipos_end = rate->ipos - 1 + frames_in;
opos_end = (uint64_t)ipos_end << 32;
/* last frame written was at rate->opos - rate->opos_inc */
if (opos_end + rate->opos_inc <= rate->opos) {
return 0;
}
opos_delta = opos_end - rate->opos + rate->opos_inc;
frames_out = opos_delta / rate->opos_inc;
return opos_delta % rate->opos_inc ? frames_out : frames_out - 1;
}
/**
* st_rate_frames_in() - returns the number of frames needed to
* get frames_out frames after resampling
*
* @opaque: pointer to struct rate
* @frames_out: number of frames
*
* When downsampling, there may be more than one correct result. In this
* case, the function returns the maximum number of input frames needed.
*/
uint32_t st_rate_frames_in(void *opaque, uint32_t frames_out)
{
struct rate *rate = opaque;
uint64_t opos_start, opos_end;
uint32_t ipos_start, ipos_end;
if (rate->opos_inc == 1ULL << 32) {
return frames_out;
}
if (frames_out) {
opos_start = rate->opos;
ipos_start = rate->ipos;
} else {
uint64_t offset;
/* add offset = ceil(opos_inc) to opos and ipos to avoid an underflow */
offset = (rate->opos_inc + (1ULL << 32) - 1) & ~((1ULL << 32) - 1);
opos_start = rate->opos + offset;
ipos_start = rate->ipos + (offset >> 32);
}
/* last frame written was at opos_start - rate->opos_inc */
opos_end = opos_start - rate->opos_inc + rate->opos_inc * frames_out;
ipos_end = (opos_end >> 32) + 1;
/* last frame read was at ipos_start - 1 */
return ipos_end + 1 > ipos_start ? ipos_end + 1 - ipos_start : 0;
}
void mixeng_clear (struct st_sample *buf, int len)
{
memset (buf, 0, len * sizeof (struct st_sample));

2
audio/mixeng.h
View File

@ -52,6 +52,8 @@ void st_rate_flow(void *opaque, st_sample *ibuf, st_sample *obuf,
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);
uint32_t st_rate_frames_out(void *opaque, uint32_t frames_in);
uint32_t st_rate_frames_in(void *opaque, uint32_t frames_out);
void mixeng_clear (struct st_sample *buf, int len);
void mixeng_volume (struct st_sample *buf, int len, struct mixeng_volume *vol);

21
audio/rate_template.h
View File

@ -40,8 +40,6 @@ void NAME (void *opaque, struct st_sample *ibuf, struct st_sample *obuf,
int64_t t;
#endif
ilast = rate->ilast;
istart = ibuf;
iend = ibuf + *isamp;
@ -59,15 +57,17 @@ void NAME (void *opaque, struct st_sample *ibuf, struct st_sample *obuf,
return;
}
while (obuf < oend) {
/* without input samples, there's nothing to do */
if (ibuf >= iend) {
*osamp = 0;
return;
}
/* Safety catch to make sure we have input samples. */
if (ibuf >= iend) {
break;
}
ilast = rate->ilast;
while (true) {
/* read as many input samples so that ipos > opos */
while (rate->ipos <= (rate->opos >> 32)) {
ilast = *ibuf++;
rate->ipos++;
@ -78,6 +78,11 @@ void NAME (void *opaque, struct st_sample *ibuf, struct st_sample *obuf,
}
}
/* make sure that the next output sample can be written */
if (obuf >= oend) {
break;
}
icur = *ibuf;
/* wrap ipos and opos around long before they overflow */