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audio: change type of mix_buf and conv_buf

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Change the type of mix_buf in struct HWVoiceOut and conv_buf
in struct HWVoiceIn from STSampleBuffer * to STSampleBuffer.
However, a buffer pointer is still needed. For this reason in
struct STSampleBuffer samples[] is changed to *buffer.

This is a preparation for the next patch. The next patch will
add this line, which is not possible with the current struct
STSampleBuffer definition.

+        sw->resample_buf.buffer = hw->mix_buf.buffer + rpos2;

There are no functional changes.

Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20230224190555.7409-1-vr_qemu@t-online.de>
This commit is contained in:
Volker Rümelin 2023-02-24 20:05:41 +01:00 committed by Marc-André Lureau
parent 5140ad8279
commit 8dbd3d1795
3 changed files with 67 additions and 64 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

106
audio/audio.c
View File

@ -523,8 +523,8 @@ 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;
@ -533,13 +533,13 @@ 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;
@ -561,12 +561,12 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
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);
rpos = audio_ring_posb(hw->conv_buf.pos, live, hw->conv_buf.size);
samples = size / sw->info.bytes_per_frame;
@ -574,11 +574,11 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
swlim = MIN (swlim, samples);
while (swlim) {
src = hw->conv_buf->samples + rpos;
if (hw->conv_buf->pos > rpos) {
isamp = hw->conv_buf->pos - rpos;
src = hw->conv_buf.buffer + rpos;
if (hw->conv_buf.pos > rpos) {
isamp = hw->conv_buf.pos - rpos;
} else {
isamp = hw->conv_buf->size - rpos;
isamp = hw->conv_buf.size - rpos;
}
if (!isamp) {
@ -588,7 +588,7 @@ 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->conv_buf->size;
rpos = (rpos + isamp) % hw->conv_buf.size;
dst += osamp;
ret += osamp;
total += isamp;
@ -636,8 +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;
@ -654,17 +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;
}
@ -683,11 +683,11 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
return size;
}
hwsamples = sw->hw->mix_buf->size;
hwsamples = sw->hw->mix_buf.size;
live = sw->total_hw_samples_mixed;
if (audio_bug(__func__, live > hwsamples)) {
dolog("live=%zu hw->mix_buf->size=%zu\n", live, hwsamples);
dolog("live=%zu hw->mix_buf.size=%zu\n", live, hwsamples);
return 0;
}
@ -698,7 +698,7 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
return 0;
}
wpos = (sw->hw->mix_buf->pos + live) % hwsamples;
wpos = (sw->hw->mix_buf.pos + live) % hwsamples;
dead = hwsamples - live;
hw_free = audio_pcm_hw_get_free(sw->hw);
@ -725,7 +725,7 @@ 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->samples + wpos,
sw->hw->mix_buf.buffer + wpos,
&isamp,
&osamp
);
@ -989,9 +989,9 @@ 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->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;
}
@ -1026,13 +1026,13 @@ 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",
@ -1056,12 +1056,12 @@ static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos,
n = samples;
while (n) {
size_t till_end_of_hw = hw->mix_buf->size - 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.bytes_per_frame;
size_t written;
sw->buf = hw->mix_buf->samples + rpos2;
sw->buf = hw->mix_buf.buffer + rpos2;
written = audio_pcm_sw_write (sw, NULL, bytes);
if (written - bytes) {
dolog("Could not mix %zu bytes into a capture "
@ -1070,14 +1070,14 @@ static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos,
break;
}
n -= to_write;
rpos2 = (rpos2 + to_write) % hw->mix_buf->size;
rpos2 = (rpos2 + to_write) % 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)
@ -1103,7 +1103,7 @@ 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;
@ -1174,8 +1174,8 @@ 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;
}
@ -1203,13 +1203,13 @@ 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
@ -1290,10 +1290,10 @@ 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;
@ -1326,14 +1326,14 @@ 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);
@ -1341,10 +1341,10 @@ 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) {
@ -1903,7 +1903,7 @@ 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];
@ -1955,7 +1955,7 @@ 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);
}

6
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;

19
audio/audio_template.h
View File

@ -71,8 +71,9 @@ static void glue(audio_init_nb_voices_, TYPE)(AudioState *s,
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)
@ -83,10 +84,12 @@ 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;
}
}
@ -111,9 +114,9 @@ static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw)
}
#ifdef DAC
samples = ((int64_t) sw->HWBUF->size << 32) / sw->ratio;
samples = ((int64_t)sw->HWBUF.size << 32) / sw->ratio;
#else
samples = (int64_t)sw->HWBUF->size * sw->ratio >> 32;
samples = (int64_t)sw->HWBUF.size * sw->ratio >> 32;
#endif
if (audio_bug(__func__, samples < 0)) {
dolog("Can not allocate buffer for `%s' (%d samples)\n",
@ -126,7 +129,7 @@ static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw)
size_t f_fe_min;
/* f_fe_min = ceil(1 [frames] * f_be [Hz] / size_be [frames]) */
f_fe_min = (hw->info.freq + HWBUF->size - 1) / HWBUF->size;
f_fe_min = (hw->info.freq + 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 >= %zu Hz are"