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

audio: replace shift in audio_pcm_info with bytes_per_frame

Browse Source 
The bit shifting trick worked because the number of bytes per frame was
always a power-of-two (since QEMU only supports mono, stereo and 8, 16
and 32 bit samples).  But if we want to add support for surround sound,
this no longer holds true.

Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 1351fd9bcce0ff20d81850c5292722194329de02.1570996490.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
This commit is contained in:
Kővágó, Zoltán 2019-10-13 21:58:02 +02:00 committed by Gerd Hoffmann
parent cecc1e79bf
commit 2b9cce8c8c
10 changed files with 66 additions and 65 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

11
audio/alsaaudio.c
View File

@ -602,7 +602,7 @@ static size_t alsa_write(HWVoiceOut *hw, void *buf, size_t len)
{
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
size_t pos = 0;
size_t len_frames = len >> hw->info.shift;
size_t len_frames = len / hw->info.bytes_per_frame;
while (len_frames) {
char *src = advance(buf, pos);
@ -648,7 +648,7 @@ static size_t alsa_write(HWVoiceOut *hw, void *buf, size_t len)
}
}
pos += written << hw->info.shift;
pos += written * hw->info.bytes_per_frame;
if (written < len_frames) {
break;
}
@ -802,7 +802,8 @@ static size_t alsa_read(HWVoiceIn *hw, void *buf, size_t len)
void *dst = advance(buf, pos);
snd_pcm_sframes_t nread;
nread = snd_pcm_readi(alsa->handle, dst, len >> hw->info.shift);
nread = snd_pcm_readi(
alsa->handle, dst, len / hw->info.bytes_per_frame);
if (nread <= 0) {
switch (nread) {
@ -828,8 +829,8 @@ static size_t alsa_read(HWVoiceIn *hw, void *buf, size_t len)
}
}
pos += nread << hw->info.shift;
len -= nread << hw->info.shift;
pos += nread * hw->info.bytes_per_frame;
len -= nread * hw->info.bytes_per_frame;
}
return pos;

74
audio/audio.c
View File

@ -299,12 +299,13 @@ static int audio_pcm_info_eq (struct audio_pcm_info *info, struct audsettings *a
void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
{
int bits = 8, sign = 0, shift = 0;
int bits = 8, sign = 0, mul;
switch (as->fmt) {
case AUDIO_FORMAT_S8:
sign = 1;
case AUDIO_FORMAT_U8:
mul = 1;
break;
case AUDIO_FORMAT_S16:
@ -312,7 +313,7 @@ void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
/* fall through */
case AUDIO_FORMAT_U16:
bits = 16;
shift = 1;
mul = 2;
break;
case AUDIO_FORMAT_S32:
@ -320,7 +321,7 @@ void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
/* fall through */
case AUDIO_FORMAT_U32:
bits = 32;
shift = 2;
mul = 4;
break;
default:
@ -331,9 +332,8 @@ void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
info->bits = bits;
info->sign = sign;
info->nchannels = as->nchannels;
info->shift = (as->nchannels == 2) + shift;
info->align = (1 << info->shift) - 1;
info->bytes_per_second = info->freq << info->shift;
info->bytes_per_frame = as->nchannels * mul;
info->bytes_per_second = info->freq * info->bytes_per_frame;
info->swap_endianness = (as->endianness != AUDIO_HOST_ENDIANNESS);
}
@ -344,26 +344,25 @@ void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len)
}
if (info->sign) {
memset (buf, 0x00, len << info->shift);
memset(buf, 0x00, len * info->bytes_per_frame);
}
else {
switch (info->bits) {
case 8:
memset (buf, 0x80, len << info->shift);
memset(buf, 0x80, len * info->bytes_per_frame);
break;
case 16:
{
int i;
uint16_t *p = buf;
int shift = info->nchannels - 1;
short s = INT16_MAX;
if (info->swap_endianness) {
s = bswap16 (s);
}
for (i = 0; i < len << shift; i++) {
for (i = 0; i < len * info->nchannels; i++) {
p[i] = s;
}
}
@ -373,14 +372,13 @@ void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len)
{
int i;
uint32_t *p = buf;
int shift = info->nchannels - 1;
int32_t s = INT32_MAX;
if (info->swap_endianness) {
s = bswap32 (s);
}
for (i = 0; i < len << shift; i++) {
for (i = 0; i < len * info->nchannels; i++) {
p[i] = s;
}
}
@ -558,7 +556,7 @@ static void audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf, size_t len)
while (len) {
st_sample *src = hw->mix_buf->samples + pos;
uint8_t *dst = advance(pcm_buf, clipped << hw->info.shift);
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_to_clip = MIN(len, samples_till_end_of_buf);
@ -607,7 +605,7 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
return 0;
}
samples = size >> sw->info.shift;
samples = size / sw->info.bytes_per_frame;
if (!live) {
return 0;
}
@ -642,7 +640,7 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
sw->clip (buf, sw->buf, ret);
sw->total_hw_samples_acquired += total;
return ret << sw->info.shift;
return ret * sw->info.bytes_per_frame;
}
/*
@ -715,7 +713,7 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
}
wpos = (sw->hw->mix_buf->pos + live) % hwsamples;
samples = size >> sw->info.shift;
samples = size / sw->info.bytes_per_frame;
dead = hwsamples - live;
swlim = ((int64_t) dead << 32) / sw->ratio;
@ -759,13 +757,13 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
dolog (
"%s: write size %zu ret %zu total sw %zu\n",
SW_NAME (sw),
size >> sw->info.shift,
size / sw->info.bytes_per_frame,
ret,
sw->total_hw_samples_mixed
);
#endif
return ret << sw->info.shift;
return ret * sw->info.bytes_per_frame;
}
#ifdef DEBUG_AUDIO
@ -882,7 +880,7 @@ size_t AUD_read(SWVoiceIn *sw, void *buf, size_t size)
int AUD_get_buffer_size_out (SWVoiceOut *sw)
{
return sw->hw->mix_buf->size << sw->hw->info.shift;
return sw->hw->mix_buf->size * sw->hw->info.bytes_per_frame;
}
void AUD_set_active_out (SWVoiceOut *sw, int on)
@ -998,10 +996,10 @@ static size_t audio_get_avail (SWVoiceIn *sw)
ldebug (
"%s: get_avail live %d ret %" PRId64 "\n",
SW_NAME (sw),
live, (((int64_t) live << 32) / sw->ratio) << sw->info.shift
live, (((int64_t) live << 32) / sw->ratio) * sw->info.bytes_per_frame
);
return (((int64_t) live << 32) / sw->ratio) << sw->info.shift;
return (((int64_t) live << 32) / sw->ratio) * sw->info.bytes_per_frame;
}
static size_t audio_get_free(SWVoiceOut *sw)
@ -1025,10 +1023,11 @@ static size_t audio_get_free(SWVoiceOut *sw)
#ifdef DEBUG_OUT
dolog ("%s: get_free live %d dead %d ret %" PRId64 "\n",
SW_NAME (sw),
live, dead, (((int64_t) dead << 32) / sw->ratio) << sw->info.shift);
live, dead, (((int64_t) dead << 32) / sw->ratio) *
sw->info.bytes_per_frame);
#endif
return (((int64_t) dead << 32) / sw->ratio) << sw->info.shift;
return (((int64_t) dead << 32) / sw->ratio) * sw->info.bytes_per_frame;
}
static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos,
@ -1047,7 +1046,7 @@ static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos,
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.shift;
size_t bytes = to_write * hw->info.bytes_per_frame;
size_t written;
sw->buf = hw->mix_buf->samples + rpos2;
@ -1082,10 +1081,11 @@ static size_t audio_pcm_hw_run_out(HWVoiceOut *hw, size_t live)
return clipped + live;
}
decr = MIN(size >> hw->info.shift, live);
decr = MIN(size / hw->info.bytes_per_frame, live);
audio_pcm_hw_clip_out(hw, buf, decr);
proc = hw->pcm_ops->put_buffer_out(hw, buf, decr << hw->info.shift) >>
hw->info.shift;
proc = hw->pcm_ops->put_buffer_out(hw, buf,
decr * hw->info.bytes_per_frame) /
hw->info.bytes_per_frame;
live -= proc;
clipped += proc;
@ -1234,16 +1234,16 @@ static size_t audio_pcm_hw_run_in(HWVoiceIn *hw, size_t samples)
while (samples) {
size_t proc;
size_t size = samples << hw->info.shift;
size_t size = samples * hw->info.bytes_per_frame;
void *buf = hw->pcm_ops->get_buffer_in(hw, &size);
assert((size & hw->info.align) == 0);
assert(size % hw->info.bytes_per_frame == 0);
if (size == 0) {
hw->pcm_ops->put_buffer_in(hw, buf, size);
break;
}
proc = MIN(size >> hw->info.shift,
proc = MIN(size / hw->info.bytes_per_frame,
conv_buf->size - conv_buf->pos);
hw->conv(conv_buf->samples + conv_buf->pos, buf, proc);
@ -1251,7 +1251,7 @@ static size_t audio_pcm_hw_run_in(HWVoiceIn *hw, size_t samples)
samples -= proc;
conv += proc;
hw->pcm_ops->put_buffer_in(hw, buf, proc << hw->info.shift);
hw->pcm_ops->put_buffer_in(hw, buf, proc * hw->info.bytes_per_frame);
}
return conv;
@ -1325,7 +1325,7 @@ static void audio_run_capture (AudioState *s)
for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
cb->ops.capture (cb->opaque, cap->buf,
to_capture << hw->info.shift);
to_capture * hw->info.bytes_per_frame);
}
rpos = (rpos + to_capture) % hw->mix_buf->size;
live -= to_capture;
@ -1378,7 +1378,7 @@ void *audio_generic_get_buffer_in(HWVoiceIn *hw, size_t *size)
ssize_t start;
if (unlikely(!hw->buf_emul)) {
size_t calc_size = hw->conv_buf->size << hw->info.shift;
size_t calc_size = hw->conv_buf->size * hw->info.bytes_per_frame;
hw->buf_emul = g_malloc(calc_size);
hw->size_emul = calc_size;
hw->pos_emul = hw->pending_emul = 0;
@ -1414,7 +1414,7 @@ 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->mix_buf->size << hw->info.shift;
size_t calc_size = hw->mix_buf->size * hw->info.bytes_per_frame;
hw->buf_emul = g_malloc(calc_size);
hw->size_emul = calc_size;
@ -1833,7 +1833,7 @@ CaptureVoiceOut *AUD_add_capture(
audio_pcm_init_info (&hw->info, as);
cap->buf = g_malloc0_n(hw->mix_buf->size, 1 << hw->info.shift);
cap->buf = g_malloc0_n(hw->mix_buf->size, hw->info.bytes_per_frame);
hw->clip = mixeng_clip
[hw->info.nchannels == 2]
@ -2153,14 +2153,14 @@ size_t audio_rate_get_bytes(struct audio_pcm_info *info, RateCtl *rate,
now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
ticks = now - rate->start_ticks;
bytes = muldiv64(ticks, info->bytes_per_second, NANOSECONDS_PER_SECOND);
samples = (bytes - rate->bytes_sent) >> info->shift;
samples = (bytes - rate->bytes_sent) / info->bytes_per_frame;
if (samples < 0 || samples > 65536) {
AUD_log(NULL, "Resetting rate control (%" PRId64 " samples)\n", samples);
audio_rate_start(rate);
samples = 0;
}
ret = MIN(samples << info->shift, bytes_avail);
ret = MIN(samples * info->bytes_per_frame, bytes_avail);
rate->bytes_sent += ret;
return ret;
}

3
audio/audio_int.h
View File

@ -43,8 +43,7 @@ struct audio_pcm_info {
int sign;
int freq;
int nchannels;
int align;
int shift;
int bytes_per_frame;
int bytes_per_second;
int swap_endianness;
};

4
audio/coreaudio.c
View File

@ -440,7 +440,7 @@ static OSStatus audioDeviceIOProc(
}
frameCount = core->audioDevicePropertyBufferFrameSize;
pending_frames = hw->pending_emul >> hw->info.shift;
pending_frames = hw->pending_emul / hw->info.bytes_per_frame;
/* if there are not enough samples, set signal and return */
if (pending_frames < frameCount) {
@ -449,7 +449,7 @@ static OSStatus audioDeviceIOProc(
return 0;
}
len = frameCount << hw->info.shift;
len = frameCount * hw->info.bytes_per_frame;
while (len) {
size_t write_len;
ssize_t start = ((ssize_t) hw->pos_emul) - hw->pending_emul;

10
audio/dsound_template.h
View File

@ -98,8 +98,8 @@ static int glue (dsound_lock_, TYPE) (
goto fail;
}
if ((p1p && *p1p && (*blen1p & info->align)) ||
(p2p && *p2p && (*blen2p & info->align))) {
if ((p1p && *p1p && (*blen1p % info->bytes_per_frame)) ||
(p2p && *p2p && (*blen2p % info->bytes_per_frame))) {
dolog("DirectSound returned misaligned buffer %ld %ld\n",
*blen1p, *blen2p);
glue(dsound_unlock_, TYPE)(buf, *p1p, p2p ? *p2p : NULL, *blen1p,
@ -247,14 +247,14 @@ static int dsound_init_out(HWVoiceOut *hw, struct audsettings *as,
obt_as.endianness = 0;
audio_pcm_init_info (&hw->info, &obt_as);
if (bc.dwBufferBytes & hw->info.align) {
if (bc.dwBufferBytes % hw->info.bytes_per_frame) {
dolog (
"GetCaps returned misaligned buffer size %ld, alignment %d\n",
bc.dwBufferBytes, hw->info.align + 1
bc.dwBufferBytes, hw->info.bytes_per_frame
);
}
hw->size_emul = bc.dwBufferBytes;
hw->samples = bc.dwBufferBytes >> hw->info.shift;
hw->samples = bc.dwBufferBytes / hw->info.bytes_per_frame;
ds->s = s;
#ifdef DEBUG_DSOUND

4
audio/dsoundaudio.c
View File

@ -320,8 +320,8 @@ static void dsound_clear_sample (HWVoiceOut *hw, LPDIRECTSOUNDBUFFER dsb,
return;
}
len1 = blen1 >> hw->info.shift;
len2 = blen2 >> hw->info.shift;
len1 = blen1 / hw->info.bytes_per_frame;
len2 = blen2 / hw->info.bytes_per_frame;
#ifdef DEBUG_DSOUND
dolog ("clear %p,%ld,%ld %p,%ld,%ld\n",

2
audio/noaudio.c
View File

@ -91,7 +91,7 @@ static size_t no_read(HWVoiceIn *hw, void *buf, size_t size)
NoVoiceIn *no = (NoVoiceIn *) hw;
int64_t bytes = audio_rate_get_bytes(&hw->info, &no->rate, size);
audio_pcm_info_clear_buf(&hw->info, buf, bytes >> hw->info.shift);
audio_pcm_info_clear_buf(&hw->info, buf, bytes / hw->info.bytes_per_frame);
return bytes;
}

14
audio/ossaudio.c
View File

@ -506,16 +506,16 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
oss->nfrags = obt.nfrags;
oss->fragsize = obt.fragsize;
if (obt.nfrags * obt.fragsize & hw->info.align) {
if (obt.nfrags * obt.fragsize % hw->info.bytes_per_frame) {
dolog ("warning: Misaligned DAC buffer, size %d, alignment %d\n",
obt.nfrags * obt.fragsize, hw->info.align + 1);
obt.nfrags * obt.fragsize, hw->info.bytes_per_frame);
}
hw->samples = (obt.nfrags * obt.fragsize) >> hw->info.shift;
hw->samples = (obt.nfrags * obt.fragsize) / hw->info.bytes_per_frame;
oss->mmapped = 0;
if (oopts->has_try_mmap && oopts->try_mmap) {
hw->size_emul = hw->samples << hw->info.shift;
hw->size_emul = hw->samples * hw->info.bytes_per_frame;
hw->buf_emul = mmap(
NULL,
hw->size_emul,
@ -644,12 +644,12 @@ static int oss_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
oss->nfrags = obt.nfrags;
oss->fragsize = obt.fragsize;
if (obt.nfrags * obt.fragsize & hw->info.align) {
if (obt.nfrags * obt.fragsize % hw->info.bytes_per_frame) {
dolog ("warning: Misaligned ADC buffer, size %d, alignment %d\n",
obt.nfrags * obt.fragsize, hw->info.align + 1);
obt.nfrags * obt.fragsize, hw->info.bytes_per_frame);
}
hw->samples = (obt.nfrags * obt.fragsize) >> hw->info.shift;
hw->samples = (obt.nfrags * obt.fragsize) / hw->info.bytes_per_frame;
oss->fd = fd;
oss->dev = dev;

3
audio/spiceaudio.c
View File

@ -131,7 +131,8 @@ static void *line_out_get_buffer(HWVoiceOut *hw, size_t *size)
if (out->frame) {
*size = audio_rate_get_bytes(
&hw->info, &out->rate, (out->fsize - out->fpos) << hw->info.shift);
&hw->info, &out->rate,
(out->fsize - out->fpos) * hw->info.bytes_per_frame);
} else {
audio_rate_start(&out->rate);
}

6
audio/wavaudio.c
View File

@ -43,14 +43,14 @@ static size_t wav_write_out(HWVoiceOut *hw, void *buf, size_t len)
{
WAVVoiceOut *wav = (WAVVoiceOut *) hw;
int64_t bytes = audio_rate_get_bytes(&hw->info, &wav->rate, len);
assert(bytes >> hw->info.shift << hw->info.shift == bytes);
assert(bytes % hw->info.bytes_per_frame == 0);
if (bytes && fwrite(buf, bytes, 1, wav->f) != 1) {
dolog("wav_write_out: fwrite of %" PRId64 " bytes failed\nReason: %s\n",
bytes, strerror(errno));
}
wav->total_samples += bytes >> hw->info.shift;
wav->total_samples += bytes / hw->info.bytes_per_frame;
return bytes;
}
@ -134,7 +134,7 @@ static void wav_fini_out (HWVoiceOut *hw)
WAVVoiceOut *wav = (WAVVoiceOut *) hw;
uint8_t rlen[4];
uint8_t dlen[4];
uint32_t datalen = wav->total_samples << hw->info.shift;
uint32_t datalen = wav->total_samples * hw->info.bytes_per_frame;
uint32_t rifflen = datalen + 36;
if (!wav->f) {