mirror of https://github.com/xemu-project/xemu.git
audio: replace shift in audio_pcm_info with bytes_per_frame
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:
parent
cecc1e79bf
commit
2b9cce8c8c
|
@ -602,7 +602,7 @@ static size_t alsa_write(HWVoiceOut *hw, void *buf, size_t len)
|
||||||
{
|
{
|
||||||
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
|
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
|
||||||
size_t pos = 0;
|
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) {
|
while (len_frames) {
|
||||||
char *src = advance(buf, pos);
|
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) {
|
if (written < len_frames) {
|
||||||
break;
|
break;
|
||||||
}
|
}
|
||||||
|
@ -802,7 +802,8 @@ static size_t alsa_read(HWVoiceIn *hw, void *buf, size_t len)
|
||||||
void *dst = advance(buf, pos);
|
void *dst = advance(buf, pos);
|
||||||
snd_pcm_sframes_t nread;
|
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) {
|
if (nread <= 0) {
|
||||||
switch (nread) {
|
switch (nread) {
|
||||||
|
@ -828,8 +829,8 @@ static size_t alsa_read(HWVoiceIn *hw, void *buf, size_t len)
|
||||||
}
|
}
|
||||||
}
|
}
|
||||||
|
|
||||||
pos += nread << hw->info.shift;
|
pos += nread * hw->info.bytes_per_frame;
|
||||||
len -= nread << hw->info.shift;
|
len -= nread * hw->info.bytes_per_frame;
|
||||||
}
|
}
|
||||||
|
|
||||||
return pos;
|
return pos;
|
||||||
|
|
|
@ -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)
|
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) {
|
switch (as->fmt) {
|
||||||
case AUDIO_FORMAT_S8:
|
case AUDIO_FORMAT_S8:
|
||||||
sign = 1;
|
sign = 1;
|
||||||
case AUDIO_FORMAT_U8:
|
case AUDIO_FORMAT_U8:
|
||||||
|
mul = 1;
|
||||||
break;
|
break;
|
||||||
|
|
||||||
case AUDIO_FORMAT_S16:
|
case AUDIO_FORMAT_S16:
|
||||||
|
@ -312,7 +313,7 @@ void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
|
||||||
/* fall through */
|
/* fall through */
|
||||||
case AUDIO_FORMAT_U16:
|
case AUDIO_FORMAT_U16:
|
||||||
bits = 16;
|
bits = 16;
|
||||||
shift = 1;
|
mul = 2;
|
||||||
break;
|
break;
|
||||||
|
|
||||||
case AUDIO_FORMAT_S32:
|
case AUDIO_FORMAT_S32:
|
||||||
|
@ -320,7 +321,7 @@ void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
|
||||||
/* fall through */
|
/* fall through */
|
||||||
case AUDIO_FORMAT_U32:
|
case AUDIO_FORMAT_U32:
|
||||||
bits = 32;
|
bits = 32;
|
||||||
shift = 2;
|
mul = 4;
|
||||||
break;
|
break;
|
||||||
|
|
||||||
default:
|
default:
|
||||||
|
@ -331,9 +332,8 @@ void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
|
||||||
info->bits = bits;
|
info->bits = bits;
|
||||||
info->sign = sign;
|
info->sign = sign;
|
||||||
info->nchannels = as->nchannels;
|
info->nchannels = as->nchannels;
|
||||||
info->shift = (as->nchannels == 2) + shift;
|
info->bytes_per_frame = as->nchannels * mul;
|
||||||
info->align = (1 << info->shift) - 1;
|
info->bytes_per_second = info->freq * info->bytes_per_frame;
|
||||||
info->bytes_per_second = info->freq << info->shift;
|
|
||||||
info->swap_endianness = (as->endianness != AUDIO_HOST_ENDIANNESS);
|
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) {
|
if (info->sign) {
|
||||||
memset (buf, 0x00, len << info->shift);
|
memset(buf, 0x00, len * info->bytes_per_frame);
|
||||||
}
|
}
|
||||||
else {
|
else {
|
||||||
switch (info->bits) {
|
switch (info->bits) {
|
||||||
case 8:
|
case 8:
|
||||||
memset (buf, 0x80, len << info->shift);
|
memset(buf, 0x80, len * info->bytes_per_frame);
|
||||||
break;
|
break;
|
||||||
|
|
||||||
case 16:
|
case 16:
|
||||||
{
|
{
|
||||||
int i;
|
int i;
|
||||||
uint16_t *p = buf;
|
uint16_t *p = buf;
|
||||||
int shift = info->nchannels - 1;
|
|
||||||
short s = INT16_MAX;
|
short s = INT16_MAX;
|
||||||
|
|
||||||
if (info->swap_endianness) {
|
if (info->swap_endianness) {
|
||||||
s = bswap16 (s);
|
s = bswap16 (s);
|
||||||
}
|
}
|
||||||
|
|
||||||
for (i = 0; i < len << shift; i++) {
|
for (i = 0; i < len * info->nchannels; i++) {
|
||||||
p[i] = s;
|
p[i] = s;
|
||||||
}
|
}
|
||||||
}
|
}
|
||||||
|
@ -373,14 +372,13 @@ void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len)
|
||||||
{
|
{
|
||||||
int i;
|
int i;
|
||||||
uint32_t *p = buf;
|
uint32_t *p = buf;
|
||||||
int shift = info->nchannels - 1;
|
|
||||||
int32_t s = INT32_MAX;
|
int32_t s = INT32_MAX;
|
||||||
|
|
||||||
if (info->swap_endianness) {
|
if (info->swap_endianness) {
|
||||||
s = bswap32 (s);
|
s = bswap32 (s);
|
||||||
}
|
}
|
||||||
|
|
||||||
for (i = 0; i < len << shift; i++) {
|
for (i = 0; i < len * info->nchannels; i++) {
|
||||||
p[i] = s;
|
p[i] = s;
|
||||||
}
|
}
|
||||||
}
|
}
|
||||||
|
@ -558,7 +556,7 @@ static void audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf, size_t len)
|
||||||
|
|
||||||
while (len) {
|
while (len) {
|
||||||
st_sample *src = hw->mix_buf->samples + pos;
|
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_till_end_of_buf = hw->mix_buf->size - pos;
|
||||||
size_t samples_to_clip = MIN(len, samples_till_end_of_buf);
|
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;
|
return 0;
|
||||||
}
|
}
|
||||||
|
|
||||||
samples = size >> sw->info.shift;
|
samples = size / sw->info.bytes_per_frame;
|
||||||
if (!live) {
|
if (!live) {
|
||||||
return 0;
|
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->clip (buf, sw->buf, ret);
|
||||||
sw->total_hw_samples_acquired += total;
|
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;
|
wpos = (sw->hw->mix_buf->pos + live) % hwsamples;
|
||||||
samples = size >> sw->info.shift;
|
samples = size / sw->info.bytes_per_frame;
|
||||||
|
|
||||||
dead = hwsamples - live;
|
dead = hwsamples - live;
|
||||||
swlim = ((int64_t) dead << 32) / sw->ratio;
|
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 (
|
dolog (
|
||||||
"%s: write size %zu ret %zu total sw %zu\n",
|
"%s: write size %zu ret %zu total sw %zu\n",
|
||||||
SW_NAME (sw),
|
SW_NAME (sw),
|
||||||
size >> sw->info.shift,
|
size / sw->info.bytes_per_frame,
|
||||||
ret,
|
ret,
|
||||||
sw->total_hw_samples_mixed
|
sw->total_hw_samples_mixed
|
||||||
);
|
);
|
||||||
#endif
|
#endif
|
||||||
|
|
||||||
return ret << sw->info.shift;
|
return ret * sw->info.bytes_per_frame;
|
||||||
}
|
}
|
||||||
|
|
||||||
#ifdef DEBUG_AUDIO
|
#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)
|
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)
|
void AUD_set_active_out (SWVoiceOut *sw, int on)
|
||||||
|
@ -998,10 +996,10 @@ static size_t audio_get_avail (SWVoiceIn *sw)
|
||||||
ldebug (
|
ldebug (
|
||||||
"%s: get_avail live %d ret %" PRId64 "\n",
|
"%s: get_avail live %d ret %" PRId64 "\n",
|
||||||
SW_NAME (sw),
|
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)
|
static size_t audio_get_free(SWVoiceOut *sw)
|
||||||
|
@ -1025,10 +1023,11 @@ static size_t audio_get_free(SWVoiceOut *sw)
|
||||||
#ifdef DEBUG_OUT
|
#ifdef DEBUG_OUT
|
||||||
dolog ("%s: get_free live %d dead %d ret %" PRId64 "\n",
|
dolog ("%s: get_free live %d dead %d ret %" PRId64 "\n",
|
||||||
SW_NAME (sw),
|
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
|
#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,
|
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) {
|
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 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;
|
size_t written;
|
||||||
|
|
||||||
sw->buf = hw->mix_buf->samples + rpos2;
|
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;
|
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);
|
audio_pcm_hw_clip_out(hw, buf, decr);
|
||||||
proc = hw->pcm_ops->put_buffer_out(hw, buf, decr << hw->info.shift) >>
|
proc = hw->pcm_ops->put_buffer_out(hw, buf,
|
||||||
hw->info.shift;
|
decr * hw->info.bytes_per_frame) /
|
||||||
|
hw->info.bytes_per_frame;
|
||||||
|
|
||||||
live -= proc;
|
live -= proc;
|
||||||
clipped += proc;
|
clipped += proc;
|
||||||
|
@ -1234,16 +1234,16 @@ static size_t audio_pcm_hw_run_in(HWVoiceIn *hw, size_t samples)
|
||||||
|
|
||||||
while (samples) {
|
while (samples) {
|
||||||
size_t proc;
|
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);
|
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) {
|
if (size == 0) {
|
||||||
hw->pcm_ops->put_buffer_in(hw, buf, size);
|
hw->pcm_ops->put_buffer_in(hw, buf, size);
|
||||||
break;
|
break;
|
||||||
}
|
}
|
||||||
|
|
||||||
proc = MIN(size >> hw->info.shift,
|
proc = MIN(size / hw->info.bytes_per_frame,
|
||||||
conv_buf->size - conv_buf->pos);
|
conv_buf->size - conv_buf->pos);
|
||||||
|
|
||||||
hw->conv(conv_buf->samples + conv_buf->pos, buf, proc);
|
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;
|
samples -= proc;
|
||||||
conv += 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;
|
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) {
|
for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
|
||||||
cb->ops.capture (cb->opaque, cap->buf,
|
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;
|
rpos = (rpos + to_capture) % hw->mix_buf->size;
|
||||||
live -= to_capture;
|
live -= to_capture;
|
||||||
|
@ -1378,7 +1378,7 @@ void *audio_generic_get_buffer_in(HWVoiceIn *hw, size_t *size)
|
||||||
ssize_t start;
|
ssize_t start;
|
||||||
|
|
||||||
if (unlikely(!hw->buf_emul)) {
|
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->buf_emul = g_malloc(calc_size);
|
||||||
hw->size_emul = calc_size;
|
hw->size_emul = calc_size;
|
||||||
hw->pos_emul = hw->pending_emul = 0;
|
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)
|
void *audio_generic_get_buffer_out(HWVoiceOut *hw, size_t *size)
|
||||||
{
|
{
|
||||||
if (unlikely(!hw->buf_emul)) {
|
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->buf_emul = g_malloc(calc_size);
|
||||||
hw->size_emul = calc_size;
|
hw->size_emul = calc_size;
|
||||||
|
@ -1833,7 +1833,7 @@ CaptureVoiceOut *AUD_add_capture(
|
||||||
|
|
||||||
audio_pcm_init_info (&hw->info, as);
|
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->clip = mixeng_clip
|
||||||
[hw->info.nchannels == 2]
|
[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);
|
now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
|
||||||
ticks = now - rate->start_ticks;
|
ticks = now - rate->start_ticks;
|
||||||
bytes = muldiv64(ticks, info->bytes_per_second, NANOSECONDS_PER_SECOND);
|
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) {
|
if (samples < 0 || samples > 65536) {
|
||||||
AUD_log(NULL, "Resetting rate control (%" PRId64 " samples)\n", samples);
|
AUD_log(NULL, "Resetting rate control (%" PRId64 " samples)\n", samples);
|
||||||
audio_rate_start(rate);
|
audio_rate_start(rate);
|
||||||
samples = 0;
|
samples = 0;
|
||||||
}
|
}
|
||||||
|
|
||||||
ret = MIN(samples << info->shift, bytes_avail);
|
ret = MIN(samples * info->bytes_per_frame, bytes_avail);
|
||||||
rate->bytes_sent += ret;
|
rate->bytes_sent += ret;
|
||||||
return ret;
|
return ret;
|
||||||
}
|
}
|
||||||
|
|
|
@ -43,8 +43,7 @@ struct audio_pcm_info {
|
||||||
int sign;
|
int sign;
|
||||||
int freq;
|
int freq;
|
||||||
int nchannels;
|
int nchannels;
|
||||||
int align;
|
int bytes_per_frame;
|
||||||
int shift;
|
|
||||||
int bytes_per_second;
|
int bytes_per_second;
|
||||||
int swap_endianness;
|
int swap_endianness;
|
||||||
};
|
};
|
||||||
|
|
|
@ -440,7 +440,7 @@ static OSStatus audioDeviceIOProc(
|
||||||
}
|
}
|
||||||
|
|
||||||
frameCount = core->audioDevicePropertyBufferFrameSize;
|
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 there are not enough samples, set signal and return */
|
||||||
if (pending_frames < frameCount) {
|
if (pending_frames < frameCount) {
|
||||||
|
@ -449,7 +449,7 @@ static OSStatus audioDeviceIOProc(
|
||||||
return 0;
|
return 0;
|
||||||
}
|
}
|
||||||
|
|
||||||
len = frameCount << hw->info.shift;
|
len = frameCount * hw->info.bytes_per_frame;
|
||||||
while (len) {
|
while (len) {
|
||||||
size_t write_len;
|
size_t write_len;
|
||||||
ssize_t start = ((ssize_t) hw->pos_emul) - hw->pending_emul;
|
ssize_t start = ((ssize_t) hw->pos_emul) - hw->pending_emul;
|
||||||
|
|
|
@ -98,8 +98,8 @@ static int glue (dsound_lock_, TYPE) (
|
||||||
goto fail;
|
goto fail;
|
||||||
}
|
}
|
||||||
|
|
||||||
if ((p1p && *p1p && (*blen1p & info->align)) ||
|
if ((p1p && *p1p && (*blen1p % info->bytes_per_frame)) ||
|
||||||
(p2p && *p2p && (*blen2p & info->align))) {
|
(p2p && *p2p && (*blen2p % info->bytes_per_frame))) {
|
||||||
dolog("DirectSound returned misaligned buffer %ld %ld\n",
|
dolog("DirectSound returned misaligned buffer %ld %ld\n",
|
||||||
*blen1p, *blen2p);
|
*blen1p, *blen2p);
|
||||||
glue(dsound_unlock_, TYPE)(buf, *p1p, p2p ? *p2p : NULL, *blen1p,
|
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;
|
obt_as.endianness = 0;
|
||||||
audio_pcm_init_info (&hw->info, &obt_as);
|
audio_pcm_init_info (&hw->info, &obt_as);
|
||||||
|
|
||||||
if (bc.dwBufferBytes & hw->info.align) {
|
if (bc.dwBufferBytes % hw->info.bytes_per_frame) {
|
||||||
dolog (
|
dolog (
|
||||||
"GetCaps returned misaligned buffer size %ld, alignment %d\n",
|
"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->size_emul = bc.dwBufferBytes;
|
||||||
hw->samples = bc.dwBufferBytes >> hw->info.shift;
|
hw->samples = bc.dwBufferBytes / hw->info.bytes_per_frame;
|
||||||
ds->s = s;
|
ds->s = s;
|
||||||
|
|
||||||
#ifdef DEBUG_DSOUND
|
#ifdef DEBUG_DSOUND
|
||||||
|
|
|
@ -320,8 +320,8 @@ static void dsound_clear_sample (HWVoiceOut *hw, LPDIRECTSOUNDBUFFER dsb,
|
||||||
return;
|
return;
|
||||||
}
|
}
|
||||||
|
|
||||||
len1 = blen1 >> hw->info.shift;
|
len1 = blen1 / hw->info.bytes_per_frame;
|
||||||
len2 = blen2 >> hw->info.shift;
|
len2 = blen2 / hw->info.bytes_per_frame;
|
||||||
|
|
||||||
#ifdef DEBUG_DSOUND
|
#ifdef DEBUG_DSOUND
|
||||||
dolog ("clear %p,%ld,%ld %p,%ld,%ld\n",
|
dolog ("clear %p,%ld,%ld %p,%ld,%ld\n",
|
||||||
|
|
|
@ -91,7 +91,7 @@ static size_t no_read(HWVoiceIn *hw, void *buf, size_t size)
|
||||||
NoVoiceIn *no = (NoVoiceIn *) hw;
|
NoVoiceIn *no = (NoVoiceIn *) hw;
|
||||||
int64_t bytes = audio_rate_get_bytes(&hw->info, &no->rate, size);
|
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;
|
return bytes;
|
||||||
}
|
}
|
||||||
|
|
||||||
|
|
|
@ -506,16 +506,16 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
|
||||||
oss->nfrags = obt.nfrags;
|
oss->nfrags = obt.nfrags;
|
||||||
oss->fragsize = obt.fragsize;
|
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",
|
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;
|
oss->mmapped = 0;
|
||||||
if (oopts->has_try_mmap && oopts->try_mmap) {
|
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(
|
hw->buf_emul = mmap(
|
||||||
NULL,
|
NULL,
|
||||||
hw->size_emul,
|
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->nfrags = obt.nfrags;
|
||||||
oss->fragsize = obt.fragsize;
|
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",
|
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->fd = fd;
|
||||||
oss->dev = dev;
|
oss->dev = dev;
|
||||||
|
|
|
@ -131,7 +131,8 @@ static void *line_out_get_buffer(HWVoiceOut *hw, size_t *size)
|
||||||
|
|
||||||
if (out->frame) {
|
if (out->frame) {
|
||||||
*size = audio_rate_get_bytes(
|
*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 {
|
} else {
|
||||||
audio_rate_start(&out->rate);
|
audio_rate_start(&out->rate);
|
||||||
}
|
}
|
||||||
|
|
|
@ -43,14 +43,14 @@ static size_t wav_write_out(HWVoiceOut *hw, void *buf, size_t len)
|
||||||
{
|
{
|
||||||
WAVVoiceOut *wav = (WAVVoiceOut *) hw;
|
WAVVoiceOut *wav = (WAVVoiceOut *) hw;
|
||||||
int64_t bytes = audio_rate_get_bytes(&hw->info, &wav->rate, len);
|
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) {
|
if (bytes && fwrite(buf, bytes, 1, wav->f) != 1) {
|
||||||
dolog("wav_write_out: fwrite of %" PRId64 " bytes failed\nReason: %s\n",
|
dolog("wav_write_out: fwrite of %" PRId64 " bytes failed\nReason: %s\n",
|
||||||
bytes, strerror(errno));
|
bytes, strerror(errno));
|
||||||
}
|
}
|
||||||
|
|
||||||
wav->total_samples += bytes >> hw->info.shift;
|
wav->total_samples += bytes / hw->info.bytes_per_frame;
|
||||||
return bytes;
|
return bytes;
|
||||||
}
|
}
|
||||||
|
|
||||||
|
@ -134,7 +134,7 @@ static void wav_fini_out (HWVoiceOut *hw)
|
||||||
WAVVoiceOut *wav = (WAVVoiceOut *) hw;
|
WAVVoiceOut *wav = (WAVVoiceOut *) hw;
|
||||||
uint8_t rlen[4];
|
uint8_t rlen[4];
|
||||||
uint8_t dlen[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;
|
uint32_t rifflen = datalen + 36;
|
||||||
|
|
||||||
if (!wav->f) {
|
if (!wav->f) {
|
||||||
|
|
Loading…
Reference in New Issue