OpenE2K/qemu-e2k
13
4
Fork 0
Code Issues 4 Pull Requests Projects 2 Releases Activity
605a8b5491
qemu-e2k/audio/alsaaudio.c
Volker Rümelin 5140ad8279 alsaaudio: reintroduce default recording settings 
Audio recording with ALSA default settings currently doesn't
work. The debug log shows updates every 0.75s and 1.5s.

audio: Elapsed since last alsa run (running): 0.743030
audio: Elapsed since last alsa run (running): 1.486048
audio: Elapsed since last alsa run (running): 0.743008
audio: Elapsed since last alsa run (running): 1.485878
audio: Elapsed since last alsa run (running): 1.486040
audio: Elapsed since last alsa run (running): 1.485886

The time between updates should be in the 10ms range. Audio
recording with ALSA has the same timing contraints as playback.
Reintroduce the default recording settings and use the same
default settings for recording as for playback.

The term "reintroduce" is correct because commit a93f328177
("alsaaudio: port to -audiodev config") removed the default
settings for recording.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Message-Id: <20230121094735.11644-11-vr_qemu@t-online.de>
2023-03-06 10:30:23 +04:00

975 lines
25 KiB
C
Raw Blame History

/*
* QEMU ALSA audio driver
*
* Copyright (c) 2005 Vassili Karpov (malc)
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
#include "qemu/osdep.h"
#include <alsa/asoundlib.h>
#include "qemu/main-loop.h"
#include "qemu/module.h"
#include "audio.h"
#include "trace.h"
#pragma GCC diagnostic ignored "-Waddress"
#define AUDIO_CAP "alsa"
#include "audio_int.h"
#define DEBUG_ALSA 0
struct pollhlp {
snd_pcm_t *handle;
struct pollfd *pfds;
int count;
int mask;
AudioState *s;
};
typedef struct ALSAVoiceOut {
HWVoiceOut hw;
snd_pcm_t *handle;
struct pollhlp pollhlp;
Audiodev *dev;
} ALSAVoiceOut;
typedef struct ALSAVoiceIn {
HWVoiceIn hw;
snd_pcm_t *handle;
struct pollhlp pollhlp;
Audiodev *dev;
} ALSAVoiceIn;
struct alsa_params_req {
int freq;
snd_pcm_format_t fmt;
int nchannels;
};
struct alsa_params_obt {
int freq;
AudioFormat fmt;
int endianness;
int nchannels;
snd_pcm_uframes_t samples;
};
static void G_GNUC_PRINTF (2, 3) alsa_logerr (int err, const char *fmt, ...)
{
va_list ap;
va_start (ap, fmt);
AUD_vlog (AUDIO_CAP, fmt, ap);
va_end (ap);
AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
}
static void G_GNUC_PRINTF (3, 4) alsa_logerr2 (
int err,
const char *typ,
const char *fmt,
...
)
{
va_list ap;
AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);
va_start (ap, fmt);
AUD_vlog (AUDIO_CAP, fmt, ap);
va_end (ap);
AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
}
static void alsa_fini_poll (struct pollhlp *hlp)
{
int i;
struct pollfd *pfds = hlp->pfds;
if (pfds) {
for (i = 0; i < hlp->count; ++i) {
qemu_set_fd_handler (pfds[i].fd, NULL, NULL, NULL);
}
g_free (pfds);
}
hlp->pfds = NULL;
hlp->count = 0;
hlp->handle = NULL;
}
static void alsa_anal_close1 (snd_pcm_t **handlep)
{
int err = snd_pcm_close (*handlep);
if (err) {
alsa_logerr (err, "Failed to close PCM handle %p\n", *handlep);
}
*handlep = NULL;
}
static void alsa_anal_close (snd_pcm_t **handlep, struct pollhlp *hlp)
{
alsa_fini_poll (hlp);
alsa_anal_close1 (handlep);
}
static int alsa_recover (snd_pcm_t *handle)
{
int err = snd_pcm_prepare (handle);
if (err < 0) {
alsa_logerr (err, "Failed to prepare handle %p\n", handle);
return -1;
}
return 0;
}
static int alsa_resume (snd_pcm_t *handle)
{
int err = snd_pcm_resume (handle);
if (err < 0) {
alsa_logerr (err, "Failed to resume handle %p\n", handle);
return -1;
}
return 0;
}
static void alsa_poll_handler (void *opaque)
{
int err, count;
snd_pcm_state_t state;
struct pollhlp *hlp = opaque;
unsigned short revents;
count = poll (hlp->pfds, hlp->count, 0);
if (count < 0) {
dolog ("alsa_poll_handler: poll %s\n", strerror (errno));
return;
}
if (!count) {
return;
}
/* XXX: ALSA example uses initial count, not the one returned by
poll, correct? */
err = snd_pcm_poll_descriptors_revents (hlp->handle, hlp->pfds,
hlp->count, &revents);
if (err < 0) {
alsa_logerr (err, "snd_pcm_poll_descriptors_revents");
return;
}
if (!(revents & hlp->mask)) {
trace_alsa_revents(revents);
return;
}
state = snd_pcm_state (hlp->handle);
switch (state) {
case SND_PCM_STATE_SETUP:
alsa_recover (hlp->handle);
break;
case SND_PCM_STATE_XRUN:
alsa_recover (hlp->handle);
break;
case SND_PCM_STATE_SUSPENDED:
alsa_resume (hlp->handle);
break;
case SND_PCM_STATE_PREPARED:
audio_run(hlp->s, "alsa run (prepared)");
break;
case SND_PCM_STATE_RUNNING:
audio_run(hlp->s, "alsa run (running)");
break;
default:
dolog ("Unexpected state %d\n", state);
}
}
static int alsa_poll_helper (snd_pcm_t *handle, struct pollhlp *hlp, int mask)
{
int i, count, err;
struct pollfd *pfds;
count = snd_pcm_poll_descriptors_count (handle);
if (count <= 0) {
dolog ("Could not initialize poll mode\n"
"Invalid number of poll descriptors %d\n", count);
return -1;
}
pfds = g_new0(struct pollfd, count);
err = snd_pcm_poll_descriptors (handle, pfds, count);
if (err < 0) {
alsa_logerr (err, "Could not initialize poll mode\n"
"Could not obtain poll descriptors\n");
g_free (pfds);
return -1;
}
for (i = 0; i < count; ++i) {
if (pfds[i].events & POLLIN) {
qemu_set_fd_handler (pfds[i].fd, alsa_poll_handler, NULL, hlp);
}
if (pfds[i].events & POLLOUT) {
trace_alsa_pollout(i, pfds[i].fd);
qemu_set_fd_handler (pfds[i].fd, NULL, alsa_poll_handler, hlp);
}
trace_alsa_set_handler(pfds[i].events, i, pfds[i].fd, err);
}
hlp->pfds = pfds;
hlp->count = count;
hlp->handle = handle;
hlp->mask = mask;
return 0;
}
static int alsa_poll_out (HWVoiceOut *hw)
{
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLOUT);
}
static int alsa_poll_in (HWVoiceIn *hw)
{
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLIN);
}
static snd_pcm_format_t aud_to_alsafmt (AudioFormat fmt, int endianness)
{
switch (fmt) {
case AUDIO_FORMAT_S8:
return SND_PCM_FORMAT_S8;
case AUDIO_FORMAT_U8:
return SND_PCM_FORMAT_U8;
case AUDIO_FORMAT_S16:
if (endianness) {
return SND_PCM_FORMAT_S16_BE;
} else {
return SND_PCM_FORMAT_S16_LE;
}
case AUDIO_FORMAT_U16:
if (endianness) {
return SND_PCM_FORMAT_U16_BE;
} else {
return SND_PCM_FORMAT_U16_LE;
}
case AUDIO_FORMAT_S32:
if (endianness) {
return SND_PCM_FORMAT_S32_BE;
} else {
return SND_PCM_FORMAT_S32_LE;
}
case AUDIO_FORMAT_U32:
if (endianness) {
return SND_PCM_FORMAT_U32_BE;
} else {
return SND_PCM_FORMAT_U32_LE;
}
case AUDIO_FORMAT_F32:
if (endianness) {
return SND_PCM_FORMAT_FLOAT_BE;
} else {
return SND_PCM_FORMAT_FLOAT_LE;
}
default:
dolog ("Internal logic error: Bad audio format %d\n", fmt);
#ifdef DEBUG_AUDIO
abort ();
#endif
return SND_PCM_FORMAT_U8;
}
}
static int alsa_to_audfmt (snd_pcm_format_t alsafmt, AudioFormat *fmt,
int *endianness)
{
switch (alsafmt) {
case SND_PCM_FORMAT_S8:
*endianness = 0;
*fmt = AUDIO_FORMAT_S8;
break;
case SND_PCM_FORMAT_U8:
*endianness = 0;
*fmt = AUDIO_FORMAT_U8;
break;
case SND_PCM_FORMAT_S16_LE:
*endianness = 0;
*fmt = AUDIO_FORMAT_S16;
break;
case SND_PCM_FORMAT_U16_LE:
*endianness = 0;
*fmt = AUDIO_FORMAT_U16;
break;
case SND_PCM_FORMAT_S16_BE:
*endianness = 1;
*fmt = AUDIO_FORMAT_S16;
break;
case SND_PCM_FORMAT_U16_BE:
*endianness = 1;
*fmt = AUDIO_FORMAT_U16;
break;
case SND_PCM_FORMAT_S32_LE:
*endianness = 0;
*fmt = AUDIO_FORMAT_S32;
break;
case SND_PCM_FORMAT_U32_LE:
*endianness = 0;
*fmt = AUDIO_FORMAT_U32;
break;
case SND_PCM_FORMAT_S32_BE:
*endianness = 1;
*fmt = AUDIO_FORMAT_S32;
break;
case SND_PCM_FORMAT_U32_BE:
*endianness = 1;
*fmt = AUDIO_FORMAT_U32;
break;
case SND_PCM_FORMAT_FLOAT_LE:
*endianness = 0;
*fmt = AUDIO_FORMAT_F32;
break;
case SND_PCM_FORMAT_FLOAT_BE:
*endianness = 1;
*fmt = AUDIO_FORMAT_F32;
break;
default:
dolog ("Unrecognized audio format %d\n", alsafmt);
return -1;
}
return 0;
}
static void alsa_dump_info (struct alsa_params_req *req,
struct alsa_params_obt *obt,
snd_pcm_format_t obtfmt,
AudiodevAlsaPerDirectionOptions *apdo)
{
dolog("parameter | requested value | obtained value\n");
dolog("format | %10d | %10d\n", req->fmt, obtfmt);
dolog("channels | %10d | %10d\n",
req->nchannels, obt->nchannels);
dolog("frequency | %10d | %10d\n", req->freq, obt->freq);
dolog("============================================\n");
dolog("requested: buffer len %" PRId32 " period len %" PRId32 "\n",
apdo->buffer_length, apdo->period_length);
dolog("obtained: samples %ld\n", obt->samples);
}
static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
{
int err;
snd_pcm_sw_params_t *sw_params;
snd_pcm_sw_params_alloca (&sw_params);
err = snd_pcm_sw_params_current (handle, sw_params);
if (err < 0) {
dolog ("Could not fully initialize DAC\n");
alsa_logerr (err, "Failed to get current software parameters\n");
return;
}
err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold);
if (err < 0) {
dolog ("Could not fully initialize DAC\n");
alsa_logerr (err, "Failed to set software threshold to %ld\n",
threshold);
return;
}
err = snd_pcm_sw_params (handle, sw_params);
if (err < 0) {
dolog ("Could not fully initialize DAC\n");
alsa_logerr (err, "Failed to set software parameters\n");
return;
}
}
static int alsa_open(bool in, struct alsa_params_req *req,
struct alsa_params_obt *obt, snd_pcm_t **handlep,
Audiodev *dev)
{
AudiodevAlsaOptions *aopts = &dev->u.alsa;
AudiodevAlsaPerDirectionOptions *apdo = in ? aopts->in : aopts->out;
snd_pcm_t *handle;
snd_pcm_hw_params_t *hw_params;
int err;
unsigned int freq, nchannels;
const char *pcm_name = apdo->dev ?: "default";
snd_pcm_uframes_t obt_buffer_size;
const char *typ = in ? "ADC" : "DAC";
snd_pcm_format_t obtfmt;
freq = req->freq;
nchannels = req->nchannels;
snd_pcm_hw_params_alloca (&hw_params);
err = snd_pcm_open (
&handle,
pcm_name,
in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
SND_PCM_NONBLOCK
);
if (err < 0) {
alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name);
return -1;
}
err = snd_pcm_hw_params_any (handle, hw_params);
if (err < 0) {
alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n");
goto err;
}
err = snd_pcm_hw_params_set_access (
handle,
hw_params,
SND_PCM_ACCESS_RW_INTERLEAVED
);
if (err < 0) {
alsa_logerr2 (err, typ, "Failed to set access type\n");
goto err;
}
err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt);
if (err < 0) {
alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
}
err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
if (err < 0) {
alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq);
goto err;
}
err = snd_pcm_hw_params_set_channels_near (
handle,
hw_params,
&nchannels
);
if (err < 0) {
alsa_logerr2 (err, typ, "Failed to set number of channels %d\n",
req->nchannels);
goto err;
}
if (apdo->buffer_length) {
int dir = 0;
unsigned int btime = apdo->buffer_length;
err = snd_pcm_hw_params_set_buffer_time_near(
handle, hw_params, &btime, &dir);
if (err < 0) {
alsa_logerr2(err, typ, "Failed to set buffer time to %" PRId32 "\n",
apdo->buffer_length);
goto err;
}
if (apdo->has_buffer_length && btime != apdo->buffer_length) {
dolog("Requested buffer time %" PRId32
" was rejected, using %u\n", apdo->buffer_length, btime);
}
}
if (apdo->period_length) {
int dir = 0;
unsigned int ptime = apdo->period_length;
err = snd_pcm_hw_params_set_period_time_near(handle, hw_params, &ptime,
&dir);
if (err < 0) {
alsa_logerr2(err, typ, "Failed to set period time to %" PRId32 "\n",
apdo->period_length);
goto err;
}
if (apdo->has_period_length && ptime != apdo->period_length) {
dolog("Requested period time %" PRId32 " was rejected, using %d\n",
apdo->period_length, ptime);
}
}
err = snd_pcm_hw_params (handle, hw_params);
if (err < 0) {
alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
goto err;
}
err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size);
if (err < 0) {
alsa_logerr2 (err, typ, "Failed to get buffer size\n");
goto err;
}
err = snd_pcm_hw_params_get_format (hw_params, &obtfmt);
if (err < 0) {
alsa_logerr2 (err, typ, "Failed to get format\n");
goto err;
}
if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) {
dolog ("Invalid format was returned %d\n", obtfmt);
goto err;
}
err = snd_pcm_prepare (handle);
if (err < 0) {
alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle);
goto err;
}
if (!in && aopts->has_threshold && aopts->threshold) {
struct audsettings as = { .freq = freq };
alsa_set_threshold(
handle,
audio_buffer_frames(qapi_AudiodevAlsaPerDirectionOptions_base(apdo),
&as, aopts->threshold));
}
obt->nchannels = nchannels;
obt->freq = freq;
obt->samples = obt_buffer_size;
*handlep = handle;
if (DEBUG_ALSA || obtfmt != req->fmt ||
obt->nchannels != req->nchannels || obt->freq != req->freq) {
dolog ("Audio parameters for %s\n", typ);
alsa_dump_info(req, obt, obtfmt, apdo);
}
return 0;
err:
alsa_anal_close1 (&handle);
return -1;
}
static size_t alsa_buffer_get_free(HWVoiceOut *hw)
{
ALSAVoiceOut *alsa = (ALSAVoiceOut *)hw;
snd_pcm_sframes_t avail;
size_t alsa_free, generic_free, generic_in_use;
avail = snd_pcm_avail_update(alsa->handle);
if (avail < 0) {
if (avail == -EPIPE) {
if (!alsa_recover(alsa->handle)) {
avail = snd_pcm_avail_update(alsa->handle);
}
}
if (avail < 0) {
alsa_logerr(avail,
"Could not obtain number of available frames\n");
avail = 0;
}
}
alsa_free = avail * hw->info.bytes_per_frame;
generic_free = audio_generic_buffer_get_free(hw);
generic_in_use = hw->samples * hw->info.bytes_per_frame - generic_free;
if (generic_in_use) {
/*
* This code can only be reached in the unlikely case that
* snd_pcm_avail_update() returned a larger number of frames
* than snd_pcm_writei() could write. Make sure that all
* remaining bytes in the generic buffer can be written.
*/
alsa_free = alsa_free > generic_in_use ? alsa_free - generic_in_use : 0;
}
return alsa_free;
}
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.bytes_per_frame;
while (len_frames) {
char *src = advance(buf, pos);
snd_pcm_sframes_t written;
written = snd_pcm_writei(alsa->handle, src, len_frames);
if (written <= 0) {
switch (written) {
case 0:
trace_alsa_wrote_zero(len_frames);
return pos;
case -EPIPE:
if (alsa_recover(alsa->handle)) {
alsa_logerr(written, "Failed to write %zu frames\n",
len_frames);
return pos;
}
trace_alsa_xrun_out();
continue;
case -ESTRPIPE:
/*
* stream is suspended and waiting for an application
* recovery
*/
if (alsa_resume(alsa->handle)) {
alsa_logerr(written, "Failed to write %zu frames\n",
len_frames);
return pos;
}
trace_alsa_resume_out();
continue;
case -EAGAIN:
return pos;
default:
alsa_logerr(written, "Failed to write %zu frames from %p\n",
len, src);
return pos;
}
}
pos += written * hw->info.bytes_per_frame;
if (written < len_frames) {
break;
}
len_frames -= written;
}
return pos;
}
static void alsa_fini_out (HWVoiceOut *hw)
{
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
ldebug ("alsa_fini\n");
alsa_anal_close (&alsa->handle, &alsa->pollhlp);
}
static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
void *drv_opaque)
{
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
struct alsa_params_req req;
struct alsa_params_obt obt;
snd_pcm_t *handle;
struct audsettings obt_as;
Audiodev *dev = drv_opaque;
req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
req.freq = as->freq;
req.nchannels = as->nchannels;
if (alsa_open(0, &req, &obt, &handle, dev)) {
return -1;
}
obt_as.freq = obt.freq;
obt_as.nchannels = obt.nchannels;
obt_as.fmt = obt.fmt;
obt_as.endianness = obt.endianness;
audio_pcm_init_info (&hw->info, &obt_as);
hw->samples = obt.samples;
alsa->pollhlp.s = hw->s;
alsa->handle = handle;
alsa->dev = dev;
return 0;
}
#define VOICE_CTL_PAUSE 0
#define VOICE_CTL_PREPARE 1
#define VOICE_CTL_START 2
static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int ctl)
{
int err;
if (ctl == VOICE_CTL_PAUSE) {
err = snd_pcm_drop (handle);
if (err < 0) {
alsa_logerr (err, "Could not stop %s\n", typ);
return -1;
}
} else {
err = snd_pcm_prepare (handle);
if (err < 0) {
alsa_logerr (err, "Could not prepare handle for %s\n", typ);
return -1;
}
if (ctl == VOICE_CTL_START) {
err = snd_pcm_start(handle);
if (err < 0) {
alsa_logerr (err, "Could not start handle for %s\n", typ);
return -1;
}
}
}
return 0;
}
static void alsa_enable_out(HWVoiceOut *hw, bool enable)
{
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.out;
if (enable) {
bool poll_mode = apdo->try_poll;
ldebug("enabling voice\n");
if (poll_mode && alsa_poll_out(hw)) {
poll_mode = 0;
}
hw->poll_mode = poll_mode;
alsa_voice_ctl(alsa->handle, "playback", VOICE_CTL_PREPARE);
} else {
ldebug("disabling voice\n");
if (hw->poll_mode) {
hw->poll_mode = 0;
alsa_fini_poll(&alsa->pollhlp);
}
alsa_voice_ctl(alsa->handle, "playback", VOICE_CTL_PAUSE);
}
}
static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
{
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
struct alsa_params_req req;
struct alsa_params_obt obt;
snd_pcm_t *handle;
struct audsettings obt_as;
Audiodev *dev = drv_opaque;
req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
req.freq = as->freq;
req.nchannels = as->nchannels;
if (alsa_open(1, &req, &obt, &handle, dev)) {
return -1;
}
obt_as.freq = obt.freq;
obt_as.nchannels = obt.nchannels;
obt_as.fmt = obt.fmt;
obt_as.endianness = obt.endianness;
audio_pcm_init_info (&hw->info, &obt_as);
hw->samples = obt.samples;
alsa->pollhlp.s = hw->s;
alsa->handle = handle;
alsa->dev = dev;
return 0;
}
static void alsa_fini_in (HWVoiceIn *hw)
{
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
alsa_anal_close (&alsa->handle, &alsa->pollhlp);
}
static size_t alsa_read(HWVoiceIn *hw, void *buf, size_t len)
{
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
size_t pos = 0;
while (len) {
void *dst = advance(buf, pos);
snd_pcm_sframes_t nread;
nread = snd_pcm_readi(
alsa->handle, dst, len / hw->info.bytes_per_frame);
if (nread <= 0) {
switch (nread) {
case 0:
trace_alsa_read_zero(len);
return pos;
case -EPIPE:
if (alsa_recover(alsa->handle)) {
alsa_logerr(nread, "Failed to read %zu frames\n", len);
return pos;
}
trace_alsa_xrun_in();
continue;
case -EAGAIN:
return pos;
default:
alsa_logerr(nread, "Failed to read %zu frames to %p\n",
len, dst);
return pos;
}
}
pos += nread * hw->info.bytes_per_frame;
len -= nread * hw->info.bytes_per_frame;
}
return pos;
}
static void alsa_enable_in(HWVoiceIn *hw, bool enable)
{
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.in;
if (enable) {
bool poll_mode = apdo->try_poll;
ldebug("enabling voice\n");
if (poll_mode && alsa_poll_in(hw)) {
poll_mode = 0;
}
hw->poll_mode = poll_mode;
alsa_voice_ctl(alsa->handle, "capture", VOICE_CTL_START);
} else {
ldebug ("disabling voice\n");
if (hw->poll_mode) {
hw->poll_mode = 0;
alsa_fini_poll(&alsa->pollhlp);
}
alsa_voice_ctl(alsa->handle, "capture", VOICE_CTL_PAUSE);
}
}
static void alsa_init_per_direction(AudiodevAlsaPerDirectionOptions *apdo)
{
if (!apdo->has_try_poll) {
apdo->try_poll = true;
apdo->has_try_poll = true;
}
}
static void *alsa_audio_init(Audiodev *dev)
{
AudiodevAlsaOptions *aopts;
assert(dev->driver == AUDIODEV_DRIVER_ALSA);
aopts = &dev->u.alsa;
alsa_init_per_direction(aopts->in);
alsa_init_per_direction(aopts->out);
/* don't set has_* so alsa_open can identify it wasn't set by the user */
if (!dev->u.alsa.out->has_period_length) {
/* 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 = 92880;
}
if (!dev->u.alsa.in->has_period_length) {
/* 256 frames assuming 44100Hz */
dev->u.alsa.in->period_length = 5805;
}
if (!dev->u.alsa.in->has_buffer_length) {
/* 4096 frames assuming 44100Hz */
dev->u.alsa.in->buffer_length = 92880;
}
return dev;
}
static void alsa_audio_fini (void *opaque)
{
}
static struct audio_pcm_ops alsa_pcm_ops = {
.init_out = alsa_init_out,
.fini_out = alsa_fini_out,
.write = alsa_write,
.buffer_get_free = alsa_buffer_get_free,
.run_buffer_out = audio_generic_run_buffer_out,
.enable_out = alsa_enable_out,
.init_in = alsa_init_in,
.fini_in = alsa_fini_in,
.read = alsa_read,
.run_buffer_in = audio_generic_run_buffer_in,
.enable_in = alsa_enable_in,
};
static struct audio_driver alsa_audio_driver = {
.name = "alsa",
.descr = "ALSA http://www.alsa-project.org",
.init = alsa_audio_init,
.fini = alsa_audio_fini,
.pcm_ops = &alsa_pcm_ops,
.can_be_default = 1,
.max_voices_out = INT_MAX,
.max_voices_in = INT_MAX,
.voice_size_out = sizeof (ALSAVoiceOut),
.voice_size_in = sizeof (ALSAVoiceIn)
};
static void register_audio_alsa(void)
{
audio_driver_register(&alsa_audio_driver);
}
type_init(register_audio_alsa);
Reference in New Issue View Git Blame Copy Permalink