libsigrok/hardware/alsa/api.c

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/*
* This file is part of the libsigrok project.
*
* Copyright (C) 2011 Daniel Ribeiro <drwyrm@gmail.com>
* Copyright (C) 2012 Uwe Hermann <uwe@hermann-uwe.de>
* Copyright (C) 2012 Alexandru Gagniuc <mr.nuke.me@gmail.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <stdlib.h>
#include <unistd.h>
#include <string.h>
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#include "libsigrok.h"
#include "libsigrok-internal.h"
#include "protocol.h"
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static const int32_t hwcaps[] = {
SR_CONF_SAMPLERATE,
SR_CONF_LIMIT_SAMPLES,
SR_CONF_CONTINUOUS,
};
SR_PRIV struct sr_dev_driver alsa_driver_info;
static struct sr_dev_driver *di = &alsa_driver_info;
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static void clear_helper(void *priv)
{
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struct dev_context *devc;
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devc = priv;
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snd_pcm_hw_params_free(devc->hw_params);
devc->hw_params = NULL;
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g_free((void *)devc->samplerates);
devc->samplerates = NULL;
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}
static int dev_clear(void)
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{
return std_dev_clear(di, clear_helper);
}
static int init(struct sr_context *sr_ctx)
{
return std_init(sr_ctx, di, LOG_PREFIX);
}
static GSList *scan(GSList *options)
{
return alsa_scan(options, di);
}
static GSList *dev_list(void)
{
return ((struct drv_context *)(di->priv))->instances;
}
static int dev_open(struct sr_dev_inst *sdi)
{
struct dev_context *devc;
int ret;
devc = sdi->priv;
if (!(devc->hwdev)) {
sr_err("devc->hwdev was NULL.");
return SR_ERR_BUG;
}
sr_dbg("Opening audio device '%s' for stream capture.", devc->hwdev);
ret = snd_pcm_open(&devc->capture_handle, devc->hwdev,
SND_PCM_STREAM_CAPTURE, 0);
if (ret < 0) {
sr_err("Can't open audio device: %s.", snd_strerror(ret));
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return SR_ERR;
}
sr_dbg("Initializing hardware parameter structure.");
ret = snd_pcm_hw_params_any(devc->capture_handle, devc->hw_params);
if (ret < 0) {
sr_err("Can't initialize hardware parameter structure: %s.",
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snd_strerror(ret));
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return SR_ERR;
}
sdi->status = SR_ST_ACTIVE;
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return SR_OK;
}
static int dev_close(struct sr_dev_inst *sdi)
{
int ret;
struct dev_context *devc;
devc = sdi->priv;
if (devc->capture_handle) {
sr_dbg("Closing PCM device.");
if ((ret = snd_pcm_close(devc->capture_handle)) < 0)
sr_err("Failed to close device: %s.",
snd_strerror(ret));
devc->capture_handle = NULL;
sdi->status = SR_ST_INACTIVE;
} else {
sr_dbg("No capture handle, no need to close audio device.");
}
return SR_OK;
}
static int cleanup(void)
{
return dev_clear();
}
static int config_get(int id, GVariant **data, const struct sr_dev_inst *sdi,
const struct sr_probe_group *probe_group)
{
struct dev_context *devc;
(void)probe_group;
switch (id) {
case SR_CONF_SAMPLERATE:
devc = sdi->priv;
*data = g_variant_new_uint64(devc->cur_samplerate);
break;
default:
return SR_ERR_NA;
}
return SR_OK;
}
static int config_set(int id, GVariant *data, const struct sr_dev_inst *sdi,
const struct sr_probe_group *probe_group)
{
struct dev_context *devc;
(void)probe_group;
if (sdi->status != SR_ST_ACTIVE)
return SR_ERR_DEV_CLOSED;
devc = sdi->priv;
switch (id) {
case SR_CONF_SAMPLERATE:
alsa_set_samplerate(sdi, g_variant_get_uint64(data));
break;
case SR_CONF_LIMIT_SAMPLES:
devc->limit_samples = g_variant_get_uint64(data);
break;
default:
return SR_ERR_NA;
}
return SR_OK;
}
static int config_list(int key, GVariant **data, const struct sr_dev_inst *sdi,
const struct sr_probe_group *probe_group)
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{
struct dev_context *devc;
GVariant *gvar;
GVariantBuilder gvb;
int i;
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(void)probe_group;
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switch (key) {
case SR_CONF_DEVICE_OPTIONS:
*data = g_variant_new_fixed_array(G_VARIANT_TYPE_INT32,
hwcaps, ARRAY_SIZE(hwcaps), sizeof(int32_t));
break;
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case SR_CONF_SAMPLERATE:
if (!sdi || !sdi->priv)
return SR_ERR_ARG;
devc = sdi->priv;
if (!devc->samplerates) {
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sr_err("Instance did not contain a samplerate list.");
return SR_ERR_ARG;
}
for (i = 0; devc->samplerates[i]; i++)
;
g_variant_builder_init(&gvb, G_VARIANT_TYPE("a{sv}"));
gvar = g_variant_new_fixed_array(G_VARIANT_TYPE("t"),
devc->samplerates, i, sizeof(uint64_t));
g_variant_builder_add(&gvb, "{sv}", "samplerates", gvar);
*data = g_variant_builder_end(&gvb);
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break;
default:
return SR_ERR_NA;
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}
return SR_OK;
}
static int dev_acquisition_start(const struct sr_dev_inst *sdi, void *cb_data)
{
struct dev_context *devc;
int count, ret;
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char *endianness;
if (sdi->status != SR_ST_ACTIVE)
return SR_ERR_DEV_CLOSED;
devc = sdi->priv;
devc->cb_data = cb_data;
devc->num_samples = 0;
sr_dbg("Setting audio access type to RW/interleaved.");
ret = snd_pcm_hw_params_set_access(devc->capture_handle,
devc->hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
if (ret < 0) {
sr_err("Can't set audio access type: %s.", snd_strerror(ret));
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return SR_ERR;
}
/* FIXME: Hardcoded for 16bits. */
if (SND_PCM_FORMAT_S16 == SND_PCM_FORMAT_S16_LE)
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endianness = "little endian";
else
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endianness = "big endian";
sr_dbg("Setting audio sample format to signed 16bit (%s).", endianness);
ret = snd_pcm_hw_params_set_format(devc->capture_handle,
devc->hw_params,
SND_PCM_FORMAT_S16);
if (ret < 0) {
sr_err("Can't set audio sample format: %s.", snd_strerror(ret));
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return SR_ERR;
}
sr_dbg("Setting audio samplerate to %" PRIu64 "Hz.",
devc->cur_samplerate);
ret = snd_pcm_hw_params_set_rate(devc->capture_handle, devc->hw_params,
(unsigned int)devc->cur_samplerate, 0);
if (ret < 0) {
sr_err("Can't set audio sample rate: %s.", snd_strerror(ret));
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return SR_ERR;
}
sr_dbg("Setting audio channel count to %d.", devc->num_probes);
ret = snd_pcm_hw_params_set_channels(devc->capture_handle,
devc->hw_params, devc->num_probes);
if (ret < 0) {
sr_err("Can't set channel count: %s.", snd_strerror(ret));
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return SR_ERR;
}
sr_dbg("Setting audio parameters.");
ret = snd_pcm_hw_params(devc->capture_handle, devc->hw_params);
if (ret < 0) {
sr_err("Can't set parameters: %s.", snd_strerror(ret));
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return SR_ERR;
}
sr_dbg("Preparing audio interface for use.");
ret = snd_pcm_prepare(devc->capture_handle);
if (ret < 0) {
sr_err("Can't prepare audio interface for use: %s.",
snd_strerror(ret));
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return SR_ERR;
}
count = snd_pcm_poll_descriptors_count(devc->capture_handle);
if (count < 1) {
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sr_err("Unable to obtain poll descriptors count.");
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return SR_ERR;
}
if (!(devc->ufds = g_try_malloc(count * sizeof(struct pollfd)))) {
sr_err("Failed to malloc ufds.");
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return SR_ERR_MALLOC;
}
sr_spew("Getting %d poll descriptors.", count);
ret = snd_pcm_poll_descriptors(devc->capture_handle, devc->ufds, count);
if (ret < 0) {
sr_err("Unable to obtain poll descriptors: %s.",
snd_strerror(ret));
g_free(devc->ufds);
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return SR_ERR;
}
/* Send header packet to the session bus. */
std_session_send_df_header(cb_data, LOG_PREFIX);
/* Poll every 10ms, or whenever some data comes in. */
sr_source_add(devc->ufds[0].fd, devc->ufds[0].events, 10,
alsa_receive_data, (void *)sdi);
// g_free(devc->ufds); /* FIXME */
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return SR_OK;
}
static int dev_acquisition_stop(struct sr_dev_inst *sdi, void *cb_data)
{
struct sr_datafeed_packet packet;
struct dev_context *devc;
devc = sdi->priv;
devc->cb_data = cb_data;
sr_source_remove(devc->ufds[0].fd);
/* Send end packet to the session bus. */
sr_dbg("Sending SR_DF_END packet.");
packet.type = SR_DF_END;
sr_session_send(cb_data, &packet);
return SR_OK;
}
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SR_PRIV struct sr_dev_driver alsa_driver_info = {
.name = "alsa",
.longname = "ALSA driver",
.api_version = 1,
.init = init,
.cleanup = cleanup,
.scan = scan,
.dev_list = dev_list,
.dev_clear = dev_clear,
.config_get = config_get,
.config_set = config_set,
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.config_list = config_list,
.dev_open = dev_open,
.dev_close = dev_close,
.dev_acquisition_start = dev_acquisition_start,
.dev_acquisition_stop = dev_acquisition_stop,
.priv = NULL,
};