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