/* * This file is part of the libsigrok project. * * Copyright (C) 2011 Daniel Ribeiro * Copyright (C) 2012 Uwe Hermann * Copyright (C) 2012 Alexandru Gagniuc * * 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 #include #include #include "libsigrok.h" #include "libsigrok-internal.h" #include "protocol.h" 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; static void clear_helper(void *priv) { struct dev_context *devc; devc = priv; snd_pcm_hw_params_free(devc->hw_params); devc->hw_params = NULL; g_free((void *)devc->samplerates); devc->samplerates = NULL; } static int dev_clear(void) { 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)); 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.", snd_strerror(ret)); return SR_ERR; } sdi->status = SR_ST_ACTIVE; 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_channel_group *cg) { struct dev_context *devc; (void)cg; 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_channel_group *cg) { struct dev_context *devc; (void)cg; 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_channel_group *cg) { struct dev_context *devc; GVariant *gvar; GVariantBuilder gvb; int i; (void)cg; 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; case SR_CONF_SAMPLERATE: if (!sdi || !sdi->priv) return SR_ERR_ARG; devc = sdi->priv; if (!devc->samplerates) { 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); break; default: return SR_ERR_NA; } return SR_OK; } static int dev_acquisition_start(const struct sr_dev_inst *sdi, void *cb_data) { struct dev_context *devc; int count, ret; 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)); return SR_ERR; } /* FIXME: Hardcoded for 16bits. */ if (SND_PCM_FORMAT_S16 == SND_PCM_FORMAT_S16_LE) endianness = "little endian"; else 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)); 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)); return SR_ERR; } sr_dbg("Setting audio channel count to %d.", devc->num_channels); ret = snd_pcm_hw_params_set_channels(devc->capture_handle, devc->hw_params, devc->num_channels); if (ret < 0) { sr_err("Can't set channel count: %s.", snd_strerror(ret)); 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)); 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)); return SR_ERR; } count = snd_pcm_poll_descriptors_count(devc->capture_handle); if (count < 1) { sr_err("Unable to obtain poll descriptors count."); return SR_ERR; } if (!(devc->ufds = g_try_malloc(count * sizeof(struct pollfd)))) { sr_err("Failed to malloc ufds."); 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); 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 */ 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; } 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, .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, };