/* * 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 "libsigrok.h" #include "libsigrok-internal.h" #include "protocol.h" /* * There is no way to get a list of supported samplerates from ALSA. We could * use the 'plughw' interface of ALSA, in which case any format and/or * samplerate conversion would be performed by ALSA. However, we are interested * in the hardware capabilities, and have the infrastructure in sigrok to do so. * We therefore use the 'hw' interface. The downside is that the code gets a * little bulkier, as we have to keep track of the hardware capabilities, and * only use those that the hardware supports. Case in point, ALSA will not give * us a list of capabilities; we have to test for each one individually. Hence, * we keep lists of the capabilities we are interested in. */ static const unsigned int rates[] = { 5512, 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000, 64000, 88200, 96000, 176400, 192000, 384000, 768000, /* Yes, there are sound cards that go this high. */ }; static void alsa_scan_handle_dev(GSList **devices, const char *cardname, const char *alsaname, struct sr_dev_driver *di, snd_pcm_info_t *pcminfo) { struct drv_context *drvc = NULL; struct sr_dev_inst *sdi = NULL; struct dev_context *devc = NULL; struct sr_channel *ch; int ret; unsigned int i, offset, channels, minrate, maxrate, rate; uint64_t hwrates[ARRAY_SIZE(rates)]; uint64_t *devrates = NULL; snd_pcm_t *temp_handle = NULL; snd_pcm_hw_params_t *hw_params = NULL; char p_name[32]; drvc = di->priv; /* * Get hardware parameters: * The number of channels, for example, are our sigrok channels. Getting * this information needs a detour. We need to open the device, then * query it and/or test different parameters. A side-effect of is that * we create a snd_pcm_hw_params_t object. We take advantage of the * situation, and pass this object in our dev_context->hw_params, * eliminating the need to free() it and malloc() it later. */ ret = snd_pcm_open(&temp_handle, alsaname, SND_PCM_STREAM_CAPTURE, 0); if (ret < 0) { sr_err("Cannot open device: %s.", snd_strerror(ret)); goto scan_error_cleanup; } ret = snd_pcm_hw_params_malloc(&hw_params); if (ret < 0) { sr_err("Error allocating hardware parameter structure: %s.", snd_strerror(ret)); goto scan_error_cleanup; } ret = snd_pcm_hw_params_any(temp_handle, hw_params); if (ret < 0) { sr_err("Error initializing hardware parameter structure: %s.", snd_strerror(ret)); goto scan_error_cleanup; } snd_pcm_hw_params_get_channels_max(hw_params, &channels); /* * We need to test if each samplerate between min and max is supported. * Unfortunately, ALSA won't just throw a list at us. */ snd_pcm_hw_params_get_rate_min(hw_params, &minrate, 0); snd_pcm_hw_params_get_rate_max(hw_params, &maxrate, 0); for (i = 0, offset = 0; i < ARRAY_SIZE(rates); i++) { rate = rates[i]; if (rate < minrate) continue; if (rate > maxrate) break; ret = snd_pcm_hw_params_test_rate(temp_handle, hw_params, rate, 0); if (ret >= 0) hwrates[offset++] = rate; } hwrates[offset++] = 0; if ((ret = snd_pcm_close(temp_handle)) < 0) sr_err("Failed to close device: %s.", snd_strerror(ret)); temp_handle = NULL; /* * Now we are done querying the hardware parameters. * If we made it here, we know everything we want to know, and it's * time to create our sigrok device. */ sr_info("Device %s has %d channels.", alsaname, channels); if (!(sdi = sr_dev_inst_new(0, SR_ST_INACTIVE, "ALSA:", cardname, snd_pcm_info_get_name(pcminfo)))) { sr_err("Failed to create device instance."); goto scan_error_cleanup; } if (!(devc = g_try_malloc0(sizeof(struct dev_context)))) { sr_err("Device context malloc failed."); goto scan_error_cleanup; } if (!(devrates = g_try_malloc(offset * sizeof(uint64_t)))) { sr_err("Samplerate list malloc failed."); goto scan_error_cleanup; } devc->hwdev = g_strdup(alsaname); devc->num_channels = channels; devc->hw_params = hw_params; memcpy(devrates, hwrates, offset * sizeof(uint64_t)); devc->samplerates = devrates; sdi->priv = devc; sdi->driver = di; for (i = 0; i < devc->num_channels; i++) { snprintf(p_name, sizeof(p_name), "Ch_%d", i); if (!(ch = sr_channel_new(i, SR_CHANNEL_ANALOG, TRUE, p_name))) goto scan_error_cleanup; sdi->channels = g_slist_append(sdi->channels, ch); } drvc->instances = g_slist_append(drvc->instances, sdi); *devices = g_slist_append(*devices, sdi); return; scan_error_cleanup: if (devc) { if (devc->hwdev) g_free(devc->hwdev); g_free(devc); } if (devrates) g_free(devrates); if (sdi) sr_dev_inst_free(sdi); if (hw_params) snd_pcm_hw_params_free(hw_params); if (temp_handle) if ((ret = snd_pcm_close(temp_handle)) < 0) { sr_err("Failed to close device: %s.", snd_strerror(ret)); } } /** * Scan all alsa devices, and translate them to sigrok devices. * * Each alsa device (not alsa card) gets its own sigrok device. * * For example, * hw:1,0 == sigrok device 0 * hw:1,1 == sigrok device 1 * hw:2,0 == sigrok device 2 * hw:2,1 == sigrok device 3 * hw:2,2 == sigrok device 4 * [...] * * We don't currently look at alsa subdevices. We only use subdevice 0. * Every input device will have its own channels (left, right, etc). Each of * those channels gets mapped to a different sigrok channel. A device with 4 * channels will have 4 channels from sigrok's perspective. */ SR_PRIV GSList *alsa_scan(GSList *options, struct sr_dev_driver *di) { GSList *devices = NULL; snd_ctl_t *handle; int card, ret, dev; snd_ctl_card_info_t *info; snd_pcm_info_t *pcminfo; const char *cardname; char hwcard[32], hwdev[32]; /* TODO */ (void)options; if ((ret = snd_ctl_card_info_malloc(&info)) < 0) { sr_dbg("Failed to malloc card info: %s.", snd_strerror(ret)); return NULL; } if ((ret = snd_pcm_info_malloc(&pcminfo) < 0)) { sr_dbg("Cannot malloc pcm info: %s.", snd_strerror(ret)); return NULL; } card = -1; while (snd_card_next(&card) >= 0 && card >= 0) { snprintf(hwcard, sizeof(hwcard), "hw:%d", card); if ((ret = snd_ctl_open(&handle, hwcard, 0)) < 0) { sr_dbg("Cannot open (%d): %s.", card, snd_strerror(ret)); continue; } if ((ret = snd_ctl_card_info(handle, info)) < 0) { sr_dbg("Cannot get hardware info (%d): %s.", card, snd_strerror(ret)); if ((ret = snd_ctl_close(handle)) < 0) { sr_dbg("Cannot close device (%d): %s.", card, snd_strerror(ret)); } continue; } dev = -1; while (snd_ctl_pcm_next_device(handle, &dev) >= 0 && dev >= 0) { snprintf(hwdev, sizeof(hwdev), "%s,%d", hwcard, dev); /* * TODO: We always use subdevice 0, but we have yet to * explore the possibilities opened up by other * subdevices. Most hardware only has subdevice 0. */ snd_pcm_info_set_device(pcminfo, dev); snd_pcm_info_set_subdevice(pcminfo, 0); snd_pcm_info_set_stream(pcminfo, SND_PCM_STREAM_CAPTURE); if ((ret = snd_ctl_pcm_info(handle, pcminfo)) < 0) { sr_dbg("Cannot get device info (%s): %s.", hwdev, snd_strerror(ret)); continue; } cardname = snd_ctl_card_info_get_name(info); sr_info("card %d: %s [%s], device %d: %s [%s]", card, snd_ctl_card_info_get_id(info), cardname, dev, snd_pcm_info_get_id(pcminfo), snd_pcm_info_get_name(pcminfo)); alsa_scan_handle_dev(&devices, cardname, hwdev, di, pcminfo); } if ((ret = snd_ctl_close(handle)) < 0) { sr_dbg("Cannot close device (%d): %s.", card, snd_strerror(ret)); } } snd_pcm_info_free(pcminfo); snd_ctl_card_info_free(info); return devices; } /** * Set the samplerate of the ALSA device. * * Changes the samplerate of the given ALSA device if the specified samplerate * is supported by the hardware. * * The new samplerate is recorded, but it is not applied to the hardware. The * samplerate is applied to the hardware only when acquisition is started via * dev_acquisition_start(), and cannot be changed during acquisition. To change * the samplerate, several steps are needed: * * 1) If acquisition is running, it must first be stopped. * 2) dev_config_set() must be called with the new samplerate. * 3) When starting a new acquisition, the new samplerate is applied. * */ SR_PRIV int alsa_set_samplerate(const struct sr_dev_inst *sdi, uint64_t newrate) { struct dev_context *devc; size_t i; uint64_t rate = 0; if (!(devc = sdi->priv)) return SR_ERR_ARG; i = 0; do { if (newrate == devc->samplerates[i]) { rate = newrate; break; } } while (devc->samplerates[i++] != 0); if (!rate) { sr_err("Sample rate %" PRIu64 " not supported.", newrate); return SR_ERR_ARG; } devc->cur_samplerate = rate; return SR_OK; } SR_PRIV int alsa_receive_data(int fd, int revents, void *cb_data) { struct sr_dev_inst *sdi; struct dev_context *devc; struct sr_datafeed_packet packet; struct sr_datafeed_analog analog; int16_t inbuf[4096]; int i, x, count, offset, samples_to_get; int16_t tmp16; const float s16norm = 1 / (float)(1 << 15); (void)fd; (void)revents; sdi = cb_data; devc = sdi->priv; memset(&analog, 0, sizeof(struct sr_datafeed_analog)); memset(inbuf, 0, sizeof(inbuf)); samples_to_get = MIN(4096 / 4, devc->limit_samples); sr_spew("Getting %d samples from audio device.", samples_to_get); count = snd_pcm_readi(devc->capture_handle, inbuf, samples_to_get); if (count < 0) { sr_err("Failed to read samples: %s.", snd_strerror(count)); return FALSE; } else if (count != samples_to_get) { sr_spew("Only got %d/%d samples.", count, samples_to_get); } analog.data = g_try_malloc0(count * sizeof(float) * devc->num_channels); if (!analog.data) { sr_err("Failed to malloc sample buffer."); return FALSE; } offset = 0; /* * It's impossible to know what voltage levels the soundcard handles. * Some handle 0 dBV rms, some 0dBV peak-to-peak, +4dbmW (600 ohm), etc * Each of these corresponds to a different voltage, and there is no * mechanism to determine this voltage. The best solution is to send all * audio data as a normalized float, and let the frontend or user worry * about the calibration. */ for (i = 0; i < count; i += devc->num_channels) { for (x = 0; x < devc->num_channels; x++) { tmp16 = inbuf[i + x]; analog.data[offset++] = tmp16 * s16norm; } } /* Send a sample packet with the analog values. */ analog.channels = sdi->channels; analog.num_samples = count; analog.mq = SR_MQ_VOLTAGE; /* FIXME */ analog.unit = SR_UNIT_VOLT; /* FIXME */ packet.type = SR_DF_ANALOG; packet.payload = &analog; sr_session_send(devc->cb_data, &packet); g_free(analog.data); devc->num_samples += count; /* Stop acquisition if we acquired enough samples. */ if (devc->limit_samples && devc->num_samples >= devc->limit_samples) { sr_info("Requested number of samples reached."); sdi->driver->dev_acquisition_stop(sdi, cb_data); } return TRUE; }