]> sigrok.org Git - libsigrok.git/commitdiff
Preliminary support to use soundcard as oscilloscope via SDL2
authorTomas Mudrunka <redacted>
Tue, 6 Feb 2024 20:07:01 +0000 (21:07 +0100)
committerSoeren Apel <redacted>
Fri, 9 Feb 2024 16:22:04 +0000 (17:22 +0100)
Makefile.am
configure.ac
src/hardware/sdl2/api.c [new file with mode: 0644]
src/hardware/sdl2/protocol.h [new file with mode: 0644]

index 62aca8ac95ad0830b9b35971fab21b28c8c6e546..2fc450777c98e4818e548255d5f42126a25db2a6 100644 (file)
@@ -341,6 +341,11 @@ src_libdrivers_la_SOURCES += \
        src/hardware/devantech-eth008/protocol.c \
        src/hardware/devantech-eth008/api.c
 endif
+if HW_SDL2
+src_libdrivers_la_SOURCES += \
+       src/hardware/sdl2/protocol.h \
+       src/hardware/sdl2/api.c
+endif
 if HW_DREAMSOURCELAB_DSLOGIC
 src_libdrivers_la_SOURCES += \
        src/hardware/dreamsourcelab-dslogic/protocol.h \
index 5c30a8165f383484c07c0eab6210d085b7a376a7..9d1ff751eb0f67a8dbfabd1ff8dd463a1f2b992a 100644 (file)
@@ -113,6 +113,8 @@ SR_ARG_OPT_PKG([libserialport], [LIBSERIALPORT], ,
 
 SR_ARG_OPT_PKG([libftdi], [LIBFTDI], , [libftdi1 >= 1.0])
 
+SR_ARG_OPT_PKG([libsdl2], [LIBSDL], , [sdl2 >= 2.0])
+
 # pkg-config file names: MinGW/MacOSX: hidapi; Linux: hidapi-hidraw/-libusb
 SR_ARG_OPT_PKG([libhidapi], [LIBHIDAPI], ,
        [hidapi >= 0.8.0], [hidapi-hidraw >= 0.8.0], [hidapi-libusb >= 0.8.0])
@@ -373,6 +375,7 @@ SR_DRIVER([Saleae Logic16], [saleae-logic16], [libusb])
 SR_DRIVER([Saleae Logic Pro], [saleae-logic-pro], [libusb])
 SR_DRIVER([SCPI DMM], [scpi-dmm])
 SR_DRIVER([SCPI PPS], [scpi-pps])
+SR_DRIVER([sdl2], [sdl2], [libsdl2])
 SR_DRIVER([serial DMM], [serial-dmm], [serial_comm])
 SR_DRIVER([serial LCR], [serial-lcr], [serial_comm])
 SR_DRIVER([Siglent SDS], [siglent-sds])
diff --git a/src/hardware/sdl2/api.c b/src/hardware/sdl2/api.c
new file mode 100644 (file)
index 0000000..835a755
--- /dev/null
@@ -0,0 +1,344 @@
+/*
+ * This file is part of the libsigrok project.
+ *
+ * Copyright (C) 2022 Tomas Mudrunka <harviecz@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 3 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, see <http://www.gnu.org/licenses/>.
+ */
+
+#include <config.h>
+#include "protocol.h"
+#include <SDL2/SDL.h>
+
+#define INPUT_BUFFER_SIZE 65536
+
+static const uint32_t drvopts[] = {
+       SR_CONF_OSCILLOSCOPE,
+       SR_CONF_LOGIC_ANALYZER,
+};
+
+static const uint32_t devopts[] = {
+       SR_CONF_LIMIT_SAMPLES | SR_CONF_SET,
+       SR_CONF_SAMPLERATE | SR_CONF_GET,
+};
+
+static const char *channel_names[] = {
+       //Channel names for 7.1 DS Audio:
+       //Front-Left, Front-Right, Center, LowFreq, Surround-Left, Surround-Right, Hearing-Impaired, Visualy-Impaired, etc...
+       "FL", "FR", "CE", "LF", "SL", "SR", "HI", "VI", "CL", "CR", "RSL", "RSR", "CH13", "CH14", "CH15", "CH16", "PLSSTOP", "SRSLY",
+};
+
+int SDL_GetAudioDeviceSpec_open(int index, int iscapture, SDL_AudioSpec *spec);
+int SDL_GetAudioDeviceSpec_open(int index, int iscapture, SDL_AudioSpec *spec)
+{
+       //ALSA does not allow to fully read specs of device without opening it.
+       //This wrapper tries to open device when SDL_GetAudioDeviceSpec() reports device to have 0 channels.
+       //See https://github.com/libsdl-org/SDL/blob/237348c772b4ff0e758ace83f471dbf8570535e2/src/audio/alsa/SDL_alsa_audio.c#L759
+
+       int ret = SDL_GetAudioDeviceSpec(index, iscapture, spec);
+       if(!ret && spec->channels == 0) {
+               sr_err("Failed SDL_GetAudioDeviceSpec(), trying to open device to get specs.");
+               SDL_AudioDeviceID d;
+               d = SDL_OpenAudioDevice(SDL_GetAudioDeviceName(index, iscapture), iscapture, spec, spec, SDL_AUDIO_ALLOW_ANY_CHANGE);
+               if(d) SDL_CloseAudioDevice(d);
+       }
+       return ret;
+}
+
+static int init(struct sr_dev_driver *di, struct sr_context *sr_ctx)
+{
+       SDL_Init(SDL_INIT_AUDIO);
+       return std_init(di, sr_ctx);
+}
+
+static int cleanup(const struct sr_dev_driver *di)
+{
+       SDL_Quit();
+       return std_cleanup(di);
+}
+
+static GSList *scan(struct sr_dev_driver *di, GSList *options)
+{
+       (void)options;
+
+       GSList                 *devices = NULL;
+       struct dev_context         *devc;
+       struct sr_dev_inst         *sdi;
+       struct sr_channel         *ch;
+       struct sr_channel_group *acg;
+
+       SDL_AudioDeviceID dev_count = SDL_GetNumAudioDevices(0);
+       SDL_AudioDeviceID dev_i;
+       SDL_AudioSpec     dev_spec;
+
+       for (dev_i = 0; dev_i < dev_count; ++dev_i) {
+               if (SDL_GetAudioDeviceSpec_open(dev_i, 1, &dev_spec)) continue;
+
+               //Create driver specific data (priv) structure for driver instance
+               devc = g_malloc0(sizeof(struct dev_context));
+               memcpy(&devc->sdl_device_spec, &dev_spec, sizeof(SDL_AudioSpec));
+               devc->sdl_device_index = dev_i;
+               devc->sdl_device_name  = SDL_GetAudioDeviceName(dev_i, 1);
+
+               //Create device instance
+               sdi         = g_malloc0(sizeof(struct sr_dev_inst));
+               sdi->status = SR_ST_INACTIVE;
+               sdi->model  = g_strdup_printf("[#%d, %dch, %dHz] %s", dev_i, dev_spec.channels, dev_spec.freq, devc->sdl_device_name);
+               sdi->priv = devc;                         //Reference to driver specific data
+               devices   = g_slist_append(devices, sdi); //Add device to list
+
+               //Create analog channel group
+               acg                 = g_malloc0(sizeof(struct sr_channel_group));
+               acg->name           = g_strdup("Analog");
+               sdi->channel_groups = g_slist_append(sdi->channel_groups, acg);
+
+               int ch_i;
+               for (ch_i = 0; ch_i < dev_spec.channels; ch_i++) {
+                       //Put new channel to group
+                       ch            = sr_channel_new(sdi, ch_i, SR_CHANNEL_ANALOG, TRUE, channel_names[ch_i]);
+                       acg->channels = g_slist_append(acg->channels, ch);
+               }
+       }
+
+       return std_scan_complete(di, devices);
+}
+
+static int dev_open(struct sr_dev_inst *sdi)
+{
+       struct dev_context *devc;
+
+       devc = sdi->priv;
+
+       //Check if SDL device is still available
+       SDL_AudioSpec dev_spec;
+       if (SDL_GetAudioDeviceSpec_open(devc->sdl_device_index, 1, &dev_spec)) return SR_ERR;
+
+       //TODO: flush buffer?
+
+       return SR_OK;
+}
+
+static int config_get(unsigned int key, GVariant **data, const struct sr_dev_inst *sdi, const struct sr_channel_group *cg)
+{
+       struct dev_context *devc;
+
+       (void)cg;
+
+       if (!sdi) return SR_ERR_ARG;
+
+       devc = sdi->priv;
+
+       switch (key) {
+               case SR_CONF_LIMIT_SAMPLES: *data = g_variant_new_uint64(devc->limit_samples); break;
+               case SR_CONF_SAMPLERATE: *data = g_variant_new_uint64(SR_HZ(devc->sdl_device_spec.freq)); break;
+               default: return SR_ERR_NA;
+       }
+
+       return SR_OK;
+}
+
+static int config_set(unsigned int key, GVariant *data, const struct sr_dev_inst *sdi, const struct sr_channel_group *cg)
+{
+       struct dev_context *devc;
+       uint64_t            num_samples;
+
+       (void)cg;
+
+       devc = sdi->priv;
+
+       switch (key) {
+               case SR_CONF_SAMPLERATE:
+                       // FIXME
+                       return SR_ERR_NA;
+               case SR_CONF_LIMIT_SAMPLES:
+                       num_samples = g_variant_get_uint64(data);
+                       sr_err("Received config to limit samples: %lu", num_samples);
+                       devc->limit_samples = num_samples;
+                       break;
+               default: return SR_ERR_NA;
+       }
+
+       return SR_OK;
+}
+
+static int config_list(unsigned int key, GVariant **data, const struct sr_dev_inst *sdi, const struct sr_channel_group *cg)
+{
+       if(cg) return SR_ERR_NA; //Cannot handle this right now
+
+       switch (key) {
+               case SR_CONF_DEVICE_OPTIONS:
+                       return STD_CONFIG_LIST(key, data, sdi, cg, NO_OPTS, drvopts, devopts);
+               default:
+                       return SR_ERR_NA;
+       }
+
+       return SR_OK;
+}
+
+static int dev_acquisition_stop(struct sr_dev_inst *sdi);
+int sdl_data_callback(int fd, int revents, void *cb_data);
+int sdl_data_callback(int fd, int revents, void *cb_data)
+{
+       (void)fd;
+       (void)revents;
+
+       struct sr_dev_inst         *sdi;
+       struct dev_context         *devc;
+       struct sr_datafeed_packet packet;
+       struct sr_datafeed_analog packet_analog;
+
+       struct sr_analog_encoding encoding;
+       struct sr_analog_meaning  meaning;
+       struct sr_analog_spec     spec;
+
+       struct sr_rational r_scale, r_offset;
+
+       sdi  = cb_data;
+       devc = sdi->priv;
+
+       if (devc->limit_samples_remaining <= 0 /* || devc->limit_samples_remaining > 65535 */ ) { //Already sent everything
+               sr_err("Loop finished");
+               std_session_send_df_end(sdi);
+               SDL_CloseAudioDevice(devc->sdl_device_handle);
+               return SR_OK;
+       }
+
+       sr_analog_init(&packet_analog, &encoding, &meaning, &spec, 0);
+
+       struct sr_channel_group *lastcg = g_slist_nth_data(sdi->channel_groups, 0);
+
+       SDL_AudioFormat sf;
+       sf = devc->sdl_device_spec.format;
+
+       //TODO: lot of stuff done here should actualy be prepared only once during aquisition start!
+
+       if(SDL_AUDIO_ISFLOAT(sf)) sr_err("SDL2 float samples are not really correctly implemented yet!");
+
+       //encoding
+       encoding.unitsize          = SDL_AUDIO_BITSIZE(sf) / 8; //???
+       encoding.is_signed         = SDL_AUDIO_ISSIGNED(sf);
+       encoding.is_float          = SDL_AUDIO_ISFLOAT(sf);
+       encoding.is_bigendian      = SDL_AUDIO_ISBIGENDIAN(sf);
+       encoding.digits            = 2;
+       encoding.is_digits_decimal = 1;
+       r_scale.p  = 1;
+       r_scale.q  = SDL_FORMAT_MAX_VAL(sf)/2; //Scale so that MAX signal is always +-1 volt //TODO: user configurable calibration
+       r_offset.p = SDL_AUDIO_ISSIGNED(sf) ? 0 : -1; //Center unsigned audio samples to enable negative voltages
+       r_offset.q = 1;
+       encoding.scale             = r_scale;
+       encoding.offset            = r_offset;
+       spec.spec_digits           = 2;
+
+       //meaning
+       meaning.mq       = SR_MQ_VOLTAGE;
+       meaning.unit     = SR_UNIT_VOLT;
+       meaning.mqflags  = 0;
+       meaning.channels = lastcg->channels;
+
+       //data
+       uint8_t data[INPUT_BUFFER_SIZE];
+       uint32_t requ_bytes = INPUT_BUFFER_SIZE;
+       if(requ_bytes > SDL_SAMPLES_TO_BYTES(devc->limit_samples_remaining, devc->sdl_device_spec))
+               requ_bytes = SDL_SAMPLES_TO_BYTES(devc->limit_samples_remaining, devc->sdl_device_spec);
+
+       uint32_t recv_bytes = 0;
+       while(!recv_bytes) {
+               recv_bytes = SDL_DequeueAudio(devc->sdl_device_handle, data, requ_bytes);
+               SDL_Delay(100);
+       }
+
+       packet_analog.data        = data;
+       packet_analog.num_samples = 4;
+       packet_analog.encoding    = &encoding;
+       packet_analog.meaning     = &meaning;
+       packet_analog.spec        = &spec;
+       packet_analog.num_samples = SDL_BYTES_TO_SAMPLES(recv_bytes, devc->sdl_device_spec);
+
+       //packet
+       packet.type    = SR_DF_ANALOG;
+       packet.payload = &packet_analog;
+
+       sr_session_send(sdi, &packet);
+       devc->limit_samples_remaining -= packet_analog.num_samples;
+
+       return G_SOURCE_CONTINUE;
+}
+
+static int dev_acquisition_start(const struct sr_dev_inst *sdi)
+{
+       struct dev_context *devc;
+       devc = sdi->priv;
+
+       devc->limit_samples_remaining = devc->limit_samples;
+       sr_err("Limiting samples to %lu", devc->limit_samples_remaining);
+
+       //Initialize SDL2 recording
+       devc->sdl_device_spec.callback=NULL;
+       devc->sdl_device_spec.samples = SDL_BYTES_TO_SAMPLES(INPUT_BUFFER_SIZE, devc->sdl_device_spec);
+
+       devc->sdl_device_handle = SDL_OpenAudioDevice(devc->sdl_device_name, 1, &devc->sdl_device_spec, NULL, 0);
+       if(!devc->sdl_device_handle) {
+               sr_err("Could not open SDL2 device for capture!");
+               return SR_ERR;
+       }
+       SDL_PauseAudioDevice(devc->sdl_device_handle, 0);
+
+       sr_session_source_add(sdi->session, -1, 0, 100, sdl_data_callback, (struct sr_dev_inst *)sdi);
+
+       std_session_send_df_header(sdi);
+
+       return SR_OK;
+}
+
+static int dev_acquisition_stop(struct sr_dev_inst *sdi)
+{
+       sr_err("STOP Initiated");
+
+       struct dev_context *devc;
+       devc = sdi->priv;
+
+       devc->limit_samples_remaining = 0;
+
+       return SR_OK;
+}
+
+static struct sr_dev_driver sdl2_driver_info = {
+       .name        = "sdl2",
+       .longname    = "SoundCard Audio Capture using SDL2",
+       .api_version = 1,
+       .init        = init,
+       .cleanup     = cleanup,
+
+       //scan
+       .scan      = scan,
+       .dev_list  = std_dev_list,
+       .dev_clear = std_dev_clear,
+
+       //config
+       .config_get  = config_get,
+       .config_set  = config_set,
+       .config_list = config_list,
+
+       //open
+       .dev_open  = dev_open,
+       .dev_close = std_dummy_dev_close,
+
+       //acq
+       .dev_acquisition_start = dev_acquisition_start,
+       .dev_acquisition_stop  = dev_acquisition_stop,
+
+       //inst
+       .context = NULL,
+};
+SR_REGISTER_DEV_DRIVER(sdl2_driver_info);
diff --git a/src/hardware/sdl2/protocol.h b/src/hardware/sdl2/protocol.h
new file mode 100644 (file)
index 0000000..49134c5
--- /dev/null
@@ -0,0 +1,46 @@
+/*
+ * This file is part of the libsigrok project.
+ *
+ * Copyright (C) 2022 Tomas Mudrunka <harviecz@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 3 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, see <http://www.gnu.org/licenses/>.
+ */
+
+#ifndef LIBSIGROK_HARDWARE_SDL2_PROTOCOL_H
+#define LIBSIGROK_HARDWARE_SDL2_PROTOCOL_H
+
+#define SDL_SAMPLES_TO_BYTES(bytes,spec) ((bytes)*((SDL_AUDIO_BITSIZE((spec).format)/8)*((spec).channels)))
+#define SDL_BYTES_TO_SAMPLES(bytes,spec) ((bytes)/SDL_SAMPLES_TO_BYTES(1,(spec)))
+#define SDL_FORMAT_MAX_VAL(f) (1ull<<(SDL_AUDIO_BITSIZE(f)-SDL_AUDIO_ISSIGNED(f)))
+
+#include <stdint.h>
+#include <string.h>
+#include <glib.h>
+#include <libsigrok/libsigrok.h>
+#include "libsigrok-internal.h"
+#include <SDL2/SDL.h>
+
+#define LOG_PREFIX "sdl2-audio-interface"
+
+struct dev_context {
+       const char*       sdl_device_name;
+       SDL_AudioDeviceID sdl_device_index;
+       SDL_AudioSpec     sdl_device_spec;
+       SDL_AudioDeviceID sdl_device_handle;
+
+       uint64_t limit_samples;
+       uint64_t limit_samples_remaining;
+};
+
+#endif