Hello,
inspired by xoscope http://xoscope.sourceforge.net/
i've created libsigrok driver which enables oscilloscope-like
acquisition via arbitrary soundcard.
My goal is to be able to decode 9600 baud UART signal captured by
soundcard in pulseview.
I've tested it only on linux, but it uses SDL2 library and therefore
should be quite portable (most likely without any modifications, can
someone please try to build on windows? there is even sdl2 on android,
but i am not sure how that works).
I think the concept is relatively usefull. Definitely for education, but
i beleive it can be even useful for serious work, since high-end
soundcards now have lots of channels and samplerates in hundreds of kHz.
Also on raspberry pi and similar devices you can use rather interresting
I2S ADCs with specs far behind your traditional audio needs (even with
DC coupling possibilities). Some people even managed to modify cheap usb
soundcards to remove DC blocking capacitors enabling them to do DC
measurements.
i think in future it might be possible to add continuous acquisition,
downsampling for long-term measurements and even triggers (should be
quite easy) and more.
Also there's PR #185 with discussion at github:
https://github.com/sigrokproject/libsigrok/pull/185
---
Makefile.am | 5 +
configure.ac | 3 +
src/hardware/sdl2/api.c | 327 +++++++++++++++++++++++++++++++++++
src/hardware/sdl2/protocol.h | 46 +++++
4 files changed, 381 insertions(+)
create mode 100644 src/hardware/sdl2/api.c
create mode 100644 src/hardware/sdl2/protocol.h
diff --git a/Makefile.am b/Makefile.am
index 280cf64d..eb0248f7 100644
--- a/Makefile.am
+++ b/Makefile.am
@@ -321,6 +321,11 @@ src_libdrivers_la_SOURCES += \
src/hardware/demo/protocol.c \
src/hardware/demo/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 \
diff --git a/configure.ac b/configure.ac
index 5c480818..0de4155f 100644
--- a/configure.ac
+++ b/configure.ac
@@ -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])
@@ -353,6 +355,7 @@ 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
index 00000000..7e25d7c9
--- /dev/null
+++ b/src/hardware/sdl2/api.c
@@ -0,0 +1,327 @@
+/*
+ * This file is part of the libsigrok project.
+ *
+ * Copyright (C) 2022 Tomas Mudrunka <harvi...@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",
+};
+
+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(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(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
index 00000000..49134c59
--- /dev/null
+++ b/src/hardware/sdl2/protocol.h
@@ -0,0 +1,46 @@
+/*
+ * This file is part of the libsigrok project.
+ *
+ * Copyright (C) 2022 Tomas Mudrunka <harvi...@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
--
2.40.1
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