From a7179b2b36688c86ed3d055d0c4119ac2d63a782 Mon Sep 17 00:00:00 2001
From: Tomas Mudrunka <tomas.mudru...@gmail.com>
Date: Tue, 6 Feb 2024 21:07:01 +0100
Subject: [PATCH v5] Support use of soundcard as oscilloscope via SDL2
PATCH v5:
- fixed freeze when using pulse protocol with pipewire
- fixed JACK compatibility (forcing integer samples for now)
PATCH v4 has modifications based on IRC discussion with abraxa_
Most notably:
- Fixed datatype for result of SDL_GetNumAudioDevices()
- Using SR_UNIT_UNITLESS instead of SR_UNIT_VOLT
- Audio subsystem initialization and scanning is more verbose,
giving out useful details for debuging purposes
- Better error handling
- Code was formated to comply with sigrok rules
Signed-off-by: Tomas Mudrunka <mudru...@spoje.net>
---
Makefile.am | 5 +
configure.ac | 3 +
src/hardware/sdl2/api.c | 390 +++++++++++++++++++++++++++++++++++
src/hardware/sdl2/protocol.h | 49 +++++
4 files changed, 447 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 62aca8ac..2fc45077 100644
--- a/Makefile.am
+++ b/Makefile.am
@@ -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 \
diff --git a/configure.ac b/configure.ac
index 5c30a816..9d1ff751 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])
@@ -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
index 00000000..1c2e442c
--- /dev/null
+++ b/src/hardware/sdl2/api.c
@@ -0,0 +1,390 @@
+/*
+ * This file is part of the libsigrok project.
+ *
+ * Copyright (C) 2022-2024 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",
+};
+
+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);
+ }
+
+ //This is hack to disable floating point samples before we figure out
how to properly pass them to sigrok
+ if(!ret && SDL_AUDIO_ISFLOAT(spec->format)) spec->format =
AUDIO_S32SYS;
+
+ return ret;
+}
+
+static int init(struct sr_dev_driver *di, struct sr_context *sr_ctx)
+{
+ SDL_version compiled;
+ SDL_version linked;
+
+ SDL_VERSION(&compiled);
+ SDL_GetVersion(&linked);
+
+ sr_err("Compiled with SDL v%u.%u.%u and linked with SDL v%u.%u.%u",
+ compiled.major, compiled.minor, compiled.patch,
+ linked.major, linked.minor, linked.patch);
+
+ if (SDL_Init(SDL_INIT_AUDIO)) {
+ sr_err("Audio init failed: %s", SDL_GetError());
+ return SR_ERR;
+ }
+ 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;
+
+ int dev_count = SDL_GetNumAudioDevices(1);
+ int dev_i;
+ SDL_AudioSpec dev_spec;
+
+ const char *audio_driver = SDL_GetCurrentAudioDriver();
+ sr_err("Audio driver %s found %d capture (and %d playback) devices",
+ audio_driver ? audio_driver : "UNKNOWN", dev_count,
+ SDL_GetNumAudioDevices(0));
+
+ 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("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_UNITLESS;
+ 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;
+ recv_bytes = SDL_DequeueAudio(devc->sdl_device_handle, data,
requ_bytes);
+
+ 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;
+
+ if(packet_analog.num_samples) 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 device for capture!");
+ return SR_ERR;
+ }
+ //SDL_ClearQueuedAudio(devc->sdl_device_handle);
+ 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..c383ba49
--- /dev/null
+++ b/src/hardware/sdl2/protocol.h
@@ -0,0 +1,49 @@
+/*
+ * 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
--
2.44.0
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