@@ -269,6 +269,7 @@ External library support:
--enable-libshine enable fixed-point MP3 encoding via libshine [no]
--enable-libsmbclient enable Samba protocol via libsmbclient [no]
--enable-libsnappy enable Snappy compression, needed for hap encoding [no]
+ --enable-libsoapysdr enable SoapySDR, needed for connecting to SDR HW [no]
--enable-libsoxr enable Include libsoxr resampling [no]
--enable-libspeex enable Speex de/encoding via libspeex [no]
--enable-libsrt enable Haivision SRT protocol via libsrt [no]
@@ -1888,6 +1889,7 @@ EXTERNAL_LIBRARY_LIST="
libshine
libsmbclient
libsnappy
+ libsoapysdr
libsoxr
libspeex
libsrt
@@ -3602,6 +3604,7 @@ oss_outdev_deps_any="sys_soundcard_h"
pulse_indev_deps="libpulse"
pulse_outdev_deps="libpulse"
sdl2_outdev_deps="sdl2"
+sdr_indev_deps="libsoapysdr"
sndio_indev_deps="sndio"
sndio_outdev_deps="sndio"
v4l2_indev_deps_any="linux_videodev2_h sys_videoio_h"
@@ -6776,6 +6779,7 @@ enabled libshine && require_pkg_config libshine shine shine/layer3.h sh
enabled libsmbclient && { check_pkg_config libsmbclient smbclient libsmbclient.h smbc_init ||
require libsmbclient libsmbclient.h smbc_init -lsmbclient; }
enabled libsnappy && require libsnappy snappy-c.h snappy_compress -lsnappy -lstdc++
+enabled libsoapysdr && require libsoapysdr SoapySDR/Device.h SoapySDRDevice_enumerate -lSoapySDR
enabled libsoxr && require libsoxr soxr.h soxr_create -lsoxr
enabled libssh && require_pkg_config libssh libssh libssh/sftp.h sftp_init
enabled libspeex && require_pkg_config libspeex speex speex/speex.h speex_decoder_init
@@ -899,6 +899,96 @@ the script is directly played, the actual times will match the absolute
timestamps up to the sound controller's clock accuracy, but if the user
somehow pauses the playback or seeks, all times will be shifted accordingly.
+@section sdr / sdrfile
+
+Software Defined Radio Demuxer. The sdr demuxer will demux radio streams
+through the use of a libsoapy compatible software defined radio. sdrfile
+will do the same from a file.
+The short seek forward and backward commands can be used to select the next and
+previous radio stations. Seeking to specific radio stations and similar will be
+supported in the future. This has been tested with a RTL-SDR blog V3 and sdrplay RSP1A,
+it should work with anything else libsoapy supports but may need minor tweaks.
+
+@table @option
+
+@item block_size
+size of FFT to use, leave at 0 to choose automatically
+
+@item video_size
+Enable vissualization with the specified video resolution. Default is
+disabled
+
+@item mode
+
+@table @samp
+
+@item single_mode
+Demodulate 1 station
+
+@item all_mode
+Demodulate all stations in view
+
+@end table
+
+@item station_freq
+Initial station to pick, if multiple are probed.
+
+@item driver
+libsoapy driver to use. Leave empty to attempt autodetection.
+
+@item sdr_sr
+SDR sample rate, this must be a value supported by the hardware. Leave 0 to attempt
+autodetection
+
+@item sdr_freq
+initial SDR center frequency, this must be a value supported by the hardware
+
+@item min_freq
+Minimum frequency, leave at 0 for autodetection
+
+@item max_freq
+Maximum frequency, leave at 0 for autodetection
+
+@item dumpurl
+URL to dump RAW SDR stream to, this can be played back later with sdrfile. Useful
+for debuging.
+
+@item kbd_alpha
+Kaiser Bessel derived window parameter
+
+@item am_mode
+AM Demodulation method. Several different methods are supported.
+@table @samp
+@item am_inphase
+Demodulate mono signal that is in phase with the carrier. This works well if there is one
+transmitter at the frequency and there is nothing disturbing the signal.
+
+@item am_midside
+Demodulate AM into a stereo signal so that the mid is from the in phase and side is from the
+quadrature signal. This can be used if there are Multiple transmitters that transmit at the same
+frequency. Or just for fun.
+
+@item am_envelope
+Demodulate mono signal that is basically the energy of the spectrum.
+
+@end table
+
+@item fm_emphasis
+FM Emphasis filter.
+@table @samp
+@item emphasis75us
+Used in the US (default)
+
+@item emphasis50us
+Used in the EU
+
+@item none
+No emphasis used
+
+@end table
+
+@end table
+
@section tedcaptions
JSON captions used for @url{http://www.ted.com/, TED Talks}.
@@ -43,6 +43,7 @@ OBJS-$(CONFIG_PULSE_INDEV) += pulse_audio_dec.o \
OBJS-$(CONFIG_PULSE_OUTDEV) += pulse_audio_enc.o \
pulse_audio_common.o
OBJS-$(CONFIG_SDL2_OUTDEV) += sdl2.o
+OBJS-$(CONFIG_SDR_INDEV) += sdrindev.o
OBJS-$(CONFIG_SNDIO_INDEV) += sndio_dec.o sndio.o
OBJS-$(CONFIG_SNDIO_OUTDEV) += sndio_enc.o sndio.o
OBJS-$(CONFIG_V4L2_INDEV) += v4l2.o v4l2-common.o timefilter.o
@@ -47,6 +47,7 @@ extern const FFOutputFormat ff_oss_muxer;
extern const AVInputFormat ff_pulse_demuxer;
extern const FFOutputFormat ff_pulse_muxer;
extern const FFOutputFormat ff_sdl2_muxer;
+extern const AVInputFormat ff_sdr_demuxer;
extern const AVInputFormat ff_sndio_demuxer;
extern const FFOutputFormat ff_sndio_muxer;
extern const AVInputFormat ff_v4l2_demuxer;
new file mode 100644
@@ -0,0 +1,369 @@
+/*
+ * SDR Input device / Demodulator
+ * Copyright (c) 2023 Michael Niedermayer
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg 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
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include <SoapySDR/Version.h>
+#include <SoapySDR/Device.h>
+#include <SoapySDR/Formats.h>
+
+#include "libavutil/avassert.h"
+#include "libavutil/channel_layout.h"
+#include "libavutil/fifo.h"
+#include "libavutil/intreadwrite.h"
+#include "libavutil/opt.h"
+#include "libavutil/time.h"
+#include "libavutil/thread.h"
+#include "libavutil/tx.h"
+#include "libavcodec/kbdwin.h"
+#include "libavformat/avformat.h"
+#include "libavformat/demux.h"
+#include "libavformat/internal.h"
+#include "libavformat/sdr.h"
+
+#define MAX_CHANNELS 4
+
+static int sdrindev_read_callback(SDRContext *sdr, FIFOElement *fifo_element, int remaining)
+{
+ AVFormatContext *avfmt = sdr->avfmt;
+ void *soapy_buffers[MAX_CHANNELS] = {(uint8_t*)fifo_element->halfblock + (sdr->block_size - remaining) * sdr->sample_size, NULL};
+ long long soapy_ts;
+ int soapy_flags;
+
+ int ret = SoapySDRDevice_readStream(sdr->soapy, sdr->soapyRxStream,
+ soapy_buffers,
+ remaining,
+ &soapy_flags,
+ &soapy_ts,
+ 10*1000);
+
+ if (ret == SOAPY_SDR_TIMEOUT) {
+ return AVERROR(EAGAIN);
+ } else if (ret == SOAPY_SDR_OVERFLOW) {
+ av_log(avfmt, AV_LOG_WARNING, "SOAPY OVERFLOW\n");
+ return AVERROR(EAGAIN);
+ } else if (ret < 0) {
+ av_log(avfmt, AV_LOG_ERROR, "SoapySDRDevice_readStream() Failed with (%d) %s\n", ret, SoapySDRDevice_lastError());
+ av_usleep(10*1000);
+ return AVERROR(EAGAIN);
+ }
+
+ return ret;
+}
+
+static int sdrindev_set_frequency_callback(SDRContext *sdr, int64_t freq)
+{
+ AVFormatContext *avfmt = sdr->avfmt;
+ SoapySDRDevice *soapy = sdr->soapy;
+ if (soapy) {
+ if (sdr->rtlsdr_fixes>0) {
+ // The official unmodified RTLSDR needs to be put in Direct sampling Q-ADC mode below 24Mhz and non direct sampling above
+ const char *value = freq < 24000000 ? "2" : "0";
+ if (!sdr->current_direct_samp || strcmp(value, sdr->current_direct_samp)) {
+ int ret = SoapySDRDevice_writeSetting(soapy, "direct_samp", value);
+ if (ret) {
+ av_log(avfmt, AV_LOG_WARNING, "SoapySDRDevice_writeSetting fail while adjusting direct sampling mode for RTLSDR\n");
+ } else
+ sdr->current_direct_samp = value;
+ }
+ }
+
+ if (SoapySDRDevice_setFrequency(soapy, SOAPY_SDR_RX, 0, freq, NULL) != 0) {
+ av_log(avfmt, AV_LOG_ERROR, "setFrequency fail: %s\n", SoapySDRDevice_lastError());
+ return AVERROR_EXTERNAL;
+ }
+ }
+ return 0;
+}
+
+/**
+ * Initial setup of SDR HW through soapy.
+ * This will go over available settings and match them up with what the user requested
+ * That is List available + check user settings + autoselect if the user made no choice
+ * It will also set callbacks and run the common init.
+ */
+static int sdrindev_initial_hw_setup(AVFormatContext *s)
+{
+ SDRContext *sdr = s->priv_data;
+ int ret, i;
+
+ SoapySDRRange *ranges, range;
+ SoapySDRKwargs *results = NULL;
+ SoapySDRKwargs args = {};
+ int has_agc, has_adcc;
+ SoapySDRArgInfo *arg_info = NULL;
+ size_t length;
+ char** names;
+ int64_t max_sample_rate;
+ SoapySDRDevice *soapy = NULL;
+ SoapySDRStream *soapyRxStream = NULL;
+ const char * soapy_format;
+
+ sdr->read_callback = sdrindev_read_callback;
+ sdr->set_frequency_callback = sdrindev_set_frequency_callback;
+
+ // Go over all available soapy devices
+ // Print the usable ones, and choose one unless the user has choosen one
+ results = SoapySDRDevice_enumerate(NULL, &length);
+ for (i = 0; i < length; i++) {
+ int usable = 1;
+ for (int j = 0; j < results[i].size; j++) {
+ if (!strcmp("driver", results[i].keys[j])) {
+ if (!strcmp("audio", results[i].vals[j])) {
+ usable = 0;
+ } else if (!sdr->driver_name) {
+ sdr->driver_name = av_strdup(results[i].vals[j]);
+ if (!sdr->driver_name)
+ return AVERROR(ENOMEM);
+ }
+ }
+ }
+ if (!usable)
+ continue;
+ av_log(s, AV_LOG_INFO, "Soapy enumeration %d\n", i);
+ for (int j = 0; j < results[i].size; j++) {
+ av_log(s, AV_LOG_INFO, " results %s = %s\n", results[i].keys[j], results[i].vals[j]);
+ }
+ }
+ SoapySDRKwargsList_clear(results, length);
+
+ av_log(s, AV_LOG_INFO, "Opening %s\n", sdr->driver_name);
+ if (!sdr->driver_name)
+ return AVERROR(EINVAL); //No driver specified and none found
+
+ if (sdr->rtlsdr_fixes < 0)
+ sdr->rtlsdr_fixes = !strcmp(sdr->driver_name, "rtlsdr");
+
+ SoapySDRKwargs_set(&args, "driver", sdr->driver_name);
+ sdr->soapy = soapy = SoapySDRDevice_make(&args);
+ SoapySDRKwargs_clear(&args);
+
+ if (soapy == NULL) {
+ av_log(s, AV_LOG_ERROR, "SoapySDRDevice_make fail: %s\n", SoapySDRDevice_lastError());
+ return AVERROR_EXTERNAL;
+ }
+
+ //Go over all Antennas and print them
+ names = SoapySDRDevice_listAntennas(soapy, SOAPY_SDR_RX, 0, &length);
+ av_log(s, AV_LOG_INFO, "Antennas: ");
+ for (i = 0; i < length; i++)
+ av_log(s, AV_LOG_INFO, "%s, ", names[i]);
+ av_log(s, AV_LOG_INFO, "\n");
+ SoapySDRStrings_clear(&names, length);
+
+ //Go over all Gain Elements and print them
+ names = SoapySDRDevice_listGains(soapy, SOAPY_SDR_RX, 0, &length);
+ av_log(s, AV_LOG_INFO, "Rx Gain Elements: ");
+ for (i = 0; i < length; i++)
+ av_log(s, AV_LOG_INFO, "%s, ", names[i]);
+ av_log(s, AV_LOG_INFO, "\n");
+ SoapySDRStrings_clear(&names, length);
+
+ //Inform the user if AGC is supported and setup AGC as requested by the user
+ has_agc = SoapySDRDevice_hasGainMode(soapy, SOAPY_SDR_RX, 0);
+ av_log(s, AV_LOG_INFO, "RX AGC Supported: %s\n", has_agc ? "yes" : "no");
+ if (has_agc && sdr->sdr_agc >= 0)
+ SoapySDRDevice_setGainMode(soapy, SOAPY_SDR_RX, 0, sdr->sdr_agc);
+
+ //Inform the user if automatic DC correction is supported and setup DC correction as requested by the user
+ has_adcc = SoapySDRDevice_hasDCOffsetMode(soapy, SOAPY_SDR_RX, 0);
+ av_log(s, AV_LOG_INFO, "RX Automatic DC Correction Supported: %s\n", has_adcc ? "yes" : "no");
+ if (has_adcc && sdr->sdr_adcc >= 0)
+ SoapySDRDevice_setDCOffsetMode(soapy, SOAPY_SDR_RX, 0, sdr->sdr_adcc);
+
+ //Inform the user about the Gain range available
+ range = SoapySDRDevice_getGainRange(soapy, SOAPY_SDR_RX, 0);
+ av_log(s, AV_LOG_INFO, "Rx Gain range: %f dB - %f dB\n", range.minimum, range.maximum);
+
+ //Inform the user about the Frequency ranges available, verify the range requested by the user and set the range if the user has not specified one
+ ranges = SoapySDRDevice_getFrequencyRange(soapy, SOAPY_SDR_RX, 0, &length);
+ av_log(s, AV_LOG_INFO, "Rx freq ranges: ");
+ for (i = 0; i < length; i++) {
+ double minimum = ranges[i].minimum;
+ av_log(s, AV_LOG_INFO, "[%g Hz -> %g Hz], ", ranges[i].minimum, ranges[i].maximum);
+
+ if (sdr->rtlsdr_fixes>0) {
+ //rtlsdr supports direct sampling down to 500khz, frequency range returned is wrong
+ minimum = FFMIN(minimum, 500000);
+ if (!sdr->min_freq)
+ sdr->min_freq = minimum;
+ }
+
+ if (sdr->max_freq > ranges[i].maximum)
+ continue;
+ if (sdr->min_freq && sdr->min_freq < minimum)
+ continue;
+ break;
+ }
+ av_log(s, AV_LOG_INFO, "\n");
+ if (i == length) {
+ av_log(s, AV_LOG_ERROR, "Invalid frequency range\n");
+ return AVERROR(EINVAL);
+ }
+ if (!sdr->max_freq)
+ sdr->max_freq = ranges[i].maximum;
+ if (!sdr->min_freq)
+ sdr->min_freq = ranges[i].minimum;
+ free(ranges); ranges = NULL;
+ av_log(s, AV_LOG_INFO, "frequency range: %"PRId64" - %"PRId64"\n", sdr->min_freq, sdr->max_freq);
+ //TODO do any drivers support multiple distinct frequency ranges ? if so we pick just one, thats not ideal
+
+ //Inform the user about the Sample rate ranges available, verify the rate requested by the user and choose the range if the user has not specified one
+ ranges = SoapySDRDevice_getSampleRateRange(soapy, SOAPY_SDR_RX, 0, &length);
+ max_sample_rate = 0;
+ av_log(s, AV_LOG_INFO, "SampleRate ranges: ");
+ for (i = 0; i < length; i++) {
+ av_log(s, AV_LOG_INFO, "[%g Hz -> %g Hz], ", ranges[i].minimum, ranges[i].maximum);
+ if (sdr->sdr_sample_rate &&
+ (sdr->sdr_sample_rate < ranges[i].minimum || sdr->sdr_sample_rate > ranges[i].maximum))
+ continue;
+ if (sdr->rtlsdr_fixes)
+ // 2.88 and 3.2 Mhz do not work reliable here so lets not automatically choose them, the user can override this
+ if (ranges[i].maximum > (2560000 + 2880000)/2)
+ continue;
+ max_sample_rate = FFMAX(max_sample_rate, ranges[i].maximum);
+ }
+ av_log(s, AV_LOG_INFO, "\n");
+ free(ranges); ranges = NULL;
+ if (!sdr->sdr_sample_rate) {
+ sdr->sdr_sample_rate = max_sample_rate;
+ }
+
+ // We disallow odd sample rates as they result in either center or endpoint frequencies to be non integer. No big deal but simpler is better
+ if (!max_sample_rate || sdr->sdr_sample_rate%2) {
+ av_log(s, AV_LOG_ERROR, "Invalid sdr sample rate\n");
+ return AVERROR(EINVAL);
+ }
+
+ //List the bandwidth ranges available
+ ranges = SoapySDRDevice_getBandwidthRange(soapy, SOAPY_SDR_RX, 0, &length);
+ av_log(s, AV_LOG_INFO, "Bandwidth ranges: ");
+ for (i = 0; i < length; i++) {
+ av_log(s, AV_LOG_INFO, "[%g Hz -> %g Hz], ", ranges[i].minimum, ranges[i].maximum);
+ }
+ av_log(s, AV_LOG_INFO, "\n");
+ free(ranges); ranges = NULL;
+
+ //apply settings
+ if (SoapySDRDevice_setSampleRate(soapy, SOAPY_SDR_RX, 0, sdr->sdr_sample_rate) != 0) {
+ av_log(s, AV_LOG_ERROR, "setSampleRate fail: %s\n", SoapySDRDevice_lastError());
+ return AVERROR_EXTERNAL;
+ }
+ ret = avpriv_sdr_set_freq(sdr, sdr->wanted_freq);
+ if (ret < 0)
+ return ret;
+
+ //List the available custom options for this specific driver to the user
+ arg_info = SoapySDRDevice_getSettingInfo(soapy, &length);
+ av_log(s, AV_LOG_INFO, "settings: ");
+ for (i=0; i<length; i++) {
+ SoapySDRArgInfo *info = arg_info + i;
+ av_log(s, AV_LOG_INFO, "key:%s desc:%s ", info->key, info->description);
+ if (info->units && *info->units)
+ av_log(s, AV_LOG_INFO, "units: %s ", info->units);
+ if (info->type == SOAPY_SDR_ARG_INFO_INT || info->type == SOAPY_SDR_ARG_INFO_FLOAT) {
+ av_log(s, AV_LOG_INFO, "range: %g - %g ", info->range.minimum, info->range.maximum);
+ }
+ if (info->numOptions) {
+ av_log(s, AV_LOG_INFO, "Options: ");
+ for (int o= 0; o<info->numOptions; o++){
+ av_log(s, AV_LOG_INFO, "%s (%s) ", info->options[o], info->optionNames[o]);
+ }
+ }
+ }
+ av_log(s, AV_LOG_INFO, "\n");
+
+ soapy_format = SoapySDRDevice_getNativeStreamFormat(soapy, SOAPY_SDR_RX, 0, &sdr->sample_scale);
+ av_log(s, AV_LOG_INFO, "native format: %s, full scale: %f\n", soapy_format, sdr->sample_scale);
+ if (!strcmp(soapy_format, SOAPY_SDR_CS8)) {
+ sdr->sample_size = 2;
+ } else if (!strcmp(soapy_format, SOAPY_SDR_CS16)) {
+ sdr->sample_size = 4;
+ } else {
+ sdr->sample_size = 8;
+ sdr->sample_scale = 1.0;
+ soapy_format = SOAPY_SDR_CF32;
+ }
+
+ //setup a stream (complex 32bit floats)
+#if SOAPY_SDR_API_VERSION < 0x00080000 // The old version is still widely used so we must support it
+ if (SoapySDRDevice_setupStream(soapy, &soapyRxStream, SOAPY_SDR_RX, soapy_format, NULL, 0, NULL))
+ soapyRxStream = NULL;
+#else
+ soapyRxStream = SoapySDRDevice_setupStream(soapy, SOAPY_SDR_RX, soapy_format, NULL, 0, NULL);
+#endif
+ if (!soapyRxStream) {
+ av_log(s, AV_LOG_ERROR, "setupStream fail: %s\n", SoapySDRDevice_lastError());
+ return AVERROR_EXTERNAL;
+ }
+ sdr->soapyRxStream = soapyRxStream;
+
+ sdr->bandwidth = SoapySDRDevice_getBandwidth(soapy, SOAPY_SDR_RX, 0);
+
+ // rtlsdr doesnt return a valid value
+ if (!sdr->bandwidth)
+ sdr->bandwidth = sdr->sdr_sample_rate;
+ av_log(s, AV_LOG_INFO, "bandwidth %"PRId64"\n", sdr->bandwidth);
+
+ SoapySDRDevice_activateStream(soapy, soapyRxStream, 0, 0, 0);
+
+ return avpriv_sdr_common_init(s);
+}
+
+static int sdrindev_read_close(AVFormatContext *s)
+{
+ SDRContext *sdr = s->priv_data;
+ SoapySDRDevice *soapy = sdr->soapy;
+
+ avpriv_sdr_stop_threading(s);
+
+ if (soapy) {
+ if (sdr->soapyRxStream) {
+ SoapySDRDevice_deactivateStream(soapy, sdr->soapyRxStream, 0, 0);
+ SoapySDRDevice_closeStream(soapy, sdr->soapyRxStream);
+ sdr->soapyRxStream = NULL;
+ }
+
+ SoapySDRDevice_unmake(soapy);
+ sdr->soapy = NULL;
+ }
+
+ return avpriv_sdr_read_close(s);
+}
+
+static const AVClass sdr_demuxer_class = {
+ .class_name = "sdr",
+ .item_name = av_default_item_name,
+ .option = avpriv_sdr_options,
+ .version = LIBAVUTIL_VERSION_INT,
+ .category = AV_CLASS_CATEGORY_DEMUXER,
+};
+
+const AVInputFormat ff_sdr_demuxer = {
+ .name = "sdr",
+ .long_name = NULL_IF_CONFIG_SMALL("Software Defined Radio Demodulator"),
+ .priv_data_size = sizeof(SDRContext),
+ .read_header = sdrindev_initial_hw_setup,
+ .read_packet = avpriv_sdr_read_packet,
+ .read_close = sdrindev_read_close,
+ .read_seek = avpriv_sdr_read_seek,
+ .flags = AVFMT_NOFILE,
+ .flags_internal = FF_FMT_INIT_CLEANUP,
+ .priv_class = &sdr_demuxer_class,
+};
@@ -528,6 +528,7 @@ OBJS-$(CONFIG_SCC_MUXER) += sccenc.o
OBJS-$(CONFIG_SCD_DEMUXER) += scd.o
OBJS-$(CONFIG_SDNS_DEMUXER) += sdns.o
OBJS-$(CONFIG_SDP_DEMUXER) += rtsp.o
+OBJS-$(CONFIG_SDRFILE_DEMUXER) += sdrdemux.o
OBJS-$(CONFIG_SDR2_DEMUXER) += sdr2.o
OBJS-$(CONFIG_SDS_DEMUXER) += sdsdec.o
OBJS-$(CONFIG_SDX_DEMUXER) += sdxdec.o pcm.o
@@ -411,6 +411,7 @@ extern const AVInputFormat ff_scd_demuxer;
extern const AVInputFormat ff_sdns_demuxer;
extern const AVInputFormat ff_sdp_demuxer;
extern const AVInputFormat ff_sdr2_demuxer;
+extern const AVInputFormat ff_sdrfile_demuxer;
extern const AVInputFormat ff_sds_demuxer;
extern const AVInputFormat ff_sdx_demuxer;
extern const AVInputFormat ff_segafilm_demuxer;
new file mode 100644
@@ -0,0 +1,239 @@
+/*
+ * SDR Input device / Demuxer / Demodulator
+ * Copyright (c) 2023 Michael Niedermayer
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg 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
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVFORMAT_SDR_H
+#define AVFORMAT_SDR_H
+
+#include <pthread.h>
+#include <stdatomic.h>
+#include <float.h>
+
+#include "libavutil/fifo.h"
+#include "libavutil/opt.h"
+#include "libavutil/time.h"
+#include "libavutil/thread.h"
+#include "libavutil/tx.h"
+#include "libavformat/avformat.h"
+
+
+#define INDEX2F(INDEX) (((INDEX) - sdr->block_size + 0.5) * 0.5 * sdr->sdr_sample_rate / sdr->block_size + sdr->block_center_freq)
+#define F2INDEX(F) ((( F) - sdr->block_center_freq) * 2 * sdr->block_size / sdr->sdr_sample_rate + sdr->block_size - 0.5)
+
+typedef enum Mode {
+ SingleStationMode, //< demodulate 1 station for Radio like usage
+ AllStationMode, //< demodulate all stations in current input
+ ModeNB,
+} Mode;
+
+typedef enum AMMode {
+ AMMidSide,
+ AMLeftRight,
+ AMInPhase,
+ AMEnvelope,
+ AMModeNB,
+} AMMode;
+
+typedef enum FMEmphasisMode {
+ EMPHASIS_50us,
+ EMPHASIS_75us,
+ EMPHASIS_NONE,
+ EMPHASISNB,
+} FMEmphasisMode;
+
+typedef enum Modulation {
+ AM,
+ FM,
+ OFDM_DQPSK, //DAB
+ //QAM, PSK, ...
+} Modulation;
+
+typedef struct Station {
+ char *name;
+ enum Modulation modulation;
+ double frequency;
+ int64_t bandwidth;
+ int64_t bandwidth_p2;
+ float score;
+ int timeout; //since how many blocks was this detectable but not detected
+ int multiplex_index; //DAB can have multiple stations on one frequency
+
+ struct SDRStream *stream;
+} Station;
+
+typedef struct FIFOElement {
+ int64_t center_frequency;
+ void *halfblock;
+} FIFOElement;
+
+typedef struct SDRStream {
+ AVTXContext *ifft_ctx;
+ AVTXContext *fft_p2_ctx;
+ AVTXContext *ifft_p2_ctx;
+ av_tx_fn ifft;
+ av_tx_fn fft_p2;
+ av_tx_fn ifft_p2;
+ int block_size;
+ int block_size_p2;
+ int processing_index;
+ float *out_buf;
+ AVComplexFloat *block;
+ AVComplexFloat *iblock;
+ AVComplexFloat *icarrier;
+ AVComplexFloat *iside;
+ float *window;
+ float *window_p2;
+ Station *station;
+
+ int frame_size;
+ int frame_buffer_line;
+ uint8_t *frame_buffer;
+} SDRStream;
+
+typedef struct SDRContext {
+ const AVClass *class;
+ AVFormatContext *avfmt;
+ void *soapy;
+ void *soapyRxStream;
+ const char *current_direct_samp;
+ AVTXContext *fft_ctx;
+ av_tx_fn fft;
+ Mode mode;
+ AVRational fps;
+ char *driver_name;
+ char *dump_url;
+ int fileheader_size;
+ AVIOContext *dump_avio;
+ Station **station;
+ int nb_stations;
+ int width, height;
+ int single_ch_audio_st_index;
+ int waterfall_st_index;
+ int64_t freq;
+ int64_t min_freq;
+ int64_t max_freq;
+ int64_t min_center_freq;
+ int64_t max_center_freq;
+ int sdr_sample_rate;
+ int sdr_agc;
+ int sdr_adcc;
+ int64_t bandwidth;
+ int64_t last_pts;
+ int64_t pts;
+ int block_size;
+ int kbd_alpha;
+ AVComplexFloat *windowed_block;
+ int64_t block_center_freq; ///< center frequency the current block contains
+ int64_t station_freq;
+ int sample_size;
+ double sample_scale;
+
+ int am_mode; ///< AMMode but using int for generic option access
+ int emphasis_mode;
+
+ pthread_t hw_thread;
+ int thread_started;
+ pthread_mutex_t mutex; ///< Mutex to protect common variable between mainthread and hw_thread, and also to protect soapy from concurrent calls
+ AVFifo *empty_block_fifo;
+ AVFifo *full_block_fifo;
+ atomic_int close_requested;
+ int64_t wanted_freq; ///< center frequency we want the hw to provide next
+ int seek_direction; ///< if a seek is requested this is -1 or 1 otherwise 0
+ int skip_probe;
+
+ /**
+ * Setup the hardware for the requested frequency
+ */
+ int (*set_frequency_callback)(struct SDRContext *sdr, int64_t frequency);
+
+ /**
+ * Read from the hardware, block if nothing available with a reasonable timeout
+ *
+ * @param remaining space remaining in fifo_element
+ *
+ * @return AVERROR(EAGAIN) in case of timeout or the number of elements read
+ */
+ int (*read_callback)(struct SDRContext *sdr, FIFOElement *fifo_element, int remaining);
+
+ /**
+ * space remaining in teh current block in the file.
+ *
+ * This is needed as the sdrdemux code can be setup to use a different block size
+ * than what was used when capturing the data
+ */
+ int remaining_file_block_size;
+
+ AVComplexFloat *block;
+ float *len2block;
+ float *window;
+
+ int missing_streams;
+
+ int rtlsdr_fixes;
+} SDRContext;
+
+typedef struct ModulationDescriptor {
+ const char *name;
+ const char *shortname;
+ enum Modulation modulation;
+ enum AVMediaType media_type;
+
+ /**
+ * Scan all of the current sdr->block and call create_station() for each found station
+ */
+ int (*probe)(SDRContext *sdr);
+
+ /**
+ * Demodulate given station into packet
+ */
+ int (*demodulate)(SDRContext *sdr, int stream_index, AVPacket *pkt);
+} ModulationDescriptor;
+
+typedef struct BandDescriptor {
+ const char *name;
+ const char *shortname;
+ int64_t freq_min;
+ int64_t freq_max;
+} BandDescriptor ;
+
+extern const AVOption avpriv_sdr_options[];
+
+/**
+ * Set the center frequency of the hardware
+ * this will check the argument and call set_frequency_callback()
+ * It can be called before the thread is started or from within the thread,
+ */
+int avpriv_sdr_set_freq(SDRContext *sdr, int64_t freq);
+
+int avpriv_sdr_common_init(AVFormatContext *s);
+
+int avpriv_sdr_read_packet(AVFormatContext *s, AVPacket *pkt);
+
+int avpriv_sdr_read_seek(AVFormatContext *s, int stream_index, int64_t target, int flags);
+
+/**
+ * shuts down threads, destroys mutex
+ * Safe to call if no thread was started or after it was shutdown
+ */
+void avpriv_sdr_stop_threading(AVFormatContext *s);
+
+int avpriv_sdr_read_close(AVFormatContext *s);
+
+#endif /* AVFORMAT_SDR_H */
new file mode 100644
@@ -0,0 +1,1843 @@
+/*
+ * SDR Demuxer / Demodulator
+ * Copyright (c) 2023 Michael Niedermayer
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg 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
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ *
+ *
+ */
+
+/**
+ * TODO
+ * * DAB
+ * * DVB
+ * * Improve probing using multiple detections and differential detection
+ *
+ */
+
+#include "sdr.h"
+
+#include <pthread.h>
+#include <stdatomic.h>
+#include <float.h>
+#include "libavutil/avassert.h"
+#include "libavutil/channel_layout.h"
+#include "libavutil/ffmath.h"
+#include "libavutil/fifo.h"
+#include "libavutil/intreadwrite.h"
+#include "libavutil/opt.h"
+#include "libavutil/time.h"
+#include "libavutil/thread.h"
+#include "libavutil/tx.h"
+#include "libavutil/xga_font_data.h"
+#include "libavcodec/kbdwin.h"
+#include "libavformat/avformat.h"
+#include "libavformat/demux.h"
+#include "libavformat/internal.h"
+
+#ifdef SYN_TEST
+#include "libavutil/lfg.h"
+#endif
+
+#define FREQ_BITS 22
+#define TIMEBASE ((48000ll / 128) << FREQ_BITS)
+
+#define STATION_TIMEOUT 100 ///< The number of frames after which a station is removed if it was not detected
+
+/*
+ * 100 detects nothing
+ * 50 detects a good bit but not all
+ */
+#define AM_THRESHOLD 20
+
+#define AM_MAX23 0.06 //smaller causes failure on synthetic signals
+#define AM_MAX4 0.02
+
+#define FM_THRESHOLD .8 //TODO adjust
+
+//Least squares fit at 1khz points of frequency response shown by Frank McClatchie, FM SYSTEMS, INC. 800-235-6960
+static double emphasis75us(int f)
+{
+ return ((((((((- 4.79546E-9 * f + 5.32101E-7) * f - 0.0000254577) * f + 0.000687225) * f
+ - 0.0114925) * f + 0.122781 ) * f - 0.827885 ) * f + 3.25025 ) * f - 4.6049) * f + 2.06937;
+}
+
+/**
+ * Apply emphasis filter in frequency domain
+ */
+static void apply_emphasis(SDRContext *sdr, AVComplexFloat *data, int len, int sample_rate, int dir)
+{
+ if (sdr->emphasis_mode == EMPHASIS_NONE)
+ return;
+
+ len = FFMIN(len, len * 19000 / sample_rate);
+
+ for (int i = 1; i < len; i++) {
+ double index = 1.0 + i / 1000;
+ double scale;
+
+ if (sdr->emphasis_mode == EMPHASIS_50us)
+ index = FFMIN(0, index - 1061);
+
+ scale = ff_exp10(-emphasis75us(index) / 20.0);
+
+ data[i*dir].re *= scale;
+ data[i*dir].im *= scale;
+ }
+}
+
+static float len2(AVComplexFloat c)
+{
+ return c.re*c.re + c.im*c.im;
+}
+
+static int create_station(SDRContext *sdr, enum Modulation modulation, double freq, int64_t bandwidth, int64_t bandwidth_p2, float score) {
+ Station *station;
+ void *tmp;
+ int i;
+ double best_distance = INT64_MAX;
+ Station *best_station = NULL;
+ int best_station_index = -1;
+ float drift = bandwidth/3.0;
+
+ //rtlsdr has artifacts at the center, we dont want to create a station there
+ if (sdr->rtlsdr_fixes>0 && fabs(freq - sdr->block_center_freq) < bandwidth/2)
+ return AVERROR(EINVAL);
+
+ for (i=0; i<sdr->nb_stations; i++) {
+ double delta = fabs(sdr->station[i]->frequency - freq);
+ // Station already added, or we have 2 rather close stations
+ //FIXME we want to make sure that the stronger station is not skiped but we also dont want to add a station twice
+ if (modulation == sdr->station[i]->modulation && delta < best_distance) {
+ best_distance = delta;
+ best_station = sdr->station[i];
+ best_station_index = i;
+ }
+ if (modulation != sdr->station[i]->modulation && delta < (bandwidth + sdr->station[i]->bandwidth)/2.1)
+ return -1;
+ }
+ if (best_station) {
+ if (best_distance <= drift) {
+ if (score > best_station->score) {
+ int log_level = fabs(best_station->frequency - freq) < 3.0 ? AV_LOG_DEBUG : AV_LOG_WARNING;
+ av_log(sdr->avfmt, log_level, "Update station %d score:%f -> %f freq: %f %f -> %f\n",
+ best_station_index,
+ best_station->score, score,
+ best_station->frequency - freq, best_station->frequency, freq
+ );
+ best_station->frequency = freq;
+ best_station->score = score;
+ }
+ return best_station_index;
+ }
+ }
+
+ tmp = av_realloc_array(sdr->station, sdr->nb_stations+1, sizeof(*sdr->station));
+ if (!tmp)
+ return AVERROR(ENOMEM);
+ sdr->station = tmp;
+
+ station = av_mallocz(sizeof(*station));
+ if (!station)
+ return AVERROR(ENOMEM);
+
+ sdr->station[sdr->nb_stations++] = station;
+
+ station->modulation = modulation;
+ station->frequency = freq;
+ station->bandwidth = bandwidth;
+ station->bandwidth_p2 = bandwidth_p2;
+ station->score = score;
+
+ // if we just found a new station lets also probe the next frame
+ sdr->skip_probe = 0;
+
+ av_log(sdr, AV_LOG_INFO, "create_station %d f:%f bw:%"PRId64"/%"PRId64" score: %f\n", modulation, freq, bandwidth, bandwidth_p2, score);
+
+ return sdr->nb_stations - 1;
+}
+
+static void free_station(Station *station)
+{
+ av_freep(&station->name);
+ if (station->stream)
+ station->stream->station = NULL;
+ av_free(station);
+}
+
+/**
+ * remove stations which we no longer receive well
+ * Especially with AM and weather conditions stations disapear, this keeps things a bit more tidy
+ */
+static void decay_stations(SDRContext *sdr)
+{
+ for (int i=0; i<sdr->nb_stations; i++) {
+ Station *station = sdr->station[i];
+
+ if (station->frequency - station->bandwidth/2 < sdr->block_center_freq - sdr->bandwidth/2 ||
+ station->frequency + station->bandwidth/2 > sdr->block_center_freq + sdr->bandwidth/2)
+ continue;
+ if (station->stream)
+ continue;
+
+ if (station->timeout++ > STATION_TIMEOUT) {
+ free_station(station);
+ sdr->station[i] = sdr->station[--sdr->nb_stations];
+ }
+ }
+}
+
+static void probe_common(SDRContext *sdr)
+{
+ for(int i = 0; i < 2*sdr->block_size; i++) {
+ sdr->len2block[i] = len2(sdr->block[i]);
+ }
+}
+
+// simple and dumb implementation for max we could do it faster but it doesnzt matter ATM
+static float max_in_range(SDRContext *sdr, unsigned start, unsigned end)
+{
+ float max = sdr->len2block[start];
+ for (; start <= end; start++)
+ max = fmax(max, sdr->len2block[start]);
+ return max;
+}
+
+static double find_peak(SDRContext *sdr, const float *data, int index, int len)
+{
+ double y[3], b, a;
+
+ if (index == 0) {
+ index = 1;
+ } else if (index >= len - 1)
+ index = len - 2;
+
+ //We use a simple quadratic to find the maximum at higher precission than the available samples
+ //ax^2 + bx + c
+ //dy/dx = 2ax + b = 0
+ y[0] = data[index-1];
+ y[1] = data[index];
+ y[2] = data[index+1];
+
+ b = (y[2] - y[0]) * 0.5;
+ a = (y[0] + y[2]) * 0.5 - y[1];
+
+ //This should not happen
+ if (a >= 0.0)
+ return INT_MIN;
+
+ return index -0.5 * b / a;
+ //TODO theres some simplification possible above but the fm_probe() using this needs to be tuned first so we dont optimize the wrong algorithm
+}
+
+static double find_peak_macleod(const SDRContext *sdr, const AVComplexFloat *data, int index, int len, float *phase) {
+ AVComplexFloat ref;
+ double r0, r1, r2, rd, g;
+ double ab[16][2] = {
+ {1.525084, 3.376388}, {1.773260, 3.129280}, {1.970200, 2.952300}, {2.122700, 2.825800},
+ {2.243600, 2.731620}, {2.341880, 2.658740}, {2.423310, 2.600750}, {2.492110, 2.553360},
+ {2.551000, 2.513930}, {2.602300, 2.480400}, {2.646880, 2.451880}, {2.686200, 2.427200},
+ {2.721880, 2.405120}, {2.753600, 2.385800}, {2.781800, 2.368900}, {2.808580, 2.352920},
+ };
+
+ double a = ab[sdr->kbd_alpha-1][0];
+ double b = ab[sdr->kbd_alpha-1][1];
+
+ if (index == 0) {
+ index = 1;
+ } else if (index >= len - 1)
+ index = len - 2;
+
+ /* Baed on Macleod, M.D., "Fast Nearly ML Estimation of the Parameters
+ * of Real or Complex Single Tones or Resolved Multiple Tones,"
+ * IEEE Trans. Sig. Proc. Vol 46 No 1,
+ * January 1998, pp141-148.
+ *
+ * We use the 3 point estimator without corrections, the corrections
+ * provide insignificant performance increase. We add compensation due to
+ * the window used, this performs substantially better than the original
+ * with a rectangular window.
+ */
+ ref = data[index];
+ r0 = data[index-1].re * ref.re + data[index-1].im * ref.im;
+ r1 = ref.re * ref.re + ref.im * ref.im;
+ r2 = data[index+1].re * ref.re + data[index+1].im * ref.im;
+ rd = 2*r1 + r0 + r2;
+ if (r2 > r1 || r0 > r1 || rd <=0) // rounding and float numeric issues
+ return -1;
+ g = (r0-r2) / rd + DBL_MIN;
+ g = (sqrt(a*a + 8*g*g)-a)/(b*g);
+
+ if (phase) {
+ AVComplexFloat t;
+ if (g < 0){
+ t.re = ref.re * (1+g) + data[index-1].re * g;
+ t.im = ref.im * (1+g) + data[index-1].im * g;
+ } else {
+ t.re = ref.re * (1-g) - data[index+1].re * g;
+ t.im = ref.im * (1-g) - data[index+1].im * g;
+ }
+ *phase = atan2(t.im, t.re) - M_PI*g;
+ }
+
+ return index + g;
+}
+
+static int probe_am(SDRContext *sdr)
+{
+ int i;
+ int bandwidth_f = 6000;
+ int half_bw_i = bandwidth_f * (int64_t)sdr->block_size / sdr->sdr_sample_rate;
+ double avg = 0;
+
+ if (2*half_bw_i > 2*sdr->block_size)
+ return 0;
+
+ for (i = 0; i<2*half_bw_i; i++)
+ avg += sdr->len2block[i];
+
+ for (i = half_bw_i; i<2*sdr->block_size - half_bw_i; i++) {
+ float mid = sdr->len2block[i];
+ double score;
+ avg += sdr->len2block[i + half_bw_i];
+ score = half_bw_i * mid / (avg - mid);
+ avg -= sdr->len2block[i - half_bw_i];
+ //TODO also check for symmetry in the spectrum
+ if (mid > 0 && score > AM_THRESHOLD &&
+ sdr->len2block[i - 1] < mid && sdr->len2block[i + 1] <= mid &&
+ sdr->len2block[i - 2] < mid*AM_MAX23 && sdr->len2block[i + 2] < mid*AM_MAX23 &&
+ sdr->len2block[i - 3] < mid*AM_MAX23 && sdr->len2block[i + 3] < mid*AM_MAX23
+ ){
+ if (max_in_range(sdr, i-half_bw_i, i-4) < mid*AM_MAX4 &&
+ max_in_range(sdr, i+4, i+half_bw_i) < mid*AM_MAX4) {
+ int station_index;
+ double peak_i = find_peak_macleod(sdr, sdr->block, i, 2*sdr->block_size, NULL);
+ if (peak_i < 0)
+ continue;
+ if (fabs(peak_i-i) > 1.0) {
+ av_log(sdr->avfmt, AV_LOG_WARNING, "peak detection failure\n");
+ continue;
+ }
+
+ station_index = create_station(sdr, AM, INDEX2F(peak_i), bandwidth_f, 0, score);
+
+ if (station_index >= 0) {
+ Station *station = sdr->station[station_index];
+ station->timeout = 0;
+ }
+ }
+ }
+ }
+
+ return 0;
+}
+
+/**
+ *
+ * @param data_len length of the data and len2block arrays
+ * @param index center of search
+ * @param search_range range on both sides to search for carrier
+ * @param len AM signal if any surrounding the carrier
+ */
+static double find_am_carrier(SDRContext *sdr, const AVComplexFloat *data, int data_len, float *len2block, int index, int search_range, int len)
+{
+ int i_max, i;
+ double mid, score;
+ double avg = 0;
+
+ if (index + len + search_range >= data_len ||
+ index - len - search_range < 0)
+ return AVERROR(ERANGE);
+
+ for (i = index - len - search_range; i < index + len + search_range; i++)
+ len2block[i] = len2(data[i]);
+
+ i_max = index;
+ for (i = index-search_range; i<index+search_range; i++) {
+ if (len2block[i] > len2block[i_max])
+ i_max = i;
+ }
+ mid = len2block[i_max];
+
+ for (i = -len; i < len+1; i++)
+ avg += len2block[i + index];
+ score = len * mid / (avg - mid);
+ //find optimal frequency for this block if we have a carrier
+ if (score > AM_THRESHOLD / 4) {
+ double peak_i = find_peak_macleod(sdr, data, i_max, data_len, NULL);
+ if (peak_i < 0)
+ return peak_i;
+ if (fabs(peak_i-i_max) > 1.0)
+ return -1;
+ return peak_i;
+ }
+ return -1;
+}
+
+/**
+ * Demodulate with a carrier that is in the middle of the signal like in AM
+ * This will normalize the signal based on the carrier amplitude and subtract the carrier
+ */
+static void synchronous_am_demodulation(AVComplexFloat *iblock, AVComplexFloat *icarrier, float *window, int len)
+{
+ for (int i = 0; i<len; i++) {
+ AVComplexFloat c = icarrier[i];
+ AVComplexFloat s = iblock[i];
+ float w = window[i];
+ float den = w / (c.re*c.re + c.im*c.im);
+ av_assert0(c.re*c.re + c.im*c.im > 0);
+
+ iblock[i].re = ( s.im*c.im + s.re*c.re) * den;
+ iblock[i].im = ( s.im*c.re - s.re*c.im) * den;
+ iblock[i].re -= w;
+ }
+}
+
+/**
+ * Demodulate with a carrier that is half the frequency and reduced amplitude
+ * This will not normalize the signal based on the carrier amplitude
+ */
+static void synchronous_am_demodulation2(AVComplexFloat *iblock, AVComplexFloat *icarrier, float *window, int len)
+{
+ for (int i = 0; i<len; i++) {
+ AVComplexFloat c = icarrier[i];
+ AVComplexFloat s = iblock[i];
+ AVComplexFloat c2;
+ float w = window[i];
+ float den = w / (c.re*c.re + c.im*c.im);
+ av_assert0(c.re*c.re + c.im*c.im > 0);
+
+ c2.re = c.re*c.re - c.im*c.im;
+ c2.im = c.re*c.im + c.im*c.re;
+
+ iblock[i].re = ( s.im*c2.im + s.re*c2.re) * den;
+ iblock[i].im = ( s.im*c2.re - s.re*c2.im) * den;
+ }
+}
+
+static int demodulate_am(SDRContext *sdr, int stream_index, AVPacket *pkt)
+{
+ AVStream *st = sdr->avfmt->streams[stream_index];
+ SDRStream *sst = st->priv_data;
+ double freq = sst->station->frequency;
+ int64_t bandwidth = sst->station->bandwidth;
+ int index = lrint(F2INDEX(freq));
+ int len = (bandwidth * 2ll * sdr->block_size + sdr->sdr_sample_rate/2) / sdr->sdr_sample_rate;
+ float *newbuf;
+ float scale;
+ int sample_rate = sdr->sdr_sample_rate * (int64_t)sst->block_size / sdr->block_size;
+ int ret, i;
+ double current_station_i;
+ float limits[2] = {-0.0, 0.0};
+ float clip = 1.0;
+ enum AMMode am_mode = sdr->am_mode;
+
+#define CARRIER_SEARCH 2
+ if (index + len + CARRIER_SEARCH>= 2*sdr->block_size ||
+ index - len - CARRIER_SEARCH < 0 ||
+ 2*len + 1 > 2*sst->block_size)
+ return AVERROR(ERANGE);
+
+ current_station_i = find_am_carrier(sdr, sdr->block, 2*sdr->block_size, sdr->len2block, index, CARRIER_SEARCH, len);
+
+ if (current_station_i >= 0) {
+ if (current_station_i >= 0) {
+ av_log(sdr->avfmt, AV_LOG_DEBUG, "adjusting frequency %f, max index: %ld\n", INDEX2F(current_station_i) - freq, lrint(F2INDEX(freq)) - index);
+ freq = INDEX2F(current_station_i);
+ index = lrint(F2INDEX(freq));
+ }
+ } else {
+ // We have no carrier so we cannot decode Synchronously to a carrier
+ am_mode = AMEnvelope;
+ }
+
+ newbuf = av_malloc(sizeof(*sst->out_buf) * 2 * sst->block_size);
+ if (!newbuf)
+ return AVERROR(ENOMEM);
+#define SEPC 4
+
+ i = 2*len+1;
+ memcpy(sst->block, sdr->block + index - len, sizeof(*sst->block) * i);
+ memset(sst->block + i, 0, sizeof(*sst->block) * (2 * sst->block_size - i));
+
+ sst->ifft(sst->ifft_ctx, sst->iblock , sst->block, sizeof(AVComplexFloat));
+
+ if (am_mode == AMEnvelope) {
+ double vdotw = 0;
+ double wdot = 0; // could be precalculated
+ for (i = 0; i<2*sst->block_size; i++) {
+ float w = sst->window[i];
+ float v = sqrt(len2(sst->iblock[i]));
+ sst->iblock[i].re = v;
+ sst->iblock[i].im = 0;
+
+ vdotw += w*v;
+ wdot += w*w;
+ }
+
+ vdotw /= wdot ;
+ for (i = 0; i<2*sst->block_size; i++) {
+ float w = sst->window[i];
+ sst->iblock[i].re -= w*vdotw;
+ }
+
+ scale = 0.9/vdotw;
+ } else {
+ // Synchronous demodulation
+ memset(sst->block, 0, sizeof(*sst->block) * i);
+ for (i = len-SEPC+1; i<len+SEPC; i++)
+ sst->block[i] = sdr->block[index + i - len];
+ sst->ifft(sst->ifft_ctx, sst->icarrier, sst->block, sizeof(AVComplexFloat));
+
+ synchronous_am_demodulation(sst->iblock, sst->icarrier, sst->window, 2*sst->block_size);
+ scale = 0.9;
+ }
+
+ for(int i = 0; i<2*sst->block_size; i++) {
+ av_assert0(isfinite(sst->iblock[i].re));
+ av_assert0(isfinite(sst->iblock[i].im));
+ limits[0] = FFMIN(limits[0], FFMIN(sst->iblock[i].re - sst->iblock[i].im, sst->iblock[i].re + sst->iblock[i].im));
+ limits[1] = FFMAX(limits[1], FFMAX(sst->iblock[i].re - sst->iblock[i].im, sst->iblock[i].re + sst->iblock[i].im));
+ }
+ av_assert1(FFMAX(limits[1], -limits[0]) >= 0);
+ scale = FFMIN(scale, 0.98 / FFMAX(limits[1], -limits[0]));
+
+ for(i = 0; i<sst->block_size; i++) {
+ float m, q;
+
+ m = sst->out_buf[2*i+0] + (sst->iblock[i ].re) * sst->window[i ] * scale;
+ newbuf[2*i+0] = (sst->iblock[i + sst->block_size].re) * sst->window[i + sst->block_size] * scale;
+
+ switch(am_mode) {
+ case AMMidSide:
+ case AMLeftRight:
+ q = sst->out_buf[2*i+1] + sst->iblock[i ].im * sst->window[i ] * scale;
+ newbuf[2*i+1] = sst->iblock[i + sst->block_size].im * sst->window[i + sst->block_size] * scale;
+ switch(am_mode) {
+ case AMMidSide:
+ q *= 0.5;
+ sst->out_buf[2*i+0] = m + q;
+ sst->out_buf[2*i+1] = m - q;
+ break;
+ case AMLeftRight:
+ sst->out_buf[2*i+0] = m;
+ sst->out_buf[2*i+1] = q;
+ break;
+ }
+ break;
+
+ case AMEnvelope:
+ case AMInPhase:
+ sst->out_buf[2*i+0] =
+ sst->out_buf[2*i+1] = m;
+ break;
+ }
+
+ if (fabs(sst->out_buf[i]) > clip) {
+ av_log(sdr->avfmt, AV_LOG_WARNING, "CLIP %f\n", sst->out_buf[i]);
+ clip = fabs(sst->out_buf[i]) * 1.1;
+ }
+ }
+
+ ret = av_packet_from_data(pkt, (void*)sst->out_buf, sizeof(*sst->out_buf) * 2 * sst->block_size);
+ if (ret < 0)
+ av_free(sst->out_buf);
+ sst->out_buf = newbuf;
+
+ if (st->codecpar->ch_layout.nb_channels != 2 ||
+ st->codecpar->sample_rate != sample_rate
+ ) {
+ av_log(sdr->avfmt, AV_LOG_INFO, "set channel parameters %d %d %d\n", st->codecpar->ch_layout.nb_channels, st->codecpar->sample_rate, sample_rate);
+ if (st->codecpar->sample_rate == 0)
+ sdr->missing_streams--;
+ ff_add_param_change(pkt, 2, 0, sample_rate, 0, 0);
+ st->codecpar->sample_rate = sample_rate;
+ }
+
+ return ret;
+}
+
+static int probe_fm(SDRContext *sdr)
+{
+ int i;
+ int bandwidth_f = 200*1000;
+ int bandwidth_p2 = 38*1000; //phase 2 bandwidth
+ int half_bw_i = bandwidth_f * (int64_t)sdr->block_size / sdr->sdr_sample_rate;
+ double avg[2] = {0}, tri = 0;
+ float last_score[3] = {FLT_MAX, FLT_MAX, FLT_MAX};
+
+ if (2*half_bw_i > 2*sdr->block_size)
+ return 0;
+
+ for (i = 0; i<half_bw_i; i++) {
+ avg[0] += sdr->len2block[i];
+ tri += i*sdr->len2block[i];
+ }
+ for (; i<2*half_bw_i; i++) {
+ avg[1] += sdr->len2block[i];
+ tri += (2*half_bw_i-i)*sdr->len2block[i];
+ }
+
+ for(i = half_bw_i; i<2*sdr->block_size - half_bw_i; i++) {
+ double b = avg[0] + sdr->len2block[i];
+ avg[0] += sdr->len2block[i] - sdr->len2block[i - half_bw_i];
+ avg[1] -= sdr->len2block[i] - sdr->len2block[i + half_bw_i];
+ b += avg[1];
+ tri += avg[1] - avg[0];
+
+ last_score[2] = last_score[1];
+ last_score[1] = last_score[0];
+ last_score[0] = tri / (b * half_bw_i);
+
+ if (last_score[1] >= last_score[0] &&
+ last_score[1] > last_score[2] &&
+ last_score[1] > FM_THRESHOLD) {
+
+ // as secondary check, we could check that without the center 3 samples we are still having a strong signal FIXME
+
+ double peak_i = find_peak(sdr, last_score, 1, 3) + i - 1;
+ if (peak_i < 0)
+ continue;
+ av_assert0(fabs(peak_i-i) < 2);
+ create_station(sdr, FM, peak_i * 0.5 * sdr->sdr_sample_rate / sdr->block_size + sdr->block_center_freq - sdr->sdr_sample_rate/2, bandwidth_f, bandwidth_p2, last_score[1]);
+ }
+ }
+
+ return 0;
+}
+
+static int demodulate_fm(SDRContext *sdr, int stream_index, AVPacket *pkt)
+{
+ AVStream *st = sdr->avfmt->streams[stream_index];
+ SDRStream *sst = st->priv_data;
+ double freq = sst->station->frequency;
+ int64_t bandwidth = sst->station->bandwidth;
+ int index = lrint(F2INDEX(freq));
+ int len = (bandwidth * 2ll * sdr->block_size + sdr->sdr_sample_rate/2) / sdr->sdr_sample_rate;
+ float *newbuf;
+ float scale;
+ int sample_rate = sdr->sdr_sample_rate * (int64_t)sst->block_size / sdr->block_size;
+ int sample_rate_p2 = sdr->sdr_sample_rate * (int64_t)sst->block_size_p2 / sdr->block_size;
+ int ret, i;
+ float clip = 1.0;
+ double last_phase = 0;
+ int carrier19_i = 2L*sst->block_size*19000 / sample_rate;
+ int len17_i = 2L*sst->block_size*17000 / sample_rate;
+ double carrier19_i_exact;
+
+ if (index + len >= 2*sdr->block_size ||
+ index - len < 0 ||
+ 2*len + 1 > 2*sst->block_size)
+ return AVERROR(ERANGE);
+
+ newbuf = av_malloc(sizeof(*sst->out_buf) * 2 * sst->block_size);
+ if (!newbuf)
+ return AVERROR(ENOMEM);
+
+ i = 2*len+1;
+ memcpy(sst->block, sdr->block + index, sizeof(*sst->block) * (len + 1));
+ memcpy(sst->block + 2 * sst->block_size - len, sdr->block + index - len, sizeof(*sst->block) * len);
+ memset(sst->block + len + 1, 0, sizeof(*sst->block) * (2 * sst->block_size - i));
+
+ sst->ifft(sst->ifft_ctx, sst->iblock, sst->block, sizeof(AVComplexFloat));
+
+ for (i = 0; i<2*sst->block_size; i++) {
+ double phase = atan2(-sst->iblock[i].im, sst->iblock[i].re);
+ sst->iblock[i].re = remainder(phase - last_phase, M_PI) * sst->window[i];
+ sst->iblock[i].im = 0;
+ last_phase = phase;
+ }
+ int W= 5;
+
+ av_assert0(sst->block_size_p2 * 2 < sst->block_size);
+ //FIXME this only needs to be a RDFT
+ //CONSIDER, this and in fact alot can be done with bandpass and lowpass filters instead of FFTs, find out which is better
+ //CONSIDER synthesizing the carrier instead of IFFT, we have all parameters for that
+ sst->fft_p2(sst->fft_p2_ctx, sst->block, sst->iblock, sizeof(AVComplexFloat));
+ // Only the low N/2+1 are used the upper is just a reflection
+
+ carrier19_i_exact = find_am_carrier(sdr, sst->block, 2*sst->block_size, (void*)(sst->block + 1 + sst->block_size), carrier19_i, 10, 10);
+ carrier19_i = lrint(carrier19_i_exact);
+
+ if (carrier19_i >= 0) {
+ int shift = carrier19_i/2;
+ i = sst->block_size;
+ memset(sst->block + i, 0, 2*sst->block_size_p2 * sizeof(AVComplexFloat));
+ memcpy(sst->block + i + carrier19_i - W - shift , sst->block + carrier19_i - W, sizeof(AVComplexFloat)*(2*W+1));
+ sst->ifft_p2(sst->ifft_p2_ctx, sst->icarrier, sst->block + i, sizeof(AVComplexFloat));
+
+ memcpy(sst->block + i + 2*carrier19_i - 2*shift - len17_i, sst->block + 2*carrier19_i - len17_i, sizeof(AVComplexFloat)*2*len17_i);
+ apply_emphasis(sdr, sst->block + i + 2*carrier19_i - 2*shift, sst->block_size_p2, sample_rate_p2, + 1);
+ apply_emphasis(sdr, sst->block + i + 2*carrier19_i - 2*shift, sst->block_size_p2, sample_rate_p2, - 1);
+ sst->ifft_p2(sst->ifft_p2_ctx, sst->iside , sst->block + i, sizeof(AVComplexFloat));
+ synchronous_am_demodulation2(sst->iside, sst->icarrier, sst->window_p2, 2*sst->block_size_p2);
+ }
+ memset(sst->block + len17_i, 0, (2*sst->block_size_p2 - len17_i) * sizeof(AVComplexFloat));
+ apply_emphasis(sdr, sst->block, 2*sst->block_size_p2, sample_rate_p2, + 1);
+ sst->ifft_p2(sst->ifft_p2_ctx, sst->iblock , sst->block, sizeof(AVComplexFloat));
+ memset(sst->iblock + 2*sst->block_size_p2, 0 ,(2*sst->block_size -2*sst->block_size_p2) * sizeof(AVComplexFloat));
+
+ scale = 5 / (M_PI * 2*sst->block_size);
+ for(i = 0; i<sst->block_size_p2; i++) {
+ float m, q;
+
+ m = sst->out_buf[2*i+0] + (sst->iblock[i ].re) * sst->window_p2[i ] * scale;
+ newbuf[2*i+0] = (sst->iblock[i + sst->block_size_p2].re) * sst->window_p2[i + sst->block_size_p2] * scale;
+
+ if (carrier19_i >= 0) {
+ q = sst->out_buf[2*i+1] + sst->iside[i ].im * sst->window_p2[i ] * scale;
+ newbuf[2*i+1] = sst->iside[i + sst->block_size_p2].im * sst->window_p2[i + sst->block_size_p2] * scale;
+
+ sst->out_buf[2*i+0] = m + q;
+ sst->out_buf[2*i+1] = m - q;
+ } else {
+ sst->out_buf[2*i+0] =
+ sst->out_buf[2*i+1] = m;
+ }
+
+ if (fabs(sst->out_buf[i]) > clip) {
+ av_log(sdr->avfmt, AV_LOG_WARNING, "CLIP %f\n", sst->out_buf[i]);
+ clip = fabs(sst->out_buf[i]) * 1.1;
+ }
+ }
+
+ ret = av_packet_from_data(pkt, (void*)sst->out_buf, sizeof(*sst->out_buf) * 2 * sst->block_size_p2);
+ if (ret < 0)
+ av_free(sst->out_buf);
+ sst->out_buf = newbuf;
+
+ if (st->codecpar->ch_layout.nb_channels != 2 ||
+ st->codecpar->sample_rate != sample_rate_p2
+ ) {
+ av_log(sdr->avfmt, AV_LOG_INFO, "set channel parameters %d %d %d\n", st->codecpar->ch_layout.nb_channels, st->codecpar->sample_rate, sample_rate);
+ if (st->codecpar->sample_rate == 0)
+ sdr->missing_streams--;
+ ff_add_param_change(pkt, 2, 0, sample_rate_p2, 0, 0);
+ st->codecpar->sample_rate = sample_rate_p2;
+ }
+
+ return ret;
+}
+
+
+BandDescriptor band_descs[] = {
+ {"Shortwave band" , "SW", 6000000, 18000000},
+ {"FM broadcast band", "FM", 88000000, 108000000},
+};
+
+ModulationDescriptor modulation_descs[] = {
+ {"Amplitude Modulation", "AM", AM, AVMEDIA_TYPE_AUDIO, probe_am, demodulate_am},
+ {"Frequency Modulation", "FM", FM, AVMEDIA_TYPE_AUDIO, probe_fm, demodulate_fm},
+};
+
+int avpriv_sdr_set_freq(SDRContext *sdr, int64_t freq)
+{
+ freq = av_clip64(freq, sdr->min_center_freq, sdr->max_center_freq);
+
+ if (sdr->set_frequency_callback) {
+ int ret = sdr->set_frequency_callback(sdr, freq);
+ if (ret < 0)
+ return ret;
+ }
+
+ sdr->freq = freq;
+
+ return 0;
+}
+
+static void free_stream(SDRContext *sdr, int stream_index)
+{
+ AVFormatContext *s = sdr->avfmt;
+ AVStream *st = s->streams[stream_index];
+ SDRStream *sst = st->priv_data;
+
+ av_tx_uninit(&sst->ifft_ctx);
+ av_tx_uninit(&sst->fft_p2_ctx);
+ av_tx_uninit(&sst->ifft_p2_ctx);
+ sst->ifft = NULL;
+ sst->fft_p2 = NULL;
+ sst->ifft_p2 = NULL;
+ sst->block_size = 0;
+
+ av_freep(&sst->out_buf);
+ av_freep(&sst->block);
+ av_freep(&sst->iblock);
+ av_freep(&sst->icarrier);
+ av_freep(&sst->iside);
+ av_freep(&sst->window);
+ av_freep(&sst->window_p2);
+
+}
+
+static int setup_stream(SDRContext *sdr, int stream_index, Station *station)
+{
+ AVFormatContext *s = sdr->avfmt;
+ AVStream *st = s->streams[stream_index];
+ SDRStream *sst = st->priv_data;
+ int ret;
+
+ //For now we expect each station to be only demodulated once, nothing should break though if its done more often
+ av_assert0(station->stream == NULL || station->stream == sst);
+
+ if (!station->stream)
+ av_log(sdr->avfmt, AV_LOG_INFO, "setup stream_index: %d to Station: %f Mhz\n", stream_index, station->frequency / 1000000);
+
+ if (sst->station)
+ sst->station->stream = NULL;
+
+ sst->station = station;
+ station->stream = sst;
+
+ if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
+ free_stream(sdr, stream_index);
+
+ for (sst->block_size = 4; 2ll *sst->station->bandwidth * sdr->block_size / sdr->sdr_sample_rate > sst->block_size; sst->block_size <<= 1)
+ ;
+ sst->block_size = FFMIN(sdr->block_size, sst->block_size);
+
+ ret = av_tx_init(&sst->ifft_ctx, &sst->ifft, AV_TX_FLOAT_FFT, 1, 2*sst->block_size, NULL, 0);
+ if (ret < 0)
+ return ret;
+
+ if (sst->station->bandwidth_p2) {
+ //Allocate 2nd stage demodulation fields if needed
+ ret = av_tx_init(&sst-> fft_p2_ctx, &sst-> fft_p2, AV_TX_FLOAT_FFT, 0, 2*sst->block_size , NULL, 0);
+ if (ret < 0)
+ return ret;
+
+ for (sst->block_size_p2 = 4; 2ll *sst->station->bandwidth_p2 * sdr->block_size / sdr->sdr_sample_rate > sst->block_size_p2; sst->block_size_p2 <<= 1)
+ ;
+ ret = av_tx_init(&sst->ifft_p2_ctx, &sst->ifft_p2, AV_TX_FLOAT_FFT, 1, 2*sst->block_size_p2, NULL, 0);
+ if (ret < 0)
+ return ret;
+
+ }
+
+ sst->out_buf = av_malloc(sizeof(*sst->out_buf) * 2 * sst->block_size);
+ sst->block = av_malloc(sizeof(*sst-> block) * 2 * sst->block_size);
+ sst->iblock = av_malloc(sizeof(*sst->iblock) * 2 * sst->block_size);
+ sst->icarrier = av_malloc(sizeof(*sst->icarrier) * 2 * sst->block_size);
+ sst->iside = av_malloc(sizeof(*sst->iside) * 2 * sst->block_size_p2);
+ sst->window = av_malloc(sizeof(*sst->window) * 2 * sst->block_size);
+ sst->window_p2 = av_malloc(sizeof(*sst->window_p2)* 2 * sst->block_size_p2);
+ if (!sst->out_buf || !sst->block || !sst->iblock || !sst->icarrier || !sst->iside || !sst->window || !sst->window_p2)
+ return AVERROR(ENOMEM);
+
+ avpriv_kbd_window_init(sst->window, sdr->kbd_alpha, sst->block_size);
+ for(int i = sst->block_size; i < 2 * sst->block_size; i++) {
+ sst->window[i] = sst->window[2*sst->block_size - i - 1];
+ }
+
+ avpriv_kbd_window_init(sst->window_p2, sdr->kbd_alpha, sst->block_size_p2);
+ for(int i = sst->block_size_p2; i < 2 * sst->block_size_p2; i++) {
+ sst->window_p2[i] = sst->window_p2[2*sst->block_size_p2 - i - 1];
+ }
+ }
+
+ return 0;
+}
+
+static void inject_block_into_fifo(SDRContext *sdr, AVFifo *fifo, FIFOElement *fifo_element, const char *error_message)
+{
+ int ret;
+
+ pthread_mutex_lock(&sdr->mutex);
+ ret = av_fifo_write(fifo, fifo_element, 1);
+ pthread_mutex_unlock(&sdr->mutex);
+ if (ret < 0) {
+ av_log(sdr->avfmt, AV_LOG_DEBUG, "%s", error_message);
+ av_freep(&fifo_element->halfblock);
+ }
+}
+
+static void flush_fifo(SDRContext *sdr, AVFifo *fifo)
+{
+ FIFOElement fifo_element;
+
+ pthread_mutex_lock(&sdr->mutex);
+ while(fifo) {
+ int ret = av_fifo_read(fifo, &fifo_element, 1);
+ if (ret < 0)
+ break;
+ av_freep(&fifo_element.halfblock);
+ }
+ pthread_mutex_unlock(&sdr->mutex);
+}
+
+static int sdrfile_read_callback(SDRContext *sdr, FIFOElement *fifo_element, int remaining)
+{
+ AVFormatContext *avfmt = sdr->avfmt;
+ int64_t size;
+ void *buffer = (uint8_t*)fifo_element->halfblock + (sdr->block_size - remaining) * sdr->sample_size;
+ int ret;
+
+ if (!sdr->remaining_file_block_size) {
+ int block_size;
+ avio_skip(avfmt->pb, 16 + 4); //FFSDR00XintYYBE
+ block_size = avio_rb32(avfmt->pb);
+ fifo_element->center_frequency = av_int2double(avio_rb64(avfmt->pb));
+ avio_rb64(avfmt->pb); //pts
+ avio_skip(avfmt->pb, sdr->fileheader_size - 40);
+ sdr->remaining_file_block_size = block_size;
+ }
+ pthread_mutex_lock(&sdr->mutex);
+
+ //We have no FIFO API to check how much we can write
+ size = sdr->sdr_sample_rate / sdr->block_size - av_fifo_can_read(sdr->full_block_fifo);
+ pthread_mutex_unlock(&sdr->mutex);
+ if (size <= 0) {
+ av_usleep(10*1000);
+ return AVERROR(EAGAIN);
+ }
+ size = FFMIN(remaining, sdr->remaining_file_block_size) * sdr->sample_size;
+ ret = avio_read(avfmt->pb, buffer, size);
+ if (ret == AVERROR_EOF || (ret > 0 && ret % sdr->sample_size)) {
+ avio_seek(avfmt->pb, SEEK_SET, 0);
+ av_log(avfmt, AV_LOG_INFO, "EOF, will wraparound\n");
+ return AVERROR(EAGAIN);
+ } else if (ret == AVERROR(EAGAIN)) {
+ av_log(avfmt, AV_LOG_DEBUG, "read EAGAIN\n");
+ return AVERROR(EAGAIN);
+ } else if (ret < 0) {
+ av_log(avfmt, AV_LOG_ERROR, "read Failed with (%d)\n", ret);
+ return AVERROR(EAGAIN);
+ }
+
+ ret /= sdr->sample_size;
+ sdr->remaining_file_block_size -= ret;
+ return ret;
+}
+
+static int snap2station(SDRContext *sdr, int *seek_direction) {
+ AVFormatContext *avfmt = sdr->avfmt;
+ AVStream *st = avfmt->streams[sdr->single_ch_audio_st_index];
+ SDRStream *sst = st->priv_data;
+ double current_freq;
+ double best_distance = INT64_MAX;
+ Station *best_station = NULL;
+
+ if (sst->station) {
+ current_freq = sst->station->frequency;
+ } else if (sdr->station_freq) {
+ current_freq = sdr->station_freq;
+ } else
+ current_freq = sdr->block_center_freq;
+
+ for(int i = 0; i<sdr->nb_stations; i++) {
+ Station *station = sdr->station[i];
+ double distance = station->frequency - current_freq;
+
+ if (distance * *seek_direction < 0 || station == sst->station)
+ continue;
+ distance = fabs(distance);
+ if (distance < best_distance) {
+ best_distance = distance;
+ best_station = station;
+ }
+ }
+ av_assert0(!best_station || best_station != sst->station);
+
+ if (best_station) {
+ int ret = setup_stream(sdr, sdr->single_ch_audio_st_index, best_station);
+ if (ret < 0) {
+ av_log(avfmt, AV_LOG_DEBUG, "setup_stream failed\n");
+ return ret;
+ }
+
+ pthread_mutex_lock(&sdr->mutex);
+ *seek_direction =
+ sdr->seek_direction = 0;
+ sdr->wanted_freq = lrint(best_station->frequency + 213*1000); // We target a bit off teh exact frequency to avoid artifacts
+ //200*1000 had artifacts
+
+ av_log(avfmt, AV_LOG_DEBUG, "request f = %"PRId64"\n", sdr->wanted_freq);
+ pthread_mutex_unlock(&sdr->mutex);
+ return 1;
+ }
+
+ return 0;
+}
+
+/**
+ * move frequency in the delta direction within a band
+ * This can be called from the thread so it must only access runtime constants from sdr->
+ */
+static int64_t snap2band(SDRContext *sdr, int64_t wanted_freq, int64_t delta) {
+ int64_t best_distance = INT64_MAX;
+ int64_t best_frequency = wanted_freq;
+ int64_t wrap_frequency = delta > 0 ? 0 : INT64_MAX;
+
+ for(int i = 0; i<FF_ARRAY_ELEMS(band_descs); i++) {
+ int64_t distance, new_freq;
+ BandDescriptor *desc = &band_descs[i];
+ int64_t min_center_freq = av_clip64(desc->freq_min, sdr->min_freq, sdr->max_freq);
+ int64_t max_center_freq = av_clip64(desc->freq_max, sdr->min_freq, sdr->max_freq);
+
+ // Is the band within the frequency range ?
+ if (min_center_freq == max_center_freq)
+ continue;
+
+ if (max_center_freq - min_center_freq <= sdr->bandwidth) {
+ min_center_freq =
+ max_center_freq = (min_center_freq + max_center_freq) / 2;
+ } else {
+ min_center_freq += sdr->bandwidth / 2;
+ max_center_freq -= sdr->bandwidth / 2;
+ }
+
+ // Is the band in the direction we want to seek to ?
+ if (FFSIGN(min_center_freq - wanted_freq) == FFSIGN(max_center_freq - wanted_freq))
+ if (FFSIGN(delta) != FFSIGN(min_center_freq - wanted_freq))
+ continue;
+
+ new_freq = av_clip64(wanted_freq + delta, min_center_freq, max_center_freq);
+ distance = FFABS(new_freq - (wanted_freq + delta));
+ if (distance < best_distance && new_freq != wanted_freq) {
+ best_distance = distance;
+ best_frequency = new_freq;
+ }
+ }
+
+ //If we cant move (that is hit the end) then wrap around unless we already wraped
+ if (best_distance == INT64_MAX && wanted_freq != wrap_frequency)
+ return snap2band(sdr, wrap_frequency, delta);
+
+ return best_frequency;
+}
+
+/**
+ * Grab data from soapy and put it in a bigger buffer.
+ * This thread would not be needed if libsoapy internal buffering was not restricted
+ * to a few milli seconds. libavformat cannot guarntee that it will get called from the user
+ * application every 10-20ms or something like that so we need a thread to pull data from soapy
+ * and put it in a larger buffer that we can then read from at the rate the code is called
+ * by the user
+ */
+static void *soapy_needs_bigger_buffers_worker(SDRContext *sdr)
+{
+ AVFormatContext *avfmt = sdr->avfmt;
+ unsigned block_counter = 0;
+
+ sdr->remaining_file_block_size = 0;
+
+ ff_thread_setname("sdrdemux - soapy rx");
+
+ while(!atomic_load(&sdr->close_requested)) {
+ FIFOElement fifo_element;
+ int remaining, ret;
+ int empty_blocks, full_blocks;
+
+ //i wish av_fifo was thread safe
+ pthread_mutex_lock(&sdr->mutex);
+ ret = av_fifo_read(sdr->empty_block_fifo, &fifo_element, 1);
+ empty_blocks = av_fifo_can_read(sdr->empty_block_fifo);
+ full_blocks = av_fifo_can_read(sdr->full_block_fifo);
+ pthread_mutex_unlock(&sdr->mutex);
+
+ if (ret < 0) {
+ av_log(avfmt, AV_LOG_WARNING, "Allocating new block due to lack of space full:%d empty:%d\n", full_blocks, empty_blocks);
+ fifo_element.halfblock = av_malloc(sdr->block_size * sdr->sample_size);
+ if (!fifo_element.halfblock) {
+ av_log(avfmt, AV_LOG_ERROR, "Allocation failed, waiting for free space\n");
+ // we wait 10ms here, tests have shown soapy to loose data when it is not serviced for 40-50ms
+ av_usleep(10*1000);
+ continue;
+ }
+ }
+
+ block_counter ++;
+ pthread_mutex_lock(&sdr->mutex);
+ // we try to get 2 clean blocks after windowing, to improve chances scanning doesnt miss too much
+ // First block after parameter change is not reliable, we do not assign it any frequency
+ // 2 blocks are needed with windowing to get a clean FFT output
+ // Thus > 3 is the minimum for the next frequency update if we want to do something reliable with the data
+ if (sdr->seek_direction && block_counter > 3) {
+ sdr->wanted_freq = snap2band(sdr, sdr->wanted_freq, sdr->seek_direction*sdr->bandwidth);
+ }
+ if (sdr->wanted_freq != sdr->freq) {
+ //We could use a seperate MUTEX for the FIFO and for soapy
+ avpriv_sdr_set_freq(sdr, sdr->wanted_freq);
+ //This shouldnt really cause any problem if we just continue on error except that we continue returning data with the previous target frequency range
+ //And theres not much else we can do, an error message was already printed by avpriv_sdr_set_freq() in that case
+ block_counter = 0; // we just changed the frequency, do not trust the next blocks content
+ }
+ pthread_mutex_unlock(&sdr->mutex);
+
+ fifo_element.center_frequency = block_counter > 0 ? sdr->freq : 0;
+
+ remaining = sdr->block_size;
+ while (remaining && !atomic_load(&sdr->close_requested)) {
+ ret = sdr->read_callback(sdr, &fifo_element, remaining);
+ if (ret == AVERROR(EAGAIN))
+ continue;
+
+ av_assert0(ret <= remaining);
+ remaining -= ret;
+ }
+
+ inject_block_into_fifo(sdr, sdr->full_block_fifo, &fifo_element, "block fifo overflow, discarding block\n");
+ }
+ av_assert0(atomic_load(&sdr->close_requested) == 1);
+
+ return NULL;
+}
+
+int avpriv_sdr_common_init(AVFormatContext *s)
+{
+ SDRContext *sdr = s->priv_data;
+ AVStream *st;
+ SDRStream *sst;
+ int ret;
+ float scale = 1.0 / sdr->sample_scale;
+
+ sdr->avfmt = s;
+
+ if (sdr->width>1 && sdr->height>1) {
+ /* video stream */
+ st = avformat_new_stream(s, NULL);
+ if (!st)
+ return AVERROR(ENOMEM);
+ sst = av_mallocz(sizeof(SDRStream));
+ if (!sst)
+ return AVERROR(ENOMEM);
+ st->priv_data = sst;
+ st->codecpar->codec_type = AVMEDIA_TYPE_VIDEO;
+ st->codecpar->codec_id = AV_CODEC_ID_RAWVIDEO;
+ st->codecpar->codec_tag = 0; /* no fourcc */
+ st->codecpar->format = AV_PIX_FMT_BGRA;
+ st->codecpar->width = sdr->width;
+ st->codecpar->height = sdr->height;
+ avpriv_set_pts_info(st, 64, 1, (48000/128) << FREQ_BITS);
+ sdr->waterfall_st_index = st->index;
+ } else
+ sdr->waterfall_st_index = -1;
+
+ if (sdr->mode == SingleStationMode) {
+ /* audio stream */
+ st = avformat_new_stream(s, NULL);
+ if (!st)
+ return AVERROR(ENOMEM);
+ sst = av_mallocz(sizeof(SDRStream));
+ if (!sst)
+ return AVERROR(ENOMEM);
+ st->priv_data = sst;
+ st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
+ st->codecpar->codec_tag = 0; /* no fourcc */
+ st->codecpar->codec_id = HAVE_BIGENDIAN ? AV_CODEC_ID_PCM_F32BE : AV_CODEC_ID_PCM_F32LE;
+ st->codecpar->ch_layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO;
+ st->codecpar->sample_rate = 0; //will be set later
+ avpriv_set_pts_info(st, 64, 1, (48000/128) << FREQ_BITS);
+ sdr->single_ch_audio_st_index = st->index;
+ sdr->missing_streams++;
+ }
+
+ sdr->min_center_freq = sdr->min_freq + sdr->sdr_sample_rate / 2;
+ sdr->max_center_freq = sdr->max_freq + sdr->sdr_sample_rate / 2;
+
+
+ if(!sdr->block_size) {
+ // The user has not set a block_size, we choose one automatically based on the sample rate
+ for(sdr->block_size = 1; 25*sdr->block_size < sdr->sdr_sample_rate; sdr->block_size <<=1)
+ ;
+ } else if(sdr->block_size & (sdr->block_size - 1)) {
+ av_log(s, AV_LOG_ERROR, "Block size must be a power of 2\n");
+ return AVERROR(EINVAL);
+ }
+ av_log(s, AV_LOG_INFO, "Block size %d\n", sdr->block_size);
+
+
+ sdr->windowed_block = av_malloc(sizeof(*sdr->windowed_block) * 2 * sdr->block_size);
+ sdr->block = av_malloc(sizeof(*sdr->block ) * 2 * sdr->block_size);
+ sdr->len2block = av_malloc(sizeof(*sdr->len2block) * 2 * sdr->block_size);
+ sdr->window = av_malloc(sizeof(*sdr->window ) * 2 * sdr->block_size);
+ if (!sdr->windowed_block || !sdr->len2block || !sdr->block || !sdr->window)
+ return AVERROR(ENOMEM);
+
+ ret = av_tx_init(&sdr->fft_ctx, &sdr->fft, AV_TX_FLOAT_FFT, 0, 2*sdr->block_size, NULL, 0);
+ if (ret < 0)
+ return ret;
+
+ avpriv_kbd_window_init(sdr->window, sdr->kbd_alpha, sdr->block_size);
+
+ for(int i = sdr->block_size; i < 2 * sdr->block_size; i++) {
+ sdr->window[i] = sdr->window[2*sdr->block_size - i - 1];
+ }
+ for (int i = 0; i < 2 * sdr->block_size; i++)
+ sdr->window[i] *= ((i&1) ? 1:-1) * scale;
+
+ if (sdr->waterfall_st_index >= 0) {
+ AVStream *st = s->streams[sdr->waterfall_st_index];
+ SDRStream *sst = st->priv_data;
+ av_assert0(st->codecpar->codec_type == AVMEDIA_TYPE_VIDEO);
+ sst->frame_size = sdr->width * sdr->height * 4;
+ sst->frame_buffer = av_malloc(sst->frame_size * 2);
+ if (!sst->frame_buffer)
+ return AVERROR(ENOMEM);
+ }
+
+ sdr->pts = 0;
+
+ sdr->empty_block_fifo = av_fifo_alloc2(1, sizeof(FIFOElement), AV_FIFO_FLAG_AUTO_GROW);
+ sdr-> full_block_fifo = av_fifo_alloc2(1, sizeof(FIFOElement), AV_FIFO_FLAG_AUTO_GROW);
+ if (!sdr->empty_block_fifo || !sdr-> full_block_fifo)
+ return AVERROR(ENOMEM);
+ //Limit fifo to 1second to avoid OOM
+ av_fifo_auto_grow_limit(sdr->empty_block_fifo, sdr->sdr_sample_rate / sdr->block_size);
+ av_fifo_auto_grow_limit(sdr-> full_block_fifo, sdr->sdr_sample_rate / sdr->block_size);
+
+ atomic_init(&sdr->close_requested, 0);
+ ret = pthread_mutex_init(&sdr->mutex, NULL);
+ if (ret) {
+ av_log(s, AV_LOG_ERROR, "pthread_mutex_init failed: %s\n", strerror(ret));
+ return AVERROR(ret);
+ }
+ sdr->thread_started = 1;
+
+ ret = pthread_create(&sdr->hw_thread, NULL, (void*)soapy_needs_bigger_buffers_worker, sdr);
+ if (ret != 0) {
+ av_log(s, AV_LOG_ERROR, "pthread_create failed : %s\n", strerror(ret));
+ pthread_mutex_destroy(&sdr->mutex);
+ return AVERROR(ret);
+ }
+ sdr->thread_started = 2;
+
+ if(sdr->dump_url) {
+ ret = avio_open2(&sdr->dump_avio, sdr->dump_url, AVIO_FLAG_WRITE, NULL, NULL);
+ if (ret < 0) {
+ fprintf(stderr, "Unable to open %s\n", sdr->dump_url);
+ return ret;
+ }
+ }
+
+ return 0;
+}
+
+/**
+ * This matches sdrindev_initial_hw_setup() but using a file and no hardware
+ */
+static int sdrfile_initial_setup(AVFormatContext *s)
+{
+ SDRContext *sdr = s->priv_data;
+
+ int version, ret, bits;
+
+ avio_skip(s->pb, 5); //FFSDR
+ version = avio_rb24(s->pb); //000
+ avio_skip(s->pb, 3); //int
+ bits = avio_rb16(s->pb);
+ if (bits == AV_RB16("16")) {
+ sdr->sample_size = 4;
+ sdr->sample_scale = 32768;
+ } else if (bits == AV_RB16("08")) {
+ sdr->sample_size = 2;
+ sdr->sample_scale = 128;
+ } else
+ return AVERROR_INVALIDDATA;
+
+ avio_skip(s->pb, 3); //BE
+ sdr->sdr_sample_rate = avio_rb32(s->pb);
+ avio_rb32(s->pb); //block_size
+ sdr->wanted_freq = av_int2double(avio_rb64(s->pb));
+ sdr->pts = avio_rb64(s->pb);
+ if (version > AV_RB24("000")) {
+ sdr->bandwidth = avio_rb32(s->pb);
+ sdr->fileheader_size = avio_rb32(s->pb);
+ } else {
+ sdr->bandwidth = sdr->sdr_sample_rate;
+ sdr->fileheader_size = 40;
+ }
+
+ //After reading the first packet header we return to the begin so the packet can be read whole
+ avio_seek(s->pb, 0, SEEK_SET);
+
+ ret = avpriv_sdr_set_freq(sdr, sdr->wanted_freq);
+ if (ret < 0)
+ return ret;
+
+ sdr->read_callback = sdrfile_read_callback;
+
+ return avpriv_sdr_common_init(s);
+}
+
+static inline void draw_point_component(uint8_t *frame_buffer, ptrdiff_t stride, int x, int y, int r, int g, int b, int w, int h)
+{
+ uint8_t *p;
+
+ if (x<0 || y<0 || x>=w || y>=h)
+ return;
+ p = frame_buffer + 4*x + stride*y;
+
+ p[0] = av_clip_uint8(p[0] + (b>>16));
+ p[1] = av_clip_uint8(p[1] + (g>>16));
+ p[2] = av_clip_uint8(p[2] + (r>>16));
+}
+
+// Draw a point with subpixel precission, (it looked bad otherwise)
+static void draw_point(uint8_t *frame_buffer, ptrdiff_t stride, int x, int y, int r, int g, int b, int w, int h)
+{
+ int px = x>>8;
+ int py = y>>8;
+ int sx = x&255;
+ int sy = y&255;
+ int s;
+
+ s = (256 - sx) * (256 - sy);
+ draw_point_component(frame_buffer, stride, px , py , r*s, g*s, b*s, w, h);
+ s = sx * (256 - sy);
+ draw_point_component(frame_buffer, stride, px+1, py , r*s, g*s, b*s, w, h);
+ s = (256 - sx) * sy;
+ draw_point_component(frame_buffer, stride, px , py+1, r*s, g*s, b*s, w, h);
+ s = sx * sy;
+ draw_point_component(frame_buffer, stride, px+1, py+1, r*s, g*s, b*s, w, h);
+}
+
+static void draw_char(uint8_t *frame_buffer, ptrdiff_t stride, char ch, int x0, int y0, int xd, int yd, int r, int g, int b, int w, int h)
+{
+ for(int y = 0; y < 16; y++) {
+ int mask = avpriv_vga16_font[16*ch + y];
+ for(int x = 0; x < 8; x++) {
+ if (mask&0x80)
+ draw_point(frame_buffer, stride, x0 + xd*x - yd*y, y0 + yd*x + xd*y, r, g, b, w, h);
+ mask<<=1;
+ }
+ }
+}
+
+static void draw_string(uint8_t *frame_buffer, ptrdiff_t stride, char *str, int x0, int y0, int xd, int yd, int r, int g, int b, int w, int h)
+{
+ while(*str) {
+ draw_char(frame_buffer, stride, *str++, x0, y0, xd, yd, r, g, b, w, h);
+ x0 += xd*9;
+ y0 += yd*9;
+ }
+}
+
+static int vissualization(SDRContext *sdr, AVStream *st, AVPacket *pkt)
+{
+ SDRStream *sst = st->priv_data;
+ int w = st->codecpar->width;
+ int h = st->codecpar->height;
+ int h2 = FFMIN(64, h / 4);
+ int frame_index = av_rescale(sdr->pts, sdr->fps.num, sdr->fps.den * TIMEBASE);
+ int last_index = av_rescale(sdr->last_pts, sdr->fps.num, sdr->fps.den * TIMEBASE);
+ int skip = frame_index == last_index || sdr->missing_streams;
+ av_assert0(sdr->missing_streams >= 0);
+
+ for(int x= 0; x<w; x++) {
+ int color;
+ int idx = 4*(x + sst->frame_buffer_line*w);
+ int bindex = x * 2ll * sdr->block_size / w;
+ int bindex2 = (x+1) * 2ll * sdr->block_size / w;
+ float a = 0;
+ av_assert0(bindex2 <= 2 * sdr->block_size);
+ for (int i = bindex; i < bindex2; i++) {
+ AVComplexFloat sample = sdr->block[i];
+ a += len2(sample);
+ }
+ color = lrintf(log(a)*8 + 32);
+
+ sst->frame_buffer[idx + 0] = color;
+ sst->frame_buffer[idx + 1] = color;
+ sst->frame_buffer[idx + 2] = color;
+ sst->frame_buffer[idx + 3] = 255;
+ }
+
+ // Display locations of all vissible stations
+// for(int station_index = 0; station_index<sdr->nb_stations; station_index++) {
+// Station *s = sdr->station[station_index];
+// double f = s->frequency;
+// // int bw = s->bandwidth;
+// // int xleft = 256*((f-bw) - sdr->block_center_freq + sdr->sdr_sample_rate/2) * w / sdr->sdr_sample_rate;
+// // int xright= 256*((f+bw) - sdr->block_center_freq + sdr->sdr_sample_rate/2) * w / sdr->sdr_sample_rate;
+// int xmid = 256*( f - sdr->block_center_freq + sdr->sdr_sample_rate/2) * w / sdr->sdr_sample_rate;
+// int g = s->modulation == AM ? 50 : 0;
+// int b = s->modulation == AM ? 0 : 70;
+// int r = s->stream ? 50 : 0;
+//
+// draw_point(sst->frame_buffer, 4*w, xmid, 256*(sst->frame_buffer_line+1), r, g, b, w, h);
+// }
+
+ if (!skip) {
+ int ret = av_new_packet(pkt, sst->frame_size);
+ if (ret < 0)
+ return ret;
+
+ for(int y= 0; y<h2; y++) {
+ for(int x= 0; x<w; x++) {
+ int color;
+ int idx = x + y*w;
+ int idx_t = (idx / h2) + (idx % h2)*w;
+ int bindex = idx * 2ll * sdr->block_size / (w * h2);
+ int bindex2 = (idx+1) * 2ll * sdr->block_size / (w * h2);
+ float a = 0;
+ av_assert0(bindex2 <= 2 * sdr->block_size);
+ for (int i = bindex; i < bindex2; i++) {
+ AVComplexFloat sample = sdr->block[i];
+ a += len2(sample);
+ }
+ color = lrintf(log(a)*9 + 64);
+
+ idx_t *= 4;
+
+ pkt->data[idx_t+0] = color;
+ pkt->data[idx_t+1] = color;
+ pkt->data[idx_t+2] = color;
+ pkt->data[idx_t+3] = 255;
+ }
+ }
+
+ for (int y= h2; y<h; y++)
+ memcpy(pkt->data + 4*y*w, sst->frame_buffer + 4*(y + sst->frame_buffer_line - h2)*w, 4*w);
+
+ for(int station_index = 0; station_index<sdr->nb_stations; station_index++) {
+ Station *s = sdr->station[station_index];
+ double f = s->frequency;
+ int xmid = 256*( f - sdr->block_center_freq + sdr->sdr_sample_rate/2) * w / sdr->sdr_sample_rate;
+ char text[20];
+ int color = s->stream ? 64 : 32;
+ int size = s->stream ? 181 : 128;
+ int xd = size, yd = size;
+ snprintf(text, sizeof(text), "%s %f Mhz",
+ modulation_descs[s->modulation].shortname,
+ f/1000000);
+ draw_string(pkt->data, 4*w, text, xmid + 8*yd, 320*h2, xd, yd, color, color, color, w, h);
+ }
+ }
+
+ if (!sst->frame_buffer_line) {
+ memcpy(sst->frame_buffer + sst->frame_size, sst->frame_buffer, sst->frame_size);
+ sst->frame_buffer_line = h-1;
+ } else
+ sst->frame_buffer_line--;
+
+//TODO
+// draw RDS*
+ return skip;
+}
+
+int avpriv_sdr_read_packet(AVFormatContext *s, AVPacket *pkt)
+{
+ SDRContext *sdr = s->priv_data;
+ int ret, i, full_blocks, seek_direction;
+ FIFOElement fifo_element[2];
+
+process_next_block:
+
+ for (int stream_index = 0; stream_index < s->nb_streams; stream_index++) {
+ AVStream *st = s->streams[stream_index];
+ SDRStream *sst = st->priv_data;
+
+ if (sst->processing_index) {
+ int skip = 1;
+ if (st->codecpar->codec_type == AVMEDIA_TYPE_VIDEO) {
+ skip = vissualization(sdr, st, pkt);
+ if (skip < 0)
+ return skip;
+ } else if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
+ if (sst->station) {
+ skip = 0;
+ ret = modulation_descs[ sst->station->modulation ].demodulate(sdr, stream_index, pkt);
+ if (ret < 0) {
+ av_log(s, AV_LOG_ERROR, "demodulation failed ret = %d\n", ret);
+ }
+ }
+ } else
+ av_assert0(0);
+ sst->processing_index = 0;
+ if (pkt && !skip) {
+ pkt->stream_index = stream_index;
+ pkt->dts = (sdr->pts & (-1<<FREQ_BITS));
+ if (sst->station)
+ pkt->dts += lrint(sst->station->frequency/1000);
+ pkt->pts = pkt->dts;
+
+ return 0;
+ }
+ }
+ }
+
+ pthread_mutex_lock(&sdr->mutex);
+ full_blocks = av_fifo_can_read(sdr->full_block_fifo) - 1;
+ ret = av_fifo_peek(sdr->full_block_fifo, &fifo_element, 2, 0);
+ if (ret >= 0)
+ av_fifo_drain2(sdr->full_block_fifo, 1);
+ seek_direction = sdr->seek_direction; //This doesnt need a mutex here at all but tools might complain
+ pthread_mutex_unlock(&sdr->mutex);
+
+ if (ret < 0) {
+ av_log(s, AV_LOG_DEBUG, "EAGAIN on not enough data\n");
+ return AVERROR(EAGAIN);
+ }
+ if (fifo_element[0].center_frequency != fifo_element[1].center_frequency) {
+ av_log(s, AV_LOG_DEBUG, "Mismatching frequency blocks\n");
+// fifo_element[0].center_frequency = 0;
+// inject_block_into_fifo(sdr, sdr->empty_block_fifo, &fifo_element, "Cannot pass next buffer, freeing it\n");
+ //Its simpler to just continue than to discard this, but we must make sure that on seeking we have matching blocks for each block we want to scan
+ sdr->block_center_freq = 0;
+ sdr->skip_probe = 1; // we want to probe the next block as it will have new frequencies
+ } else
+ sdr->block_center_freq = fifo_element[0].center_frequency;
+
+ if (sdr->dump_avio) {
+ uint8_t header[48] = "FFSDR001int16BE";
+ uint8_t *tmp = (void*)sdr->windowed_block; //We use an unused array as temporary here
+
+ if (sdr->sample_size == 2)
+ memcpy(header + 11, "08", 2);
+
+ AV_WB32(header+16, sdr->sdr_sample_rate);
+ AV_WB32(header+20, sdr->block_size);
+ AV_WB64(header+24, av_double2int(fifo_element[0].center_frequency));
+ AV_WB64(header+32, sdr->pts);
+ AV_WB32(header+40, sdr->bandwidth);
+ AV_WB32(header+44, sizeof(header));
+
+ avio_write(sdr->dump_avio, header, sizeof(header));
+
+ if (sdr->sample_size == 2) {
+ avio_write(sdr->dump_avio, fifo_element[0].halfblock, sdr->block_size * sdr->sample_size);
+ } else if (sdr->sample_size == 4){
+ const int16_t *halfblock = fifo_element[0].halfblock;
+ for(int i=0; i<sdr->block_size; i++) {
+ AV_WB16(tmp + 4*i + 0, halfblock[2*i+0]);
+ AV_WB16(tmp + 4*i + 2, halfblock[2*i+1]);
+ }
+ avio_write(sdr->dump_avio, tmp, sdr->block_size * sdr->sample_size);
+ } else {
+ //We convert to 16bit to safe space, i have no hardware which can do more than 14bit
+ const AVComplexFloat *halfblock = fifo_element[0].halfblock;
+ for(int i=0; i<sdr->block_size; i++) {
+ int re = lrint(halfblock[i].re * 32768);
+ int im = lrint(halfblock[i].im * 32768);
+ AV_WB16(tmp + 4*i + 0, re);
+ AV_WB16(tmp + 4*i + 2, im);
+ }
+ avio_write(sdr->dump_avio, tmp, sdr->block_size * 4);
+ }
+
+ }
+//TODO merge scale into window
+ if (sdr->sample_size == 2) {
+ const int8_t *halfblock0 = fifo_element[0].halfblock;
+ const int8_t *halfblock1 = fifo_element[1].halfblock;
+ for (i = 0; i<sdr->block_size; i++) {
+ sdr->windowed_block[i].re = halfblock0[2*i+0] * sdr->window[i];
+ sdr->windowed_block[i].im = halfblock0[2*i+1] * sdr->window[i];
+ }
+ for (i = sdr->block_size; i<2*sdr->block_size; i++) {
+ sdr->windowed_block[i].re = halfblock1[2*(i - sdr->block_size)+0] * sdr->window[i];
+ sdr->windowed_block[i].im = halfblock1[2*(i - sdr->block_size)+1] * sdr->window[i];
+ }
+ } else if (sdr->sample_size == 4){
+ const int16_t *halfblock0 = fifo_element[0].halfblock;
+ const int16_t *halfblock1 = fifo_element[1].halfblock;
+ for (i = 0; i<sdr->block_size; i++) {
+ sdr->windowed_block[i].re = halfblock0[2*i+0] * sdr->window[i];
+ sdr->windowed_block[i].im = halfblock0[2*i+1] * sdr->window[i];
+ }
+ for (i = sdr->block_size; i<2*sdr->block_size; i++) {
+ sdr->windowed_block[i].re = halfblock1[2*(i - sdr->block_size)+0] * sdr->window[i];
+ sdr->windowed_block[i].im = halfblock1[2*(i - sdr->block_size)+1] * sdr->window[i];
+ }
+ } else {
+ const AVComplexFloat *halfblock0 = fifo_element[0].halfblock;
+ const AVComplexFloat *halfblock1 = fifo_element[1].halfblock;
+ for (i = 0; i<sdr->block_size; i++) {
+ sdr->windowed_block[i].re = halfblock0[i].re * sdr->window[i];
+ sdr->windowed_block[i].im = halfblock0[i].im * sdr->window[i];
+ }
+ for (i = sdr->block_size; i<2*sdr->block_size; i++) {
+ sdr->windowed_block[i].re = halfblock1[i - sdr->block_size].re * sdr->window[i];
+ sdr->windowed_block[i].im = halfblock1[i - sdr->block_size].im * sdr->window[i];
+ }
+ }
+
+ inject_block_into_fifo(sdr, sdr->empty_block_fifo, &fifo_element[0], "Cannot pass next buffer, freeing it\n");
+#ifdef SYN_TEST //synthetic test signal
+ static int64_t synp=0;
+ AVLFG avlfg;
+ if(!synp)
+ av_lfg_init(&avlfg, 0);
+ for(int i = 0; i<2*sdr->block_size; i++) {
+ double f = F2INDEX(7123456.78901234567);
+ int64_t synp2 = synp % (40*sdr->block_size);
+ double fsig0 = 0.00002;
+ double fsig2= 123;
+
+ if (!i)
+ av_log(0,0, "i= %f %f\n", f, INDEX2F(f));
+ double noise[2] = {0,0};
+ double sig0 = 1.0 + 0.25*sin(synp2*fsig0*synp2*M_PI / sdr->block_size);
+ double sig2 = 0.1 + 0.03*cos(synp*fsig2*M_PI / sdr->block_size);
+ if (i & 256)
+ av_bmg_get(&avlfg, noise);
+ sdr->windowed_block[i].re = (noise[0]*0.0000001 + 0.00001*sig0*sin(synp*M_PI*(f)/sdr->block_size)
+ + 0.00001*sig2*cos(synp*M_PI*(f)/sdr->block_size)
+ ) * fabs(sdr->window[i]);
+ sdr->windowed_block[i].im = noise[1]*0.0000001 * sdr->window[i];
+ synp++;
+ }
+ synp -= sdr->block_size;
+#endif
+
+
+ sdr->fft(sdr->fft_ctx, sdr->block, sdr->windowed_block, sizeof(AVComplexFloat));
+ // windowed_block is unused now, we can fill it with the next blocks data
+
+ if (sdr->block_center_freq) {
+ if (sdr->skip_probe-- <= 0) {
+ //Probing takes a bit of time, lets not do it every time
+ sdr->skip_probe = 5;
+ probe_common(sdr);
+
+ for(int i = 0; i < FF_ARRAY_ELEMS(modulation_descs); i++) {
+ ModulationDescriptor *md = &modulation_descs[i];
+ md->probe(sdr);
+ av_assert0(i == md->modulation);
+ }
+
+ decay_stations(sdr);
+ }
+
+ if (sdr->mode == SingleStationMode) {
+ AVStream *st = s->streams[sdr->single_ch_audio_st_index];
+ SDRStream *sst = st->priv_data;
+
+ if (!sst->station || seek_direction) {
+ ret = snap2station(sdr, &seek_direction);
+ if (ret < 0)
+ return ret;
+ }
+ } else {
+ av_assert0(sdr->mode == AllStationMode);
+ for(int i = 0; i<sdr->nb_stations; i++) {
+ Station *station = sdr->station[i];
+ if (!station->stream) {
+ /* audio stream */
+ AVStream *st = avformat_new_stream(s, NULL);
+ SDRStream *sst;
+ if (!st)
+ return AVERROR(ENOMEM);
+ sst = av_mallocz(sizeof(*sst));
+ if (!sst)
+ return AVERROR(ENOMEM);
+ st->priv_data = sst;
+ st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
+ st->codecpar->codec_tag = 0; /* no fourcc */
+ st->codecpar->codec_id = HAVE_BIGENDIAN ? AV_CODEC_ID_PCM_F32BE : AV_CODEC_ID_PCM_F32LE;
+ st->codecpar->ch_layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO;
+ st->codecpar->sample_rate = 0; // will be set later
+ avpriv_set_pts_info(st, 64, 1, (48000/128) << FREQ_BITS);
+ ret = setup_stream(sdr, st->index, station);
+ if (ret < 0)
+ return ret;
+ sdr->missing_streams++;
+ }
+ }
+ }
+
+ // new data is available for all streams, lets tell them
+ //TODO with mixed blocks during scanning theres more than one block_center_freq, we may want to try to support them as they likely have demodulatable data
+ for (int stream_index = 0; stream_index < s->nb_streams; stream_index++) {
+ AVStream *st = s->streams[stream_index];
+ SDRStream *sst = st->priv_data;
+ sst->processing_index += sdr->block_size;
+ }
+ }
+
+ sdr->last_pts = sdr->pts;
+ sdr->pts += av_rescale(sdr->block_size, TIMEBASE, sdr->sdr_sample_rate);
+
+ //some user apps force a delay on EAGAIN so we cannot always return EAGAIN or we overflow the fifos
+ if (full_blocks >= 2)
+ goto process_next_block;
+
+ return AVERROR(EAGAIN);
+}
+
+int avpriv_sdr_read_seek(AVFormatContext *s, int stream_index,
+ int64_t target, int flags)
+{
+ SDRContext *sdr = s->priv_data;
+ int64_t step;
+ int dir = (flags & AVSEEK_FLAG_BACKWARD) ? -1 : 1;
+ AVStream *st = s->streams[stream_index];
+ SDRStream *sst = st->priv_data;
+ int ret;
+
+ if (sdr->mode != SingleStationMode) {
+ return AVERROR(ENOTSUP);
+ }
+
+ av_assert0(stream_index >= 0);
+
+ step = FFABS(target - sdr->pts);
+ if (step < 35 * TIMEBASE) {
+ target = (sdr->pts & (-1<<FREQ_BITS)) + dir;
+ if (sst->station)
+ target += lrint(sst->station->frequency/1000);
+ }else if (step < 330 * TIMEBASE)
+ return AVERROR(ENOTSUP); // Seek to next/prev band
+ else
+ return AVERROR(ENOTSUP); // Reserved for future ideas
+
+ ret = snap2station(sdr, &dir);
+ if (ret < 0)
+ return ret;
+ //snap2station found no station lets command the thread to seek
+ if (!ret) {
+ pthread_mutex_lock(&sdr->mutex);
+ sdr->seek_direction = dir;
+ pthread_mutex_unlock(&sdr->mutex);
+ flush_fifo(sdr, sdr->full_block_fifo);
+ }
+
+ return 0;
+}
+
+void avpriv_sdr_stop_threading(AVFormatContext *s)
+{
+ SDRContext *sdr = s->priv_data;
+
+ atomic_store(&sdr->close_requested, 1);
+
+ if(sdr->thread_started >= 2)
+ pthread_join(sdr->hw_thread, NULL);
+
+ if(sdr->thread_started >= 1) {
+ flush_fifo(sdr, sdr->empty_block_fifo);
+ flush_fifo(sdr, sdr-> full_block_fifo);
+
+ pthread_mutex_destroy(&sdr->mutex);
+ }
+
+ sdr->thread_started = 0;
+}
+
+int avpriv_sdr_read_close(AVFormatContext *s)
+{
+ SDRContext *sdr = s->priv_data;
+ int i;
+
+ avpriv_sdr_stop_threading(s);
+
+ av_fifo_freep2(&sdr->empty_block_fifo);
+ av_fifo_freep2(&sdr->full_block_fifo);
+
+ for (i = 0; i < s->nb_streams; i++) {
+ AVStream *st = s->streams[i];
+ SDRStream *sst = st->priv_data;
+
+ free_stream(sdr, i);
+
+ av_freep(&sst->frame_buffer);
+ sst->frame_size = 0;
+ }
+
+ for (i = 0; i < sdr->nb_stations; i++) {
+ free_station(sdr->station[i]);
+ }
+ sdr->nb_stations = 0;
+ av_freep(&sdr->station);
+
+ av_freep(&sdr->windowed_block);
+ av_freep(&sdr->block);
+ av_freep(&sdr->len2block);
+ av_freep(&sdr->window);
+
+ av_tx_uninit(&sdr->fft_ctx);
+ sdr->fft = NULL;
+
+ avio_close(sdr->dump_avio);
+
+ return 0;
+}
+
+static int sdrfile_probe(const AVProbeData *p)
+{
+ if (memcmp(p->buf , "FFSDR00", 7))
+ return 0;
+ if (memcmp(p->buf+8, "int16BE", 7) &&
+ memcmp(p->buf+8, "int08BE", 7))
+ return 0;
+ return AVPROBE_SCORE_MAX;
+}
+
+#define OFFSET(x) offsetof(SDRContext, x)
+#define DEC AV_OPT_FLAG_DECODING_PARAM
+
+const AVOption avpriv_sdr_options[] = {
+ { "video_size", "set frame size", OFFSET(width), AV_OPT_TYPE_IMAGE_SIZE, {.str = "0x0"}, 0, 0, DEC },
+ { "framerate" , "set frame rate", OFFSET(fps), AV_OPT_TYPE_VIDEO_RATE, {.str = "25"}, 0, INT_MAX,DEC },
+ { "block_size", "FFT block size", OFFSET(block_size), AV_OPT_TYPE_INT, {.i64 = 0}, 0, INT_MAX, DEC},
+ { "mode", "" , OFFSET(mode), AV_OPT_TYPE_INT, {.i64 = SingleStationMode}, 0, ModeNB-1, DEC, "mode"},
+ { "single_mode", "Demodulate 1 station", 0, AV_OPT_TYPE_CONST, {.i64 = SingleStationMode}, 0, 0, DEC, "mode"},
+ { "all_mode" , "Demodulate all station", 0, AV_OPT_TYPE_CONST, {.i64 = AllStationMode}, 0, 0, DEC, "mode"},
+
+ { "station_freq", "current station/channel/stream frequency", OFFSET(station_freq), AV_OPT_TYPE_INT64, {.i64 = 88000000}, 0, INT64_MAX, DEC},
+
+ { "driver" , "sdr driver name" , OFFSET(driver_name), AV_OPT_TYPE_STRING, {.str = NULL}, 0, 0, DEC},
+ { "rtlsdr_fixes" , "workaround rtlsdr issues", OFFSET(rtlsdr_fixes), AV_OPT_TYPE_INT , {.i64 = -1}, -1, 1, DEC},
+ { "sdr_sr" , "sdr sample rate" , OFFSET(sdr_sample_rate ), AV_OPT_TYPE_INT , {.i64 = 0}, 0, INT_MAX, DEC},
+ { "sdr_freq", "sdr frequency" , OFFSET(wanted_freq), AV_OPT_TYPE_INT64 , {.i64 = 9000000}, 0, INT64_MAX, DEC},
+ { "sdr_agc" , "sdr automatic gain control", OFFSET(sdr_agc), AV_OPT_TYPE_BOOL , {.i64 = 1}, -1, 1, DEC},
+ { "sdr_adcc" ,"sdr automatic dc correction", OFFSET(sdr_adcc), AV_OPT_TYPE_BOOL , {.i64 = -1}, -1, 1, DEC},
+ { "min_freq", "minimum frequency", OFFSET(min_freq ), AV_OPT_TYPE_INT64 , {.i64 = 0}, 0, INT64_MAX, DEC},
+ { "max_freq", "maximum frequency", OFFSET(max_freq ), AV_OPT_TYPE_INT64 , {.i64 = 0}, 0, INT64_MAX, DEC},
+
+ { "dumpurl", "url to dump soapy output to" , OFFSET(dump_url), AV_OPT_TYPE_STRING , {.str = NULL}, 0, 0, DEC},
+ { "kbd_alpha", "Kaiser Bessel window parameter" , OFFSET(kbd_alpha), AV_OPT_TYPE_INT , {.i64 = 8}, 1, 16, DEC},
+
+
+ { "am_mode", "AM Demodulation Mode", OFFSET(am_mode ), AV_OPT_TYPE_INT , {.i64 = AMMidSide}, 0, AMModeNB-1, DEC, "am_mode"},
+ { "am_leftright", "AM Demodulation Left Right", 0, AV_OPT_TYPE_CONST, {.i64 = AMLeftRight}, 0, 0, DEC, "am_mode"},
+ { "am_midside", "AM Demodulation Mid Side", 0, AV_OPT_TYPE_CONST, {.i64 = AMMidSide}, 0, 0, DEC, "am_mode"},
+ { "am_inphase", "AM Demodulation In Phase", 0, AV_OPT_TYPE_CONST, {.i64 = AMInPhase}, 0, 0, DEC, "am_mode"},
+ { "am_envelope","AM Demodulation EnvelopeDC", 0, AV_OPT_TYPE_CONST, {.i64 = AMEnvelope}, 0, 0, DEC, "am_mode"},
+
+
+ { "fm_emphasis", "FM Emphasis", OFFSET(emphasis_mode), AV_OPT_TYPE_INT, {.i64 = EMPHASIS_75us}, 0, EMPHASISNB - 1, DEC, "fm_emphasis"},
+ { "emphasis75us", "FM Emphasis 75us", 0, AV_OPT_TYPE_CONST, {.i64 = EMPHASIS_75us}, 0, 0, DEC, "fm_emphasis"},
+ { "emphasis50us", "FM Emphasis 50us", 0, AV_OPT_TYPE_CONST, {.i64 = EMPHASIS_50us}, 0, 0, DEC, "fm_emphasis"},
+ { "none" , "No FM Emphasis" , 0, AV_OPT_TYPE_CONST, {.i64 = EMPHASIS_NONE}, 0, 0, DEC, "fm_emphasis"},
+ { NULL },
+};
+
+static const AVClass sdrfile_demuxer_class = {
+ .class_name = "sdrfile",
+ .item_name = av_default_item_name,
+ .option = avpriv_sdr_options,
+ .version = LIBAVUTIL_VERSION_INT,
+ .category = AV_CLASS_CATEGORY_DEMUXER,
+};
+
+const AVInputFormat ff_sdrfile_demuxer = {
+ .name = "sdrfile",
+ .long_name = NULL_IF_CONFIG_SMALL("Software Defined Radio Demodulator (Using a file for testing)"),
+ .priv_data_size = sizeof(SDRContext),
+ .read_probe = sdrfile_probe,
+ .read_header = sdrfile_initial_setup,
+ .read_packet = avpriv_sdr_read_packet,
+ .read_close = avpriv_sdr_read_close,
+ .read_seek = avpriv_sdr_read_seek,
+ .flags_internal = FF_FMT_INIT_CLEANUP,
+ .priv_class = &sdrfile_demuxer_class,
+};
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc> --- configure | 4 + doc/demuxers.texi | 90 ++ libavdevice/Makefile | 1 + libavdevice/alldevices.c | 1 + libavdevice/sdrindev.c | 369 ++++++++ libavformat/Makefile | 1 + libavformat/allformats.c | 1 + libavformat/sdr.h | 239 +++++ libavformat/sdrdemux.c | 1843 ++++++++++++++++++++++++++++++++++++++ 9 files changed, 2549 insertions(+) create mode 100644 libavdevice/sdrindev.c create mode 100644 libavformat/sdr.h create mode 100644 libavformat/sdrdemux.c