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shairport-sync/audio_jack.c

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/*
* jack output driver. This file is part of Shairport Sync.
* Copyright (c) 2019 -- 2022 Mike Brady <4265913+mikebrady@users.noreply.github.com>,
* Jörn Nettingsmeier <nettings@luchtbeweging.nl>
*
* All rights reserved.
*
* Permission to use, copy, modify, and distribute this software for any
* purpose with or without fee is hereby granted, provided that the above
* copyright notice and this permission notice appear in all copies.
*
* THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
* WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR
* ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
* WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN
* ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF
* OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
*/
#include "audio.h"
#include "common.h"
#include <errno.h>
#include <limits.h>
#include <pthread.h>
#include <stdlib.h>
#include <string.h>
#include <jack/jack.h>
#include <jack/ringbuffer.h>
#ifdef CONFIG_SOXR
#include <soxr.h>
#endif
#define NPORTS 2
typedef jack_default_audio_sample_t sample_t;
#define jack_sample_size sizeof(sample_t)
// Two-channel, 32bit audio:
const int bytes_per_frame = NPORTS * jack_sample_size;
pthread_mutex_t buffer_mutex = PTHREAD_MUTEX_INITIALIZER;
pthread_mutex_t client_mutex = PTHREAD_MUTEX_INITIALIZER;
// This also affects deinterlacing.
// So make it exactly the number of incoming audio channels!
jack_port_t *port[NPORTS];
const char *port_name[NPORTS] = {"out_L", "out_R"};
jack_client_t *client;
jack_nframes_t sample_rate;
jack_nframes_t jack_latency;
jack_ringbuffer_t *jackbuf;
int flush_please = 0;
jack_latency_range_t latest_latency_range[NPORTS];
int64_t time_of_latest_transfer;
#ifdef CONFIG_SOXR
typedef struct soxr_quality {
int quality;
const char *name;
} soxr_quality_t;
soxr_quality_t soxr_quality_table[] = {{SOXR_VHQ, "very high"}, {SOXR_HQ, "high"},
{SOXR_MQ, "medium"}, {SOXR_LQ, "low"},
{SOXR_QQ, "quick"}, {-1, NULL}};
static int parse_soxr_quality_name(const char *name) {
for (soxr_quality_t *s = soxr_quality_table; s->name != NULL; ++s) {
if (!strcmp(s->name, name)) {
return s->quality;
}
}
return -1;
}
soxr_t soxr = NULL;
soxr_quality_spec_t quality_spec;
soxr_io_spec_t io_spec;
#endif
static inline sample_t sample_conv(short sample) {
// It sounds correct, but I don't understand it.
// Zero int needs to be zero float. Check.
// Plus 32767 int is 1.0. Check.
// Minus 32767 int is -0.99997. And here my brain shuts down.
// In my head, it should be 1.0, and we should tolerate an overflow
// at minus 32768. But I'm sure there's a textbook explanation somewhere.
return ((sample < 0) ? (-1.0 * sample / SHRT_MIN) : (1.0 * sample / SHRT_MAX));
}
static void deinterleave(const char *interleaved_input_buffer, sample_t *jack_output_buffer[],
jack_nframes_t offset, jack_nframes_t nframes) {
jack_nframes_t f;
// We're dealing with 16bit audio here:
sample_t *ifp = (sample_t *)interleaved_input_buffer;
// Zero-copy, we're working directly on the target and destination buffers,
// so deal with an offset for the second part of the input ringbuffer
for (f = offset; f < (nframes + offset); f++) {
for (int i = 0; i < NPORTS; i++) {
jack_output_buffer[i][f] = *ifp++;
}
}
}
// This is the JACK process callback. We don't decide when it runs.
// It must be hard-realtime safe (i.e. fully deterministic, with constant CPU
// usage. No calls to anything that could ever block: no syscalls, no screen
// output, no file access, no mutexes...
// The JACK ringbuffer we use to get the data in here is explicitly lock-free.
static int process(jack_nframes_t nframes, __attribute__((unused)) void *arg) {
sample_t *buffer[NPORTS];
// Expect an array of two elements because of possible ringbuffer wrap-around:
jack_ringbuffer_data_t v[2] = {0};
jack_nframes_t i, thisbuf;
int frames_written = 0;
int frames_required = 0;
for (i = 0; i < NPORTS; i++) {
buffer[i] = (sample_t *)jack_port_get_buffer(port[i], nframes);
}
if (flush_please) {
// We just move the read pointer ahead without doing anything with the data.
jack_ringbuffer_read_advance(jackbuf, jack_ringbuffer_read_space(jackbuf));
flush_please = 0;
// Since we don't change nframes, the whole buffer will be zeroed later.
} else {
jack_ringbuffer_get_read_vector(jackbuf, v);
for (i = 0; i < 2; i++) {
thisbuf = v[i].len / bytes_per_frame;
if (thisbuf > nframes) {
frames_required = nframes;
} else {
frames_required = thisbuf;
}
deinterleave(v[i].buf, buffer, frames_written, frames_required);
frames_written += frames_required;
nframes -= frames_required;
}
jack_ringbuffer_read_advance(jackbuf, frames_written * bytes_per_frame);
}
// If there are any more frames to put into the buffer, fill them with
// silence. This is a critical underflow situation. Let's at least keep the JACK
// graph humming along while preventing the motorboat sound of a repeating buffer.
while (nframes > 0) {
for (i = 0; i < NPORTS; i++) {
buffer[i][frames_written] = 0.0;
}
frames_written++;
nframes--;
}
return 0; // Tell JACK that all is well.
}
// This is the JACK graph reorder callback. Now we know some JACK connections
// have changed, so we recompute the latency.
static int graph(__attribute__((unused)) void *arg) {
int latency = 0;
debug(2, "JACK graph reorder callback called.");
for (int i = 0; i < NPORTS; i++) {
jack_port_get_latency_range(port[i], JackPlaybackLatency, &latest_latency_range[i]);
debug(2, "JACK latency for port %s\tmin: %d\t max: %d", port_name[i],
latest_latency_range[i].min, latest_latency_range[i].max);
latency += latest_latency_range[i].max;
}
latency /= NPORTS;
jack_latency = latency;
debug(1, "Average maximum JACK latency across all ports: %d", jack_latency);
return 0;
}
// This the function JACK will call in case of an error in the library.
static void error(const char *desc) { warn("JACK error: \"%s\"", desc); }
// This is the function JACK will call in case of a non-critical event in the library.
static void info(const char *desc) { inform("JACK information: \"%s\"", desc); }
static int jack_init(__attribute__((unused)) int argc, __attribute__((unused)) char **argv) {
int i;
int bufsz = -1;
config.audio_backend_latency_offset = 0;
config.audio_backend_buffer_desired_length = 0.500;
// Below this, soxr interpolation will not occur -- it'll be basic interpolation
// instead.
config.audio_backend_buffer_interpolation_threshold_in_seconds = 0.25;
// Do the "general" audio options. Note, these options are in the "general" stanza!
parse_general_audio_options();
#ifdef CONFIG_SOXR
config.jack_soxr_resample_quality = -1; // don't resample by default
#endif
// Now the options specific to the backend, from the "jack" stanza:
if (config.cfg != NULL) {
const char *str;
if (config_lookup_string(config.cfg, "jack.client_name", &str)) {
config.jack_client_name = (char *)str;
}
if (config_lookup_string(config.cfg, "jack.autoconnect_pattern", &str)) {
config.jack_autoconnect_pattern = (char *)str;
}
#ifdef CONFIG_SOXR
if (config_lookup_string(config.cfg, "jack.soxr_resample_quality", &str)) {
debug(1, "SOXR quality %s", str);
config.jack_soxr_resample_quality = parse_soxr_quality_name(str);
}
#endif
if (config_lookup_int(config.cfg, "jack.bufsz", &bufsz) && bufsz <= 0)
die("jack: bufsz must be > 0");
}
if (config.jack_client_name == NULL)
config.jack_client_name = strdup("shairport-sync");
// by default a buffer that can hold up to 4 seconds of 48kHz samples
if (bufsz <= 0)
bufsz = 48000 * 4 * bytes_per_frame;
jackbuf = jack_ringbuffer_create((size_t)bufsz);
if (jackbuf == NULL)
die("Can't allocate %d bytes for the JACK ringbuffer.", bufsz);
// Lock the ringbuffer into memory so that it never gets paged out, which would
// break realtime constraints.
jack_ringbuffer_mlock(jackbuf);
// This mutex should not be necessary, but removing it causes segfaults on
// shutdown. Apparently, there are multiple threads in the main program trying
// to do stuff. FIXME: Try to consolidate into one thread and get rid of this lock.
pthread_mutex_lock(&client_mutex);
jack_status_t status;
client = jack_client_open(config.jack_client_name, JackNoStartServer, &status);
if (!client) {
die("Could not start JACK server. JackStatus is %x", status);
}
sample_rate = jack_get_sample_rate(client);
#ifdef CONFIG_SOXR
if (config.jack_soxr_resample_quality >= SOXR_QQ) {
quality_spec = soxr_quality_spec(config.jack_soxr_resample_quality, 0);
io_spec = soxr_io_spec(SOXR_INT16_I, SOXR_FLOAT32_I);
} else
#endif
if (sample_rate != 44100) {
die("The JACK server is running at the wrong sample rate (%d) for Shairport Sync."
" Must be 44100 Hz.",
sample_rate);
}
jack_set_process_callback(client, &process, NULL);
jack_set_graph_order_callback(client, &graph, NULL);
jack_set_error_function(&error);
jack_set_info_function(&info);
for (i = 0; i < NPORTS; i++) {
port[i] =
jack_port_register(client, port_name[i], JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0);
}
if (jack_activate(client)) {
die("Could not activate %s JACK client.", config.jack_client_name);
} else {
debug(2, "JACK client %s activated successfully.", config.jack_client_name);
}
if (config.jack_autoconnect_pattern != NULL) {
inform("config.jack_autoconnect_pattern is %s. If you see the program die after this,"
"you made a syntax error.",
config.jack_autoconnect_pattern);
// Sadly, this will throw a segfault if the user provides a syntactically incorrect regex.
// I've reported it to the jack-devel mailing list, they're in a better place to fix it.
const char **port_list = jack_get_ports(client, config.jack_autoconnect_pattern,
JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput);
if (port_list != NULL) {
for (i = 0; i < NPORTS; i++) {
char *full_port_name[NPORTS];
full_port_name[i] = malloc(sizeof(char) * jack_port_name_size());
sprintf(full_port_name[i], "%s:%s", config.jack_client_name, port_name[i]);
if (port_list[i] != NULL) {
int err;
debug(2, "Connecting %s to %s.", full_port_name[i], port_list[i]);
err = jack_connect(client, full_port_name[i], port_list[i]);
switch (err) {
case EEXIST:
inform("The requested connection from %s to %s already exists.", full_port_name[i],
port_list[i]);
break;
case 0:
// success
break;
default:
warn("JACK error no. %d occurred while trying to connect %s to %s.", err,
full_port_name[i], port_list[i]);
break;
}
} else {
inform("No matching port found in %s to connect %s to. You may not hear audio.",
config.jack_autoconnect_pattern, full_port_name[i]);
}
free(full_port_name[i]);
}
while (port_list[i++] != NULL) {
inform(
"Additional matching port %s found. Check that the connections are what you intended.",
port_list[i - 1]);
}
jack_free(port_list);
}
}
pthread_mutex_unlock(&client_mutex);
return 0;
}
static void jack_deinit() {
pthread_mutex_lock(&client_mutex);
if (jack_deactivate(client))
warn("Error deactivating jack client");
if (jack_client_close(client))
warn("Error closing jack client");
pthread_mutex_unlock(&client_mutex);
jack_ringbuffer_free(jackbuf);
#ifdef CONFIG_SOXR
if (soxr) {
soxr_delete(soxr);
soxr = NULL;
}
#endif
}
static void jack_start(int i_sample_rate, __attribute__((unused)) int i_sample_format) {
// Nothing to do, JACK client has already been set up at jack_init().
// Also, we have no say over the sample rate or sample format of JACK,
// We convert the 16bit samples to float, and die if the sample rate is != 44k1 without soxr.
#ifdef CONFIG_SOXR
if (config.jack_soxr_resample_quality >= SOXR_QQ) {
// we might improve a bit with soxr_clear if the sample_rate doesn't change
if (soxr) {
soxr_delete(soxr);
}
soxr_error_t e = NULL;
soxr = soxr_create(i_sample_rate, sample_rate, NPORTS, &e, &io_spec, &quality_spec, NULL);
if (!soxr) {
die("Unable to create soxr resampler for JACK: %s", e);
}
}
#endif
}
static void jack_flush() {
debug(2, "Only the consumer can safely flush a lock-free ringbuffer. Asking the"
" process callback to do it...");
flush_please = 1;
}
static int jack_delay(long *the_delay) {
// Semantics change: we now look at the last transfer into the lock-free
// ringbuffer, not into the jack buffers directly (because locking those would
// violate real-time constraints). On average, that should lead to just a
// constant additional latency.
// Without the mutex, we could get the time of what is the last transfer of data
// to a jack buffer, but then a transfer could occur and we would get the buffer
// occupancy after another transfer had occurred, so we could "lose" a full transfer
// (e.g. 1024 frames @ 44,100 fps ~ 23.2 milliseconds)
pthread_mutex_lock(&buffer_mutex);
int64_t time_now = get_absolute_time_in_ns();
int64_t delta = time_now - time_of_latest_transfer; // nanoseconds
size_t audio_occupancy_now = jack_ringbuffer_read_space(jackbuf) / bytes_per_frame;
debug(2, "audio_occupancy_now is %d.", audio_occupancy_now);
pthread_mutex_unlock(&buffer_mutex);
int64_t frames_processed_since_latest_latency_check = (delta * sample_rate) / 1000000000;
// debug(1,"delta: %" PRId64 " frames.",frames_processed_since_latest_latency_check);
// jack_latency is set by the graph() callback, it's the average of the maximum
// latencies of all our output ports. Adjust this constant baseline delay according
// to the buffer fill level:
*the_delay = jack_latency + audio_occupancy_now - frames_processed_since_latest_latency_check;
// debug(1,"reporting a delay of %d frames",*the_delay);
return 0;
}
static int play(void *buf, int samples, __attribute__((unused)) int sample_type,
__attribute__((unused)) uint32_t timestamp,
__attribute__((unused)) uint64_t playtime) {
jack_ringbuffer_data_t v[2] = {0};
size_t i, j, c;
jack_nframes_t thisbuf;
// It's ok to lock here since we're not in the realtime callback:
pthread_mutex_lock(&buffer_mutex);
jack_ringbuffer_get_write_vector(jackbuf, v);
short *in = (short *)buf;
sample_t *out;
for (i = 0; i < 2; ++i) {
thisbuf = v[i].len / (jack_sample_size * NPORTS); // #samples per channel
out = (sample_t *)v[i].buf;
#ifdef CONFIG_SOXR
if (soxr) {
size_t i_done, o_done;
soxr_error_t e;
while (samples > 0 && thisbuf > 0) {
e = soxr_process(soxr, (soxr_in_t)in, samples, &i_done, (soxr_out_t)out, thisbuf, &o_done);
if (e)
die("Error during soxr process: %s", e);
in += i_done * NPORTS; // advance our input buffer
samples -= i_done;
thisbuf -= o_done;
jack_ringbuffer_write_advance(jackbuf, o_done * jack_sample_size * NPORTS);
}
} else {
#endif
j = 0;
for (j = 0; j < thisbuf && samples > 0; ++j) {
for (c = 0; c < NPORTS; ++c)
out[j * NPORTS + c] = sample_conv(*in++);
--samples;
}
jack_ringbuffer_write_advance(jackbuf, j * jack_sample_size * NPORTS);
#ifdef CONFIG_SOXR
}
#endif
}
time_of_latest_transfer = get_absolute_time_in_ns();
pthread_mutex_unlock(&buffer_mutex);
if (samples) {
warn("JACK ringbuffer overrun. Dropped %d samples.", samples);
}
return 0;
}
audio_output audio_jack = {.name = "jack",
.help = NULL,
.init = &jack_init,
.deinit = &jack_deinit,
.prepare = NULL,
.start = &jack_start,
.stop = NULL,
.is_running = NULL,
.flush = &jack_flush,
.delay = &jack_delay,
.stats = NULL,
.play = &play,
.volume = NULL,
.parameters = NULL,
.mute = NULL};
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