audio_jack.c

/*
 * jack output driver. This file is part of Shairport Sync.
 * Copyright (c) 2018 Mike Brady <mikebrady@iercom.net>
 *
 * 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 <getopt.h>
#include <limits.h>
#include <math.h>
#include <pthread.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <unistd.h>

#include <jack/jack.h>
#include <jack/transport.h>
#include <jack/ringbuffer.h>

// Two-channel, 16bit audio:
static const int bytes_per_frame = 4;
// Four seconds buffer -- should be plenty
#define buffer_size (44100 * 4 * bytes_per_frame)

static pthread_mutex_t buffer_mutex = PTHREAD_MUTEX_INITIALIZER;
static pthread_mutex_t client_mutex = PTHREAD_MUTEX_INITIALIZER;

pthread_t *open_client_if_necessary_thread = NULL;

int jack_init(int, char **);
void jack_deinit(void);
void jack_start(int, int);
int play(void *, int);
void jack_stop(void);
int jack_is_running(void);
int jack_delay(long *);
void jack_flush(void);

audio_output audio_jack = {.name = "jack",
                           .help = NULL,
                           .init = &jack_init,
                           .deinit = &jack_deinit,
                           .start = &jack_start,
                           .stop = &jack_stop,
                           .is_running = &jack_is_running,
                           .flush = &jack_flush,
                           .delay = &jack_delay,
                           .play = &play,
                           .volume = NULL,
                           .parameters = NULL,
                           .mute = NULL};

jack_port_t *left_port;
jack_port_t *right_port;

int client_is_open;
jack_client_t *client;
jack_nframes_t sample_rate;
jack_nframes_t jack_latency;

static jack_ringbuffer_t *jackbuf;
static int flush_please = 0;

jack_latency_range_t latest_left_latency_range, latest_right_latency_range;
int64_t time_of_latest_transfer;


static inline jack_default_audio_sample_t sample_conv(short sample) {
  return ((sample < 0) ? (-1.0 * sample / SHRT_MIN) : (1.0 * sample / SHRT_MAX));
}

static void deinterleave_and_convert(const char *interleaved_frames,
                                     jack_default_audio_sample_t * const jack_buffer_left,
                                     jack_default_audio_sample_t * const jack_buffer_right,
                                     jack_nframes_t nframes) {
  jack_nframes_t i;
  short *ifp = (short *)interleaved_frames; // we're dealing with 16bit audio here
  jack_default_audio_sample_t *fpl = jack_buffer_left;
  jack_default_audio_sample_t *fpr = jack_buffer_right;
  for (i=0; i<nframes; i++) {
    fpl[i] = sample_conv(*ifp++);
    fpr[i] = sample_conv(*ifp++);
  }
}

static int jack_stream_write_cb(jack_nframes_t nframes, __attribute__((unused)) void *arg) {

  jack_default_audio_sample_t *left_buffer =
      (jack_default_audio_sample_t *)jack_port_get_buffer(left_port, nframes);
  jack_default_audio_sample_t *right_buffer =
      (jack_default_audio_sample_t *)jack_port_get_buffer(right_port, nframes);

  jack_ringbuffer_data_t v[2] = { 0 };
  jack_nframes_t i, thisbuf;
  int frames_written = 0;
  int frames_required = 0;

  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); // an array of two elements because of possible ringbuffer wrap-around
    for (i=0; i<2; i++) {
      thisbuf = v[i].len / bytes_per_frame;
      if (thisbuf > nframes) {
        frames_required = nframes;
	    } else {
        frames_required = thisbuf;
      }
      deinterleave_and_convert(v[i].buf, &left_buffer[frames_written], &right_buffer[frames_written], frames_required);
      frames_written += frames_required;
      nframes -= frames_required;
    }
    jack_ringbuffer_read_advance(jackbuf, frames_written * bytes_per_frame);
  }
  // now, if there are any more frames to put into the buffer, fill them with
  // silence
  while (nframes > 0) {
    left_buffer[frames_written] = 0.0;
    right_buffer[frames_written] = 0.0;
    frames_written++;
    nframes--;
  }
  return 0;
}

// FIXME: set_graph_order_callback(), recompute latencies here!

static void default_jack_error_callback(const char *desc) { debug(2, "jackd error: \"%s\"", desc); }

static void default_jack_info_callback(const char *desc) { inform("jackd information: \"%s\"", desc); }

static int jack_client_open_if_needed(void) {
  pthread_mutex_lock(&client_mutex);
  if (client_is_open == 0) {
    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);
    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, jack_stream_write_cb, 0);
    left_port = jack_port_register(client, "out_L", JACK_DEFAULT_AUDIO_TYPE,
                                   JackPortIsOutput, 0);
    right_port = jack_port_register(client, "out_R",
                                    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 sucessfully.", config.jack_client_name);
      client_is_open = 1;
    }
  }
  pthread_mutex_unlock(&client_mutex);
  return client_is_open;
}

static void jack_close(void) {
  pthread_mutex_lock(&client_mutex);
  if (client_is_open) {
    if (jack_deactivate(client))
      debug(1, "Error deactivating jack client");
    if (jack_client_close(client))
      debug(1, "Error closing jack client");
    client_is_open = 0;
  }
  pthread_mutex_unlock(&client_mutex);
}

int jack_init(__attribute__((unused)) int argc, __attribute__((unused)) char **argv) {
  config.audio_backend_latency_offset = 0;
  config.audio_backend_buffer_desired_length = 0.500;
  config.audio_backend_buffer_interpolation_threshold_in_seconds =
      0.25; // below this, soxr interpolation will not occur -- it'll be basic interpolation
            // instead.

  // get settings from settings file first, allow them to be overridden by
  // command line options

  // do the "general" audio  options. Note, these options are in the "general" stanza!
  parse_general_audio_options();

  // other options would be picked up here...

  // now the specific options
  if (config.cfg != NULL) {
    const char *str;
    /* Get the Client Name. */
    if (config_lookup_string(config.cfg, "jack.client_name", &str)) {
      config.jack_client_name = (char *)str;
    }
  }

  if (config.jack_client_name == NULL)
    config.jack_client_name = strdup("Shairport Sync");

  jackbuf = jack_ringbuffer_create(buffer_size);
  if (jackbuf == NULL)
    die("Can't allocate %d bytes for the JACK ringbuffer.", buffer_size);
  jack_ringbuffer_mlock(jackbuf);		 // lock buffer into memory so that it never gets paged out

  jack_set_error_function(default_jack_error_callback);
  jack_set_info_function(default_jack_info_callback);

  client_is_open = 0;

  jack_client_open_if_needed();

  return 0;
}

void jack_deinit() {
  jack_close();
  jack_ringbuffer_free(jackbuf);
}

void jack_start(__attribute__((unused)) int i_sample_rate,
                __attribute__((unused)) int i_sample_format) {
  // debug(1, "jack start");
  // see if the client is running. If not, try to open and initialise it

  if (jack_client_open_if_needed() == 0)
    debug(1, "cannot open a jack client for a play session");
}

void jack_stop(void) {
  // debug(1, "jack stop");
}

int jack_is_running() {
  int reply = -1; // meaning jack is not running
  if (client_is_open) {

    // check if the ports have a zero latency -- if they both have, then it's disconnected.

    // FIXME: this causes a segfault when shairport-sync is exited with CTRL-C, because
    // the client_is_open flag is stale by then. Also, this test is not necessary.
    // shairport-sync should not worry what's reading its ports. As long as jack is alive,
    // deliver audio, even if nothing is connected. This behaviour probably stems from 
    // the wish to not hog an audio device if not needed, which is no longer an issue with
    // jack. Moreover, don't "conserve" CPU this way, because in a realtime system you want
    // deterministic  CPU load more than anything else.
    //    jack_latency_range_t left_latency_range, right_latency_range;
    //	  jack_port_get_latency_range(left_port, JackPlaybackLatency, &left_latency_range);
    //    jack_port_get_latency_range(right_port, JackPlaybackLatency, &right_latency_range);

    //    if ((left_latency_range.min == 0) && (left_latency_range.max == 0) &&
    //        (right_latency_range.min == 0) && (right_latency_range.max == 0)) {
    //      reply = -2; // meaning Shairport Sync is not connected
    //    } else {

    // FIXME: For now, we assume JACK is always running, as it should.
    // Still need to understand why shairport-sync needs this function.
    reply = 0; // meaning jack is open and Shairport Sync is connected to it
               //    }
  }
  return reply;
}

void jack_flush() {
  // debug(1, "Only the consumer can safely flush a lock-free ringbuffer. Asking the process callback to do it...");
  flush_please = 1;
}

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. the old comment still applies:

  // 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_fp();
    // this is the time back to the last time data
    // was transferred into a jack buffer
    int64_t delta = time_now - time_of_latest_transfer;
    // this is the buffer occupancy before any
    // subsequent transfer because transfer is blocked
    // by the mutex
    size_t audio_occupancy_now = jack_ringbuffer_read_space(jackbuf) / bytes_per_frame;
    // debug(1, "audio_occupancy_now is %d.", audio_occupancy_now);
  pthread_mutex_unlock(&buffer_mutex);

  int64_t frames_processed_since_latest_latency_check = (delta * 44100) >> 32;

  // FIXME: this should only be done if there was an actual change, i.e. on the jack graph reorder
  // callback, to update a static variable which can be checked here. For now, use a fixed arbitrary value:
  jack_nframes_t base_latency = 0;

  // debug(1,"delta: %" PRId64 " frames.",frames_processed_since_latest_latency_check);

  // jack_nframes_t base_latency = (latest_left_latency_range.min + latest_left_latency_range.max) / 2;
  // if (base_latency == 0)
  //   base_latency = (latest_right_latency_range.min + latest_right_latency_range.max) / 2;
  *the_delay = base_latency + audio_occupancy_now - frames_processed_since_latest_latency_check;
  // debug(1,"reporting a delay of %d frames",*the_delay);
  return 0;
}

int play(void *buf, int samples) {
  // debug(1,"jack_play of %d samples.",samples);
  // copy the samples into the queue
  size_t bytes_to_transfer, bytes_transferred;
  bytes_to_transfer = samples * bytes_per_frame;
  pthread_mutex_lock(&buffer_mutex); // it's ok to lock here since we're not in the realtime callback
    bytes_transferred = jack_ringbuffer_write(jackbuf, buf, bytes_to_transfer);
    // semantics change: we now measure the last time audio was moved into the ringbuffer, not the jack output buffers.
    time_of_latest_transfer = get_absolute_time_in_fp();
  pthread_mutex_unlock(&buffer_mutex);
  if (bytes_transferred < bytes_to_transfer) {
    debug(1, "JACK ringbuffer overrun. Only wrote %d of %d bytes.", bytes_transferred, bytes_to_transfer);
  }
  return 0;
}