view libao2/ao_jack.c @ 22616:09dc129234a0

Matroska seeking fixes If a relative seek forward went past the last index position the Matroska demuxer did not seek to any index position. It did however set the mkv_d->skip_to_timecode variable which meant that the next fill_buffer() call would read from the current position until the target position (probably the end of the file). Fix this by changing the code to seek to the last index position if that is between the current and target positions. Also change backwards relative seek to accept an exactly matching index position (<= vs <) and reorganize the seeking conditionals to allow making the above change without turning the code into a complete mess.

/* 
 * ao_jack.c - libao2 JACK Audio Output Driver for MPlayer
 *
 * This driver is under the same license as MPlayer.
 * (http://www.mplayerhq.hu)
 *
 * Copyleft 2001 by Felix Bünemann ([e-mail])
 * and Reimar Döffinger ([e-mail])
 */

#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <unistd.h>

#include "config.h"
#include "mp_msg.h"
#include "help_mp.h"

#include "audio_out.h"
#include "audio_out_internal.h"
#include "libaf/af_format.h"
#include "osdep/timer.h"
#include "subopt-helper.h"

#include "libvo/fastmemcpy.h"

#include <jack/jack.h>

static ao_info_t info = 
{
  "JACK audio output",
  "jack",
  "Reimar Döffinger <[e-mail]>",
  "based on ao_sdl.c"
};

LIBAO_EXTERN(jack)

//! maximum number of channels supported, avoids lots of mallocs
#define MAX_CHANS 6
static jack_port_t *ports[MAX_CHANS];
static int num_ports; ///< Number of used ports == number of channels
static jack_client_t *client;
static float jack_latency;
static int estimate;
static volatile int paused = 0; ///< set if paused
static volatile int underrun = 0; ///< signals if an underrun occured

static volatile float callback_interval = 0;
static volatile float callback_time = 0;

//! size of one chunk, if this is too small MPlayer will start to "stutter"
//! after a short time of playback
#define CHUNK_SIZE (16 * 1024)
//! number of "virtual" chunks the buffer consists of
#define NUM_CHUNKS 8
// This type of ring buffer may never fill up completely, at least
// one byte must always be unused.
// For performance reasons (alignment etc.) one whole chunk always stays
// empty, not only one byte.
#define BUFFSIZE ((NUM_CHUNKS + 1) * CHUNK_SIZE)

//! buffer for audio data
static unsigned char *buffer = NULL;

//! buffer read position, may only be modified by playback thread or while it is stopped
static volatile int read_pos;
//! buffer write position, may only be modified by MPlayer's thread
static volatile int write_pos;

/**
 * \brief get the number of free bytes in the buffer
 * \return number of free bytes in buffer
 * 
 * may only be called by MPlayer's thread
 * return value may change between immediately following two calls,
 * and the real number of free bytes might be larger!
 */
static int buf_free(void) {
  int free = read_pos - write_pos - CHUNK_SIZE;
  if (free < 0) free += BUFFSIZE;
  return free;
}

/**
 * \brief get amount of data available in the buffer
 * \return number of bytes available in buffer
 *
 * may only be called by the playback thread
 * return value may change between immediately following two calls,
 * and the real number of buffered bytes might be larger!
 */
static int buf_used(void) {
  int used = write_pos - read_pos;
  if (used < 0) used += BUFFSIZE;
  return used;
}

/**
 * \brief insert len bytes into buffer
 * \param data data to insert
 * \param len length of data
 * \return number of bytes inserted into buffer
 *
 * If there is not enough room, the buffer is filled up
 */
static int write_buffer(unsigned char* data, int len) {
  int first_len = BUFFSIZE - write_pos;
  int free = buf_free();
  if (len > free) len = free;
  if (first_len > len) first_len = len;
  // till end of buffer
  memcpy (&buffer[write_pos], data, first_len);
  if (len > first_len) { // we have to wrap around
    // remaining part from beginning of buffer
    memcpy (buffer, &data[first_len], len - first_len);
  }
  write_pos = (write_pos + len) % BUFFSIZE;
  return len;
}

/**
 * \brief read data from buffer and splitting it into channels
 * \param bufs num_bufs float buffers, each will contain the data of one channel
 * \param cnt number of samples to read per channel
 * \param num_bufs number of channels to split the data into
 * \return number of samples read per channel, equals cnt unless there was too
 *         little data in the buffer
 *
 * Assumes the data in the buffer is of type float, the number of bytes
 * read is res * num_bufs * sizeof(float), where res is the return value.
 * If there is not enough data in the buffer remaining parts will be filled
 * with silence.
 */
static int read_buffer(float **bufs, int cnt, int num_bufs) {
  int buffered = buf_used();
  int i, j;
  int orig_cnt = cnt;
  if (cnt * sizeof(float) * num_bufs > buffered)
    cnt = buffered / sizeof(float) / num_bufs;
  for (i = 0; i < cnt; i++) {
    for (j = 0; j < num_bufs; j++) {
      bufs[j][i] = *((float *)(&buffer[read_pos]));
      read_pos = (read_pos + sizeof(float)) % BUFFSIZE;
    }
  }
  for (i = cnt; i < orig_cnt; i++)
    for (j = 0; j < num_bufs; j++)
      bufs[j][i] = 0;
  return cnt;
}

// end ring buffer stuff

static int control(int cmd, void *arg) {
  return CONTROL_UNKNOWN;
}

/**
 * \brief fill the buffers with silence
 * \param bufs num_bufs float buffers, each will contain the data of one channel
 * \param cnt number of samples in each buffer
 * \param num_bufs number of buffers
 */
static void silence(float **bufs, int cnt, int num_bufs) {
  int i, j;
  for (i = 0; i < cnt; i++)
    for (j = 0; j < num_bufs; j++)
      bufs[j][i] = 0;
}

/**
 * \brief JACK Callback function
 * \param nframes number of frames to fill into buffers
 * \param arg unused
 * \return currently always 0
 *
 * Write silence into buffers if paused or an underrun occured
 */
static int outputaudio(jack_nframes_t nframes, void *arg) {
  float *bufs[MAX_CHANS];
  int i;
  for (i = 0; i < num_ports; i++)
    bufs[i] = jack_port_get_buffer(ports[i], nframes);
  if (paused || underrun)
    silence(bufs, nframes, num_ports);
  else
    if (read_buffer(bufs, nframes, num_ports) < nframes)
      underrun = 1;
  if (estimate) {
    float now = (float)GetTimer() / 1000000.0;
    float diff = callback_time + callback_interval - now;
    if ((diff > -0.002) && (diff < 0.002))
      callback_time += callback_interval;
    else
      callback_time = now;
    callback_interval = (float)nframes / (float)ao_data.samplerate;
  }
  return 0;
}

/**
 * \brief print suboption usage help
 */
static void print_help (void)
{
  mp_msg (MSGT_AO, MSGL_FATAL,
           "\n-ao jack commandline help:\n"
           "Example: mplayer -ao jack:port=myout\n"
           "  connects MPlayer to the jack ports named myout\n"
           "\nOptions:\n"
           "  port=<port name>\n"
           "    Connects to the given ports instead of the default physical ones\n"
           "  name=<client name>\n"
           "    Client name to pass to JACK\n"
           "  estimate\n"
           "    Estimates the amount of data in buffers (experimental)\n");
}

static int init(int rate, int channels, int format, int flags) {
  const char **matching_ports = NULL;
  char *port_name = NULL;
  char *client_name = NULL;
  opt_t subopts[] = {
    {"port", OPT_ARG_MSTRZ, &port_name, NULL},
    {"name", OPT_ARG_MSTRZ, &client_name, NULL},
    {"estimate", OPT_ARG_BOOL, &estimate, NULL},
    {NULL}
  };
  int port_flags = JackPortIsInput;
  int i;
  estimate = 1;
  if (subopt_parse(ao_subdevice, subopts) != 0) {
    print_help();
    return 0;
  }
  if (channels > MAX_CHANS) {
    mp_msg(MSGT_AO, MSGL_FATAL, "[JACK] Invalid number of channels: %i\n", channels);
    goto err_out;
  }
  if (!client_name) {
    client_name = malloc(40);
  sprintf(client_name, "MPlayer [%d]", getpid());
  }
  client = jack_client_new(client_name);
  if (!client) {
    mp_msg(MSGT_AO, MSGL_FATAL, "[JACK] cannot open server\n");
    goto err_out;
  }
  reset();
  jack_set_process_callback(client, outputaudio, 0);

  // list matching ports
  if (!port_name)
    port_flags |= JackPortIsPhysical;
  matching_ports = jack_get_ports(client, port_name, NULL, port_flags);
  for (num_ports = 0; matching_ports && matching_ports[num_ports]; num_ports++) ;
  if (!num_ports) {
    mp_msg(MSGT_AO, MSGL_FATAL, "[JACK] no physical ports available\n");
    goto err_out;
  }
  if (channels > num_ports) channels = num_ports;
  num_ports = channels;

  // create out output ports
  for (i = 0; i < num_ports; i++) {
    char pname[30];
    snprintf(pname, 30, "out_%d", i);
    ports[i] = jack_port_register(client, pname, JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0);
    if (!ports[i]) {
      mp_msg(MSGT_AO, MSGL_FATAL, "[JACK] not enough ports available\n");
      goto err_out;
    }
  }
  if (jack_activate(client)) {
    mp_msg(MSGT_AO, MSGL_FATAL, "[JACK] activate failed\n");
    goto err_out;
  }
  for (i = 0; i < num_ports; i++) {
    if (jack_connect(client, jack_port_name(ports[i]), matching_ports[i])) {
      mp_msg(MSGT_AO, MSGL_FATAL, "[JACK] connecting failed\n");
      goto err_out;
    }
  }
  rate = jack_get_sample_rate(client);
  jack_latency = (float)(jack_port_get_total_latency(client, ports[0]) +
                         jack_get_buffer_size(client)) / (float)rate;
  callback_interval = 0;
  buffer = (unsigned char *) malloc(BUFFSIZE);

  ao_data.channels = channels;
  ao_data.samplerate = rate;
  ao_data.format = AF_FORMAT_FLOAT_NE;
  ao_data.bps = channels * rate * sizeof(float);
  ao_data.buffersize = CHUNK_SIZE * NUM_CHUNKS;
  ao_data.outburst = CHUNK_SIZE;
  free(matching_ports);
  free(port_name);
  free(client_name);
  return 1;

err_out:
  free(matching_ports);
  free(port_name);
  free(client_name);
  if (client)
    jack_client_close(client);
  free(buffer);
  buffer = NULL;
  return 0;
}

// close audio device
static void uninit(int immed) {
  if (!immed)
    usec_sleep(get_delay() * 1000 * 1000);
  // HACK, make sure jack doesn't loop-output dirty buffers
  reset();
  usec_sleep(100 * 1000);
  jack_client_close(client);
  free(buffer);
  buffer = NULL;
}

/**
 * \brief stop playing and empty buffers (for seeking/pause)
 */
static void reset(void) {
  paused = 1;
  read_pos = 0;
  write_pos = 0;
  paused = 0;
}

/**
 * \brief stop playing, keep buffers (for pause)
 */
static void audio_pause(void) {
  paused = 1;
}

/**
 * \brief resume playing, after audio_pause()
 */
static void audio_resume(void) {
  paused = 0;
}

static int get_space(void) {
  return buf_free();
}

/**
 * \brief write data into buffer and reset underrun flag
 */
static int play(void *data, int len, int flags) {
  if (!(flags & AOPLAY_FINAL_CHUNK))
  len -= len % ao_data.outburst;
  underrun = 0;
  return write_buffer(data, len);
}

static float get_delay(void) {
  int buffered = BUFFSIZE - CHUNK_SIZE - buf_free(); // could be less
  float in_jack = jack_latency;
  if (estimate && callback_interval > 0) {
    float elapsed = (float)GetTimer() / 1000000.0 - callback_time;
    in_jack += callback_interval - elapsed;
    if (in_jack < 0) in_jack = 0;
  }
  return (float)buffered / (float)ao_data.bps + in_jack;
}