mpv/libao2/ao_jack.c

369 lines
10 KiB
C

/*
* 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 (atmosfear@users.sf.net)
* and Reimar Döffinger (Reimar.Doeffinger@stud.uni-karlsruhe.de)
*/
#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 <Reimar.Doeffinger@stud.uni-karlsruhe.de>",
"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() {
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() {
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.
*/
static int read_buffer(float **bufs, int cnt, int num_bufs) {
int first_len = BUFFSIZE - read_pos;
int buffered = buf_used();
int i, j;
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;
}
}
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)
if (read_buffer(bufs, nframes, num_ports) < nframes)
underrun = 1;
if (paused || underrun)
silence(bufs, nframes, num_ports);
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 ()
{
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 = (char *)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() {
paused = 1;
read_pos = 0;
write_pos = 0;
paused = 0;
}
/**
* \brief stop playing, keep buffers (for pause)
*/
static void audio_pause() {
paused = 1;
}
/**
* \brief resume playing, after audio_pause()
*/
static void audio_resume() {
paused = 0;
}
static int get_space() {
return buf_free();
}
/**
* \brief write data into buffer and reset underrun flag
*/
static int play(void *data, int len, int flags) {
len -= len % ao_data.outburst;
underrun = 0;
return write_buffer(data, len);
}
static float get_delay() {
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;
}