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mirror of https://github.com/mpv-player/mpv synced 2024-10-30 04:46:41 +01:00
mpv/libao2/ao_jack.c
tack 24c36f29ec ao_jack: increase maximum allowed channels from 6 to 8.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@31358 b3059339-0415-0410-9bf9-f77b7e298cf2
2010-11-02 04:07:42 +02:00

362 lines
10 KiB
C

/*
* JACK audio output driver for MPlayer
*
* Copyleft 2001 by Felix Bünemann (atmosfear@users.sf.net)
* and Reimar Döffinger (Reimar.Doeffinger@stud.uni-karlsruhe.de)
*
* This file is part of MPlayer.
*
* MPlayer is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* MPlayer 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 General Public License for more details.
*
* You should have received a copy of the GNU General Public License along
* along with MPlayer; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <unistd.h>
#include "config.h"
#include "mp_msg.h"
#include "audio_out.h"
#include "audio_out_internal.h"
#include "libaf/af_format.h"
#include "osdep/timer.h"
#include "subopt-helper.h"
#include "libavutil/fifo.h"
#include <jack/jack.h>
static const 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 8
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
#define BUFFSIZE (NUM_CHUNKS * CHUNK_SIZE)
//! buffer for audio data
static AVFifoBuffer *buffer;
/**
* \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 free = av_fifo_space(buffer);
if (len > free) len = free;
return av_fifo_generic_write(buffer, data, len, NULL);
}
static void silence(float **bufs, int cnt, int num_bufs);
struct deinterleave {
float **bufs;
int num_bufs;
int cur_buf;
int pos;
};
static void deinterleave(void *info, void *src, int len) {
struct deinterleave *di = info;
float *s = src;
int i;
len /= sizeof(float);
for (i = 0; i < len; i++) {
di->bufs[di->cur_buf++][di->pos] = s[i];
if (di->cur_buf >= di->num_bufs) {
di->cur_buf = 0;
di->pos++;
}
}
}
/**
* \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) {
struct deinterleave di = {bufs, num_bufs, 0, 0};
int buffered = av_fifo_size(buffer);
if (cnt * sizeof(float) * num_bufs > buffered) {
silence(bufs, cnt, num_bufs);
cnt = buffered / sizeof(float) / num_bufs;
}
av_fifo_generic_read(buffer, &di, cnt * num_bufs * sizeof(float), deinterleave);
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;
for (i = 0; i < num_bufs; i++)
memset(bufs[i], 0, cnt * sizeof(float));
}
/**
* \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"
" autostart\n"
" Automatically start JACK server if necessary\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;
int autostart = 0;
const opt_t subopts[] = {
{"port", OPT_ARG_MSTRZ, &port_name, NULL},
{"name", OPT_ARG_MSTRZ, &client_name, NULL},
{"estimate", OPT_ARG_BOOL, &estimate, NULL},
{"autostart", OPT_ARG_BOOL, &autostart, NULL},
{NULL}
};
jack_options_t open_options = JackUseExactName;
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());
}
if (!autostart)
open_options |= JackNoStartServer;
client = jack_client_open(client_name, open_options, NULL);
if (!client) {
mp_msg(MSGT_AO, MSGL_FATAL, "[JACK] cannot open server\n");
goto err_out;
}
buffer = av_fifo_alloc(BUFFSIZE);
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);
if (!matching_ports || !matching_ports[0]) {
mp_msg(MSGT_AO, MSGL_FATAL, "[JACK] no physical ports available\n");
goto err_out;
}
i = 1;
while (matching_ports[i]) i++;
if (channels > i) channels = i;
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;
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);
av_fifo_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);
av_fifo_free(buffer);
buffer = NULL;
}
/**
* \brief stop playing and empty buffers (for seeking/pause)
*/
static void reset(void) {
paused = 1;
av_fifo_reset(buffer);
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 av_fifo_space(buffer);
}
/**
* \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 = av_fifo_size(buffer); // 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;
}