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mpv/audio/out/ao_jack.c

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/*
* 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 "core/mp_msg.h"
#include "ao.h"
#include "audio/format.h"
#include "osdep/timer.h"
#include "core/subopt-helper.h"
#include "libavutil/fifo.h"
#include <jack/jack.h>
//! maximum number of channels supported, avoids lots of mallocs
#define MAX_CHANS MP_NUM_CHANNELS
//! 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
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struct priv {
jack_port_t * ports[MAX_CHANS];
int num_ports; // Number of used ports == number of channels
jack_client_t *client;
float jack_latency;
int estimate;
volatile int paused;
volatile int underrun; // signals if an underrun occured
volatile float callback_interval;
volatile float callback_time;
AVFifoBuffer *buffer; // buffer for audio data
};
/**
* \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
*/
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static int write_buffer(AVFifoBuffer *buffer, unsigned char *data, int len)
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{
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 {
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float **bufs;
int num_bufs;
int cur_buf;
int pos;
};
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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.
*/
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static int read_buffer(AVFifoBuffer *buffer, float **bufs, int cnt, int num_bufs)
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{
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
/**
* \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
*/
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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
*/
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static int outputaudio(jack_nframes_t nframes, void *arg)
{
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struct ao *ao = arg;
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struct priv *p = ao->priv;
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float *bufs[MAX_CHANS];
int i;
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for (i = 0; i < p->num_ports; i++)
bufs[i] = jack_port_get_buffer(p->ports[i], nframes);
if (p->paused || p->underrun || !p->buffer)
silence(bufs, nframes, p->num_ports);
else if (read_buffer(p->buffer, bufs, nframes, p->num_ports) < nframes)
p->underrun = 1;
if (p->estimate) {
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float now = mp_time_us() / 1000000.0;
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float diff = p->callback_time + p->callback_interval - now;
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if ((diff > -0.002) && (diff < 0.002))
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p->callback_time += p->callback_interval;
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else
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p->callback_time = now;
p->callback_interval = (float)nframes / (float)ao->samplerate;
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}
return 0;
}
/**
* \brief print suboption usage help
*/
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static void print_help(void)
{
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mp_msg(
MSGT_AO, MSGL_FATAL,
"\n-ao jack commandline help:\n"
"Example: mpv -ao jack:port=myout\n"
" connects mpv to the jack ports named myout\n"
"\nOptions:\n"
" connect\n"
" Automatically connect to output ports\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"
);
}
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static int init(struct ao *ao, char *params)
{
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const char **matching_ports = NULL;
char *port_name = NULL;
char *client_name = NULL;
int autostart = 0;
int connect = 1;
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struct priv *p = talloc_zero(ao, struct priv);
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const opt_t subopts[] = {
{"port", OPT_ARG_MSTRZ, &port_name, NULL},
{"name", OPT_ARG_MSTRZ, &client_name, NULL},
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{"estimate", OPT_ARG_BOOL, &p->estimate, NULL},
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{"autostart", OPT_ARG_BOOL, &autostart, NULL},
{"connect", OPT_ARG_BOOL, &connect, NULL},
{NULL}
};
jack_options_t open_options = JackUseExactName;
int port_flags = JackPortIsInput;
int i;
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ao->priv = p;
p->estimate = 1;
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if (subopt_parse(params, subopts) != 0) {
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print_help();
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return -1;
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}
struct mp_chmap_sel sel = {0};
mp_chmap_sel_add_waveext(&sel);
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if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels))
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goto err_out;
if (!client_name) {
client_name = malloc(40);
sprintf(client_name, "mpv [%d]", getpid());
}
if (!autostart)
open_options |= JackNoStartServer;
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p->client = jack_client_open(client_name, open_options, NULL);
if (!p->client) {
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mp_msg(MSGT_AO, MSGL_FATAL, "[JACK] cannot open server\n");
goto err_out;
}
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jack_set_process_callback(p->client, outputaudio, ao);
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// list matching ports if connections should be made
if (connect) {
if (!port_name)
port_flags |= JackPortIsPhysical;
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matching_ports = jack_get_ports(p->client, port_name, NULL, port_flags);
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if (!matching_ports || !matching_ports[0]) {
mp_msg(MSGT_AO, MSGL_FATAL, "[JACK] no physical ports available\n");
goto err_out;
}
i = 1;
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p->num_ports = ao->channels.num;
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while (matching_ports[i])
i++;
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if (p->num_ports > i)
p->num_ports = i;
}
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// create out output ports
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for (i = 0; i < p->num_ports; i++) {
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char pname[30];
snprintf(pname, 30, "out_%d", i);
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p->ports[i] =
jack_port_register(p->client, pname, JACK_DEFAULT_AUDIO_TYPE,
JackPortIsOutput, 0);
if (!p->ports[i]) {
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mp_msg(MSGT_AO, MSGL_FATAL, "[JACK] not enough ports available\n");
goto err_out;
}
}
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if (jack_activate(p->client)) {
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mp_msg(MSGT_AO, MSGL_FATAL, "[JACK] activate failed\n");
goto err_out;
}
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for (i = 0; i < p->num_ports; i++) {
if (jack_connect(p->client, jack_port_name(p->ports[i]),
matching_ports[i]))
{
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mp_msg(MSGT_AO, MSGL_FATAL, "[JACK] connecting failed\n");
goto err_out;
}
}
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ao->samplerate = jack_get_sample_rate(p->client);
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jack_latency_range_t jack_latency_range;
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jack_port_get_latency_range(p->ports[0], JackPlaybackLatency,
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&jack_latency_range);
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p->jack_latency = (float)(jack_latency_range.max + jack_get_buffer_size(p->client))
/ (float)ao->samplerate;
p->callback_interval = 0;
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if (!ao_chmap_sel_get_def(ao, &sel, &ao->channels, p->num_ports))
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goto err_out;
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ao->format = AF_FORMAT_FLOAT_NE;
ao->bps = ao->channels.num * ao->samplerate * sizeof(float);
int unitsize = ao->channels.num * sizeof(float);
ao->outburst = CHUNK_SIZE / unitsize * unitsize;
ao->buffersize = NUM_CHUNKS * ao->outburst;
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p->buffer = av_fifo_alloc(ao->buffersize);
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free(matching_ports);
free(port_name);
free(client_name);
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return 0;
err_out:
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free(matching_ports);
free(port_name);
free(client_name);
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if (p->client)
jack_client_close(p->client);
av_fifo_free(p->buffer);
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return -1;
}
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static float get_delay(struct ao *ao)
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{
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struct priv *p = ao->priv;
int buffered = av_fifo_size(p->buffer); // could be less
float in_jack = p->jack_latency;
if (p->estimate && p->callback_interval > 0) {
float elapsed = mp_time_us() / 1000000.0 - p->callback_time;
in_jack += p->callback_interval - elapsed;
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if (in_jack < 0)
in_jack = 0;
}
return (float)buffered / (float)ao->bps + in_jack;
}
/**
* \brief stop playing and empty buffers (for seeking/pause)
*/
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static void reset(struct ao *ao)
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{
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struct priv *p = ao->priv;
p->paused = 1;
av_fifo_reset(p->buffer);
p->paused = 0;
}
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// close audio device
static void uninit(struct ao *ao, bool immed)
{
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struct priv *p = ao->priv;
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if (!immed)
mp_sleep_us(get_delay(ao) * 1000 * 1000);
// HACK, make sure jack doesn't loop-output dirty buffers
reset(ao);
mp_sleep_us(100 * 1000);
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jack_client_close(p->client);
av_fifo_free(p->buffer);
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}
/**
* \brief stop playing, keep buffers (for pause)
*/
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static void audio_pause(struct ao *ao)
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{
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struct priv *p = ao->priv;
p->paused = 1;
}
/**
* \brief resume playing, after audio_pause()
*/
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static void audio_resume(struct ao *ao)
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{
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struct priv *p = ao->priv;
p->paused = 0;
}
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static int get_space(struct ao *ao)
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{
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struct priv *p = ao->priv;
return av_fifo_space(p->buffer);
}
/**
* \brief write data into buffer and reset underrun flag
*/
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static int play(struct ao *ao, void *data, int len, int flags)
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{
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struct priv *p = ao->priv;
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if (!(flags & AOPLAY_FINAL_CHUNK))
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len -= len % ao->outburst;
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p->underrun = 0;
return write_buffer(p->buffer, data, len);
}
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const struct ao_driver audio_out_jack = {
.info = &(const struct ao_info) {
"JACK audio output",
"jack",
"Reimar Döffinger <Reimar.Doeffinger@stud.uni-karlsruhe.de>",
"based on ao_sdl.c"
},
.init = init,
.uninit = uninit,
.get_space = get_space,
.play = play,
.get_delay = get_delay,
.pause = audio_pause,
.resume = audio_resume,
.reset = reset,
};