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mirror of https://github.com/mpv-player/mpv synced 2024-12-26 17:12:36 +00:00
mpv/audio/out/ao_jack.c
wm4 edd36a3afc audio/out: do some mp_msg conversions
Use the new MP_ macros for some AOs instead of mp_msg.

Not all AOs are converted, and some only partially. In some cases, some
additional cosmetic changes are made.
2013-08-22 23:12:35 +02:00

371 lines
11 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 "mpvcore/mp_msg.h"
#include "ao.h"
#include "audio/format.h"
#include "osdep/timer.h"
#include "mpvcore/m_option.h"
#include "mpvcore/mp_ring.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 (24 * 1024)
//! number of "virtual" chunks the buffer consists of
#define NUM_CHUNKS 8
struct priv {
jack_port_t * ports[MAX_CHANS];
int num_ports; // Number of used ports == number of channels
jack_client_t *client;
int outburst;
float jack_latency;
char *cfg_port;
char *cfg_client_name;
int estimate;
int connect;
int autostart;
int stdlayout;
volatile int paused;
volatile int underrun; // signals if an underrun occured
volatile float callback_interval;
volatile float callback_time;
struct mp_ring *ring; // buffer for audio data
};
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(struct mp_ring *ring, float **bufs, int cnt, int num_bufs)
{
struct deinterleave di = {
bufs, num_bufs, 0, 0
};
int buffered = mp_ring_buffered(ring);
if (cnt * sizeof(float) * num_bufs > buffered) {
silence(bufs, cnt, num_bufs);
cnt = buffered / sizeof(float) / num_bufs;
}
mp_ring_read_cb(ring, &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
*/
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)
{
struct ao *ao = arg;
struct priv *p = ao->priv;
float *bufs[MAX_CHANS];
int i;
for (i = 0; i < p->num_ports; i++)
bufs[i] = jack_port_get_buffer(p->ports[i], nframes);
if (p->paused || p->underrun || !p->ring)
silence(bufs, nframes, p->num_ports);
else if (read_buffer(p->ring, bufs, nframes, p->num_ports) < nframes)
p->underrun = 1;
if (p->estimate) {
float now = mp_time_us() / 1000000.0;
float diff = p->callback_time + p->callback_interval - now;
if ((diff > -0.002) && (diff < 0.002))
p->callback_time += p->callback_interval;
else
p->callback_time = now;
p->callback_interval = (float)nframes / (float)ao->samplerate;
}
return 0;
}
static int init(struct ao *ao)
{
struct priv *p = ao->priv;
const char **matching_ports = NULL;
char *port_name = p->cfg_port && p->cfg_port[0] ? p->cfg_port : NULL;
jack_options_t open_options = JackUseExactName;
int port_flags = JackPortIsInput;
int i;
struct mp_chmap_sel sel = {0};
if (p->stdlayout == 0) {
mp_chmap_sel_add_waveext(&sel);
} else if (p->stdlayout == 1) {
mp_chmap_sel_add_alsa_def(&sel);
} else {
mp_chmap_sel_add_any(&sel);
}
if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels))
goto err_out;
if (!p->autostart)
open_options |= JackNoStartServer;
p->client = jack_client_open(p->cfg_client_name, open_options, NULL);
if (!p->client) {
MP_FATAL(ao, "cannot open server\n");
goto err_out;
}
jack_set_process_callback(p->client, outputaudio, ao);
// list matching ports if connections should be made
if (p->connect) {
if (!port_name)
port_flags |= JackPortIsPhysical;
matching_ports = jack_get_ports(p->client, port_name, NULL, port_flags);
if (!matching_ports || !matching_ports[0]) {
MP_FATAL(ao, "no physical ports available\n");
goto err_out;
}
i = 1;
p->num_ports = ao->channels.num;
while (matching_ports[i])
i++;
if (p->num_ports > i)
p->num_ports = i;
}
// create out output ports
for (i = 0; i < p->num_ports; i++) {
char pname[30];
snprintf(pname, 30, "out_%d", i);
p->ports[i] =
jack_port_register(p->client, pname, JACK_DEFAULT_AUDIO_TYPE,
JackPortIsOutput, 0);
if (!p->ports[i]) {
MP_FATAL(ao, "not enough ports available\n");
goto err_out;
}
}
if (jack_activate(p->client)) {
MP_FATAL(ao, "activate failed\n");
goto err_out;
}
for (i = 0; i < p->num_ports; i++) {
if (jack_connect(p->client, jack_port_name(p->ports[i]),
matching_ports[i]))
{
MP_FATAL(ao, "connecting failed\n");
goto err_out;
}
}
ao->samplerate = jack_get_sample_rate(p->client);
jack_latency_range_t jack_latency_range;
jack_port_get_latency_range(p->ports[0], JackPlaybackLatency,
&jack_latency_range);
p->jack_latency = (float)(jack_latency_range.max + jack_get_buffer_size(p->client))
/ (float)ao->samplerate;
p->callback_interval = 0;
if (!ao_chmap_sel_get_def(ao, &sel, &ao->channels, p->num_ports))
goto err_out;
ao->format = AF_FORMAT_FLOAT_NE;
int unitsize = ao->channels.num * sizeof(float);
p->outburst = (CHUNK_SIZE + unitsize - 1) / unitsize * unitsize;
p->ring = mp_ring_new(p, NUM_CHUNKS * p->outburst);
free(matching_ports);
return 0;
err_out:
free(matching_ports);
if (p->client)
jack_client_close(p->client);
return -1;
}
static float get_delay(struct ao *ao)
{
struct priv *p = ao->priv;
int buffered = mp_ring_buffered(p->ring); // 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;
if (in_jack < 0)
in_jack = 0;
}
return (float)buffered / (float)ao->bps + in_jack;
}
/**
* \brief stop playing and empty buffers (for seeking/pause)
*/
static void reset(struct ao *ao)
{
struct priv *p = ao->priv;
p->paused = 1;
mp_ring_reset(p->ring);
p->paused = 0;
}
// close audio device
static void uninit(struct ao *ao, bool immed)
{
struct priv *p = ao->priv;
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);
jack_client_close(p->client);
}
/**
* \brief stop playing, keep buffers (for pause)
*/
static void audio_pause(struct ao *ao)
{
struct priv *p = ao->priv;
p->paused = 1;
}
/**
* \brief resume playing, after audio_pause()
*/
static void audio_resume(struct ao *ao)
{
struct priv *p = ao->priv;
p->paused = 0;
}
static int get_space(struct ao *ao)
{
struct priv *p = ao->priv;
return mp_ring_available(p->ring);
}
/**
* \brief write data into buffer and reset underrun flag
*/
static int play(struct ao *ao, void *data, int len, int flags)
{
struct priv *p = ao->priv;
if (!(flags & AOPLAY_FINAL_CHUNK))
len -= len % p->outburst;
p->underrun = 0;
return mp_ring_write(p->ring, data, len);
}
#define OPT_BASE_STRUCT struct priv
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,
.priv_size = sizeof(struct priv),
.priv_defaults = &(const struct priv) {
.cfg_client_name = "mpv",
.estimate = 1,
.connect = 1,
},
.options = (const struct m_option[]) {
OPT_STRING("port", cfg_port, 0),
OPT_STRING("name", cfg_client_name, 0),
OPT_FLAG("estimate", estimate, 0),
OPT_FLAG("autostart", autostart, 0),
OPT_FLAG("connect", connect, 0),
OPT_CHOICE("std-channel-layout", stdlayout, 0,
({"waveext", 0}, {"alsa", 1}, {"any", 2})),
{0}
},
};