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mirror of https://github.com/mpv-player/mpv.git synced 2024-09-20 12:02:23 +02:00
mpv/audio/out/ao_jack.c
wm4 e56d8a200d Replace all calls to GetTimer()/GetTimerMS()
GetTimer() is generally replaced with mp_time_us(). Both calls return
microseconds, but the latter uses int64_t, us defined to never wrap,
and never returns 0 or negative values.

GetTimerMS() has no direct replacement. Instead the other functions are
used.

For some code, switch to mp_time_sec(), which returns the time as double
float value in seconds. The returned time is offset to program start
time, so there is enough precision left to deliver microsecond
resolution for at least 100 years. Unless it's casted to a float
(or the CPU reduces precision), which is why we still use mp_time_us()
out of paranoia in places where precision is clearly needed.

Always switch to the correct time. The whole point of the new timer
calls is that they don't wrap, and storing microseconds in unsigned int
variables would negate this.

In some cases, remove wrap-around handling for time values.
2013-05-26 16:44:20 +02:00

370 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 "core/mp_msg.h"
#include "ao.h"
#include "audio_out_internal.h"
#include "audio/format.h"
#include "osdep/timer.h"
#include "core/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 MP_NUM_CHANNELS
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 = mp_time_us() / 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: mpv -ao jack:port=myout\n"
" connects mpv 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, const struct mp_chmap *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;
}
struct mp_chmap_sel sel = {0};
mp_chmap_sel_add_waveext(&sel);
if (!ao_chmap_sel_adjust(&ao_data, &sel, &ao_data.channels))
goto err_out;
if (!client_name) {
client_name = malloc(40);
sprintf(client_name, "mpv [%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;
num_ports = ao_data.channels.num;
while (matching_ports[i]) i++;
if (num_ports > i) num_ports = i;
// 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_range_t jack_latency_range;
jack_port_get_latency_range(ports[0], JackPlaybackLatency,
&jack_latency_range);
jack_latency = (float)(jack_latency_range.max + jack_get_buffer_size(client))
/ (float)rate;
callback_interval = 0;
if (!ao_chmap_sel_get_def(&ao_data, &sel, &ao_data.channels, num_ports))
goto err_out;
ao_data.samplerate = rate;
ao_data.format = AF_FORMAT_FLOAT_NE;
ao_data.bps = ao_data.channels.num * 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 = mp_time_us() / 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;
}