mirror of
https://github.com/mpv-player/mpv
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![tack](/assets/img/avatar_default.png)
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@31358 b3059339-0415-0410-9bf9-f77b7e298cf2
362 lines
10 KiB
C
362 lines
10 KiB
C
/*
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* JACK audio output driver for MPlayer
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*
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* Copyleft 2001 by Felix Bünemann (atmosfear@users.sf.net)
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* and Reimar Döffinger (Reimar.Doeffinger@stud.uni-karlsruhe.de)
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*
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* This file is part of MPlayer.
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*
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* MPlayer is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* MPlayer is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along
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* along with MPlayer; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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#include <unistd.h>
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#include "config.h"
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#include "mp_msg.h"
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#include "audio_out.h"
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#include "audio_out_internal.h"
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#include "libaf/af_format.h"
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#include "osdep/timer.h"
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#include "subopt-helper.h"
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#include "libavutil/fifo.h"
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#include <jack/jack.h>
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static const ao_info_t info =
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{
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"JACK audio output",
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"jack",
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"Reimar Döffinger <Reimar.Doeffinger@stud.uni-karlsruhe.de>",
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"based on ao_sdl.c"
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};
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LIBAO_EXTERN(jack)
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//! maximum number of channels supported, avoids lots of mallocs
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#define MAX_CHANS 8
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static jack_port_t *ports[MAX_CHANS];
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static int num_ports; ///< Number of used ports == number of channels
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static jack_client_t *client;
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static float jack_latency;
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static int estimate;
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static volatile int paused = 0; ///< set if paused
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static volatile int underrun = 0; ///< signals if an underrun occured
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static volatile float callback_interval = 0;
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static volatile float callback_time = 0;
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//! size of one chunk, if this is too small MPlayer will start to "stutter"
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//! after a short time of playback
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#define CHUNK_SIZE (16 * 1024)
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//! number of "virtual" chunks the buffer consists of
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#define NUM_CHUNKS 8
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#define BUFFSIZE (NUM_CHUNKS * CHUNK_SIZE)
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//! buffer for audio data
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static AVFifoBuffer *buffer;
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/**
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* \brief insert len bytes into buffer
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* \param data data to insert
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* \param len length of data
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* \return number of bytes inserted into buffer
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*
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* If there is not enough room, the buffer is filled up
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*/
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static int write_buffer(unsigned char* data, int len) {
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int free = av_fifo_space(buffer);
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if (len > free) len = free;
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return av_fifo_generic_write(buffer, data, len, NULL);
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}
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static void silence(float **bufs, int cnt, int num_bufs);
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struct deinterleave {
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float **bufs;
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int num_bufs;
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int cur_buf;
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int pos;
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};
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static void deinterleave(void *info, void *src, int len) {
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struct deinterleave *di = info;
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float *s = src;
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int i;
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len /= sizeof(float);
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for (i = 0; i < len; i++) {
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di->bufs[di->cur_buf++][di->pos] = s[i];
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if (di->cur_buf >= di->num_bufs) {
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di->cur_buf = 0;
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di->pos++;
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}
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}
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}
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/**
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* \brief read data from buffer and splitting it into channels
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* \param bufs num_bufs float buffers, each will contain the data of one channel
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* \param cnt number of samples to read per channel
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* \param num_bufs number of channels to split the data into
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* \return number of samples read per channel, equals cnt unless there was too
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* little data in the buffer
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*
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* Assumes the data in the buffer is of type float, the number of bytes
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* read is res * num_bufs * sizeof(float), where res is the return value.
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* If there is not enough data in the buffer remaining parts will be filled
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* with silence.
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*/
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static int read_buffer(float **bufs, int cnt, int num_bufs) {
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struct deinterleave di = {bufs, num_bufs, 0, 0};
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int buffered = av_fifo_size(buffer);
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if (cnt * sizeof(float) * num_bufs > buffered) {
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silence(bufs, cnt, num_bufs);
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cnt = buffered / sizeof(float) / num_bufs;
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}
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av_fifo_generic_read(buffer, &di, cnt * num_bufs * sizeof(float), deinterleave);
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return cnt;
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}
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// end ring buffer stuff
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static int control(int cmd, void *arg) {
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return CONTROL_UNKNOWN;
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}
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/**
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* \brief fill the buffers with silence
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* \param bufs num_bufs float buffers, each will contain the data of one channel
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* \param cnt number of samples in each buffer
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* \param num_bufs number of buffers
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*/
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static void silence(float **bufs, int cnt, int num_bufs) {
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int i;
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for (i = 0; i < num_bufs; i++)
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memset(bufs[i], 0, cnt * sizeof(float));
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}
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/**
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* \brief JACK Callback function
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* \param nframes number of frames to fill into buffers
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* \param arg unused
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* \return currently always 0
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*
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* Write silence into buffers if paused or an underrun occured
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*/
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static int outputaudio(jack_nframes_t nframes, void *arg) {
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float *bufs[MAX_CHANS];
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int i;
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for (i = 0; i < num_ports; i++)
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bufs[i] = jack_port_get_buffer(ports[i], nframes);
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if (paused || underrun)
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silence(bufs, nframes, num_ports);
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else
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if (read_buffer(bufs, nframes, num_ports) < nframes)
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underrun = 1;
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if (estimate) {
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float now = (float)GetTimer() / 1000000.0;
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float diff = callback_time + callback_interval - now;
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if ((diff > -0.002) && (diff < 0.002))
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callback_time += callback_interval;
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else
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callback_time = now;
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callback_interval = (float)nframes / (float)ao_data.samplerate;
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}
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return 0;
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}
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/**
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* \brief print suboption usage help
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*/
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static void print_help (void)
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{
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mp_msg (MSGT_AO, MSGL_FATAL,
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"\n-ao jack commandline help:\n"
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"Example: mplayer -ao jack:port=myout\n"
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" connects MPlayer to the jack ports named myout\n"
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"\nOptions:\n"
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" port=<port name>\n"
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" Connects to the given ports instead of the default physical ones\n"
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" name=<client name>\n"
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" Client name to pass to JACK\n"
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" estimate\n"
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" Estimates the amount of data in buffers (experimental)\n"
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" autostart\n"
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" Automatically start JACK server if necessary\n"
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);
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}
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static int init(int rate, int channels, int format, int flags) {
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const char **matching_ports = NULL;
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char *port_name = NULL;
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char *client_name = NULL;
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int autostart = 0;
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const opt_t subopts[] = {
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{"port", OPT_ARG_MSTRZ, &port_name, NULL},
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{"name", OPT_ARG_MSTRZ, &client_name, NULL},
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{"estimate", OPT_ARG_BOOL, &estimate, NULL},
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{"autostart", OPT_ARG_BOOL, &autostart, NULL},
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{NULL}
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};
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jack_options_t open_options = JackUseExactName;
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int port_flags = JackPortIsInput;
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int i;
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estimate = 1;
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if (subopt_parse(ao_subdevice, subopts) != 0) {
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print_help();
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return 0;
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}
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if (channels > MAX_CHANS) {
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mp_msg(MSGT_AO, MSGL_FATAL, "[JACK] Invalid number of channels: %i\n", channels);
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goto err_out;
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}
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if (!client_name) {
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client_name = malloc(40);
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sprintf(client_name, "MPlayer [%d]", getpid());
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}
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if (!autostart)
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open_options |= JackNoStartServer;
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client = jack_client_open(client_name, open_options, NULL);
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if (!client) {
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mp_msg(MSGT_AO, MSGL_FATAL, "[JACK] cannot open server\n");
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goto err_out;
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}
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buffer = av_fifo_alloc(BUFFSIZE);
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jack_set_process_callback(client, outputaudio, 0);
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// list matching ports
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if (!port_name)
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port_flags |= JackPortIsPhysical;
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matching_ports = jack_get_ports(client, port_name, NULL, port_flags);
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if (!matching_ports || !matching_ports[0]) {
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mp_msg(MSGT_AO, MSGL_FATAL, "[JACK] no physical ports available\n");
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goto err_out;
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}
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i = 1;
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while (matching_ports[i]) i++;
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if (channels > i) channels = i;
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num_ports = channels;
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// create out output ports
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for (i = 0; i < num_ports; i++) {
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char pname[30];
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snprintf(pname, 30, "out_%d", i);
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ports[i] = jack_port_register(client, pname, JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0);
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if (!ports[i]) {
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mp_msg(MSGT_AO, MSGL_FATAL, "[JACK] not enough ports available\n");
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goto err_out;
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}
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}
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if (jack_activate(client)) {
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mp_msg(MSGT_AO, MSGL_FATAL, "[JACK] activate failed\n");
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goto err_out;
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}
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for (i = 0; i < num_ports; i++) {
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if (jack_connect(client, jack_port_name(ports[i]), matching_ports[i])) {
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mp_msg(MSGT_AO, MSGL_FATAL, "[JACK] connecting failed\n");
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goto err_out;
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}
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}
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rate = jack_get_sample_rate(client);
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jack_latency = (float)(jack_port_get_total_latency(client, ports[0]) +
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jack_get_buffer_size(client)) / (float)rate;
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callback_interval = 0;
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ao_data.channels = channels;
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ao_data.samplerate = rate;
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ao_data.format = AF_FORMAT_FLOAT_NE;
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ao_data.bps = channels * rate * sizeof(float);
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ao_data.buffersize = CHUNK_SIZE * NUM_CHUNKS;
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ao_data.outburst = CHUNK_SIZE;
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free(matching_ports);
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free(port_name);
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free(client_name);
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return 1;
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err_out:
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free(matching_ports);
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free(port_name);
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free(client_name);
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if (client)
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jack_client_close(client);
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av_fifo_free(buffer);
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buffer = NULL;
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return 0;
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}
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// close audio device
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static void uninit(int immed) {
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if (!immed)
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usec_sleep(get_delay() * 1000 * 1000);
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// HACK, make sure jack doesn't loop-output dirty buffers
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reset();
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usec_sleep(100 * 1000);
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jack_client_close(client);
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av_fifo_free(buffer);
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buffer = NULL;
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}
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/**
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* \brief stop playing and empty buffers (for seeking/pause)
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*/
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static void reset(void) {
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paused = 1;
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av_fifo_reset(buffer);
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paused = 0;
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}
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/**
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* \brief stop playing, keep buffers (for pause)
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*/
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static void audio_pause(void) {
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paused = 1;
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}
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/**
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* \brief resume playing, after audio_pause()
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*/
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static void audio_resume(void) {
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paused = 0;
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}
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static int get_space(void) {
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return av_fifo_space(buffer);
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}
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/**
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* \brief write data into buffer and reset underrun flag
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*/
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static int play(void *data, int len, int flags) {
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if (!(flags & AOPLAY_FINAL_CHUNK))
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len -= len % ao_data.outburst;
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underrun = 0;
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return write_buffer(data, len);
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}
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static float get_delay(void) {
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int buffered = av_fifo_size(buffer); // could be less
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float in_jack = jack_latency;
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if (estimate && callback_interval > 0) {
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float elapsed = (float)GetTimer() / 1000000.0 - callback_time;
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in_jack += callback_interval - elapsed;
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if (in_jack < 0) in_jack = 0;
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}
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return (float)buffered / (float)ao_data.bps + in_jack;
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}
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