mirror of
https://github.com/mpv-player/mpv
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4873b32c59
Finish renaming directories and moving files. Adjust all include statements to make the previous commit compile. The two commits are separate, because git is bad at tracking renames and content changes at the same time. Also take this as an opportunity to remove the separation between "common" and "mplayer" sources in the Makefile. ("common" used to be shared between mplayer and mencoder.)
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 "core/mp_msg.h"
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#include "ao.h"
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#include "audio_out_internal.h"
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#include "audio/format.h"
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#include "osdep/timer.h"
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#include "core/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: mpv -ao jack:port=myout\n"
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" connects mpv 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, "mpv [%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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