mirror of
https://github.com/mpv-player/mpv
synced 2025-01-03 05:22:23 +00:00
380fc765e4
This comes with two internal AO API changes: 1. ao_driver.play now can take non-interleaved audio. For this purpose, the data pointer is changed to void **data, where data[0] corresponds to the pointer in the old API. Also, the len argument as well as the return value are now in samples, not bytes. "Sample" in this context means the unit of the smallest possible audio frame, i.e. sample_size * channels. 2. ao_driver.get_space now returns samples instead of bytes. (Similar to the play function.) Change all AOs to use the new API. The AO API as exposed to the rest of the player still uses the old API. It's emulated in ao.c. This is purely to split the commits changing all AOs and the commits adding actual support for outputting N-I audio.
368 lines
10 KiB
C
368 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 "mpvcore/mp_msg.h"
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#include "ao.h"
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#include "audio/format.h"
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#include "osdep/timer.h"
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#include "mpvcore/m_option.h"
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#include "mpvcore/mp_ring.h"
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#include <jack/jack.h>
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//! maximum number of channels supported, avoids lots of mallocs
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#define MAX_CHANS MP_NUM_CHANNELS
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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 (24 * 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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struct priv {
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jack_port_t * ports[MAX_CHANS];
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int num_ports; // Number of used ports == number of channels
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jack_client_t *client;
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int outburst;
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float jack_latency;
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char *cfg_port;
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char *cfg_client_name;
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int estimate;
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int connect;
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int autostart;
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int stdlayout;
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volatile int paused;
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volatile int underrun; // signals if an underrun occured
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volatile float callback_interval;
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volatile float callback_time;
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struct mp_ring *ring; // buffer for audio data
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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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{
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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(struct mp_ring *ring, float **bufs, int cnt, int num_bufs)
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{
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struct deinterleave di = {
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bufs, num_bufs, 0, 0
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};
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int buffered = mp_ring_buffered(ring);
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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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mp_ring_read_cb(ring, &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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/**
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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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{
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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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{
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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];
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int i;
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for (i = 0; i < p->num_ports; i++)
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bufs[i] = jack_port_get_buffer(p->ports[i], nframes);
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if (p->paused || p->underrun || !p->ring)
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silence(bufs, nframes, p->num_ports);
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else if (read_buffer(p->ring, bufs, nframes, p->num_ports) < nframes)
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p->underrun = 1;
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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;
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p->callback_interval = (float)nframes / (float)ao->samplerate;
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}
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return 0;
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}
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static int init(struct ao *ao)
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{
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struct priv *p = ao->priv;
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const char **matching_ports = NULL;
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char *port_name = p->cfg_port && p->cfg_port[0] ? p->cfg_port : NULL;
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jack_options_t open_options = JackNullOption;
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int port_flags = JackPortIsInput;
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int i;
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struct mp_chmap_sel sel = {0};
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if (p->stdlayout == 0) {
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mp_chmap_sel_add_waveext(&sel);
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} else if (p->stdlayout == 1) {
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mp_chmap_sel_add_alsa_def(&sel);
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} else {
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mp_chmap_sel_add_any(&sel);
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}
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if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels))
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goto err_out;
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if (!p->autostart)
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open_options |= JackNoStartServer;
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p->client = jack_client_open(p->cfg_client_name, open_options, NULL);
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if (!p->client) {
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MP_FATAL(ao, "cannot open server\n");
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goto err_out;
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}
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jack_set_process_callback(p->client, outputaudio, ao);
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// list matching ports if connections should be made
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if (p->connect) {
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if (!port_name)
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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]) {
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MP_FATAL(ao, "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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p->num_ports = ao->channels.num;
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while (matching_ports[i])
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i++;
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if (p->num_ports > i)
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p->num_ports = i;
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}
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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];
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snprintf(pname, 30, "out_%d", i);
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p->ports[i] =
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jack_port_register(p->client, pname, JACK_DEFAULT_AUDIO_TYPE,
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JackPortIsOutput, 0);
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if (!p->ports[i]) {
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MP_FATAL(ao, "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(p->client)) {
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MP_FATAL(ao, "activate failed\n");
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goto err_out;
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}
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for (i = 0; i < p->num_ports; i++) {
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if (jack_connect(p->client, jack_port_name(p->ports[i]),
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matching_ports[i]))
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{
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MP_FATAL(ao, "connecting failed\n");
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goto err_out;
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}
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}
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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))
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/ (float)ao->samplerate;
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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;
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int unitsize = ao->channels.num * sizeof(float);
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p->outburst = (CHUNK_SIZE + unitsize - 1) / unitsize * unitsize;
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p->ring = mp_ring_new(p, NUM_CHUNKS * p->outburst);
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free(matching_ports);
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return 0;
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err_out:
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free(matching_ports);
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if (p->client)
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jack_client_close(p->client);
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return -1;
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}
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static float get_delay(struct ao *ao)
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{
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struct priv *p = ao->priv;
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int buffered = mp_ring_buffered(p->ring); // could be less
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float in_jack = p->jack_latency;
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if (p->estimate && p->callback_interval > 0) {
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float elapsed = mp_time_us() / 1000000.0 - p->callback_time;
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in_jack += p->callback_interval - elapsed;
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if (in_jack < 0)
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in_jack = 0;
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}
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return (float)buffered / (float)ao->bps + in_jack;
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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(struct ao *ao)
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{
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struct priv *p = ao->priv;
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p->paused = 1;
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mp_ring_reset(p->ring);
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p->paused = 0;
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}
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// close audio device
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static void uninit(struct ao *ao, bool immed)
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{
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struct priv *p = ao->priv;
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if (!immed)
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mp_sleep_us(get_delay(ao) * 1000 * 1000);
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// HACK, make sure jack doesn't loop-output dirty buffers
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reset(ao);
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mp_sleep_us(100 * 1000);
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jack_client_close(p->client);
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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(struct ao *ao)
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{
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struct priv *p = ao->priv;
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p->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(struct ao *ao)
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{
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struct priv *p = ao->priv;
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p->paused = 0;
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}
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static int get_space(struct ao *ao)
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{
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struct priv *p = ao->priv;
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return mp_ring_available(p->ring) / ao->sstride;
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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(struct ao *ao, void **data, int samples, int flags)
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{
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struct priv *p = ao->priv;
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int len = samples * ao->sstride;
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if (!(flags & AOPLAY_FINAL_CHUNK))
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len -= len % p->outburst;
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p->underrun = 0;
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return mp_ring_write(p->ring, data[0], len) / ao->sstride;
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}
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#define OPT_BASE_STRUCT struct priv
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const struct ao_driver audio_out_jack = {
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.description = "JACK audio output",
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.name = "jack",
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.init = init,
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.uninit = uninit,
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.get_space = get_space,
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.play = play,
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.get_delay = get_delay,
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.pause = audio_pause,
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.resume = audio_resume,
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.reset = reset,
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.priv_size = sizeof(struct priv),
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.priv_defaults = &(const struct priv) {
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.cfg_client_name = "mpv",
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.estimate = 1,
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.connect = 1,
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},
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.options = (const struct m_option[]) {
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OPT_STRING("port", cfg_port, 0),
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OPT_STRING("name", cfg_client_name, 0),
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OPT_FLAG("estimate", estimate, 0),
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OPT_FLAG("autostart", autostart, 0),
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OPT_FLAG("connect", connect, 0),
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OPT_CHOICE("std-channel-layout", stdlayout, 0,
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({"waveext", 0}, {"alsa", 1}, {"any", 2})),
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{0}
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},
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};
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