/* * 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 #include #include #include #include "config.h" #include "core/mp_msg.h" #include "ao.h" #include "audio/format.h" #include "osdep/timer.h" #include "core/subopt-helper.h" #include "libavutil/fifo.h" #include //! maximum number of channels supported, avoids lots of mallocs #define MAX_CHANS MP_NUM_CHANNELS //! size of one chunk, if this is too small MPlayer will start to "stutter" //! after a short time of playback #define CHUNK_SIZE (16 * 1024) //! number of "virtual" chunks the buffer consists of #define NUM_CHUNKS 8 struct priv { jack_port_t * ports[MAX_CHANS]; int num_ports; // Number of used ports == number of channels jack_client_t *client; float jack_latency; int estimate; volatile int paused; volatile int underrun; // signals if an underrun occured volatile float callback_interval; volatile float callback_time; AVFifoBuffer *buffer; // buffer for audio data }; /** * \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(AVFifoBuffer *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(AVFifoBuffer *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 /** * \brief fill the buffers with silence * \param bufs num_bufs float buffers, each will contain the data of one channel * \param cnt number of samples in each buffer * \param num_bufs number of buffers */ static void silence(float **bufs, int cnt, int num_bufs) { int i; for (i = 0; i < num_bufs; i++) memset(bufs[i], 0, cnt * sizeof(float)); } /** * \brief JACK Callback function * \param nframes number of frames to fill into buffers * \param arg unused * \return currently always 0 * * Write silence into buffers if paused or an underrun occured */ static int outputaudio(jack_nframes_t nframes, void *arg) { struct ao *ao = arg; struct priv *p = ao->priv; float *bufs[MAX_CHANS]; int i; for (i = 0; i < p->num_ports; i++) bufs[i] = jack_port_get_buffer(p->ports[i], nframes); if (p->paused || p->underrun || !p->buffer) silence(bufs, nframes, p->num_ports); else if (read_buffer(p->buffer, bufs, nframes, p->num_ports) < nframes) p->underrun = 1; if (p->estimate) { float now = mp_time_us() / 1000000.0; float diff = p->callback_time + p->callback_interval - now; if ((diff > -0.002) && (diff < 0.002)) p->callback_time += p->callback_interval; else p->callback_time = now; p->callback_interval = (float)nframes / (float)ao->samplerate; } return 0; } /** * \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" " connect\n" " Automatically connect to output ports\n" " port=\n" " Connects to the given ports instead of the default physical ones\n" " 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(struct ao *ao, char *params) { const char **matching_ports = NULL; char *port_name = NULL; char *client_name = NULL; int autostart = 0; int connect = 1; struct priv *p = talloc_zero(ao, struct priv); const opt_t subopts[] = { {"port", OPT_ARG_MSTRZ, &port_name, NULL}, {"name", OPT_ARG_MSTRZ, &client_name, NULL}, {"estimate", OPT_ARG_BOOL, &p->estimate, NULL}, {"autostart", OPT_ARG_BOOL, &autostart, NULL}, {"connect", OPT_ARG_BOOL, &connect, NULL}, {NULL} }; jack_options_t open_options = JackUseExactName; int port_flags = JackPortIsInput; int i; ao->priv = p; p->estimate = 1; if (subopt_parse(params, subopts) != 0) { print_help(); return -1; } struct mp_chmap_sel sel = {0}; mp_chmap_sel_add_waveext(&sel); if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels)) goto err_out; if (!client_name) { client_name = malloc(40); sprintf(client_name, "mpv [%d]", getpid()); } if (!autostart) open_options |= JackNoStartServer; p->client = jack_client_open(client_name, open_options, NULL); if (!p->client) { mp_msg(MSGT_AO, MSGL_FATAL, "[JACK] cannot open server\n"); goto err_out; } jack_set_process_callback(p->client, outputaudio, ao); // list matching ports if connections should be made if (connect) { if (!port_name) port_flags |= JackPortIsPhysical; matching_ports = jack_get_ports(p->client, port_name, NULL, port_flags); if (!matching_ports || !matching_ports[0]) { mp_msg(MSGT_AO, MSGL_FATAL, "[JACK] no physical ports available\n"); goto err_out; } i = 1; p->num_ports = ao->channels.num; while (matching_ports[i]) i++; if (p->num_ports > i) p->num_ports = i; } // create out output ports for (i = 0; i < p->num_ports; i++) { char pname[30]; snprintf(pname, 30, "out_%d", i); p->ports[i] = jack_port_register(p->client, pname, JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0); if (!p->ports[i]) { mp_msg(MSGT_AO, MSGL_FATAL, "[JACK] not enough ports available\n"); goto err_out; } } if (jack_activate(p->client)) { mp_msg(MSGT_AO, MSGL_FATAL, "[JACK] activate failed\n"); goto err_out; } for (i = 0; i < p->num_ports; i++) { if (jack_connect(p->client, jack_port_name(p->ports[i]), matching_ports[i])) { mp_msg(MSGT_AO, MSGL_FATAL, "[JACK] connecting failed\n"); goto err_out; } } ao->samplerate = jack_get_sample_rate(p->client); jack_latency_range_t jack_latency_range; jack_port_get_latency_range(p->ports[0], JackPlaybackLatency, &jack_latency_range); p->jack_latency = (float)(jack_latency_range.max + jack_get_buffer_size(p->client)) / (float)ao->samplerate; p->callback_interval = 0; if (!ao_chmap_sel_get_def(ao, &sel, &ao->channels, p->num_ports)) goto err_out; ao->format = AF_FORMAT_FLOAT_NE; ao->bps = ao->channels.num * ao->samplerate * sizeof(float); int unitsize = ao->channels.num * sizeof(float); ao->outburst = CHUNK_SIZE / unitsize * unitsize; ao->buffersize = NUM_CHUNKS * ao->outburst; p->buffer = av_fifo_alloc(ao->buffersize); free(matching_ports); free(port_name); free(client_name); return 0; err_out: free(matching_ports); free(port_name); free(client_name); if (p->client) jack_client_close(p->client); av_fifo_free(p->buffer); return -1; } static float get_delay(struct ao *ao) { struct priv *p = ao->priv; int buffered = av_fifo_size(p->buffer); // could be less float in_jack = p->jack_latency; if (p->estimate && p->callback_interval > 0) { float elapsed = mp_time_us() / 1000000.0 - p->callback_time; in_jack += p->callback_interval - elapsed; if (in_jack < 0) in_jack = 0; } return (float)buffered / (float)ao->bps + in_jack; } /** * \brief stop playing and empty buffers (for seeking/pause) */ static void reset(struct ao *ao) { struct priv *p = ao->priv; p->paused = 1; av_fifo_reset(p->buffer); p->paused = 0; } // close audio device static void uninit(struct ao *ao, bool immed) { struct priv *p = ao->priv; if (!immed) mp_sleep_us(get_delay(ao) * 1000 * 1000); // HACK, make sure jack doesn't loop-output dirty buffers reset(ao); mp_sleep_us(100 * 1000); jack_client_close(p->client); av_fifo_free(p->buffer); } /** * \brief stop playing, keep buffers (for pause) */ static void audio_pause(struct ao *ao) { struct priv *p = ao->priv; p->paused = 1; } /** * \brief resume playing, after audio_pause() */ static void audio_resume(struct ao *ao) { struct priv *p = ao->priv; p->paused = 0; } static int get_space(struct ao *ao) { struct priv *p = ao->priv; return av_fifo_space(p->buffer); } /** * \brief write data into buffer and reset underrun flag */ static int play(struct ao *ao, void *data, int len, int flags) { struct priv *p = ao->priv; if (!(flags & AOPLAY_FINAL_CHUNK)) len -= len % ao->outburst; p->underrun = 0; return write_buffer(p->buffer, data, len); } const struct ao_driver audio_out_jack = { .info = &(const struct ao_info) { "JACK audio output", "jack", "Reimar Döffinger ", "based on ao_sdl.c" }, .init = init, .uninit = uninit, .get_space = get_space, .play = play, .get_delay = get_delay, .pause = audio_pause, .resume = audio_resume, .reset = reset, };