Mercurial > mplayer.hg
view libao2/ao_jack.c @ 29658:6235d300cf7e
Allow playback of dnxhd files, as produced by FFmpeg regression test.
author | cehoyos |
---|---|
date | Tue, 22 Sep 2009 00:02:05 +0000 |
parents | 2eff450157cd |
children | 32725ca88fed |
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/* * JACK audio output driver for MPlayer * * Copyleft 2001 by Felix Bünemann (atmosfear@users.sf.net) * and Reimar Döffinger (Reimar.Doeffinger@stud.uni-karlsruhe.de) * * This file is part of MPlayer. * * MPlayer is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or * (at your option) any later version. * * MPlayer is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. * * You should have received a copy of the GNU General Public License along * along with MPlayer; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ #include <stdio.h> #include <stdlib.h> #include <string.h> #include <unistd.h> #include "config.h" #include "mp_msg.h" #include "help_mp.h" #include "audio_out.h" #include "audio_out_internal.h" #include "libaf/af_format.h" #include "osdep/timer.h" #include "subopt-helper.h" #include "libavutil/fifo.h" #include <jack/jack.h> static const ao_info_t info = { "JACK audio output", "jack", "Reimar Döffinger <Reimar.Doeffinger@stud.uni-karlsruhe.de>", "based on ao_sdl.c" }; LIBAO_EXTERN(jack) //! maximum number of channels supported, avoids lots of mallocs #define MAX_CHANS 6 static jack_port_t *ports[MAX_CHANS]; static int num_ports; ///< Number of used ports == number of channels static jack_client_t *client; static float jack_latency; static int estimate; static volatile int paused = 0; ///< set if paused static volatile int underrun = 0; ///< signals if an underrun occured static volatile float callback_interval = 0; static volatile float callback_time = 0; //! size of one chunk, if this is too small MPlayer will start to "stutter" //! after a short time of playback #define CHUNK_SIZE (16 * 1024) //! number of "virtual" chunks the buffer consists of #define NUM_CHUNKS 8 #define BUFFSIZE (NUM_CHUNKS * CHUNK_SIZE) //! buffer for audio data static AVFifoBuffer *buffer; /** * \brief insert len bytes into buffer * \param data data to insert * \param len length of data * \return number of bytes inserted into buffer * * If there is not enough room, the buffer is filled up */ static int write_buffer(unsigned char* data, int len) { int free = av_fifo_space(buffer); if (len > free) len = free; return av_fifo_generic_write(buffer, data, len, NULL); } static void silence(float **bufs, int cnt, int num_bufs); struct deinterleave { float **bufs; int num_bufs; int cur_buf; int pos; }; static void deinterleave(void *info, void *src, int len) { struct deinterleave *di = info; float *s = src; int i; len /= sizeof(float); for (i = 0; i < len; i++) { di->bufs[di->cur_buf++][di->pos] = s[i]; if (di->cur_buf >= di->num_bufs) { di->cur_buf = 0; di->pos++; } } } /** * \brief read data from buffer and splitting it into channels * \param bufs num_bufs float buffers, each will contain the data of one channel * \param cnt number of samples to read per channel * \param num_bufs number of channels to split the data into * \return number of samples read per channel, equals cnt unless there was too * little data in the buffer * * Assumes the data in the buffer is of type float, the number of bytes * read is res * num_bufs * sizeof(float), where res is the return value. * If there is not enough data in the buffer remaining parts will be filled * with silence. */ static int read_buffer(float **bufs, int cnt, int num_bufs) { struct deinterleave di = {bufs, num_bufs, 0, 0}; int buffered = av_fifo_size(buffer); if (cnt * sizeof(float) * num_bufs > buffered) { silence(bufs, cnt, num_bufs); cnt = buffered / sizeof(float) / num_bufs; } av_fifo_generic_read(buffer, &di, cnt * num_bufs * sizeof(float), deinterleave); return cnt; } // end ring buffer stuff static int control(int cmd, void *arg) { return CONTROL_UNKNOWN; } /** * \brief fill the buffers with silence * \param bufs num_bufs float buffers, each will contain the data of one channel * \param cnt number of samples in each buffer * \param num_bufs number of buffers */ static void silence(float **bufs, int cnt, int num_bufs) { int i; for (i = 0; i < num_bufs; i++) memset(bufs[i], 0, cnt * sizeof(float)); } /** * \brief JACK Callback function * \param nframes number of frames to fill into buffers * \param arg unused * \return currently always 0 * * Write silence into buffers if paused or an underrun occured */ static int outputaudio(jack_nframes_t nframes, void *arg) { float *bufs[MAX_CHANS]; int i; for (i = 0; i < num_ports; i++) bufs[i] = jack_port_get_buffer(ports[i], nframes); if (paused || underrun) silence(bufs, nframes, num_ports); else if (read_buffer(bufs, nframes, num_ports) < nframes) underrun = 1; if (estimate) { float now = (float)GetTimer() / 1000000.0; float diff = callback_time + callback_interval - now; if ((diff > -0.002) && (diff < 0.002)) callback_time += callback_interval; else callback_time = now; callback_interval = (float)nframes / (float)ao_data.samplerate; } return 0; } /** * \brief print suboption usage help */ static void print_help (void) { mp_msg (MSGT_AO, MSGL_FATAL, "\n-ao jack commandline help:\n" "Example: mplayer -ao jack:port=myout\n" " connects MPlayer to the jack ports named myout\n" "\nOptions:\n" " port=<port name>\n" " Connects to the given ports instead of the default physical ones\n" " name=<client name>\n" " Client name to pass to JACK\n" " estimate\n" " Estimates the amount of data in buffers (experimental)\n" " autostart\n" " Automatically start JACK server if necessary\n" ); } static int init(int rate, int channels, int format, int flags) { const char **matching_ports = NULL; char *port_name = NULL; char *client_name = NULL; int autostart = 0; const opt_t subopts[] = { {"port", OPT_ARG_MSTRZ, &port_name, NULL}, {"name", OPT_ARG_MSTRZ, &client_name, NULL}, {"estimate", OPT_ARG_BOOL, &estimate, NULL}, {"autostart", OPT_ARG_BOOL, &autostart, NULL}, {NULL} }; jack_options_t open_options = JackUseExactName; int port_flags = JackPortIsInput; int i; estimate = 1; if (subopt_parse(ao_subdevice, subopts) != 0) { print_help(); return 0; } if (channels > MAX_CHANS) { mp_msg(MSGT_AO, MSGL_FATAL, "[JACK] Invalid number of channels: %i\n", channels); goto err_out; } if (!client_name) { client_name = malloc(40); sprintf(client_name, "MPlayer [%d]", getpid()); } if (!autostart) open_options |= JackNoStartServer; client = jack_client_open(client_name, open_options, NULL); if (!client) { mp_msg(MSGT_AO, MSGL_FATAL, "[JACK] cannot open server\n"); goto err_out; } buffer = av_fifo_alloc(BUFFSIZE); jack_set_process_callback(client, outputaudio, 0); // list matching ports if (!port_name) port_flags |= JackPortIsPhysical; matching_ports = jack_get_ports(client, port_name, NULL, port_flags); if (!matching_ports || !matching_ports[0]) { mp_msg(MSGT_AO, MSGL_FATAL, "[JACK] no physical ports available\n"); goto err_out; } i = 1; while (matching_ports[i]) i++; if (channels > i) channels = i; num_ports = channels; // create out output ports for (i = 0; i < num_ports; i++) { char pname[30]; snprintf(pname, 30, "out_%d", i); ports[i] = jack_port_register(client, pname, JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0); if (!ports[i]) { mp_msg(MSGT_AO, MSGL_FATAL, "[JACK] not enough ports available\n"); goto err_out; } } if (jack_activate(client)) { mp_msg(MSGT_AO, MSGL_FATAL, "[JACK] activate failed\n"); goto err_out; } for (i = 0; i < num_ports; i++) { if (jack_connect(client, jack_port_name(ports[i]), matching_ports[i])) { mp_msg(MSGT_AO, MSGL_FATAL, "[JACK] connecting failed\n"); goto err_out; } } rate = jack_get_sample_rate(client); jack_latency = (float)(jack_port_get_total_latency(client, ports[0]) + jack_get_buffer_size(client)) / (float)rate; callback_interval = 0; ao_data.channels = channels; ao_data.samplerate = rate; ao_data.format = AF_FORMAT_FLOAT_NE; ao_data.bps = channels * rate * sizeof(float); ao_data.buffersize = CHUNK_SIZE * NUM_CHUNKS; ao_data.outburst = CHUNK_SIZE; free(matching_ports); free(port_name); free(client_name); return 1; err_out: free(matching_ports); free(port_name); free(client_name); if (client) jack_client_close(client); av_fifo_free(buffer); buffer = NULL; return 0; } // close audio device static void uninit(int immed) { if (!immed) usec_sleep(get_delay() * 1000 * 1000); // HACK, make sure jack doesn't loop-output dirty buffers reset(); usec_sleep(100 * 1000); jack_client_close(client); av_fifo_free(buffer); buffer = NULL; } /** * \brief stop playing and empty buffers (for seeking/pause) */ static void reset(void) { paused = 1; av_fifo_reset(buffer); paused = 0; } /** * \brief stop playing, keep buffers (for pause) */ static void audio_pause(void) { paused = 1; } /** * \brief resume playing, after audio_pause() */ static void audio_resume(void) { paused = 0; } static int get_space(void) { return av_fifo_space(buffer); } /** * \brief write data into buffer and reset underrun flag */ static int play(void *data, int len, int flags) { if (!(flags & AOPLAY_FINAL_CHUNK)) len -= len % ao_data.outburst; underrun = 0; return write_buffer(data, len); } static float get_delay(void) { int buffered = av_fifo_size(buffer); // could be less float in_jack = jack_latency; if (estimate && callback_interval > 0) { float elapsed = (float)GetTimer() / 1000000.0 - callback_time; in_jack += callback_interval - elapsed; if (in_jack < 0) in_jack = 0; } return (float)buffered / (float)ao_data.bps + in_jack; }