mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-12-23 12:43:46 +02:00
Implement JACK input device.
Patch by Olivier Guilyardi list samalyse com. See the thread: "[FFmpeg-devel] [PATCH] libavdevice: JACK demuxer". Originally committed as revision 18322 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
parent
7b09db3522
commit
80ff8a16f5
4
configure
vendored
4
configure
vendored
@ -1112,6 +1112,8 @@ avisynth_demuxer_deps="avisynth"
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bktr_demuxer_deps_any="dev_bktr_ioctl_bt848_h machine_ioctl_bt848_h dev_video_bktr_ioctl_bt848_h dev_ic_bt8xx_h"
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dirac_demuxer_deps="dirac_parser"
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dv1394_demuxer_deps="dv1394 dv_demuxer"
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jack_demuxer_deps="jack_jack_h"
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jack_demuxer_extralibs="-ljack"
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libdc1394_demuxer_deps="libdc1394"
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libnut_demuxer_deps="libnut"
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libnut_muxer_deps="libnut"
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@ -2149,6 +2151,8 @@ check_header soundcard.h
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check_lib2 alsa/asoundlib.h snd_pcm_htimestamp -lasound
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check_lib2 jack/jack.h jack_client_open -ljack
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# deal with the X11 frame grabber
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enabled x11grab &&
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check_header X11/Xlib.h &&
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@ -653,6 +653,7 @@ performance on systems without hardware floating point support).
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@item BEOS audio @tab X @tab X
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@item BKTR @tab X @tab
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@item DV1394 @tab X @tab
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@item JACK @tab X @tab
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@item LIBDC1394 @tab X @tab
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@item OSS @tab X @tab X
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@item Video4Linux @tab X @tab
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@ -12,6 +12,7 @@ OBJS-$(CONFIG_ALSA_DEMUXER) += alsa-audio-common.o alsa-audio-dec.o
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OBJS-$(CONFIG_ALSA_MUXER) += alsa-audio-common.o alsa-audio-enc.o
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OBJS-$(CONFIG_BKTR_DEMUXER) += bktr.o
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OBJS-$(CONFIG_DV1394_DEMUXER) += dv1394.o
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OBJS-$(CONFIG_JACK_DEMUXER) += jack_audio.o
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OBJS-$(CONFIG_OSS_DEMUXER) += oss_audio.o
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OBJS-$(CONFIG_OSS_MUXER) += oss_audio.o
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OBJS-$(CONFIG_V4L2_DEMUXER) += v4l2.o
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@ -48,6 +48,7 @@ void avdevice_register_all(void)
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REGISTER_MUXDEMUX (AUDIO_BEOS, audio_beos);
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REGISTER_DEMUXER (BKTR, bktr);
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REGISTER_DEMUXER (DV1394, dv1394);
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REGISTER_DEMUXER (JACK, jack);
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REGISTER_MUXDEMUX (OSS, oss);
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REGISTER_DEMUXER (V4L2, v4l2);
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REGISTER_DEMUXER (V4L, v4l);
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@ -20,7 +20,7 @@
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#define AVDEVICE_AVDEVICE_H
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#define LIBAVDEVICE_VERSION_MAJOR 52
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#define LIBAVDEVICE_VERSION_MINOR 1
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#define LIBAVDEVICE_VERSION_MINOR 2
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#define LIBAVDEVICE_VERSION_MICRO 0
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#define LIBAVDEVICE_VERSION_INT AV_VERSION_INT(LIBAVDEVICE_VERSION_MAJOR, \
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326
libavdevice/jack_audio.c
Normal file
326
libavdevice/jack_audio.c
Normal file
@ -0,0 +1,326 @@
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/*
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* JACK Audio Connection Kit input device
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* Copyright (c) 2009 Samalyse
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* Author: Olivier Guilyardi <olivier samalyse com>
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg 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 GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; 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 "config.h"
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#include <semaphore.h>
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#include <jack/jack.h>
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#include "libavutil/log.h"
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#include "libavutil/fifo.h"
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#include "libavcodec/avcodec.h"
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#include "libavformat/avformat.h"
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#include "libavformat/timefilter.h"
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/**
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* Size of the internal FIFO buffers as a number of audio packets
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*/
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#define FIFO_PACKETS_NUM 16
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typedef struct {
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jack_client_t * client;
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int activated;
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sem_t packet_count;
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jack_nframes_t sample_rate;
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jack_nframes_t buffer_size;
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jack_port_t ** ports;
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int nports;
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TimeFilter * timefilter;
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AVFifoBuffer * new_pkts;
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AVFifoBuffer * filled_pkts;
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int pkt_xrun;
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int jack_xrun;
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} JackData;
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static int process_callback(jack_nframes_t nframes, void *arg)
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{
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/* Warning: this function runs in realtime. One mustn't allocate memory here
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* or do any other thing that could block. */
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int i, j;
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JackData *self = arg;
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float * buffer;
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jack_nframes_t latency, cycle_delay;
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AVPacket pkt;
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float *pkt_data;
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double cycle_time;
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if (!self->client)
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return 0;
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/* The approximate delay since the hardware interrupt as a number of frames */
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cycle_delay = jack_frames_since_cycle_start(self->client);
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/* Retrieve filtered cycle time */
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cycle_time = ff_timefilter_update(self->timefilter,
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av_gettime() / 1000000.0 - (double) cycle_delay / self->sample_rate,
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self->buffer_size);
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/* Check if an empty packet is available, and if there's enough space to send it back once filled */
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if ((av_fifo_size(self->new_pkts) < sizeof(pkt)) || (av_fifo_space(self->filled_pkts) < sizeof(pkt))) {
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self->pkt_xrun = 1;
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return 0;
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}
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/* Retrieve empty (but allocated) packet */
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av_fifo_generic_read(self->new_pkts, &pkt, sizeof(pkt), NULL);
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pkt_data = (float *) pkt.data;
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latency = 0;
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/* Copy and interleave audio data from the JACK buffer into the packet */
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for (i = 0; i < self->nports; i++) {
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latency += jack_port_get_total_latency(self->client, self->ports[i]);
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buffer = jack_port_get_buffer(self->ports[i], self->buffer_size);
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for (j = 0; j < self->buffer_size; j++)
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pkt_data[j * self->nports + i] = buffer[j];
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}
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/* Timestamp the packet with the cycle start time minus the average latency */
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pkt.pts = (cycle_time - (double) latency / (self->nports * self->sample_rate)) * 1000000.0;
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/* Send the now filled packet back, and increase packet counter */
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av_fifo_generic_write(self->filled_pkts, &pkt, sizeof(pkt), NULL);
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sem_post(&self->packet_count);
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return 0;
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}
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static void shutdown_callback(void *arg)
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{
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JackData *self = arg;
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self->client = NULL;
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}
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static int xrun_callback(void *arg)
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{
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JackData *self = arg;
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self->jack_xrun = 1;
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ff_timefilter_reset(self->timefilter);
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return 0;
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}
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static int supply_new_packets(JackData *self, AVFormatContext *context)
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{
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AVPacket pkt;
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int test, pkt_size = self->buffer_size * self->nports * sizeof(float);
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/* Supply the process callback with new empty packets, by filling the new
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* packets FIFO buffer with as many packets as possible. process_callback()
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* can't do this by itself, because it can't allocate memory in realtime. */
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while (av_fifo_space(self->new_pkts) >= sizeof(pkt)) {
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if ((test = av_new_packet(&pkt, pkt_size)) < 0) {
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av_log(context, AV_LOG_ERROR, "Could not create packet of size %d\n", pkt_size);
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return test;
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}
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av_fifo_generic_write(self->new_pkts, &pkt, sizeof(pkt), NULL);
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}
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return 0;
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}
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static int start_jack(AVFormatContext *context, AVFormatParameters *params)
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{
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JackData *self = context->priv_data;
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jack_status_t status;
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int i, test;
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double o, period;
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/* Register as a JACK client, using the context filename as client name. */
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self->client = jack_client_open(context->filename, 0, &status);
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if (!self->client) {
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av_log(context, AV_LOG_ERROR, "Unable to register as a JACK client\n");
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return AVERROR(EIO);
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}
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sem_init(&self->packet_count, 0, 0);
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self->sample_rate = jack_get_sample_rate(self->client);
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self->nports = params->channels;
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self->ports = av_malloc(self->nports * sizeof(*self->ports));
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self->buffer_size = jack_get_buffer_size(self->client);
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/* Register JACK ports */
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for (i = 0; i < self->nports; i++) {
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char str[16];
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snprintf(str, sizeof(str), "input_%d", i + 1);
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self->ports[i] = jack_port_register(self->client, str,
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JACK_DEFAULT_AUDIO_TYPE,
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JackPortIsInput, 0);
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if (!self->ports[i]) {
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av_log(context, AV_LOG_ERROR, "Unable to register port %s:%s\n",
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context->filename, str);
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jack_client_close(self->client);
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return AVERROR(EIO);
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}
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}
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/* Register JACK callbacks */
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jack_set_process_callback(self->client, process_callback, self);
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jack_on_shutdown(self->client, shutdown_callback, self);
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jack_set_xrun_callback(self->client, xrun_callback, self);
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/* Create time filter */
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period = (double) self->buffer_size / self->sample_rate;
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o = 2 * M_PI * 1.5 * period; /// bandwidth: 1.5Hz
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self->timefilter = ff_timefilter_new (1.0 / self->sample_rate, sqrt(2 * o), o * o);
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/* Create FIFO buffers */
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self->filled_pkts = av_fifo_alloc(FIFO_PACKETS_NUM * sizeof(AVPacket));
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/* New packets FIFO with one extra packet for safety against underruns */
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self->new_pkts = av_fifo_alloc((FIFO_PACKETS_NUM + 1) * sizeof(AVPacket));
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if ((test = supply_new_packets(self, context))) {
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jack_client_close(self->client);
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return test;
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}
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return 0;
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}
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static void free_pkt_fifo(AVFifoBuffer *fifo)
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{
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AVPacket pkt;
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while (av_fifo_size(fifo)) {
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av_fifo_generic_read(fifo, &pkt, sizeof(pkt), NULL);
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av_free_packet(&pkt);
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}
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av_fifo_free(fifo);
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}
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static void stop_jack(JackData *self)
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{
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if (self->client) {
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if (self->activated)
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jack_deactivate(self->client);
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jack_client_close(self->client);
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}
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sem_destroy(&self->packet_count);
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free_pkt_fifo(self->new_pkts);
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free_pkt_fifo(self->filled_pkts);
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av_freep(&self->ports);
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ff_timefilter_destroy(self->timefilter);
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}
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static int audio_read_header(AVFormatContext *context, AVFormatParameters *params)
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{
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JackData *self = context->priv_data;
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AVStream *stream;
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int test;
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if (params->sample_rate <= 0 || params->channels <= 0)
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return -1;
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if ((test = start_jack(context, params)))
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return test;
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stream = av_new_stream(context, 0);
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if (!stream) {
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stop_jack(self);
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return AVERROR(ENOMEM);
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}
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stream->codec->codec_type = CODEC_TYPE_AUDIO;
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#ifdef WORDS_BIGENDIAN
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stream->codec->codec_id = CODEC_ID_PCM_F32BE;
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#else
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stream->codec->codec_id = CODEC_ID_PCM_F32LE;
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#endif
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stream->codec->sample_rate = self->sample_rate;
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stream->codec->channels = self->nports;
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av_set_pts_info(stream, 64, 1, 1000000); /* 64 bits pts in us */
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return 0;
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}
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static int audio_read_packet(AVFormatContext *context, AVPacket *pkt)
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{
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JackData *self = context->priv_data;
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struct timespec timeout = {0, 0};
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int test;
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/* Activate the JACK client on first packet read. Activating the JACK client
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* means that process_callback() starts to get called at regular interval.
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* If we activate it in audio_read_header(), we're actually reading audio data
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* from the device before instructed to, and that may result in an overrun. */
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if (!self->activated) {
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if (!jack_activate(self->client)) {
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self->activated = 1;
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av_log(context, AV_LOG_INFO,
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"JACK client registered and activated (rate=%dHz, buffer_size=%d frames)\n",
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self->sample_rate, self->buffer_size);
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} else {
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av_log(context, AV_LOG_ERROR, "Unable to activate JACK client\n");
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return AVERROR(EIO);
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}
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}
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/* Wait for a packet comming back from process_callback(), if one isn't available yet */
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timeout.tv_sec = av_gettime() / 1000000 + 2;
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if (sem_timedwait(&self->packet_count, &timeout)) {
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if (errno == ETIMEDOUT) {
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av_log(context, AV_LOG_ERROR,
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"Input error: timed out when waiting for JACK process callback output\n");
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} else {
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av_log(context, AV_LOG_ERROR, "Error while waiting for audio packet: %s\n",
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strerror(errno));
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}
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if (!self->client)
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av_log(context, AV_LOG_ERROR, "Input error: JACK server is gone\n");
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return AVERROR(EIO);
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}
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if (self->pkt_xrun) {
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av_log(context, AV_LOG_WARNING, "Audio packet xrun\n");
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self->pkt_xrun = 0;
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}
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if (self->jack_xrun) {
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av_log(context, AV_LOG_WARNING, "JACK xrun\n");
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self->jack_xrun = 0;
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}
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/* Retrieve the packet filled with audio data by process_callback() */
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av_fifo_generic_read(self->filled_pkts, pkt, sizeof(*pkt), NULL);
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if ((test = supply_new_packets(self, context)))
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return test;
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return 0;
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}
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static int audio_read_close(AVFormatContext *context)
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{
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JackData *self = context->priv_data;
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stop_jack(self);
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return 0;
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}
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AVInputFormat jack_demuxer = {
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"jack",
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NULL_IF_CONFIG_SMALL("JACK Audio Connection Kit"),
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sizeof(JackData),
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NULL,
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audio_read_header,
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audio_read_packet,
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audio_read_close,
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.flags = AVFMT_NOFILE,
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};
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@ -243,6 +243,9 @@ OBJS-$(CONFIG_RTP_PROTOCOL) += rtpproto.o
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OBJS-$(CONFIG_TCP_PROTOCOL) += tcp.o
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OBJS-$(CONFIG_UDP_PROTOCOL) += udp.o
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# libavdevice dependencies
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OBJS-$(CONFIG_JACK_DEMUXER) += timefilter.o
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EXAMPLES = output
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TESTPROGS = timefilter
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