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e37f161e66
* qatar/master: (71 commits) movenc: Allow writing to a non-seekable output if using empty moov movenc: Support adding isml (smooth streaming live) metadata libavcodec: Don't crash in avcodec_encode_audio if time_base isn't set sunrast: Document the different Sun Raster file format types. sunrast: Add a check for experimental type. libspeexenc: use AVSampleFormat instead of deprecated/removed SampleFormat lavf: remove disabled FF_API_SET_PTS_INFO cruft lavf: remove disabled FF_API_OLD_INTERRUPT_CB cruft lavf: remove disabled FF_API_REORDER_PRIVATE cruft lavf: remove disabled FF_API_SEEK_PUBLIC cruft lavf: remove disabled FF_API_STREAM_COPY cruft lavf: remove disabled FF_API_PRELOAD cruft lavf: remove disabled FF_API_NEW_STREAM cruft lavf: remove disabled FF_API_RTSP_URL_OPTIONS cruft lavf: remove disabled FF_API_MUXRATE cruft lavf: remove disabled FF_API_FILESIZE cruft lavf: remove disabled FF_API_TIMESTAMP cruft lavf: remove disabled FF_API_LOOP_OUTPUT cruft lavf: remove disabled FF_API_LOOP_INPUT cruft lavf: remove disabled FF_API_AVSTREAM_QUALITY cruft ... Conflicts: doc/APIchanges libavcodec/8bps.c libavcodec/avcodec.h libavcodec/libx264.c libavcodec/mjpegbdec.c libavcodec/options.c libavcodec/sunrast.c libavcodec/utils.c libavcodec/version.h libavcodec/x86/h264_deblock.asm libavdevice/libdc1394.c libavdevice/v4l2.c libavformat/avformat.h libavformat/avio.c libavformat/avio.h libavformat/aviobuf.c libavformat/dv.c libavformat/mov.c libavformat/utils.c libavformat/version.h libavformat/wtv.c libavutil/Makefile libavutil/file.c libswscale/x86/input.asm libswscale/x86/swscale_mmx.c libswscale/x86/swscale_template.c tests/ref/lavf/ffm Merged-by: Michael Niedermayer <michaelni@gmx.at>
340 lines
11 KiB
C
340 lines
11 KiB
C
/*
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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 "libavutil/opt.h"
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#include "libavcodec/avcodec.h"
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#include "libavformat/avformat.h"
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#include "libavformat/internal.h"
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#include "timefilter.h"
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#include "avdevice.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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AVClass *class;
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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)
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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, JackNullOption, &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->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)
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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 ((test = start_jack(context)))
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return test;
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stream = avformat_new_stream(context, NULL);
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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 = AVMEDIA_TYPE_AUDIO;
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#if HAVE_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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avpriv_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 coming 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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#define OFFSET(x) offsetof(JackData, x)
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static const AVOption options[] = {
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{ "channels", "Number of audio channels.", OFFSET(nports), AV_OPT_TYPE_INT, { 2 }, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
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{ NULL },
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};
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static const AVClass jack_indev_class = {
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.class_name = "JACK indev",
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.item_name = av_default_item_name,
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.option = options,
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.version = LIBAVUTIL_VERSION_INT,
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};
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AVInputFormat ff_jack_demuxer = {
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.name = "jack",
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.long_name = NULL_IF_CONFIG_SMALL("JACK Audio Connection Kit"),
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.priv_data_size = sizeof(JackData),
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.read_header = audio_read_header,
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.read_packet = audio_read_packet,
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.read_close = audio_read_close,
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.flags = AVFMT_NOFILE,
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.priv_class = &jack_indev_class,
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};
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