ffmpeg/libavdevice/jack_audio.c

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/*
* JACK Audio Connection Kit input device
* Copyright (c) 2009 Samalyse
* Author: Olivier Guilyardi <olivier samalyse com>
*
* This file is part of Libav.
*
* Libav is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* Libav 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "config.h"
#include <semaphore.h>
#include <jack/jack.h>
#include "libavutil/log.h"
#include "libavutil/fifo.h"
#include "libavutil/opt.h"
#include "libavcodec/avcodec.h"
#include "libavformat/avformat.h"
#include "timefilter.h"
/**
* Size of the internal FIFO buffers as a number of audio packets
*/
#define FIFO_PACKETS_NUM 16
typedef struct {
AVClass *class;
jack_client_t * client;
int activated;
sem_t packet_count;
jack_nframes_t sample_rate;
jack_nframes_t buffer_size;
jack_port_t ** ports;
int nports;
TimeFilter * timefilter;
AVFifoBuffer * new_pkts;
AVFifoBuffer * filled_pkts;
int pkt_xrun;
int jack_xrun;
} JackData;
static int process_callback(jack_nframes_t nframes, void *arg)
{
/* Warning: this function runs in realtime. One mustn't allocate memory here
* or do any other thing that could block. */
int i, j;
JackData *self = arg;
float * buffer;
jack_nframes_t latency, cycle_delay;
AVPacket pkt;
float *pkt_data;
double cycle_time;
if (!self->client)
return 0;
/* The approximate delay since the hardware interrupt as a number of frames */
cycle_delay = jack_frames_since_cycle_start(self->client);
/* Retrieve filtered cycle time */
cycle_time = ff_timefilter_update(self->timefilter,
av_gettime() / 1000000.0 - (double) cycle_delay / self->sample_rate,
self->buffer_size);
/* Check if an empty packet is available, and if there's enough space to send it back once filled */
if ((av_fifo_size(self->new_pkts) < sizeof(pkt)) || (av_fifo_space(self->filled_pkts) < sizeof(pkt))) {
self->pkt_xrun = 1;
return 0;
}
/* Retrieve empty (but allocated) packet */
av_fifo_generic_read(self->new_pkts, &pkt, sizeof(pkt), NULL);
pkt_data = (float *) pkt.data;
latency = 0;
/* Copy and interleave audio data from the JACK buffer into the packet */
for (i = 0; i < self->nports; i++) {
latency += jack_port_get_total_latency(self->client, self->ports[i]);
buffer = jack_port_get_buffer(self->ports[i], self->buffer_size);
for (j = 0; j < self->buffer_size; j++)
pkt_data[j * self->nports + i] = buffer[j];
}
/* Timestamp the packet with the cycle start time minus the average latency */
pkt.pts = (cycle_time - (double) latency / (self->nports * self->sample_rate)) * 1000000.0;
/* Send the now filled packet back, and increase packet counter */
av_fifo_generic_write(self->filled_pkts, &pkt, sizeof(pkt), NULL);
sem_post(&self->packet_count);
return 0;
}
static void shutdown_callback(void *arg)
{
JackData *self = arg;
self->client = NULL;
}
static int xrun_callback(void *arg)
{
JackData *self = arg;
self->jack_xrun = 1;
ff_timefilter_reset(self->timefilter);
return 0;
}
static int supply_new_packets(JackData *self, AVFormatContext *context)
{
AVPacket pkt;
int test, pkt_size = self->buffer_size * self->nports * sizeof(float);
/* Supply the process callback with new empty packets, by filling the new
* packets FIFO buffer with as many packets as possible. process_callback()
* can't do this by itself, because it can't allocate memory in realtime. */
while (av_fifo_space(self->new_pkts) >= sizeof(pkt)) {
if ((test = av_new_packet(&pkt, pkt_size)) < 0) {
av_log(context, AV_LOG_ERROR, "Could not create packet of size %d\n", pkt_size);
return test;
}
av_fifo_generic_write(self->new_pkts, &pkt, sizeof(pkt), NULL);
}
return 0;
}
static int start_jack(AVFormatContext *context)
{
JackData *self = context->priv_data;
jack_status_t status;
int i, test;
double o, period;
/* Register as a JACK client, using the context filename as client name. */
self->client = jack_client_open(context->filename, JackNullOption, &status);
if (!self->client) {
av_log(context, AV_LOG_ERROR, "Unable to register as a JACK client\n");
return AVERROR(EIO);
}
sem_init(&self->packet_count, 0, 0);
self->sample_rate = jack_get_sample_rate(self->client);
self->ports = av_malloc(self->nports * sizeof(*self->ports));
self->buffer_size = jack_get_buffer_size(self->client);
/* Register JACK ports */
for (i = 0; i < self->nports; i++) {
char str[16];
snprintf(str, sizeof(str), "input_%d", i + 1);
self->ports[i] = jack_port_register(self->client, str,
JACK_DEFAULT_AUDIO_TYPE,
JackPortIsInput, 0);
if (!self->ports[i]) {
av_log(context, AV_LOG_ERROR, "Unable to register port %s:%s\n",
context->filename, str);
jack_client_close(self->client);
return AVERROR(EIO);
}
}
/* Register JACK callbacks */
jack_set_process_callback(self->client, process_callback, self);
jack_on_shutdown(self->client, shutdown_callback, self);
jack_set_xrun_callback(self->client, xrun_callback, self);
/* Create time filter */
period = (double) self->buffer_size / self->sample_rate;
o = 2 * M_PI * 1.5 * period; /// bandwidth: 1.5Hz
self->timefilter = ff_timefilter_new (1.0 / self->sample_rate, sqrt(2 * o), o * o);
/* Create FIFO buffers */
self->filled_pkts = av_fifo_alloc(FIFO_PACKETS_NUM * sizeof(AVPacket));
/* New packets FIFO with one extra packet for safety against underruns */
self->new_pkts = av_fifo_alloc((FIFO_PACKETS_NUM + 1) * sizeof(AVPacket));
if ((test = supply_new_packets(self, context))) {
jack_client_close(self->client);
return test;
}
return 0;
}
static void free_pkt_fifo(AVFifoBuffer *fifo)
{
AVPacket pkt;
while (av_fifo_size(fifo)) {
av_fifo_generic_read(fifo, &pkt, sizeof(pkt), NULL);
av_free_packet(&pkt);
}
av_fifo_free(fifo);
}
static void stop_jack(JackData *self)
{
if (self->client) {
if (self->activated)
jack_deactivate(self->client);
jack_client_close(self->client);
}
sem_destroy(&self->packet_count);
free_pkt_fifo(self->new_pkts);
free_pkt_fifo(self->filled_pkts);
av_freep(&self->ports);
ff_timefilter_destroy(self->timefilter);
}
static int audio_read_header(AVFormatContext *context, AVFormatParameters *params)
{
JackData *self = context->priv_data;
AVStream *stream;
int test;
if ((test = start_jack(context)))
return test;
stream = avformat_new_stream(context, NULL);
if (!stream) {
stop_jack(self);
return AVERROR(ENOMEM);
}
stream->codec->codec_type = AVMEDIA_TYPE_AUDIO;
#if HAVE_BIGENDIAN
stream->codec->codec_id = CODEC_ID_PCM_F32BE;
#else
stream->codec->codec_id = CODEC_ID_PCM_F32LE;
#endif
stream->codec->sample_rate = self->sample_rate;
stream->codec->channels = self->nports;
av_set_pts_info(stream, 64, 1, 1000000); /* 64 bits pts in us */
return 0;
}
static int audio_read_packet(AVFormatContext *context, AVPacket *pkt)
{
JackData *self = context->priv_data;
struct timespec timeout = {0, 0};
int test;
/* Activate the JACK client on first packet read. Activating the JACK client
* means that process_callback() starts to get called at regular interval.
* If we activate it in audio_read_header(), we're actually reading audio data
* from the device before instructed to, and that may result in an overrun. */
if (!self->activated) {
if (!jack_activate(self->client)) {
self->activated = 1;
av_log(context, AV_LOG_INFO,
"JACK client registered and activated (rate=%dHz, buffer_size=%d frames)\n",
self->sample_rate, self->buffer_size);
} else {
av_log(context, AV_LOG_ERROR, "Unable to activate JACK client\n");
return AVERROR(EIO);
}
}
/* Wait for a packet comming back from process_callback(), if one isn't available yet */
timeout.tv_sec = av_gettime() / 1000000 + 2;
if (sem_timedwait(&self->packet_count, &timeout)) {
if (errno == ETIMEDOUT) {
av_log(context, AV_LOG_ERROR,
"Input error: timed out when waiting for JACK process callback output\n");
} else {
av_log(context, AV_LOG_ERROR, "Error while waiting for audio packet: %s\n",
strerror(errno));
}
if (!self->client)
av_log(context, AV_LOG_ERROR, "Input error: JACK server is gone\n");
return AVERROR(EIO);
}
if (self->pkt_xrun) {
av_log(context, AV_LOG_WARNING, "Audio packet xrun\n");
self->pkt_xrun = 0;
}
if (self->jack_xrun) {
av_log(context, AV_LOG_WARNING, "JACK xrun\n");
self->jack_xrun = 0;
}
/* Retrieve the packet filled with audio data by process_callback() */
av_fifo_generic_read(self->filled_pkts, pkt, sizeof(*pkt), NULL);
if ((test = supply_new_packets(self, context)))
return test;
return 0;
}
static int audio_read_close(AVFormatContext *context)
{
JackData *self = context->priv_data;
stop_jack(self);
return 0;
}
#define OFFSET(x) offsetof(JackData, x)
static const AVOption options[] = {
{ "channels", "Number of audio channels.", OFFSET(nports), AV_OPT_TYPE_INT, { 2 }, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
{ NULL },
};
static const AVClass jack_indev_class = {
.class_name = "JACK indev",
.item_name = av_default_item_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
};
AVInputFormat ff_jack_demuxer = {
.name = "jack",
.long_name = NULL_IF_CONFIG_SMALL("JACK Audio Connection Kit"),
.priv_data_size = sizeof(JackData),
.read_header = audio_read_header,
.read_packet = audio_read_packet,
.read_close = audio_read_close,
.flags = AVFMT_NOFILE,
.priv_class = &jack_indev_class,
};