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/*
* Copyright (c) 2009 Samalyse
* Copyright (c) 2016 Josh de Kock
* This file is part of FFmpeg.
*
* FFmpeg 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.
*
* FFmpeg 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <semaphore.h>
#include <jack/jack.h>
#include "libavutil/internal.h"
#include "libavutil/log.h"
#include "libavutil/fifo.h"
#include "libavutil/opt.h"
#include "libavutil/time.h"
#include "libavcodec/avcodec.h"
#include "libavformat/avformat.h"
#include "libavformat/internal.h"
#include "timefilter.h"
#include "avdevice.h"
#include "jack.h"
int ff_start_jack(AVFormatContext *context)
{
JackData *self = context->priv_data;
jack_status_t status;
int i, test;
/* 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);
if(!self->sample_rate)
self->sample_rate = jack_get_sample_rate(self->client);
if(!self->ports)
self->ports = av_malloc_array(self->nports, sizeof(*self->ports));
if (!self->ports)
return AVERROR(ENOMEM);
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,
self->port_type, 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_on_shutdown(self->client, ff_shutdown_callback, self);
jack_set_xrun_callback(self->client, ff_xrun_callback, self);
/* Create time filter */
self->timefilter = ff_timefilter_new (1.0 / self->sample_rate, self->buffer_size, 1.5);
if (!self->timefilter) {
jack_client_close(self->client);
return AVERROR(ENOMEM);
}
/* Create FIFO buffers */
self->filled_pkts = av_fifo_alloc_array(FIFO_PACKETS_NUM, sizeof(AVPacket));
/* New packets FIFO with one extra packet for safety against underruns */
self->new_pkts = av_fifo_alloc_array((FIFO_PACKETS_NUM + 1), sizeof(AVPacket));
if (!self->new_pkts) {
jack_client_close(self->client);
return AVERROR(ENOMEM);
}
return 0;
}
void ff_shutdown_callback(void *arg)
{
JackData *self = arg;
self->client = NULL;
}
int ff_xrun_callback(void *arg)
{
JackData *self = arg;
self->jack_xrun = 1;
ff_timefilter_reset(self->timefilter);
return 0;
}
void ff_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);
}
void free_pkt_fifo(AVFifoBuffer **fifo)
{
AVPacket pkt;
while (av_fifo_size(*fifo)) {
av_fifo_generic_read(*fifo, &pkt, sizeof(pkt), NULL);
av_packet_unref(&pkt);
}
av_fifo_freep(fifo);
}
/*
* Copyright (c) 2009 Samalyse
* Copyright (c) 2016 Josh de Kock
* This file is part of FFmpeg.
*
* FFmpeg 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.
*
* FFmpeg 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVDEVICE_JACK_H
#if HAVE_DISPATCH_DISPATCH_H
#include <dispatch/dispatch.h>
#define sem_t dispatch_semaphore_t
#define sem_init(psem,x,val) *psem = dispatch_semaphore_create(val)
#define sem_post(psem) dispatch_semaphore_signal(*psem)
#define sem_wait(psem) dispatch_semaphore_wait(*psem, DISPATCH_TIME_FOREVER)
#define sem_timedwait(psem, val) dispatch_semaphore_wait(*psem, dispatch_walltime(val, 0))
#define sem_destroy(psem) dispatch_release(*psem)
#endif
/**
* Size of the internal FIFO buffers as a number of audio packets
*/
#define FIFO_PACKETS_NUM 16
typedef struct JackData {
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;
int port_type; ///< JACK port type, either JackPortIsInput or JackPortIsOutput
TimeFilter * timefilter;
AVFifoBuffer * new_pkts;
AVFifoBuffer * filled_pkts;
int pkt_xrun;
int jack_xrun;
} JackData;
int ff_start_jack(AVFormatContext *context);
void ff_stop_jack(JackData *self);
void ff_shutdown_callback(void *arg);
void free_pkt_fifo(AVFifoBuffer **fifo);
int ff_xrun_callback(void *arg);
#endif
/*
* JACK Audio Connection Kit input device
* Copyright (c) 2009 Samalyse
* Copyright (c) 2016 Josh de Kock
* Author: Olivier Guilyardi <olivier samalyse com>
*
* This file is part of FFmpeg.
*
* FFmpeg 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.
*
* FFmpeg 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* JACK input and output: input
*
*/
#include "config.h"
#include <semaphore.h>
#include <jack/jack.h>
#include "libavutil/internal.h"
#include "libavutil/log.h"
#include "libavutil/fifo.h"
#include "libavutil/opt.h"
#include "libavutil/time.h"
#include "libavcodec/avcodec.h"
#include "libavformat/avformat.h"
#include "libavformat/internal.h"
#include "timefilter.h"
#include "avdevice.h"
#include "jack.h"
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++) {
#if HAVE_JACK_PORT_GET_LATENCY_RANGE
jack_latency_range_t range;
jack_port_get_latency_range(self->ports[i], JackCaptureLatency, &range);
latency += range.max;
#else
latency += jack_port_get_total_latency(self->client, self->ports[i]);
#endif
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 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 audio_read_header(AVFormatContext *context)
{
JackData *self = context->priv_data;
AVStream *stream;
int test;
self->port_type = JackPortIsInput;
if ((test = ff_start_jack(context)))
return test;
jack_set_process_callback(self->client, process_callback, self);
if ((test = supply_new_packets(self, context))) {
jack_client_close(self->client);
return test;
}
stream = avformat_new_stream(context, NULL);
if (!stream) {
ff_stop_jack(self);
return AVERROR(ENOMEM);
}
stream->codec->codec_type = AVMEDIA_TYPE_AUDIO;
#if HAVE_BIGENDIAN
stream->codec->codec_id = AV_CODEC_ID_PCM_F32BE;
#else
stream->codec->codec_id = AV_CODEC_ID_PCM_F32LE;
#endif
stream->codec->sample_rate = self->sample_rate;
stream->codec->channels = self->nports;
avpriv_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 coming 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 {
char errbuf[128];
int ret = AVERROR(errno);
av_strerror(ret, errbuf, sizeof(errbuf));
av_log(context, AV_LOG_ERROR, "Error while waiting for audio packet: %s\n",
errbuf);
}
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;
ff_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, { .i64 = 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,
.category = AV_CLASS_CATEGORY_DEVICE_AUDIO_INPUT,
};
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,
};
/*
* JACK Audio Connection Kit input device
* Copyright (c) 2009 Samalyse
* Copyright (c) 2016 Josh de Kock
* Author: Olivier Guilyardi <olivier samalyse com>
*
* This file is part of FFmpeg.
*
* FFmpeg 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.
*
* FFmpeg 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* JACK input and output: input
*
*/
#include "config.h"
#include <semaphore.h>
#include <jack/jack.h>
#include "libavutil/internal.h"
#include "libavutil/log.h"
#include "libavutil/fifo.h"
#include "libavutil/opt.h"
#include "libavutil/time.h"
#include "libavcodec/avcodec.h"
#include "libavformat/avformat.h"
#include "libavformat/internal.h"
#include "timefilter.h"
#include "avdevice.h"
#include "jack.h"
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;
jack_nframes_t latency;
AVPacket pkt;
float *pkt_data;
if (!self->client)
return 0;
/* Retrieve filed packet */
av_fifo_generic_read(self->filled_pkts, &pkt, sizeof(pkt), NULL);
pkt_data = (float *) pkt.data;
latency = 0;
/* Copy and interleave audio data from the the packet buffer into JACK */
for (i = 0; i < self->nports; i++) {
float * buffer;
#if HAVE_JACK_PORT_GET_LATENCY_RANGE
jack_latency_range_t range;
jack_port_get_latency_range(self->ports[i], JackCaptureLatency, &range);
latency += range.max;
#else
latency += jack_port_get_total_latency(self->client, self->ports[i]);
#endif
buffer = jack_port_get_buffer(self->ports[i], self->buffer_size);
for (j = 0; j < self->buffer_size; j++)
buffer[j] = pkt_data[j * self->nports + i];
}
/* 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 int audio_write_header(AVFormatContext *context)
{
av_log(context, AV_LOG_INFO, "JACK audio_write_header called");
JackData *self = context->priv_data;
if (context->nb_streams != 1 || context->streams[0]->codec->codec_type != AVMEDIA_TYPE_AUDIO) {
av_log(context, AV_LOG_ERROR, "Only a single audio stream is supported.\n");
return AVERROR(EINVAL);
}
AVStream *stream = context->streams[0];
self->port_type = JackPortIsOutput;
int test;
if ((test = ff_start_jack(context)))
return test;
jack_set_process_callback(self->client, process_callback, self);
self->sample_rate = stream->codec->sample_rate;
self->nports = stream->codec->channels;
return 0;
}
static int audio_write_packet(AVFormatContext *context, AVPacket *pkt)
{
av_log(context, AV_LOG_INFO, "JACK audio_write_packet called");
JackData *self = context->priv_data;
// struct timespec timeout = {0, 0};
/* Activate the JACK client on first packet write. 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 writing 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 coming 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 {
// char errbuf[128];
// int ret = AVERROR(errno);
// av_strerror(ret, errbuf, sizeof(errbuf));
// av_log(context, AV_LOG_ERROR, "Error while waiting for audio packet: %s\n",
// errbuf);
// }
// 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;
}
/* Write the filled packet filled with audio data by process_callback() */
av_fifo_generic_write(self->filled_pkts, pkt, sizeof(*pkt), NULL);
return 0;
}
static int audio_write_close(AVFormatContext *context)
{
av_log(context, AV_LOG_INFO, "JACK audio_write_close called");
JackData *self = context->priv_data;
ff_stop_jack(self);
return 0;
}
static int audio_write_frame(AVFormatContext *context, int stream_index,
AVFrame **frame, unsigned flags)
{
av_log(context, AV_LOG_INFO, "JACK audio_write_frame called");
JackData *self = context->priv_data;
AVPacket pkt;
/* ff_alsa_open() should have accepted only supported formats */
if ((flags & AV_WRITE_UNCODED_FRAME_QUERY))
return av_sample_fmt_is_planar(context->streams[stream_index]->codec->sample_fmt) ?
AVERROR(EINVAL) : 0;
/* set only used fields */
pkt.data = (*frame)->data[0];
pkt.size = (*frame)->nb_samples * self->buffer_size;
pkt.dts = (*frame)->pkt_dts;
pkt.duration = av_frame_get_pkt_duration(*frame);
return audio_write_packet(context, &pkt);
}
#define OFFSET(x) offsetof(JackData, x)
static const AVClass jack_outdev_class = {
.class_name = "JACK outdev",
.item_name = av_default_item_name,
.version = LIBAVUTIL_VERSION_INT,
.category = AV_CLASS_CATEGORY_DEVICE_AUDIO_OUTPUT,
};
AVOutputFormat ff_jack_muxer = {
.name = "jack",
.long_name = NULL_IF_CONFIG_SMALL("JACK Audio Connection Kit"),
.priv_data_size = sizeof(JackData),
.video_codec = AV_CODEC_ID_NONE,
.write_header = audio_write_header,
.write_packet = audio_write_packet,
.write_uncoded_frame = audio_write_frame,
.write_trailer = audio_write_close,
.flags = AVFMT_NOFILE,
.priv_class = &jack_outdev_class,
};
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