/* * 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, };