diff --git a/configure b/configure index 0f32fbdd10..a9cc2fd1e0 100755 --- a/configure +++ b/configure @@ -1112,6 +1112,8 @@ avisynth_demuxer_deps="avisynth" bktr_demuxer_deps_any="dev_bktr_ioctl_bt848_h machine_ioctl_bt848_h dev_video_bktr_ioctl_bt848_h dev_ic_bt8xx_h" dirac_demuxer_deps="dirac_parser" dv1394_demuxer_deps="dv1394 dv_demuxer" +jack_demuxer_deps="jack_jack_h" +jack_demuxer_extralibs="-ljack" libdc1394_demuxer_deps="libdc1394" libnut_demuxer_deps="libnut" libnut_muxer_deps="libnut" @@ -2149,6 +2151,8 @@ check_header soundcard.h check_lib2 alsa/asoundlib.h snd_pcm_htimestamp -lasound +check_lib2 jack/jack.h jack_client_open -ljack + # deal with the X11 frame grabber enabled x11grab && check_header X11/Xlib.h && diff --git a/doc/general.texi b/doc/general.texi index a19c3b180b..9ec20ae434 100644 --- a/doc/general.texi +++ b/doc/general.texi @@ -653,6 +653,7 @@ performance on systems without hardware floating point support). @item BEOS audio @tab X @tab X @item BKTR @tab X @tab @item DV1394 @tab X @tab +@item JACK @tab X @tab @item LIBDC1394 @tab X @tab @item OSS @tab X @tab X @item Video4Linux @tab X @tab diff --git a/libavdevice/Makefile b/libavdevice/Makefile index 8bd706b03f..b769d16c76 100644 --- a/libavdevice/Makefile +++ b/libavdevice/Makefile @@ -12,6 +12,7 @@ OBJS-$(CONFIG_ALSA_DEMUXER) += alsa-audio-common.o alsa-audio-dec.o OBJS-$(CONFIG_ALSA_MUXER) += alsa-audio-common.o alsa-audio-enc.o OBJS-$(CONFIG_BKTR_DEMUXER) += bktr.o OBJS-$(CONFIG_DV1394_DEMUXER) += dv1394.o +OBJS-$(CONFIG_JACK_DEMUXER) += jack_audio.o OBJS-$(CONFIG_OSS_DEMUXER) += oss_audio.o OBJS-$(CONFIG_OSS_MUXER) += oss_audio.o OBJS-$(CONFIG_V4L2_DEMUXER) += v4l2.o diff --git a/libavdevice/alldevices.c b/libavdevice/alldevices.c index 38ce6f1eca..7efc42a897 100644 --- a/libavdevice/alldevices.c +++ b/libavdevice/alldevices.c @@ -48,6 +48,7 @@ void avdevice_register_all(void) REGISTER_MUXDEMUX (AUDIO_BEOS, audio_beos); REGISTER_DEMUXER (BKTR, bktr); REGISTER_DEMUXER (DV1394, dv1394); + REGISTER_DEMUXER (JACK, jack); REGISTER_MUXDEMUX (OSS, oss); REGISTER_DEMUXER (V4L2, v4l2); REGISTER_DEMUXER (V4L, v4l); diff --git a/libavdevice/avdevice.h b/libavdevice/avdevice.h index 7e6d461146..61400c111b 100644 --- a/libavdevice/avdevice.h +++ b/libavdevice/avdevice.h @@ -20,7 +20,7 @@ #define AVDEVICE_AVDEVICE_H #define LIBAVDEVICE_VERSION_MAJOR 52 -#define LIBAVDEVICE_VERSION_MINOR 1 +#define LIBAVDEVICE_VERSION_MINOR 2 #define LIBAVDEVICE_VERSION_MICRO 0 #define LIBAVDEVICE_VERSION_INT AV_VERSION_INT(LIBAVDEVICE_VERSION_MAJOR, \ diff --git a/libavdevice/jack_audio.c b/libavdevice/jack_audio.c new file mode 100644 index 0000000000..8232a187e2 --- /dev/null +++ b/libavdevice/jack_audio.c @@ -0,0 +1,326 @@ +/* + * JACK Audio Connection Kit input device + * Copyright (c) 2009 Samalyse + * Author: Olivier Guilyardi + * + * 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 "config.h" +#include +#include + +#include "libavutil/log.h" +#include "libavutil/fifo.h" +#include "libavcodec/avcodec.h" +#include "libavformat/avformat.h" +#include "libavformat/timefilter.h" + +/** + * Size of the internal FIFO buffers as a number of audio packets + */ +#define FIFO_PACKETS_NUM 16 + +typedef struct { + 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, AVFormatParameters *params) +{ + 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, 0, &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->nports = params->channels; + 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 (params->sample_rate <= 0 || params->channels <= 0) + return -1; + + if ((test = start_jack(context, params))) + return test; + + stream = av_new_stream(context, 0); + if (!stream) { + stop_jack(self); + return AVERROR(ENOMEM); + } + + stream->codec->codec_type = CODEC_TYPE_AUDIO; +#ifdef WORDS_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; +} + +AVInputFormat jack_demuxer = { + "jack", + NULL_IF_CONFIG_SMALL("JACK Audio Connection Kit"), + sizeof(JackData), + NULL, + audio_read_header, + audio_read_packet, + audio_read_close, + .flags = AVFMT_NOFILE, +}; diff --git a/libavformat/Makefile b/libavformat/Makefile index aad67cc056..1bdc976bc1 100644 --- a/libavformat/Makefile +++ b/libavformat/Makefile @@ -243,6 +243,9 @@ OBJS-$(CONFIG_RTP_PROTOCOL) += rtpproto.o OBJS-$(CONFIG_TCP_PROTOCOL) += tcp.o OBJS-$(CONFIG_UDP_PROTOCOL) += udp.o +# libavdevice dependencies +OBJS-$(CONFIG_JACK_DEMUXER) += timefilter.o + EXAMPLES = output TESTPROGS = timefilter