diff options
Diffstat (limited to 'libavdevice/pulse_audio_dec.c')
| -rw-r--r-- | libavdevice/pulse_audio_dec.c | 180 |
1 files changed, 180 insertions, 0 deletions
diff --git a/libavdevice/pulse_audio_dec.c b/libavdevice/pulse_audio_dec.c new file mode 100644 index 0000000000..3b5cebb7f6 --- /dev/null +++ b/libavdevice/pulse_audio_dec.c @@ -0,0 +1,180 @@ +/* + * Pulseaudio input + * Copyright (c) 2011 Luca Barbato <lu_zero@gentoo.org> + * + * 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 + * PulseAudio input using the simple API. + * @author Luca Barbato <lu_zero@gentoo.org> + */ + +#include <pulse/simple.h> +#include <pulse/rtclock.h> +#include <pulse/error.h> +#include "libavformat/avformat.h" +#include "libavformat/internal.h" +#include "libavutil/opt.h" +#include "pulse_audio_common.h" + +#define DEFAULT_CODEC_ID AV_NE(AV_CODEC_ID_PCM_S16BE, AV_CODEC_ID_PCM_S16LE) + +typedef struct PulseData { + AVClass *class; + char *server; + char *name; + char *stream_name; + int sample_rate; + int channels; + int frame_size; + int fragment_size; + pa_simple *s; + int64_t pts; + int64_t frame_duration; +} PulseData; + +static av_cold int pulse_read_header(AVFormatContext *s) +{ + PulseData *pd = s->priv_data; + AVStream *st; + char *device = NULL; + int ret, sample_bytes; + enum AVCodecID codec_id = + s->audio_codec_id == AV_CODEC_ID_NONE ? DEFAULT_CODEC_ID : s->audio_codec_id; + const pa_sample_spec ss = { ff_codec_id_to_pulse_format(codec_id), + pd->sample_rate, + pd->channels }; + + pa_buffer_attr attr = { -1 }; + + st = avformat_new_stream(s, NULL); + + if (!st) { + av_log(s, AV_LOG_ERROR, "Cannot add stream\n"); + return AVERROR(ENOMEM); + } + + attr.fragsize = pd->fragment_size; + + if (strcmp(s->filename, "default")) + device = s->filename; + + pd->s = pa_simple_new(pd->server, pd->name, + PA_STREAM_RECORD, + device, pd->stream_name, &ss, + NULL, &attr, &ret); + + if (!pd->s) { + av_log(s, AV_LOG_ERROR, "pa_simple_new failed: %s\n", + pa_strerror(ret)); + return AVERROR(EIO); + } + /* take real parameters */ + st->codec->codec_type = AVMEDIA_TYPE_AUDIO; + st->codec->codec_id = codec_id; + st->codec->sample_rate = pd->sample_rate; + st->codec->channels = pd->channels; + avpriv_set_pts_info(st, 64, 1, pd->sample_rate); /* 64 bits pts in us */ + + pd->pts = AV_NOPTS_VALUE; + sample_bytes = (av_get_bits_per_sample(codec_id) >> 3) * pd->channels; + + if (pd->frame_size % sample_bytes) { + av_log(s, AV_LOG_WARNING, "frame_size %i is not divisible by %i " + "(channels * bytes_per_sample) \n", pd->frame_size, sample_bytes); + } + + pd->frame_duration = pd->frame_size / sample_bytes; + + return 0; +} + +static int pulse_read_packet(AVFormatContext *s, AVPacket *pkt) +{ + PulseData *pd = s->priv_data; + int res; + + if (av_new_packet(pkt, pd->frame_size) < 0) { + return AVERROR(ENOMEM); + } + + if ((pa_simple_read(pd->s, pkt->data, pkt->size, &res)) < 0) { + av_log(s, AV_LOG_ERROR, "pa_simple_read failed: %s\n", + pa_strerror(res)); + av_free_packet(pkt); + return AVERROR(EIO); + } + + if (pd->pts == AV_NOPTS_VALUE) { + pa_usec_t latency; + + if ((latency = pa_simple_get_latency(pd->s, &res)) == (pa_usec_t) -1) { + av_log(s, AV_LOG_ERROR, "pa_simple_get_latency() failed: %s\n", + pa_strerror(res)); + return AVERROR(EIO); + } + + pd->pts = -latency; + } + + pkt->pts = pd->pts; + + pd->pts += pd->frame_duration; + + return 0; +} + +static av_cold int pulse_close(AVFormatContext *s) +{ + PulseData *pd = s->priv_data; + pa_simple_free(pd->s); + return 0; +} + +#define OFFSET(a) offsetof(PulseData, a) +#define D AV_OPT_FLAG_DECODING_PARAM + +static const AVOption options[] = { + { "server", "set PulseAudio server", OFFSET(server), AV_OPT_TYPE_STRING, {.str = NULL}, 0, 0, D }, + { "name", "set application name", OFFSET(name), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, D }, + { "stream_name", "set stream description", OFFSET(stream_name), AV_OPT_TYPE_STRING, {.str = "record"}, 0, 0, D }, + { "sample_rate", "set sample rate in Hz", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, D }, + { "channels", "set number of audio channels", OFFSET(channels), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT_MAX, D }, + { "frame_size", "set number of bytes per frame", OFFSET(frame_size), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, D }, + { "fragment_size", "set buffering size, affects latency and cpu usage", OFFSET(fragment_size), AV_OPT_TYPE_INT, {.i64 = -1}, -1, INT_MAX, D }, + { NULL }, +}; + +static const AVClass pulse_demuxer_class = { + .class_name = "Pulse demuxer", + .item_name = av_default_item_name, + .option = options, + .version = LIBAVUTIL_VERSION_INT, +}; + +AVInputFormat ff_pulse_demuxer = { + .name = "pulse", + .long_name = NULL_IF_CONFIG_SMALL("Pulse audio input"), + .priv_data_size = sizeof(PulseData), + .read_header = pulse_read_header, + .read_packet = pulse_read_packet, + .read_close = pulse_close, + .flags = AVFMT_NOFILE, + .priv_class = &pulse_demuxer_class, +}; |