/* * Pulseaudio input * Copyright (c) 2011 Luca Barbato * * 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 */ /** * @file * PulseAudio input using the simple API. * @author Luca Barbato */ #include #include #include #include "libavutil/internal.h" #include "libavutil/opt.h" #include "libavutil/time.h" #include "libavformat/avformat.h" #include "libavformat/internal.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; int wallclock; } PulseData; static pa_sample_format_t codec_id_to_pulse_format(int codec_id) { switch (codec_id) { case AV_CODEC_ID_PCM_U8: return PA_SAMPLE_U8; case AV_CODEC_ID_PCM_ALAW: return PA_SAMPLE_ALAW; case AV_CODEC_ID_PCM_MULAW: return PA_SAMPLE_ULAW; case AV_CODEC_ID_PCM_S16LE: return PA_SAMPLE_S16LE; case AV_CODEC_ID_PCM_S16BE: return PA_SAMPLE_S16BE; case AV_CODEC_ID_PCM_F32LE: return PA_SAMPLE_FLOAT32LE; case AV_CODEC_ID_PCM_F32BE: return PA_SAMPLE_FLOAT32BE; case AV_CODEC_ID_PCM_S32LE: return PA_SAMPLE_S32LE; case AV_CODEC_ID_PCM_S32BE: return PA_SAMPLE_S32BE; case AV_CODEC_ID_PCM_S24LE: return PA_SAMPLE_S24LE; case AV_CODEC_ID_PCM_S24BE: return PA_SAMPLE_S24BE; default: return PA_SAMPLE_INVALID; } } static av_cold int pulse_read_header(AVFormatContext *s) { PulseData *pd = s->priv_data; AVStream *st; char *device = NULL; int ret; enum AVCodecID codec_id = s->audio_codec_id == AV_CODEC_ID_NONE ? DEFAULT_CODEC_ID : s->audio_codec_id; const pa_sample_spec ss = { 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->codecpar->codec_type = AVMEDIA_TYPE_AUDIO; st->codecpar->codec_id = codec_id; st->codecpar->sample_rate = pd->sample_rate; st->codecpar->channels = pd->channels; avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */ pd->pts = AV_NOPTS_VALUE; pd->frame_duration = (pd->frame_size * 1000000LL * 8) / (pd->sample_rate * pd->channels * av_get_bits_per_sample(codec_id)); return 0; } static int pulse_read_packet(AVFormatContext *s, AVPacket *pkt) { PulseData *pd = s->priv_data; int res; pa_usec_t latency; 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_packet_unref(pkt); return AVERROR(EIO); } 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); } if (pd->pts == AV_NOPTS_VALUE) { pd->pts = -latency; if (pd->wallclock) pd->pts += av_gettime(); } 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", "pulse server name", OFFSET(server), AV_OPT_TYPE_STRING, {.str = NULL}, 0, 0, D }, { "name", "application name", OFFSET(name), AV_OPT_TYPE_STRING, {.str = "libav"}, 0, 0, D }, { "stream_name", "stream description", OFFSET(stream_name), AV_OPT_TYPE_STRING, {.str = "record"}, 0, 0, D }, { "sample_rate", "sample rate in Hz", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, D }, { "channels", "number of audio channels", OFFSET(channels), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT_MAX, D }, { "frame_size", "number of bytes per frame", OFFSET(frame_size), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, D }, { "fragment_size", "buffering size, affects latency and cpu usage", OFFSET(fragment_size), AV_OPT_TYPE_INT, {.i64 = -1}, -1, INT_MAX, D }, { "wallclock", "set the initial pts using the current time", OFFSET(wallclock), AV_OPT_TYPE_INT, {.i64 = 1}, -1, 1, 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, };