mirror of https://github.com/FFmpeg/FFmpeg.git
It currently use the simple api and is using the latency information provided only to offset the stream start. Signed-off-by: Luca Barbato <lu_zero@gentoo.org>pull/2/head
Luca Barbato
13 years ago
parent
82ed4f1ed8
commit
0de9c41ff4
6 changed files with 198 additions and 1 deletions
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/*
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* Pulseaudio input |
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* Copyright (c) 2011 Luca Barbato <lu_zero@gentoo.org> |
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* |
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* This file is part of Libav. |
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* |
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* Libav is free software; you can redistribute it and/or |
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* modify it under the terms of the GNU Lesser General Public |
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* License as published by the Free Software Foundation; either |
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* version 2.1 of the License, or (at your option) any later version. |
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* |
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* Libav is distributed in the hope that it will be useful, |
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* but WITHOUT ANY WARRANTY; without even the implied warranty of |
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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* Lesser General Public License for more details. |
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* |
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* You should have received a copy of the GNU Lesser General Public |
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* License along with Libav; if not, write to the Free Software |
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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*/ |
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/**
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* @file |
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* PulseAudio input using the simple API. |
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* @author Luca Barbato <lu_zero@gentoo.org> |
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* |
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*/ |
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#include <pulse/simple.h> |
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#include <pulse/rtclock.h> |
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#include <pulse/error.h> |
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#include "libavformat/avformat.h" |
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#include "libavutil/opt.h" |
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#define DEFAULT_CODEC_ID AV_NE(CODEC_ID_PCM_S16BE, CODEC_ID_PCM_S16LE) |
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typedef struct PulseData { |
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AVClass *class; |
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char *server; |
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char *name; |
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char *stream_name; |
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int sample_rate; |
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int channels; |
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int frame_size; |
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int fragment_size; |
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pa_simple *s; |
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int64_t pts; |
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} PulseData; |
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static pa_sample_format_t codec_id_to_pulse_format(int codec_id) { |
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switch (codec_id) { |
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case CODEC_ID_PCM_U8: return PA_SAMPLE_U8; |
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case CODEC_ID_PCM_ALAW: return PA_SAMPLE_ALAW; |
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case CODEC_ID_PCM_MULAW: return PA_SAMPLE_ULAW; |
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case CODEC_ID_PCM_S16LE: return PA_SAMPLE_S16LE; |
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case CODEC_ID_PCM_S16BE: return PA_SAMPLE_S16BE; |
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case CODEC_ID_PCM_F32LE: return PA_SAMPLE_FLOAT32LE; |
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case CODEC_ID_PCM_F32BE: return PA_SAMPLE_FLOAT32BE; |
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case CODEC_ID_PCM_S32LE: return PA_SAMPLE_S32LE; |
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case CODEC_ID_PCM_S32BE: return PA_SAMPLE_S32BE; |
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case CODEC_ID_PCM_S24LE: return PA_SAMPLE_S24LE; |
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case CODEC_ID_PCM_S24BE: return PA_SAMPLE_S24BE; |
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default: return PA_SAMPLE_INVALID; |
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} |
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} |
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static av_cold int pulse_read_header(AVFormatContext *s, |
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AVFormatParameters *ap) |
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{ |
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PulseData *pd = s->priv_data; |
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AVStream *st; |
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char *device = NULL; |
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int ret; |
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enum CodecID codec_id = |
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s->audio_codec_id == CODEC_ID_NONE ? DEFAULT_CODEC_ID : s->audio_codec_id; |
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const pa_sample_spec ss = { codec_id_to_pulse_format(codec_id), |
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pd->sample_rate, |
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pd->channels }; |
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pa_buffer_attr attr = { -1 }; |
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st = avformat_new_stream(s, NULL); |
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if (!st) { |
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av_log(s, AV_LOG_ERROR, "Cannot add stream\n"); |
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return AVERROR(ENOMEM); |
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} |
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attr.fragsize = pd->fragment_size; |
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if (strcmp(s->filename, "default")) |
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device = s->filename; |
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pd->s = pa_simple_new(pd->server, pd->name, |
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PA_STREAM_RECORD, |
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device, pd->stream_name, &ss, |
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NULL, &attr, &ret); |
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if (!pd->s) { |
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av_log(s, AV_LOG_ERROR, "pa_simple_new failed: %s\n", |
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pa_strerror(ret)); |
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return AVERROR(EIO); |
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} |
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/* take real parameters */ |
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st->codec->codec_type = AVMEDIA_TYPE_AUDIO; |
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st->codec->codec_id = codec_id; |
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st->codec->sample_rate = pd->sample_rate; |
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st->codec->channels = pd->channels; |
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av_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */ |
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pd->pts = AV_NOPTS_VALUE; |
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return 0; |
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} |
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static int pulse_read_packet(AVFormatContext *s, AVPacket *pkt) |
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{ |
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PulseData *pd = s->priv_data; |
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int res; |
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pa_usec_t latency; |
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uint64_t frame_duration = |
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(pd->frame_size*1000000LL) / (pd->sample_rate * pd->channels); |
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if (av_new_packet(pkt, pd->frame_size) < 0) { |
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return AVERROR(ENOMEM); |
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} |
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if ((pa_simple_read(pd->s, pkt->data, pkt->size, &res)) < 0) { |
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av_log(s, AV_LOG_ERROR, "pa_simple_read failed: %s\n", |
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pa_strerror(res)); |
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av_free_packet(pkt); |
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return AVERROR(EIO); |
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} |
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if ((latency = pa_simple_get_latency(pd->s, &res)) == (pa_usec_t) -1) { |
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av_log(s, AV_LOG_ERROR, "pa_simple_get_latency() failed: %s\n", |
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pa_strerror(res)); |
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return AVERROR(EIO); |
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} |
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if (pd->pts == AV_NOPTS_VALUE) { |
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pd->pts = -latency; |
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} |
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pkt->pts = pd->pts; |
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pd->pts += frame_duration; |
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return 0; |
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} |
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static av_cold int pulse_close(AVFormatContext *s) |
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{ |
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PulseData *pd = s->priv_data; |
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pa_simple_free(pd->s); |
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return 0; |
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} |
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#define OFFSET(a) offsetof(PulseData, a) |
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#define D AV_OPT_FLAG_DECODING_PARAM |
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static const AVOption options[] = { |
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{ "server", "pulse server name", OFFSET(server), AV_OPT_TYPE_STRING, {.str = NULL}, 0, 0, D }, |
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{ "name", "application name", OFFSET(name), AV_OPT_TYPE_STRING, {.str = "libav"}, 0, 0, D }, |
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{ "stream_name", "stream description", OFFSET(stream_name), AV_OPT_TYPE_STRING, {.str = "record"}, 0, 0, D }, |
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{ "sample_rate", "sample rate in Hz", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.dbl = 48000}, 1, INT_MAX, D }, |
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{ "channels", "number of audio channels", OFFSET(channels), AV_OPT_TYPE_INT, {.dbl = 2}, 1, INT_MAX, D }, |
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{ "frame_size", "number of bytes per frame", OFFSET(frame_size), AV_OPT_TYPE_INT, {.dbl = 1024}, 1, INT_MAX, D }, |
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{ "fragment_size", "buffering size, affects latency and cpu usage", OFFSET(fragment_size), AV_OPT_TYPE_INT, {.dbl = -1}, -1, INT_MAX, D }, |
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{ NULL }, |
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}; |
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static const AVClass pulse_demuxer_class = { |
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.class_name = "Pulse demuxer", |
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.item_name = av_default_item_name, |
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.option = options, |
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.version = LIBAVUTIL_VERSION_INT, |
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}; |
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AVInputFormat ff_pulse_demuxer = { |
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.name = "pulse", |
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.long_name = NULL_IF_CONFIG_SMALL("Pulse audio input"), |
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.priv_data_size = sizeof(PulseData), |
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.read_header = pulse_read_header, |
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.read_packet = pulse_read_packet, |
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.read_close = pulse_close, |
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.flags = AVFMT_NOFILE, |
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.priv_class = &pulse_demuxer_class, |
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}; |
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