diff options
Diffstat (limited to 'libavdevice/alsa_enc.c')
| -rw-r--r-- | libavdevice/alsa_enc.c | 174 |
1 files changed, 174 insertions, 0 deletions
diff --git a/libavdevice/alsa_enc.c b/libavdevice/alsa_enc.c new file mode 100644 index 0000000000..fb428f0623 --- /dev/null +++ b/libavdevice/alsa_enc.c @@ -0,0 +1,174 @@ +/* + * ALSA input and output + * Copyright (c) 2007 Luca Abeni ( lucabe72 email it ) + * Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr ) + * + * 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 + * ALSA input and output: output + * @author Luca Abeni ( lucabe72 email it ) + * @author Benoit Fouet ( benoit fouet free fr ) + * + * This avdevice encoder allows to play audio to an ALSA (Advanced Linux + * Sound Architecture) device. + * + * The filename parameter is the name of an ALSA PCM device capable of + * capture, for example "default" or "plughw:1"; see the ALSA documentation + * for naming conventions. The empty string is equivalent to "default". + * + * The playback period is set to the lower value available for the device, + * which gives a low latency suitable for real-time playback. + */ + +#include <alsa/asoundlib.h> + +#include "libavutil/internal.h" +#include "libavutil/time.h" + + +#include "libavformat/internal.h" +#include "avdevice.h" +#include "alsa.h" + +static av_cold int audio_write_header(AVFormatContext *s1) +{ + AlsaData *s = s1->priv_data; + AVStream *st = NULL; + unsigned int sample_rate; + enum AVCodecID codec_id; + int res; + + if (s1->nb_streams != 1 || s1->streams[0]->codec->codec_type != AVMEDIA_TYPE_AUDIO) { + av_log(s1, AV_LOG_ERROR, "Only a single audio stream is supported.\n"); + return AVERROR(EINVAL); + } + st = s1->streams[0]; + + sample_rate = st->codec->sample_rate; + codec_id = st->codec->codec_id; + res = ff_alsa_open(s1, SND_PCM_STREAM_PLAYBACK, &sample_rate, + st->codec->channels, &codec_id); + if (sample_rate != st->codec->sample_rate) { + av_log(s1, AV_LOG_ERROR, + "sample rate %d not available, nearest is %d\n", + st->codec->sample_rate, sample_rate); + goto fail; + } + avpriv_set_pts_info(st, 64, 1, sample_rate); + + return res; + +fail: + snd_pcm_close(s->h); + return AVERROR(EIO); +} + +static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt) +{ + AlsaData *s = s1->priv_data; + int res; + int size = pkt->size; + uint8_t *buf = pkt->data; + + size /= s->frame_size; + if (pkt->dts != AV_NOPTS_VALUE) + s->timestamp = pkt->dts; + s->timestamp += pkt->duration ? pkt->duration : size; + + if (s->reorder_func) { + if (size > s->reorder_buf_size) + if (ff_alsa_extend_reorder_buf(s, size)) + return AVERROR(ENOMEM); + s->reorder_func(buf, s->reorder_buf, size); + buf = s->reorder_buf; + } + while ((res = snd_pcm_writei(s->h, buf, size)) < 0) { + if (res == -EAGAIN) { + + return AVERROR(EAGAIN); + } + + if (ff_alsa_xrun_recover(s1, res) < 0) { + av_log(s1, AV_LOG_ERROR, "ALSA write error: %s\n", + snd_strerror(res)); + + return AVERROR(EIO); + } + } + + return 0; +} + +static int audio_write_frame(AVFormatContext *s1, int stream_index, + AVFrame **frame, unsigned flags) +{ + AlsaData *s = s1->priv_data; + AVPacket pkt; + + /* ff_alsa_open() should have accepted only supported formats */ + if ((flags & AV_WRITE_UNCODED_FRAME_QUERY)) + return av_sample_fmt_is_planar(s1->streams[stream_index]->codec->sample_fmt) ? + AVERROR(EINVAL) : 0; + /* set only used fields */ + pkt.data = (*frame)->data[0]; + pkt.size = (*frame)->nb_samples * s->frame_size; + pkt.dts = (*frame)->pkt_dts; + pkt.duration = av_frame_get_pkt_duration(*frame); + return audio_write_packet(s1, &pkt); +} + +static void +audio_get_output_timestamp(AVFormatContext *s1, int stream, + int64_t *dts, int64_t *wall) +{ + AlsaData *s = s1->priv_data; + snd_pcm_sframes_t delay = 0; + *wall = av_gettime(); + snd_pcm_delay(s->h, &delay); + *dts = s->timestamp - delay; +} + +static int audio_get_device_list(AVFormatContext *h, AVDeviceInfoList *device_list) +{ + return ff_alsa_get_device_list(device_list, SND_PCM_STREAM_PLAYBACK); +} + +static const AVClass alsa_muxer_class = { + .class_name = "ALSA muxer", + .item_name = av_default_item_name, + .version = LIBAVUTIL_VERSION_INT, + .category = AV_CLASS_CATEGORY_DEVICE_AUDIO_OUTPUT, +}; + +AVOutputFormat ff_alsa_muxer = { + .name = "alsa", + .long_name = NULL_IF_CONFIG_SMALL("ALSA audio output"), + .priv_data_size = sizeof(AlsaData), + .audio_codec = DEFAULT_CODEC_ID, + .video_codec = AV_CODEC_ID_NONE, + .write_header = audio_write_header, + .write_packet = audio_write_packet, + .write_trailer = ff_alsa_close, + .write_uncoded_frame = audio_write_frame, + .get_device_list = audio_get_device_list, + .get_output_timestamp = audio_get_output_timestamp, + .flags = AVFMT_NOFILE, + .priv_class = &alsa_muxer_class, +}; |