[FFmpeg-devel] [PATCH] Add support for sndio to libavdevice
Stefano Sabatini
stefano.sabatini-lala
Sun Aug 8 00:57:01 CEST 2010
On date Monday 2010-08-02 19:42:28 -0400, Brad encoded:
> sndio is a relatively new audio API utilized by OpenBSD.
>
> Below is a patch to add sndio playback and record support
> to FFmpeg.
>
> I believe I have touched everything that needs updating
> including documentation such as the recently added
> indevs.texi and outdevs.texi, but if not please let
> me know.
>
> The sndio code was written by Jacob Meuser <jakemsr sdf lonestar org>
>
> Please provide any feedback.
>
>
> Index: configure
> ===================================================================
> --- configure (revision 24666)
> +++ configure (working copy)
> @@ -1030,6 +1030,7 @@
> sdl
> sdl_video_size
> setmode
> + sndio_h
> socklen_t
> soundcard_h
> poll_h
> @@ -1361,6 +1362,10 @@
> libdc1394_indev_deps="libdc1394"
> oss_indev_deps_any="soundcard_h sys_soundcard_h"
> oss_outdev_deps_any="soundcard_h sys_soundcard_h"
> +sndio_indev_deps="sndio_h"
> +sndio_indev_extralibs="-lsndio"
> +sndio_outdev_deps="sndio_h"
> +sndio_outdev_extralibs="-lsndio"
> v4l_indev_deps="linux_videodev_h"
> v4l2_indev_deps_any="linux_videodev2_h sys_videoio_h"
> vfwcap_indev_deps="capCreateCaptureWindow vfwcap_defines"
> @@ -2772,11 +2777,14 @@
>
> check_header sys/soundcard.h
> check_header soundcard.h
> +check_header sndio.h
>
> enabled_any alsa_indev alsa_outdev && check_lib2 alsa/asoundlib.h snd_pcm_htimestamp -lasound
>
> enabled jack_indev && check_lib2 jack/jack.h jack_client_open -ljack
>
> +enabled_any sndio_indev sndio_outdev && check_lib2 sndio.h sio_open -lsndio
> +
> enabled x11grab &&
> check_header X11/Xlib.h &&
> check_header X11/extensions/XShm.h &&
> Index: Changelog
> ===================================================================
> --- Changelog (revision 24666)
> +++ Changelog (working copy)
> @@ -27,9 +27,9 @@
> - SubRip subtitle file muxer and demuxer
> - Chinese AVS encoding via libxavs
> - ffprobe -show_packets option added
> +- sndio support for playback and record
>
>
> -
> version 0.6:
>
> - PB-frame decoding for H.263
> Index: doc/indevs.texi
> ===================================================================
> --- doc/indevs.texi (revision 24666)
> +++ doc/indevs.texi (working copy)
> @@ -133,6 +133,23 @@
> For more information about OSS see:
> @url{http://manuals.opensound.com/usersguide/dsp.html}
>
> + at section sndio
> +
> +sndio input device.
> +
> +To enable this input device during configuration you need libsndio
> +installed on your system.
> +
> +The filename to provide to the input device is the device node
> +representing the sndio input device, and is usually set to
> + at file{/dev/audio0/}.
@file{/dev/audio0}, no trailing "/", here and below.
> +
> +For example to grab from @file{/dev/audio0/} using @file{ffmpeg} use the
> +command:
> + at example
> +ffmpeg -f sndio -i /dev/audio0 /tmp/oss.wav
> + at end example
> +
> @section video4linux and video4linux2
>
> Video4Linux and Video4Linux2 input video devices.
> Index: doc/outdevs.texi
> ===================================================================
> --- doc/outdevs.texi (revision 24666)
> +++ doc/outdevs.texi (working copy)
> @@ -30,4 +30,8 @@
>
> OSS (Open Sound System) output device.
>
> + at section sndio
> +
> +sndio audio output device.
> +
> @c man end OUTPUT DEVICES
> Index: libavdevice/alldevices.c
> ===================================================================
> --- libavdevice/alldevices.c (revision 24666)
> +++ libavdevice/alldevices.c (working copy)
> @@ -44,6 +44,7 @@
> REGISTER_INDEV (DV1394, dv1394);
> REGISTER_INDEV (JACK, jack);
> REGISTER_INOUTDEV (OSS, oss);
> + REGISTER_INOUTDEV (SNDIO, sndio);
> REGISTER_INDEV (V4L2, v4l2);
> REGISTER_INDEV (V4L, v4l);
> REGISTER_INDEV (VFWCAP, vfwcap);
> Index: libavdevice/sndio.c
> ===================================================================
> --- libavdevice/sndio.c (revision 0)
> +++ libavdevice/sndio.c (revision 0)
> @@ -0,0 +1,299 @@
> +/*
> + * sndio play and grab interface
> + * Copyright (c) 2010 Jacob Meuser
> + *
> + * 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 <stdlib.h>
> +#include <stdio.h>
> +#include <stdint.h>
> +#include <string.h>
> +#include <unistd.h>
> +
> +#include <sndio.h>
> +
> +#include "libavutil/log.h"
> +#include "libavcodec/avcodec.h"
> +#include "libavformat/avformat.h"
> +
> +
> +typedef struct {
> + struct sio_hdl *hdl;
> + int sample_rate;
> + int channels;
> + int bps;
> + enum CodecID codec_id;
> + int buffer_size;
> + uint8_t *buffer;
> + int buffer_ptr;
> + long long hwpos, softpos;
> +} AudioData;
> +
> +static void movecb(void *addr, int delta)
> +{
> + AudioData *s = addr;
> +
> + s->hwpos += delta * s->channels * s->bps;
> +}
> +
> +static int audio_open(AVFormatContext *s1, int is_output, const char *audio_device)
> +{
> + AudioData *s = s1->priv_data;
> + struct sio_hdl *hdl = NULL;
> + struct sio_par par;
> +
> + if (is_output)
> + hdl = sio_open(audio_device, SIO_PLAY, 0);
> + else
> + hdl = sio_open(audio_device, SIO_REC, 0);
simpler:
hdl = sio_open(audio_device, is_output ? SIO_PLAY : SIO_REC, 0);
> + if (hdl == NULL) {
> + av_log(s1, AV_LOG_ERROR, "could not open sndio device\n");
> + return AVERROR(EIO);
> + }
> +
> + sio_initpar(&par);
> + par.bits = 16;
> + par.sig = 1;
> + par.le = SIO_LE_NATIVE;
> + if (is_output)
> + par.pchan = s->channels;
> + else
> + par.rchan = s->channels;
This also could be simplified.
> + par.rate = s->sample_rate;
> +
> + if (!sio_setpar(hdl, &par) || !sio_getpar(hdl, &par)) {
> + av_log(s1, AV_LOG_ERROR, "error setting sndio parameters\n");
> + goto fail;
> + }
"Impossible to set sndio parameters\n"
even better would be to print an error description.
> +
> + if (par.bits != 16 || par.sig != 1 || par.le != SIO_LE_NATIVE ||
> + (is_output && (par.pchan != s->channels)) ||
> + (!is_output && (par.rchan != s->channels)) ||
> + (par.rate != s->sample_rate)) {
> + av_log(s1, AV_LOG_ERROR, "could not set appropriate sndio parameters\n");
> + goto fail;
> + }
> +
> + s->buffer_size = par.round * par.bps *
> + (is_output ? par.pchan : par.rchan);
> +
> + s->buffer = av_malloc(s->buffer_size);
> + if (s->buffer == NULL) {
> + av_log(s1, AV_LOG_ERROR, "could not allocate buffer\n");
> + goto fail;
> + }
> +
> + if (par.le)
> + s->codec_id = CODEC_ID_PCM_S16LE;
> + else
> + s->codec_id = CODEC_ID_PCM_S16BE;
> +
> + if (is_output)
> + s->channels = par.pchan;
> + else
> + s->channels = par.rchan;
-6 lines using the ? : construct.
> +
> + s->sample_rate = par.rate;
> + s->bps = par.bps;
> +
> + sio_onmove(hdl, movecb, s);
> +
> + if (!sio_start(hdl)) {
> + av_log(s1, AV_LOG_ERROR, "could not start sndio\n");
> + goto fail;
> + }
> +
> + s->hdl = hdl;
> +
> + return 0;
> +
> +fail:
> + if (s->buffer)
> + av_free(s->buffer);
> + if (hdl)
> + sio_close(hdl);
> + return AVERROR(EIO);
> +}
> +
> +static int audio_close(AudioData *s)
> +{
> + if (s->buffer)
> + av_free(s->buffer);
> + if (s->hdl)
> + sio_close(s->hdl);
> + return 0;
> +}
> +
> +/* sound output support */
> +static int audio_write_header(AVFormatContext *s1)
> +{
> + AudioData *s = s1->priv_data;
> + AVStream *st;
> + int ret;
> +
> + st = s1->streams[0];
> + s->sample_rate = st->codec->sample_rate;
> + s->channels = st->codec->channels;
> + ret = audio_open(s1, 1, s1->filename);
> + if (ret < 0) {
> + return AVERROR(EIO);
> + } else {
> + return 0;
> + }
> +}
> +
> +static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
> +{
> + AudioData *s = s1->priv_data;
> + int len, ret;
> + int size = pkt->size;
> + uint8_t *buf= pkt->data;
> +
> + while (size > 0) {
> + len = s->buffer_size - s->buffer_ptr;
> + if (len > size)
> + len = size;
> + memcpy(s->buffer + s->buffer_ptr, buf, len);
> + buf += len;
> + size -= len;
> + s->buffer_ptr += len;
> + if (s->buffer_ptr >= s->buffer_size) {
> + ret = sio_write(s->hdl, s->buffer, s->buffer_size);
> + if (ret == 0 || sio_eof(s->hdl))
> + return AVERROR(EIO);
> + s->softpos += ret;
> + s->buffer_ptr = 0;
> + }
> + }
> + return 0;
> +}
> +
> +static int audio_write_trailer(AVFormatContext *s1)
> +{
> + AudioData *s = s1->priv_data;
> +
> + audio_close(s);
> + return 0;
> +}
> +
> +/* grab support */
> +
> +static int audio_read_header(AVFormatContext *s1, AVFormatParameters *ap)
> +{
> + AudioData *s = s1->priv_data;
> + AVStream *st;
> + int ret;
> +
> + if (ap->sample_rate <= 0 || ap->channels <= 0)
> + return -1;
AVERROR(EINVAL);
> +
> + st = av_new_stream(s1, 0);
> + if (!st) {
> + return AVERROR(ENOMEM);
> + }
> + s->sample_rate = ap->sample_rate;
> + s->channels = ap->channels;
> +
> + ret = audio_open(s1, 0, s1->filename);
> + if (ret < 0) {
> + return AVERROR(EIO);
> + }
> +
> + /* take real parameters */
> + st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
> + st->codec->codec_id = s->codec_id;
> + st->codec->sample_rate = s->sample_rate;
> + st->codec->channels = s->channels;
> +
> + av_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
> + return 0;
> +}
> +
> +static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
> +{
> + AudioData *s = s1->priv_data;
> + int ret, bdelay;
> + int64_t cur_time;
> +
> + if ((ret=av_new_packet(pkt, s->buffer_size)) < 0)
> + return ret;
> +
> + ret = sio_read(s->hdl, pkt->data, pkt->size);
> + if (ret == 0 || sio_eof(s->hdl)) {
> + av_free_packet(pkt);
> + pkt->size = 0;
> + return AVERROR_EOF;
> + }
> + pkt->size = ret;
> + s->softpos += ret;
> +
> + /* compute pts of the start of the packet */
> + cur_time = av_gettime();
> +
> + bdelay = ret + s->hwpos - s->softpos;
> +
> + /* convert to wanted units */
> + pkt->pts = cur_time;
> +
> + return 0;
> +}
> +
> +static int audio_read_close(AVFormatContext *s1)
> +{
> + AudioData *s = s1->priv_data;
> +
> + audio_close(s);
> + return 0;
> +}
> +
> +#if CONFIG_SNDIO_INDEV
> +AVInputFormat sndio_demuxer = {
> + "sndio",
> + NULL_IF_CONFIG_SMALL("sndio audio capture"),
> + sizeof(AudioData),
> + NULL,
> + audio_read_header,
> + audio_read_packet,
> + audio_read_close,
> + .flags = AVFMT_NOFILE,
> +};
> +#endif
> +
> +#if CONFIG_SNDIO_OUTDEV
> +AVOutputFormat sndio_muxer = {
> + "sndio",
> + NULL_IF_CONFIG_SMALL("sndio audio playback"),
> + "",
> + "",
> + sizeof(AudioData),
> + /* XXX: we make the assumption that the soundcard accepts this format */
> + /* XXX: find better solution with "preinit" method, needed also in
> + other formats */
> +#if HAVE_BIGENDIAN
> + CODEC_ID_PCM_S16BE,
> +#else
> + CODEC_ID_PCM_S16LE,
> +#endif
> + CODEC_ID_NONE,
> + audio_write_header,
> + audio_write_packet,
> + audio_write_trailer,
> + .flags = AVFMT_NOFILE,
> +};
maybe we should use designated init for all the fields, more robust
and don't break in case of ABI breaks (e.g. in case of major bumps).
Regards.
--
FFmpeg = Fostering and Fanciful Mystic Practical Extroverse Ghost
More information about the ffmpeg-devel
mailing list