[FFmpeg-devel] [PATCH 1/4] lavd: pulse audio encoder
Lukasz M
lukasz.m.luki at gmail.com
Mon Oct 7 23:05:09 CEST 2013
On 7 October 2013 12:22, Stefano Sabatini <stefasab at gmail.com> wrote:
> On date Friday 2013-10-04 18:24:28 +0200, Lukasz M encoded:
> > >
> > > > + at subsection Options
> > >
> > > + at table @option
> > > > +
> > > > + at item server
> > > > +Connects to a specific server. Default server is used when not
> provided.
> > >
> > > Connect.
> > >
> > > What's exactly the server?
> > >
> >
> > This is good question. TBH I'm not very familiar with pulse audio, but
> > after quick research and tests it seems to be an address of the host with
> > pulseaudio server.
> > Not sure if it can mean name of the server, not just address.
> > I made some tests on Debian 7.1 (installed few days ago so it is fresh)
> and
> > ffmpeg failed to connect with "-server localhost" option.
> > After loading a module with
> > pactl load-module module-native-protocol-tcp auth-ip-acl=LOCAL_IP
> > it worked.
>
> > I'm not quite sure name is adequate, but in pulse audio API it is called
> > this way.
>
> We should follow the pulse API.
>
> What about: connect to a specific PulseAudio server, specified by an
> IP address.
>
> >
> >
> > > > + at item fragment_size
> > > > +Specify the minimal buffering fragment in PulseAudio, it will
> affect the
> > > > +audio latency. By default it is unset.
> > >
> > > expressed in which unit?
> >
> >
> > It is in bytes, but I rushed with copying it from decoder file. This
> > parameter is relevant for recording only so I removed it.
> > There are some option for playback, I will add them later, but they
> > recommend to use default values anyway.
> >
> > Rest of remark fixed.
>
> > From 404d8a3ad94c50a61c2550eb0c871b868814801f Mon Sep 17 00:00:00 2001
> > From: Lukasz Marek <lukasz.m.luki at gmail.com>
> > Date: Fri, 4 Oct 2013 11:49:07 +0200
> > Subject: [PATCH 1/4] lavd: pulse audio encoder
> >
> > Signed-off-by: Lukasz Marek <lukasz.m.luki at gmail.com>
> > ---
> > Changelog | 1 +
> > configure | 1 +
> > doc/outdevs.texi | 31 +++++++++
> > libavdevice/Makefile | 1 +
> > libavdevice/alldevices.c | 2 +-
> > libavdevice/pulse_audio_enc.c | 154
> +++++++++++++++++++++++++++++++++++++++++
> > 6 files changed, 189 insertions(+), 1 deletion(-)
> > create mode 100644 libavdevice/pulse_audio_enc.c
> >
> > diff --git a/Changelog b/Changelog
> > index b63e036..8311c88 100644
> > --- a/Changelog
> > +++ b/Changelog
> > @@ -37,6 +37,7 @@ version <next>
> > the skip_alpha flag.
> > - ladspa wrapper filter
> > - native VP9 decoder
> > +- PulseAudio output device
> >
> >
> > version 2.0:
> > diff --git a/configure b/configure
> > index 7b8cc81..c147522 100755
> > --- a/configure
> > +++ b/configure
> > @@ -2132,6 +2132,7 @@ openal_indev_deps="openal"
> > oss_indev_deps_any="soundcard_h sys_soundcard_h"
> > oss_outdev_deps_any="soundcard_h sys_soundcard_h"
> > pulse_indev_deps="libpulse"
> > +pulse_outdev_deps="libpulse"
> > sdl_outdev_deps="sdl"
> > sndio_indev_deps="sndio_h"
> > sndio_outdev_deps="sndio_h"
> > diff --git a/doc/outdevs.texi b/doc/outdevs.texi
> > index 0946276..a54f4ea 100644
> > --- a/doc/outdevs.texi
> > +++ b/doc/outdevs.texi
> > @@ -108,6 +108,37 @@ ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_dither
> colors -
> >
> > OSS (Open Sound System) output device.
> >
> > + at section pulse
> > +
> > +PulseAudio output device.
> > +
> > +To enable this output device you need to configure FFmpeg with
> @code{--enable-libpulse}.
> > +
> > + at subsection Options
> > + at table @option
> > +
> > + at item server
> > +Connect to a specific server. Default server is used when not provided.
> > +
> > + at item name
> > +Specify the application name PulseAudio will use when showing active
> clients,
> > +by default it is the @code{LIBAVFORMAT_IDENT} string.
> > +
>
> > + at item stream_name
> > +Specify the stream name PulseAudio will use when showing active streams,
> > +by default it is set to output name.
>
> nit: to the specified output name.
>
> > +
> > + at item device
> > +Specify the device to use. Default device is used when not provided.
> > +
> > + at end table
> > +
>
> > + at subsection Examples
> > +Play a file using PulseAudio:
> > + at example
> > +ffmpeg -i INPUT -f pulse -
>
> description can be slightly more detailed, for example you could
> specify that it will send the stream to the default server.
>
> > + at end example
>
> Also I suggest to place here a link to pulse audio docs/website.
>
> > +
> > @section sdl
> >
> > SDL (Simple DirectMedia Layer) output device.
> > diff --git a/libavdevice/Makefile b/libavdevice/Makefile
> > index 424ce98..2fdc47b 100644
> > --- a/libavdevice/Makefile
> > +++ b/libavdevice/Makefile
> > @@ -31,6 +31,7 @@ OBJS-$(CONFIG_OPENAL_INDEV) +=
> openal-dec.o
> > OBJS-$(CONFIG_OSS_INDEV) += oss_audio.o
> > OBJS-$(CONFIG_OSS_OUTDEV) += oss_audio.o
> > OBJS-$(CONFIG_PULSE_INDEV) += pulse.o
> > +OBJS-$(CONFIG_PULSE_OUTDEV) += pulse_audio_enc.o
> > OBJS-$(CONFIG_SDL_OUTDEV) += sdl.o
> > OBJS-$(CONFIG_SNDIO_INDEV) += sndio_common.o sndio_dec.o
> > OBJS-$(CONFIG_SNDIO_OUTDEV) += sndio_common.o sndio_enc.o
> > diff --git a/libavdevice/alldevices.c b/libavdevice/alldevices.c
> > index fc8d3ce..33ce155 100644
> > --- a/libavdevice/alldevices.c
> > +++ b/libavdevice/alldevices.c
> > @@ -57,7 +57,7 @@ void avdevice_register_all(void)
> > REGISTER_INDEV (LAVFI, lavfi);
> > REGISTER_INDEV (OPENAL, openal);
> > REGISTER_INOUTDEV(OSS, oss);
> > - REGISTER_INDEV (PULSE, pulse);
> > + REGISTER_INOUTDEV(PULSE, pulse);
> > REGISTER_OUTDEV (SDL, sdl);
> > REGISTER_INOUTDEV(SNDIO, sndio);
> > REGISTER_INOUTDEV(V4L2, v4l2);
> > diff --git a/libavdevice/pulse_audio_enc.c
> b/libavdevice/pulse_audio_enc.c
> > new file mode 100644
> > index 0000000..37dca9e
> > --- /dev/null
> > +++ b/libavdevice/pulse_audio_enc.c
> > @@ -0,0 +1,154 @@
> > +/*
> > + * Copyright (c) 2013 Lukasz Marek <lukasz.m.luki at gmail.com>
> > + *
> > + * 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 <pulse/simple.h>
> > +#include <pulse/error.h>
> > +#include "libavutil/opt.h"
> > +#include "libavutil/time.h"
> > +#include "libavutil/log.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)
>
> you can move the define right in the codec definition
>
> > +
> > +typedef struct PulseData {
> > + AVClass *class;
> > + const char *server;
> > + const char *name;
> > + const char *stream_name;
> > + const char *device;
> > + pa_simple *pa;
> > +} PulseData;
> > +
> > +static pa_sample_format_t codec_id_to_pulse_format(enum AVCodecID
> 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_write_header(AVFormatContext *h)
> > +{
> > + PulseData *s = h->priv_data;
> > + AVStream *st = h->streams[0];
> > + int ret;
> > + pa_sample_spec ss = { codec_id_to_pulse_format(st->codec->codec_id),
> > + st->codec->sample_rate,
> > + st->codec->channels };
> > + pa_buffer_attr attr = { -1, -1, -1, -1, -1 };
> > + const char *stream_name = s->stream_name;
> > +
> > + if (!stream_name)
> > + stream_name = h->filename;
> > +
> > + s->pa = pa_simple_new(s->server, // Server
> > + s->name, // Application name
> > + PA_STREAM_PLAYBACK,
> > + s->device, // Device
> > + stream_name, // Description of
> a stream
> > + &ss, // Sample format
> > + NULL, // Use default
> channel map
> > + &attr, // Buffering
> attributes
> > + &ret); // Result
> > +
> > + if (!s->pa) {
> > + av_log(s, AV_LOG_ERROR, "pa_simple_new failed: %s\n",
> pa_strerror(ret));
> > + return AVERROR(EIO);
> > + }
> > +
> > + avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
> > +
> > + return 0;
>
> Not sure if you should check the number and type of the streams. What
> happens if you send two audio streams, or one video stream?
>
> > +}
> > +
> > +static av_cold int pulse_write_trailer(AVFormatContext *h)
> > +{
> > + PulseData *s = h->priv_data;
> > + pa_simple_flush(s->pa, NULL);
> > + pa_simple_free(s->pa);
> > + s->pa = NULL;
> > + return 0;
> > +}
> > +
> > +static int pulse_write_packet(AVFormatContext *h, AVPacket *pkt)
> > +{
> > + PulseData *s = h->priv_data;
> > + int size = pkt->size;
> > + uint8_t *buf = pkt->data;
> > + int error;
> > +
> > + if ((error = pa_simple_write(s->pa, buf, size, &error))) {
> > + av_log(s, AV_LOG_ERROR, "pa_simple_write failed: %s\n",
> pa_strerror(error));
> > + return AVERROR(EIO);
> > + }
> > +
> > + return 0;
> > +}
> > +
> > +static void pulse_get_output_timestamp(AVFormatContext *h, int stream,
> int64_t *dts, int64_t *wall)
> > +{
> > + PulseData *s = h->priv_data;
> > + pa_usec_t latency = pa_simple_get_latency(s->pa, NULL);
> > + *wall = av_gettime();
> > + *dts = h->streams[0]->cur_dts - latency;
> > +}
> > +
> > +#define OFFSET(a) offsetof(PulseData, a)
> > +#define E AV_OPT_FLAG_ENCODING_PARAM
> > +
> > +static const AVOption options[] = {
> > + { "server", "set pulse server name", OFFSET(server),
> AV_OPT_TYPE_STRING, {.str = NULL}, 0, 0, E },
> > + { "name", "set application name", OFFSET(name),
> AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, E },
> > + { "stream_name", "set stream description", OFFSET(stream_name),
> AV_OPT_TYPE_STRING, {.str = NULL}, 0, 0, E },
> > + { "device", "set device name", OFFSET(device),
> AV_OPT_TYPE_STRING, {.str = NULL}, 0, 0, E },
> > + { NULL }
> > +};
> > +
> > +static const AVClass pulse_muxer_class = {
> > + .class_name = "Pulse muxer",
> > + .item_name = av_default_item_name,
> > + .option = options,
> > + .version = LIBAVUTIL_VERSION_INT,
> > +};
> > +
> > +AVOutputFormat ff_pulse_muxer = {
> > + .name = "pulse",
> > + .long_name = NULL_IF_CONFIG_SMALL("Pulse audio output"),
> > + .priv_data_size = sizeof(PulseData),
> > + .audio_codec = DEFAULT_CODEC_ID,
> > + .video_codec = AV_CODEC_ID_NONE,
> > + .write_header = pulse_write_header,
> > + .write_packet = pulse_write_packet,
> > + .write_trailer = pulse_write_trailer,
> > + .get_output_timestamp = pulse_get_output_timestamp,
> > + .flags = AVFMT_NOFILE,
> > + .priv_class = &pulse_muxer_class,
> > +};
>
> LGTM otherwise.
Updated patch attached
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