Merge remote-tracking branch 'newdev/master'

* newdev/master:
  ac3enc: Add codec-specific options for writing AC-3 metadata.
NOT MERGED:  Remove arrozcru URL from documentation
  sndio support for playback and record

Conflicts:
	doc/faq.texi
	doc/general.texi
Merged-by: NMichael Niedermayer <michaelni@gmx.at>

Changelog

@@ -80,6 +80,7 @@ version <next>:
 - Bitmap Brothers JV playback system
 - Linux framebuffer input device added
 - Apple HTTP Live Streaming protocol handler
+- sndio support for playback and record
 
 
 version 0.6:

configure

@@ -1098,6 +1098,7 @@ HAVE_LIST="
     sdl
     sdl_video_size
     setmode
+    sndio_h
     socklen_t
     soundcard_h
     poll_h
@@ -1448,6 +1449,8 @@ jack_indev_deps="jack_jack_h"
 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_outdev_deps="sndio_h"
 v4l_indev_deps="linux_videodev_h"
 v4l2_indev_deps_any="linux_videodev2_h sys_videoio_h"
 vfwcap_indev_deps="capCreateCaptureWindow vfwcap_defines"
@@ -2934,6 +2937,7 @@ check_cpp_condition vfw.h "WM_CAP_DRIVER_CONNECT > WM_USER" && enable vfwcap_def
   check_header dev/video/bktr/ioctl_bt848.h; } ||
 check_header dev/ic/bt8xx.h
 
+check_header sndio.h
 check_header sys/soundcard.h
 check_header soundcard.h
 
@@ -2941,6 +2945,8 @@ enabled_any alsa_indev alsa_outdev && check_lib2 alsa/asoundlib.h snd_pcm_htimes
 
 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      &&

doc/encoders.texi

@@ -17,4 +17,340 @@ with the options @code{--enable-encoder=@var{ENCODER}} /
 The option @code{-codecs} of the ff* tools will display the list of
 enabled encoders.
 
+A description of some of the currently available encoders follows.
+
+@section Audio Encoders
+
+@subsection ac3 and ac3_fixed
+
+AC-3 audio encoders.
+
+These encoders implement part of ATSC A/52:2010 and ETSI TS 102 366, as well as
+the undocumented RealAudio 3 (a.k.a. dnet).
+
+The @var{ac3} encoder uses floating-point math, while the @var{ac3_fixed}
+encoder only uses fixed-point integer math. This does not mean that one is
+always faster, just that one or the other may be better suited to a
+particular system. The floating-point encoder will generally produce better
+quality audio for a given bitrate. The @var{ac3_fixed} encoder is not the
+default codec for any of the output formats, so it must be specified explicitly
+using the option @code{-acodec ac3_fixed} in order to use it.
+
+@subheading AC-3 Metadata
+
+The AC-3 metadata options are used to set parameters that describe the audio,
+but in most cases do not affect the audio encoding itself. Some of the options
+do directly affect or influence the decoding and playback of the resulting
+bitstream, while others are just for informational purposes. A few of the
+options will add bits to the output stream that could otherwise be used for
+audio data, and will thus affect the quality of the output. Those will be
+indicated accordingly with a note in the option list below.
+
+These parameters are described in detail in several publicly-available
+documents.
+@itemize
+@item @uref{http://www.atsc.org/cms/standards/a_52-2010.pdf,A/52:2010 - Digital Audio Compression (AC-3) (E-AC-3) Standard}
+@item @uref{http://www.atsc.org/cms/standards/a_54a_with_corr_1.pdf,A/54 - Guide to the Use of the ATSC Digital Television Standard}
+@item @uref{http://www.dolby.com/uploadedFiles/zz-_Shared_Assets/English_PDFs/Professional/18_Metadata.Guide.pdf,Dolby Metadata Guide}
+@item @uref{http://www.dolby.com/uploadedFiles/zz-_Shared_Assets/English_PDFs/Professional/46_DDEncodingGuidelines.pdf,Dolby Digital Professional Encoding Guidelines}
+@end itemize
+
+@subsubheading Metadata Control Options
+
+@table @option
+
+@item -per_frame_metadata @var{boolean}
+Allow Per-Frame Metadata. Specifies if the encoder should check for changing
+metadata for each frame.
+@table @option
+@item 0
+The metadata values set at initialization will be used for every frame in the
+stream. (default)
+@item 1
+Metadata values can be changed before encoding each frame.
+@end table
+
+@end table
+
+@subsubheading Downmix Levels
+
+@table @option
+
+@item -center_mixlev @var{level}
+Center Mix Level. The amount of gain the decoder should apply to the center
+channel when downmixing to stereo. This field will only be written to the
+bitstream if a center channel is present. The value is specified as a scale
+factor. There are 3 valid values:
+@table @option
+@item 0.707
+Apply -3dB gain
+@item 0.595
+Apply -4.5dB gain (default)
+@item 0.500
+Apply -6dB gain
+@end table
+
+@item -surround_mixlev @var{level}
+Surround Mix Level. The amount of gain the decoder should apply to the surround
+channel(s) when downmixing to stereo. This field will only be written to the
+bitstream if one or more surround channels are present. The value is specified
+as a scale factor.  There are 3 valid values:
+@table @option
+@item 0.707
+Apply -3dB gain
+@item 0.500
+Apply -6dB gain (default)
+@item 0.000
+Silence Surround Channel(s)
+@end table
+
+@end table
+
+@subsubheading Audio Production Information
+Audio Production Information is optional information describing the mixing
+environment.  Either none or both of the fields are written to the bitstream.
+
+@table @option
+
+@item -mixing_level @var{number}
+Mixing Level. Specifies peak sound pressure level (SPL) in the production
+environment when the mix was mastered. Valid values are 80 to 111, or -1 for
+unknown or not indicated. The default value is -1, but that value cannot be
+used if the Audio Production Information is written to the bitstream. Therefore,
+if the @code{room_type} option is not the default value, the @code{mixing_level}
+option must not be -1.
+
+@item -room_type @var{type}
+Room Type. Describes the equalization used during the final mixing session at
+the studio or on the dubbing stage. A large room is a dubbing stage with the
+industry standard X-curve equalization; a small room has flat equalization.
+This field will not be written to the bitstream if both the @code{mixing_level}
+option and the @code{room_type} option have the default values.
+@table @option
+@item 0
+@itemx notindicated
+Not Indicated (default)
+@item 1
+@itemx large
+Large Room
+@item 2
+@itemx small
+Small Room
+@end table
+
+@end table
+
+@subsubheading Other Metadata Options
+
+@table @option
+
+@item -copyright @var{boolean}
+Copyright Indicator. Specifies whether a copyright exists for this audio.
+@table @option
+@item 0
+@itemx off
+No Copyright Exists (default)
+@item 1
+@itemx on
+Copyright Exists
+@end table
+
+@item -dialnorm @var{value}
+Dialogue Normalization. Indicates how far the average dialogue level of the
+program is below digital 100% full scale (0 dBFS). This parameter determines a
+level shift during audio reproduction that sets the average volume of the
+dialogue to a preset level. The goal is to match volume level between program
+sources. A value of -31dB will result in no volume level change, relative to
+the source volume, during audio reproduction. Valid values are whole numbers in
+the range -31 to -1, with -31 being the default.
+
+@item -dsur_mode @var{mode}
+Dolby Surround Mode. Specifies whether the stereo signal uses Dolby Surround
+(Pro Logic). This field will only be written to the bitstream if the audio
+stream is stereo. Using this option does @b{NOT} mean the encoder will actually
+apply Dolby Surround processing.
+@table @option
+@item 0
+@itemx notindicated
+Not Indicated (default)
+@item 1
+@itemx off
+Not Dolby Surround Encoded
+@item 2
+@itemx on
+Dolby Surround Encoded
+@end table
+
+@item -original @var{boolean}
+Original Bit Stream Indicator. Specifies whether this audio is from the
+original source and not a copy.
+@table @option
+@item 0
+@itemx off
+Not Original Source
+@item 1
+@itemx on
+Original Source (default)
+@end table
+
+@end table
+
+@subsubheading Extended Bitstream Information
+The extended bitstream options are part of the Alternate Bit Stream Syntax as
+specified in Annex D of the A/52:2010 standard. It is grouped into 2 parts.
+If any one parameter in a group is specified, all values in that group will be
+written to the bitstream.  Default values are used for those that are written
+but have not been specified.  If the mixing levels are written, the decoder
+will use these values instead of the ones specified in the @code{center_mixlev}
+and @code{surround_mixlev} options if it supports the Alternate Bit Stream
+Syntax.
+
+@subsubheading Extended Bitstream Information - Part 1
+
+@table @option
+
+@item -dmix_mode @var{mode}
+Preferred Stereo Downmix Mode. Allows the user to select either Lt/Rt
+(Dolby Surround) or Lo/Ro (normal stereo) as the preferred stereo downmix mode.
+@table @option
+@item 0
+@itemx notindicated
+Not Indicated (default)
+@item 1
+@itemx ltrt
+Lt/Rt Downmix Preferred
+@item 2
+@itemx loro
+Lo/Ro Downmix Preferred
+@end table
+
+@item -ltrt_cmixlev @var{level}
+Lt/Rt Center Mix Level. The amount of gain the decoder should apply to the
+center channel when downmixing to stereo in Lt/Rt mode.
+@table @option
+@item 1.414
+Apply +3dB gain
+@item 1.189
+Apply +1.5dB gain
+@item 1.000
+Apply 0dB gain
+@item 0.841
+Apply -1.5dB gain
+@item 0.707
+Apply -3.0dB gain
+@item 0.595
+Apply -4.5dB gain (default)
+@item 0.500
+Apply -6.0dB gain
+@item 0.000
+Silence Center Channel
+@end table
+
+@item -ltrt_surmixlev @var{level}
+Lt/Rt Surround Mix Level. The amount of gain the decoder should apply to the
+surround channel(s) when downmixing to stereo in Lt/Rt mode.
+@table @option
+@item 0.841
+Apply -1.5dB gain
+@item 0.707
+Apply -3.0dB gain
+@item 0.595
+Apply -4.5dB gain
+@item 0.500
+Apply -6.0dB gain (default)
+@item 0.000
+Silence Surround Channel(s)
+@end table
+
+@item -loro_cmixlev @var{level}
+Lo/Ro Center Mix Level. The amount of gain the decoder should apply to the
+center channel when downmixing to stereo in Lo/Ro mode.
+@table @option
+@item 1.414
+Apply +3dB gain
+@item 1.189
+Apply +1.5dB gain
+@item 1.000
+Apply 0dB gain
+@item 0.841
+Apply -1.5dB gain
+@item 0.707
+Apply -3.0dB gain
+@item 0.595
+Apply -4.5dB gain (default)
+@item 0.500
+Apply -6.0dB gain
+@item 0.000
+Silence Center Channel
+@end table
+
+@item -loro_surmixlev @var{level}
+Lo/Ro Surround Mix Level. The amount of gain the decoder should apply to the
+surround channel(s) when downmixing to stereo in Lo/Ro mode.
+@table @option
+@item 0.841
+Apply -1.5dB gain
+@item 0.707
+Apply -3.0dB gain
+@item 0.595
+Apply -4.5dB gain
+@item 0.500
+Apply -6.0dB gain (default)
+@item 0.000
+Silence Surround Channel(s)
+@end table
+
+@end table
+
+@subsubheading Extended Bitstream Information - Part 2
+
+@table @option
+
+@item -dsurex_mode @var{mode}
+Dolby Surround EX Mode. Indicates whether the stream uses Dolby Surround EX
+(7.1 matrixed to 5.1). Using this option does @b{NOT} mean the encoder will actually
+apply Dolby Surround EX processing.
+@table @option
+@item 0
+@itemx notindicated
+Not Indicated (default)
+@item 1
+@itemx on
+Dolby Surround EX On
+@item 2
+@itemx off
+Dolby Surround EX Off
+@end table
+
+@item -dheadphone_mode @var{mode}
+Dolby Headphone Mode. Indicates whether the stream uses Dolby Headphone
+encoding (multi-channel matrixed to 2.0 for use with headphones). Using this
+option does @b{NOT} mean the encoder will actually apply Dolby Headphone
+processing.
+@table @option
+@item 0
+@itemx notindicated
+Not Indicated (default)
+@item 1
+@itemx on
+Dolby Headphone On
+@item 2
+@itemx off
+Dolby Headphone Off
+@end table
+
+@item -ad_conv_type @var{type}
+A/D Converter Type. Indicates whether the audio has passed through HDCD A/D
+conversion.
+@table @option
+@item 0
+@itemx standard
+Standard A/D Converter (default)
+@item 1
+@itemx hdcd
+HDCD A/D Converter
+@end table
+
+@end table
+
 @c man end ENCODERS

doc/indevs.texi

@@ -154,6 +154,23 @@ ffmpeg -f oss -i /dev/dsp /tmp/oss.wav
 For more information about OSS see:
 @url{http://manuals.opensound.com/usersguide/dsp.html}
 
+@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
+@file{/dev/audio0}.
+
+For example to grab from @file{/dev/audio0} using @file{ffmpeg} use the
+command:
+@example
+ffmpeg -f sndio -i /dev/audio0 /tmp/oss.wav
+@end example
+
 @section video4linux and video4linux2
 
 Video4Linux and Video4Linux2 input video devices.

doc/outdevs.texi

@@ -26,4 +26,8 @@ ALSA (Advanced Linux Sound Architecture) output device.
 
 OSS (Open Sound System) output device.
 
+@section sndio
+
+sndio audio output device.
+
 @c man end OUTPUT DEVICES

libavcodec/ac3.h

@@ -48,6 +48,17 @@
 #define EXP_D25   2
 #define EXP_D45   3
 
+/* pre-defined gain values */
+#define LEVEL_PLUS_3DB          1.4142135623730950
+#define LEVEL_PLUS_1POINT5DB    1.1892071150027209
+#define LEVEL_MINUS_1POINT5DB   0.8408964152537145
+#define LEVEL_MINUS_3DB         0.7071067811865476
+#define LEVEL_MINUS_4POINT5DB   0.5946035575013605
+#define LEVEL_MINUS_6DB         0.5000000000000000
+#define LEVEL_MINUS_9DB         0.3535533905932738
+#define LEVEL_ZERO              0.0000000000000000
+#define LEVEL_ONE               1.0000000000000000
+
 /** Delta bit allocation strategy */
 typedef enum {
     DBA_REUSE = 0,

libavcodec/ac3dec.c

@@ -67,16 +67,6 @@ static const uint8_t quantization_tab[16] = {
 static float dynamic_range_tab[256];
 
 /** Adjustments in dB gain */
-#define LEVEL_PLUS_3DB          1.4142135623730950
-#define LEVEL_PLUS_1POINT5DB    1.1892071150027209
-#define LEVEL_MINUS_1POINT5DB   0.8408964152537145
-#define LEVEL_MINUS_3DB         0.7071067811865476
-#define LEVEL_MINUS_4POINT5DB   0.5946035575013605
-#define LEVEL_MINUS_6DB         0.5000000000000000
-#define LEVEL_MINUS_9DB         0.3535533905932738
-#define LEVEL_ZERO              0.0000000000000000
-#define LEVEL_ONE               1.0000000000000000
-
 static const float gain_levels[9] = {
     LEVEL_PLUS_3DB,
     LEVEL_PLUS_1POINT5DB,

libavcodec/ac3enc_fixed.c

@@ -410,5 +410,6 @@ AVCodec ff_ac3_fixed_encoder = {
     NULL,
     .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
     .long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
+    .priv_class = &ac3enc_class,
     .channel_layouts = ac3_channel_layouts,
 };

libavcodec/ac3enc_float.c

@@ -120,5 +120,6 @@ AVCodec ff_ac3_encoder = {
     NULL,
     .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE},
     .long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
+    .priv_class = &ac3enc_class,
     .channel_layouts = ac3_channel_layouts,
 };

libavdevice/Makefile

@@ -18,6 +18,8 @@ OBJS-$(CONFIG_FBDEV_INDEV)               += fbdev.o
 OBJS-$(CONFIG_JACK_INDEV)                += jack_audio.o
 OBJS-$(CONFIG_OSS_INDEV)                 += oss_audio.o
 OBJS-$(CONFIG_OSS_OUTDEV)                += oss_audio.o
+OBJS-$(CONFIG_SNDIO_INDEV)               += sndio_common.o sndio_dec.o
+OBJS-$(CONFIG_SNDIO_OUTDEV)              += sndio_common.o sndio_enc.o
 OBJS-$(CONFIG_V4L2_INDEV)                += v4l2.o
 OBJS-$(CONFIG_V4L_INDEV)                 += v4l.o
 OBJS-$(CONFIG_VFWCAP_INDEV)              += vfwcap.o
@@ -27,5 +29,6 @@ OBJS-$(CONFIG_X11_GRAB_DEVICE_INDEV)     += x11grab.o
 OBJS-$(CONFIG_LIBDC1394_INDEV)           += libdc1394.o
 
 SKIPHEADERS-$(HAVE_ALSA_ASOUNDLIB_H)     += alsa-audio.h
+SKIPHEADERS-$(HAVE_SNDIO_H)              += sndio_common.h
 
 include $(SUBDIR)../subdir.mak

libavdevice/alldevices.c

@@ -45,6 +45,7 @@ void avdevice_register_all(void)
     REGISTER_INDEV    (FBDEV, fbdev);
     REGISTER_INDEV    (JACK, jack);
     REGISTER_INOUTDEV (OSS, oss);
+    REGISTER_INOUTDEV (SNDIO, sndio);
     REGISTER_INDEV    (V4L2, v4l2);
     REGISTER_INDEV    (V4L, v4l);
     REGISTER_INDEV    (VFWCAP, vfwcap);

libavdevice/sndio_common.c

+/*
+ * sndio play and grab interface
+ * Copyright (c) 2010 Jacob Meuser
+ *
+ * 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
+ */
+
+#include <stdint.h>
+#include <sndio.h>
+
+#include "libavformat/avformat.h"
+
+#include "sndio_common.h"
+
+static inline void movecb(void *addr, int delta)
+{
+    SndioData *s = addr;
+
+    s->hwpos += delta * s->channels * s->bps;
+}
+
+av_cold int ff_sndio_open(AVFormatContext *s1, int is_output,
+                          const char *audio_device)
+{
+    SndioData *s = s1->priv_data;
+    struct sio_hdl *hdl;
+    struct sio_par par;
+
+    hdl = sio_open(audio_device, is_output ? SIO_PLAY : SIO_REC, 0);
+    if (!hdl) {
+        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;
+    par.rate = s->sample_rate;
+
+    if (!sio_setpar(hdl, &par) || !sio_getpar(hdl, &par)) {
+        av_log(s1, AV_LOG_ERROR, "Impossible to set sndio parameters, "
+               "channels: %d sample rate: %d\n", s->channels, s->sample_rate);
+        goto fail;
+    }
+
+    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, "
+               "channels: %d sample rate: %d\n", s->channels, s->sample_rate);
+        goto fail;
+    }
+
+    s->buffer_size = par.round * par.bps *
+                     (is_output ? par.pchan : par.rchan);
+
+    if (is_output) {
+        s->buffer = av_malloc(s->buffer_size);
+        if (!s->buffer) {
+            av_log(s1, AV_LOG_ERROR, "Could not allocate buffer\n");
+            goto fail;
+        }
+    }
+
+    s->codec_id    = par.le ? CODEC_ID_PCM_S16LE : CODEC_ID_PCM_S16BE;
+    s->channels    = is_output ? par.pchan : par.rchan;
+    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:
+    av_freep(&s->buffer);
+
+    if (hdl)
+        sio_close(hdl);
+
+    return AVERROR(EIO);
+}
+
+int ff_sndio_close(SndioData *s)
+{
+    av_freep(&s->buffer);
+
+    if (s->hdl)
+        sio_close(s->hdl);
+
+    return 0;
+}

libavdevice/sndio_common.h

+/*
+ * sndio play and grab interface
+ * Copyright (c) 2010 Jacob Meuser
+ *
+ * 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
+ */
+
+#ifndef AVDEVICE_SNDIO_COMMON_H
+#define AVDEVICE_SNDIO_COMMON_H
+
+#include <stdint.h>
+#include <sndio.h>
+
+#include "libavformat/avformat.h"
+
+typedef struct {
+    struct sio_hdl *hdl;
+    enum CodecID codec_id;
+    int64_t hwpos;
+    int64_t softpos;
+    uint8_t *buffer;
+    int bps;
+    int buffer_size;
+    int buffer_offset;
+    int channels;
+    int sample_rate;
+} SndioData;
+
+int ff_sndio_open(AVFormatContext *s1, int is_output, const char *audio_device);
+int ff_sndio_close(SndioData *s);
+
+#endif /* AVDEVICE_SNDIO_COMMON_H */

libavdevice/sndio_dec.c

+/*
+ * sndio play and grab interface
+ * Copyright (c) 2010 Jacob Meuser
+ *
+ * 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
+ */
+
+#include <stdint.h>
+#include <sndio.h>
+
+#include "libavformat/avformat.h"
+
+#include "sndio_common.h"
+
+static av_cold int audio_read_header(AVFormatContext *s1,
+                                     AVFormatParameters *ap)
+{
+    SndioData *s = s1->priv_data;
+    AVStream *st;
+    int ret;
+
+    if (ap->sample_rate <= 0 || ap->channels <= 0)
+        return AVERROR(EINVAL);
+
+    st = av_new_stream(s1, 0);
+    if (!st)
+        return AVERROR(ENOMEM);
+
+    s->sample_rate = ap->sample_rate;
+    s->channels    = ap->channels;
+
+    ret = ff_sndio_open(s1, 0, s1->filename);
+    if (ret < 0)
+        return ret;
+
+    /* 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)
+{
+    SndioData *s = s1->priv_data;
+    int64_t bdelay, cur_time;
+    int ret;
+
+    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);
+        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 pts */
+    pkt->pts = cur_time - ((bdelay * 1000000) /
+        (s->bps * s->channels * s->sample_rate));
+
+    return 0;
+}
+
+static av_cold int audio_read_close(AVFormatContext *s1)
+{
+    SndioData *s = s1->priv_data;
+
+    ff_sndio_close(s);
+
+    return 0;
+}
+
+AVInputFormat ff_sndio_demuxer = {
+    .name           = "sndio",
+    .long_name      = NULL_IF_CONFIG_SMALL("sndio audio capture"),
+    .priv_data_size = sizeof(SndioData),
+    .read_header    = audio_read_header,
+    .read_packet    = audio_read_packet,
+    .read_close     = audio_read_close,
+    .flags          = AVFMT_NOFILE,
+};

libavdevice/sndio_enc.c

+/*
+ * sndio play and grab interface
+ * Copyright (c) 2010 Jacob Meuser
+ *
+ * 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
+ */
+
+#include <stdint.h>
+#include <sndio.h>
+
+#include "libavformat/avformat.h"
+
+#include "sndio_common.h"
+
+static av_cold int audio_write_header(AVFormatContext *s1)
+{
+    SndioData *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 = ff_sndio_open(s1, 1, s1->filename);
+
+    return ret;
+}
+
+static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
+{
+    SndioData *s = s1->priv_data;
+    uint8_t *buf= pkt->data;
+    int size = pkt->size;
+    int len, ret;
+
+    while (size > 0) {
+        len = s->buffer_size - s->buffer_offset;
+        if (len > size)
+            len = size;
+        memcpy(s->buffer + s->buffer_offset, buf, len);
+        buf  += len;
+        size -= len;
+        s->buffer_offset += len;
+        if (s->buffer_offset >= 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_offset = 0;
+        }
+    }
+
+    return 0;
+}
+
+static int audio_write_trailer(AVFormatContext *s1)
+{
+    SndioData *s = s1->priv_data;
+
+    sio_write(s->hdl, s->buffer, s->buffer_offset);
+
+    ff_sndio_close(s);
+
+    return 0;
+}
+
+AVOutputFormat ff_sndio_muxer = {
+    .name           = "sndio",
+    .long_name      = NULL_IF_CONFIG_SMALL("sndio audio playback"),
+    .priv_data_size = sizeof(SndioData),
+    /* XXX: we make the assumption that the soundcard accepts this format */
+    /* XXX: find better solution with "preinit" method, needed also in
+       other formats */
+    .audio_codec    = AV_NE(CODEC_ID_PCM_S16BE, CODEC_ID_PCM_S16LE),
+    .video_codec    = CODEC_ID_NONE,
+    .write_header   = audio_write_header,
+    .write_packet   = audio_write_packet,
+    .write_trailer  = audio_write_trailer,
+    .flags          = AVFMT_NOFILE,
+};