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: Michael Niedermayer <michaelni@gmx.at>

16 files changed, 1183 insertions, 18 deletions

diff --git a/Changelog b/Changelog
index c4e2194979..c91faabfe6 100644
--- a/Changelog
+++ b/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:
diff --git a/configure b/configure
index 54973e959b..1750141a34 100755
--- a/configure
+++ b/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      &&
diff --git a/doc/encoders.texi b/doc/encoders.texi
index cab98fb0bd..2f347f4fb1 100644
--- a/doc/encoders.texi
+++ b/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
diff --git a/doc/indevs.texi b/doc/indevs.texi
index 1cd2dd63cb..5a8a8fa9b0 100644
--- a/doc/indevs.texi
+++ b/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.
diff --git a/doc/outdevs.texi b/doc/outdevs.texi
index 3c0acee984..fbb312363c 100644
--- a/doc/outdevs.texi
+++ b/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
diff --git a/libavcodec/ac3.h b/libavcodec/ac3.h
index 1a8cce9a6d..b4092c4be6 100644
--- a/libavcodec/ac3.h
+++ b/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,
diff --git a/libavcodec/ac3dec.c b/libavcodec/ac3dec.c
index c4365170f9..b1f09f28b1 100644
--- a/libavcodec/ac3dec.c
+++ b/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,
diff --git a/libavcodec/ac3enc.c b/libavcodec/ac3enc.c
index f41fd2da61..debd63d56a 100644
--- a/libavcodec/ac3enc.c
+++ b/libavcodec/ac3enc.c
@@ -32,6 +32,7 @@
 #include "libavutil/audioconvert.h"
 #include "libavutil/avassert.h"
 #include "libavutil/crc.h"
+#include "libavutil/opt.h"
 #include "avcodec.h"
 #include "put_bits.h"
 #include "dsputil.h"
@@ -66,6 +67,36 @@
 
 
 /**
+ * Encoding Options used by AVOption.
+ */
+typedef struct AC3EncOptions {
+    /* AC-3 metadata options*/
+    int dialogue_level;
+    int bitstream_mode;
+    float center_mix_level;
+    float surround_mix_level;
+    int dolby_surround_mode;
+    int audio_production_info;
+    int mixing_level;
+    int room_type;
+    int copyright;
+    int original;
+    int extended_bsi_1;
+    int preferred_stereo_downmix;
+    float ltrt_center_mix_level;
+    float ltrt_surround_mix_level;
+    float loro_center_mix_level;
+    float loro_surround_mix_level;
+    int extended_bsi_2;
+    int dolby_surround_ex_mode;
+    int dolby_headphone_mode;
+    int ad_converter_type;
+
+    /* other encoding options */
+    int allow_per_frame_metadata;
+} AC3EncOptions;
+
+/**
  * Data for a single audio block.
  */
 typedef struct AC3Block {
@@ -87,6 +118,8 @@ typedef struct AC3Block {
  * AC-3 encoder private context.
  */
 typedef struct AC3EncodeContext {
+    AVClass *av_class;                      ///< AVClass used for AVOption
+    AC3EncOptions options;                  ///< encoding options
     PutBitContext pb;                       ///< bitstream writer context
     DSPContext dsp;
     AC3DSPContext ac3dsp;                   ///< AC-3 optimized functions
@@ -111,9 +144,18 @@ typedef struct AC3EncodeContext {
     int channels;                           ///< total number of channels               (nchans)
     int lfe_on;                             ///< indicates if there is an LFE channel   (lfeon)
     int lfe_channel;                        ///< channel index of the LFE channel
+    int has_center;                         ///< indicates if there is a center channel
+    int has_surround;                       ///< indicates if there are one or more surround channels
     int channel_mode;                       ///< channel mode                           (acmod)
     const uint8_t *channel_map;             ///< channel map used to reorder channels
 
+    int center_mix_level;                   ///< center mix level code
+    int surround_mix_level;                 ///< surround mix level code
+    int ltrt_center_mix_level;              ///< Lt/Rt center mix level code
+    int ltrt_surround_mix_level;            ///< Lt/Rt surround mix level code
+    int loro_center_mix_level;              ///< Lo/Ro center mix level code
+    int loro_surround_mix_level;            ///< Lo/Ro surround mix level code
+
     int cutoff;                             ///< user-specified cutoff frequency, in Hz
     int bandwidth_code[AC3_MAX_CHANNELS];   ///< bandwidth code (0 to 60)               (chbwcod)
     int nb_coefs[AC3_MAX_CHANNELS];
@@ -157,6 +199,78 @@ typedef struct AC3EncodeContext {
 } AC3EncodeContext;
 
 
+#define CMIXLEV_NUM_OPTIONS 3
+static const float cmixlev_options[CMIXLEV_NUM_OPTIONS] = {
+    LEVEL_MINUS_3DB, LEVEL_MINUS_4POINT5DB, LEVEL_MINUS_6DB
+};
+
+#define SURMIXLEV_NUM_OPTIONS 3
+static const float surmixlev_options[SURMIXLEV_NUM_OPTIONS] = {
+    LEVEL_MINUS_3DB, LEVEL_MINUS_6DB, LEVEL_ZERO
+};
+
+#define EXTMIXLEV_NUM_OPTIONS 8
+static const float extmixlev_options[EXTMIXLEV_NUM_OPTIONS] = {
+    LEVEL_PLUS_3DB,  LEVEL_PLUS_1POINT5DB,  LEVEL_ONE,       LEVEL_MINUS_4POINT5DB,
+    LEVEL_MINUS_3DB, LEVEL_MINUS_4POINT5DB, LEVEL_MINUS_6DB, LEVEL_ZERO
+};
+
+
+#define OFFSET(param) offsetof(AC3EncodeContext, options.param)
+#define AC3ENC_PARAM (AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM)
+
+static const AVOption options[] = {
+/* Metadata Options */
+{"per_frame_metadata", "Allow Changing Metadata Per-Frame", OFFSET(allow_per_frame_metadata), FF_OPT_TYPE_INT, 0, 0, 1, AC3ENC_PARAM},
+/* downmix levels */
+{"center_mixlev", "Center Mix Level", OFFSET(center_mix_level), FF_OPT_TYPE_FLOAT, LEVEL_MINUS_4POINT5DB, 0.0, 1.0, AC3ENC_PARAM},
+{"surround_mixlev", "Surround Mix Level", OFFSET(surround_mix_level), FF_OPT_TYPE_FLOAT, LEVEL_MINUS_6DB, 0.0, 1.0, AC3ENC_PARAM},
+/* audio production information */
+{"mixing_level", "Mixing Level", OFFSET(mixing_level), FF_OPT_TYPE_INT, -1, -1, 111, AC3ENC_PARAM},
+{"room_type", "Room Type", OFFSET(room_type), FF_OPT_TYPE_INT, -1, -1, 2, AC3ENC_PARAM, "room_type"},
+    {"notindicated", "Not Indicated (default)", 0, FF_OPT_TYPE_CONST, 0, INT_MIN, INT_MAX, AC3ENC_PARAM, "room_type"},
+    {"large",        "Large Room",              0, FF_OPT_TYPE_CONST, 1, INT_MIN, INT_MAX, AC3ENC_PARAM, "room_type"},
+    {"small",        "Small Room",              0, FF_OPT_TYPE_CONST, 2, INT_MIN, INT_MAX, AC3ENC_PARAM, "room_type"},
+/* other metadata options */
+{"copyright", "Copyright Bit", OFFSET(copyright), FF_OPT_TYPE_INT, 0, 0, 1, AC3ENC_PARAM},
+{"dialnorm", "Dialogue Level (dB)", OFFSET(dialogue_level), FF_OPT_TYPE_INT, -31, -31, -1, AC3ENC_PARAM},
+{"dsur_mode", "Dolby Surround Mode", OFFSET(dolby_surround_mode), FF_OPT_TYPE_INT, 0, 0, 2, AC3ENC_PARAM, "dsur_mode"},
+    {"notindicated", "Not Indicated (default)",    0, FF_OPT_TYPE_CONST, 0, INT_MIN, INT_MAX, AC3ENC_PARAM, "dsur_mode"},
+    {"on",           "Dolby Surround Encoded",     0, FF_OPT_TYPE_CONST, 1, INT_MIN, INT_MAX, AC3ENC_PARAM, "dsur_mode"},
+    {"off",          "Not Dolby Surround Encoded", 0, FF_OPT_TYPE_CONST, 2, INT_MIN, INT_MAX, AC3ENC_PARAM, "dsur_mode"},
+{"original", "Original Bit Stream", OFFSET(original), FF_OPT_TYPE_INT, 1, 0, 1, AC3ENC_PARAM},
+/* extended bitstream information */
+{"dmix_mode", "Preferred Stereo Downmix Mode", OFFSET(preferred_stereo_downmix), FF_OPT_TYPE_INT, -1, -1, 2, AC3ENC_PARAM, "dmix_mode"},
+    {"notindicated", "Not Indicated (default)", 0, FF_OPT_TYPE_CONST, 0, INT_MIN, INT_MAX, AC3ENC_PARAM, "dmix_mode"},
+    {"ltrt", "Lt/Rt Downmix Preferred",         0, FF_OPT_TYPE_CONST, 1, INT_MIN, INT_MAX, AC3ENC_PARAM, "dmix_mode"},
+    {"loro", "Lo/Ro Downmix Preferred",         0, FF_OPT_TYPE_CONST, 2, INT_MIN, INT_MAX, AC3ENC_PARAM, "dmix_mode"},
+{"ltrt_cmixlev", "Lt/Rt Center Mix Level", OFFSET(ltrt_center_mix_level), FF_OPT_TYPE_FLOAT, -1.0, -1.0, 2.0, AC3ENC_PARAM},
+{"ltrt_surmixlev", "Lt/Rt Surround Mix Level", OFFSET(ltrt_surround_mix_level), FF_OPT_TYPE_FLOAT, -1.0, -1.0, 2.0, AC3ENC_PARAM},
+{"loro_cmixlev", "Lo/Ro Center Mix Level", OFFSET(loro_center_mix_level), FF_OPT_TYPE_FLOAT, -1.0, -1.0, 2.0, AC3ENC_PARAM},
+{"loro_surmixlev", "Lo/Ro Surround Mix Level", OFFSET(loro_surround_mix_level), FF_OPT_TYPE_FLOAT, -1.0, -1.0, 2.0, AC3ENC_PARAM},
+{"dsurex_mode", "Dolby Surround EX Mode", OFFSET(dolby_surround_ex_mode), FF_OPT_TYPE_INT, -1, -1, 2, AC3ENC_PARAM, "dsurex_mode"},
+    {"notindicated", "Not Indicated (default)",       0, FF_OPT_TYPE_CONST, 0, INT_MIN, INT_MAX, AC3ENC_PARAM, "dsurex_mode"},
+    {"on",           "Dolby Surround EX Encoded",     0, FF_OPT_TYPE_CONST, 1, INT_MIN, INT_MAX, AC3ENC_PARAM, "dsurex_mode"},
+    {"off",          "Not Dolby Surround EX Encoded", 0, FF_OPT_TYPE_CONST, 2, INT_MIN, INT_MAX, AC3ENC_PARAM, "dsurex_mode"},
+{"dheadphone_mode", "Dolby Headphone Mode", OFFSET(dolby_headphone_mode), FF_OPT_TYPE_INT, -1, -1, 2, AC3ENC_PARAM, "dheadphone_mode"},
+    {"notindicated", "Not Indicated (default)",     0, FF_OPT_TYPE_CONST, 0, INT_MIN, INT_MAX, AC3ENC_PARAM, "dheadphone_mode"},
+    {"on",           "Dolby Headphone Encoded",     0, FF_OPT_TYPE_CONST, 1, INT_MIN, INT_MAX, AC3ENC_PARAM, "dheadphone_mode"},
+    {"off",          "Not Dolby Headphone Encoded", 0, FF_OPT_TYPE_CONST, 2, INT_MIN, INT_MAX, AC3ENC_PARAM, "dheadphone_mode"},
+{"ad_conv_type", "A/D Converter Type", OFFSET(ad_converter_type), FF_OPT_TYPE_INT, -1, -1, 1, AC3ENC_PARAM, "ad_conv_type"},
+    {"standard", "Standard (default)", 0, FF_OPT_TYPE_CONST, 0, INT_MIN, INT_MAX, AC3ENC_PARAM, "ad_conv_type"},
+    {"hdcd",     "HDCD",               0, FF_OPT_TYPE_CONST, 1, INT_MIN, INT_MAX, AC3ENC_PARAM, "ad_conv_type"},
+{NULL}
+};
+
+#if CONFIG_AC3ENC_FLOAT
+static AVClass ac3enc_class = { "AC-3 Encoder", av_default_item_name,
+                                options, LIBAVUTIL_VERSION_INT };
+#else
+static AVClass ac3enc_class = { "Fixed-Point AC-3 Encoder", av_default_item_name,
+                                options, LIBAVUTIL_VERSION_INT };
+#endif
+
+
 /* prototypes for functions in ac3enc_fixed.c and ac3enc_float.c */
 
 static av_cold void mdct_end(AC3MDCTContext *mdct);
@@ -786,9 +900,19 @@ static void bit_alloc_init(AC3EncodeContext *s)
  */
 static void count_frame_bits(AC3EncodeContext *s)
 {
+    AC3EncOptions *opt = &s->options;
     int blk, ch;
     int frame_bits = 0;
 
+    if (opt->audio_production_info)
+        frame_bits += 7;
+    if (s->bitstream_id == 6) {
+        if (opt->extended_bsi_1)
+            frame_bits += 14;
+        if (opt->extended_bsi_2)
+            frame_bits += 14;
+    }
+
     for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
         /* stereo rematrixing */
         if (s->channel_mode == AC3_CHMODE_STEREO &&
@@ -1245,6 +1369,8 @@ static void quantize_mantissas(AC3EncodeContext *s)
  */
 static void output_frame_header(AC3EncodeContext *s)
 {
+    AC3EncOptions *opt = &s->options;
+
     put_bits(&s->pb, 16, 0x0b77);   /* frame header */
     put_bits(&s->pb, 16, 0);        /* crc1: will be filled later */
     put_bits(&s->pb, 2,  s->bit_alloc.sr_code);
@@ -1253,20 +1379,43 @@ static void output_frame_header(AC3EncodeContext *s)
     put_bits(&s->pb, 3,  s->bitstream_mode);
     put_bits(&s->pb, 3,  s->channel_mode);
     if ((s->channel_mode & 0x01) && s->channel_mode != AC3_CHMODE_MONO)
-        put_bits(&s->pb, 2, 1);     /* XXX -4.5 dB */
+        put_bits(&s->pb, 2, s->center_mix_level);
     if (s->channel_mode & 0x04)
-        put_bits(&s->pb, 2, 1);     /* XXX -6 dB */
+        put_bits(&s->pb, 2, s->surround_mix_level);
     if (s->channel_mode == AC3_CHMODE_STEREO)
-        put_bits(&s->pb, 2, 0);     /* surround not indicated */
+        put_bits(&s->pb, 2, opt->dolby_surround_mode);
     put_bits(&s->pb, 1, s->lfe_on); /* LFE */
-    put_bits(&s->pb, 5, 31);        /* dialog norm: -31 db */
+    put_bits(&s->pb, 5, -opt->dialogue_level);
     put_bits(&s->pb, 1, 0);         /* no compression control word */
     put_bits(&s->pb, 1, 0);         /* no lang code */
-    put_bits(&s->pb, 1, 0);         /* no audio production info */
-    put_bits(&s->pb, 1, 0);         /* no copyright */
-    put_bits(&s->pb, 1, 1);         /* original bitstream */
+    put_bits(&s->pb, 1, opt->audio_production_info);
+    if (opt->audio_production_info) {
+        put_bits(&s->pb, 5, opt->mixing_level - 80);
+        put_bits(&s->pb, 2, opt->room_type);
+    }
+    put_bits(&s->pb, 1, opt->copyright);
+    put_bits(&s->pb, 1, opt->original);
+    if (s->bitstream_id == 6) {
+        /* alternate bit stream syntax */
+        put_bits(&s->pb, 1, opt->extended_bsi_1);
+        if (opt->extended_bsi_1) {
+            put_bits(&s->pb, 2, opt->preferred_stereo_downmix);
+            put_bits(&s->pb, 3, s->ltrt_center_mix_level);
+            put_bits(&s->pb, 3, s->ltrt_surround_mix_level);
+            put_bits(&s->pb, 3, s->loro_center_mix_level);
+            put_bits(&s->pb, 3, s->loro_surround_mix_level);
+        }
+        put_bits(&s->pb, 1, opt->extended_bsi_2);
+        if (opt->extended_bsi_2) {
+            put_bits(&s->pb, 2, opt->dolby_surround_ex_mode);
+            put_bits(&s->pb, 2, opt->dolby_headphone_mode);
+            put_bits(&s->pb, 1, opt->ad_converter_type);
+            put_bits(&s->pb, 9, 0);     /* xbsi2 and encinfo : reserved */
+        }
+    } else {
     put_bits(&s->pb, 1, 0);         /* no time code 1 */
     put_bits(&s->pb, 1, 0);         /* no time code 2 */
+    }
     put_bits(&s->pb, 1, 0);         /* no additional bit stream info */
 }
 
@@ -1479,6 +1628,268 @@ static void output_frame(AC3EncodeContext *s, unsigned char *frame)
 }
 
 
+static void dprint_options(AVCodecContext *avctx)
+{
+#ifdef DEBUG
+    AC3EncodeContext *s = avctx->priv_data;
+    AC3EncOptions *opt = &s->options;
+    char strbuf[32];
+
+    switch (s->bitstream_id) {
+    case  6:  strncpy(strbuf, "AC-3 (alt syntax)", 32);      break;
+    case  8:  strncpy(strbuf, "AC-3 (standard)", 32);        break;
+    case  9:  strncpy(strbuf, "AC-3 (dnet half-rate)", 32);  break;
+    case 10:  strncpy(strbuf, "AC-3 (dnet quater-rate", 32); break;
+    default: snprintf(strbuf, 32, "ERROR");
+    }
+    av_dlog(avctx, "bitstream_id: %s (%d)\n", strbuf, s->bitstream_id);
+    av_dlog(avctx, "sample_fmt: %s\n", av_get_sample_fmt_name(avctx->sample_fmt));
+    av_get_channel_layout_string(strbuf, 32, s->channels, avctx->channel_layout);
+    av_dlog(avctx, "channel_layout: %s\n", strbuf);
+    av_dlog(avctx, "sample_rate: %d\n", s->sample_rate);
+    av_dlog(avctx, "bit_rate: %d\n", s->bit_rate);
+    if (s->cutoff)
+        av_dlog(avctx, "cutoff: %d\n", s->cutoff);
+
+    av_dlog(avctx, "per_frame_metadata: %s\n",
+            opt->allow_per_frame_metadata?"on":"off");
+    if (s->has_center)
+        av_dlog(avctx, "center_mixlev: %0.3f (%d)\n", opt->center_mix_level,
+                s->center_mix_level);
+    else
+        av_dlog(avctx, "center_mixlev: {not written}\n");
+    if (s->has_surround)
+        av_dlog(avctx, "surround_mixlev: %0.3f (%d)\n", opt->surround_mix_level,
+                s->surround_mix_level);
+    else
+        av_dlog(avctx, "surround_mixlev: {not written}\n");
+    if (opt->audio_production_info) {
+        av_dlog(avctx, "mixing_level: %ddB\n", opt->mixing_level);
+        switch (opt->room_type) {
+        case 0:  strncpy(strbuf, "notindicated", 32); break;
+        case 1:  strncpy(strbuf, "large", 32);        break;
+        case 2:  strncpy(strbuf, "small", 32);        break;
+        default: snprintf(strbuf, 32, "ERROR (%d)", opt->room_type);
+        }
+        av_dlog(avctx, "room_type: %s\n", strbuf);
+    } else {
+        av_dlog(avctx, "mixing_level: {not written}\n");
+        av_dlog(avctx, "room_type: {not written}\n");
+    }
+    av_dlog(avctx, "copyright: %s\n", opt->copyright?"on":"off");
+    av_dlog(avctx, "dialnorm: %ddB\n", opt->dialogue_level);
+    if (s->channel_mode == AC3_CHMODE_STEREO) {
+        switch (opt->dolby_surround_mode) {
+        case 0:  strncpy(strbuf, "notindicated", 32); break;
+        case 1:  strncpy(strbuf, "on", 32);           break;
+        case 2:  strncpy(strbuf, "off", 32);          break;
+        default: snprintf(strbuf, 32, "ERROR (%d)", opt->dolby_surround_mode);
+        }
+        av_dlog(avctx, "dsur_mode: %s\n", strbuf);
+    } else {
+        av_dlog(avctx, "dsur_mode: {not written}\n");
+    }
+    av_dlog(avctx, "original: %s\n", opt->original?"on":"off");
+
+    if (s->bitstream_id == 6) {
+        if (opt->extended_bsi_1) {
+            switch (opt->preferred_stereo_downmix) {
+            case 0:  strncpy(strbuf, "notindicated", 32); break;
+            case 1:  strncpy(strbuf, "ltrt", 32);         break;
+            case 2:  strncpy(strbuf, "loro", 32);         break;
+            default: snprintf(strbuf, 32, "ERROR (%d)", opt->preferred_stereo_downmix);
+            }
+            av_dlog(avctx, "dmix_mode: %s\n", strbuf);
+            av_dlog(avctx, "ltrt_cmixlev: %0.3f (%d)\n",
+                    opt->ltrt_center_mix_level, s->ltrt_center_mix_level);
+            av_dlog(avctx, "ltrt_surmixlev: %0.3f (%d)\n",
+                    opt->ltrt_surround_mix_level, s->ltrt_surround_mix_level);
+            av_dlog(avctx, "loro_cmixlev: %0.3f (%d)\n",
+                    opt->loro_center_mix_level, s->loro_center_mix_level);
+            av_dlog(avctx, "loro_surmixlev: %0.3f (%d)\n",
+                    opt->loro_surround_mix_level, s->loro_surround_mix_level);
+        } else {
+            av_dlog(avctx, "extended bitstream info 1: {not written}\n");
+        }
+        if (opt->extended_bsi_2) {
+            switch (opt->dolby_surround_ex_mode) {
+            case 0:  strncpy(strbuf, "notindicated", 32); break;
+            case 1:  strncpy(strbuf, "on", 32);           break;
+            case 2:  strncpy(strbuf, "off", 32);          break;
+            default: snprintf(strbuf, 32, "ERROR (%d)", opt->dolby_surround_ex_mode);
+            }
+            av_dlog(avctx, "dsurex_mode: %s\n", strbuf);
+            switch (opt->dolby_headphone_mode) {
+            case 0:  strncpy(strbuf, "notindicated", 32); break;
+            case 1:  strncpy(strbuf, "on", 32);           break;
+            case 2:  strncpy(strbuf, "off", 32);          break;
+            default: snprintf(strbuf, 32, "ERROR (%d)", opt->dolby_headphone_mode);
+            }
+            av_dlog(avctx, "dheadphone_mode: %s\n", strbuf);
+
+            switch (opt->ad_converter_type) {
+            case 0:  strncpy(strbuf, "standard", 32); break;
+            case 1:  strncpy(strbuf, "hdcd", 32);     break;
+            default: snprintf(strbuf, 32, "ERROR (%d)", opt->ad_converter_type);
+            }
+            av_dlog(avctx, "ad_conv_type: %s\n", strbuf);
+        } else {
+            av_dlog(avctx, "extended bitstream info 2: {not written}\n");
+        }
+    }
+#endif
+}
+
+
+#define FLT_OPTION_THRESHOLD 0.01
+
+static int validate_float_option(float v, const float *v_list, int v_list_size)
+{
+    int i;
+
+    for (i = 0; i < v_list_size; i++) {
+        if (v < (v_list[i] + FLT_OPTION_THRESHOLD) &&
+            v > (v_list[i] - FLT_OPTION_THRESHOLD))
+            break;
+    }
+    if (i == v_list_size)
+        return -1;
+
+    return i;
+}
+
+
+static void validate_mix_level(void *log_ctx, const char *opt_name,
+                               float *opt_param, const float *list,
+                               int list_size, int default_value, int min_value,
+                               int *ctx_param)
+{
+    int mixlev = validate_float_option(*opt_param, list, list_size);
+    if (mixlev < min_value) {
+        mixlev = default_value;
+        if (*opt_param >= 0.0) {
+            av_log(log_ctx, AV_LOG_WARNING, "requested %s is not valid. using "
+                   "default value: %0.3f\n", opt_name, list[mixlev]);
+        }
+    }
+    *opt_param = list[mixlev];
+    *ctx_param = mixlev;
+}
+
+
+/**
+ * Validate metadata options as set by AVOption system.
+ * These values can optionally be changed per-frame.
+ */
+static int validate_metadata(AVCodecContext *avctx)
+{
+    AC3EncodeContext *s = avctx->priv_data;
+    AC3EncOptions *opt = &s->options;
+
+    /* validate mixing levels */
+    if (s->has_center) {
+        validate_mix_level(avctx, "center_mix_level", &opt->center_mix_level,
+                           cmixlev_options, CMIXLEV_NUM_OPTIONS, 1, 0,
+                           &s->center_mix_level);
+    }
+    if (s->has_surround) {
+        validate_mix_level(avctx, "surround_mix_level", &opt->surround_mix_level,
+                           surmixlev_options, SURMIXLEV_NUM_OPTIONS, 1, 0,
+                           &s->surround_mix_level);
+    }
+
+    /* set audio production info flag */
+    if (opt->mixing_level >= 0 || opt->room_type >= 0) {
+        if (opt->mixing_level < 0) {
+            av_log(avctx, AV_LOG_ERROR, "mixing_level must be set if "
+                   "room_type is set\n");
+            return AVERROR(EINVAL);
+        }
+        if (opt->mixing_level < 80) {
+            av_log(avctx, AV_LOG_ERROR, "invalid mixing level. must be between "
+                   "80dB and 111dB\n");
+            return AVERROR(EINVAL);
+        }
+        /* default room type */
+        if (opt->room_type < 0)
+            opt->room_type = 0;
+        opt->audio_production_info = 1;
+    } else {
+        opt->audio_production_info = 0;
+    }
+
+    /* set extended bsi 1 flag */
+    if ((s->has_center || s->has_surround) &&
+        (opt->preferred_stereo_downmix >= 0 ||
+         opt->ltrt_center_mix_level   >= 0 ||
+         opt->ltrt_surround_mix_level >= 0 ||
+         opt->loro_center_mix_level   >= 0 ||
+         opt->loro_surround_mix_level >= 0)) {
+        /* default preferred stereo downmix */
+        if (opt->preferred_stereo_downmix < 0)
+            opt->preferred_stereo_downmix = 0;
+        /* validate Lt/Rt center mix level */
+        validate_mix_level(avctx, "ltrt_center_mix_level",
+                           &opt->ltrt_center_mix_level, extmixlev_options,
+                           EXTMIXLEV_NUM_OPTIONS, 5, 0,
+                           &s->ltrt_center_mix_level);
+        /* validate Lt/Rt surround mix level */
+        validate_mix_level(avctx, "ltrt_surround_mix_level",
+                           &opt->ltrt_surround_mix_level, extmixlev_options,
+                           EXTMIXLEV_NUM_OPTIONS, 6, 3,
+                           &s->ltrt_surround_mix_level);
+        /* validate Lo/Ro center mix level */
+        validate_mix_level(avctx, "loro_center_mix_level",
+                           &opt->loro_center_mix_level, extmixlev_options,
+                           EXTMIXLEV_NUM_OPTIONS, 5, 0,
+                           &s->loro_center_mix_level);
+        /* validate Lo/Ro surround mix level */
+        validate_mix_level(avctx, "loro_surround_mix_level",
+                           &opt->loro_surround_mix_level, extmixlev_options,
+                           EXTMIXLEV_NUM_OPTIONS, 6, 3,
+                           &s->loro_surround_mix_level);
+        opt->extended_bsi_1 = 1;
+    } else {
+        opt->extended_bsi_1 = 0;
+    }
+
+    /* set extended bsi 2 flag */
+    if (opt->dolby_surround_ex_mode >= 0 ||
+        opt->dolby_headphone_mode   >= 0 ||
+        opt->ad_converter_type      >= 0) {
+        /* default dolby surround ex mode */
+        if (opt->dolby_surround_ex_mode < 0)
+            opt->dolby_surround_ex_mode = 0;
+        /* default dolby headphone mode */
+        if (opt->dolby_headphone_mode < 0)
+            opt->dolby_headphone_mode = 0;
+        /* default A/D converter type */
+        if (opt->ad_converter_type < 0)
+            opt->ad_converter_type = 0;
+        opt->extended_bsi_2 = 1;
+    } else {
+        opt->extended_bsi_2 = 0;
+    }
+
+    /* set bitstream id for alternate bitstream syntax */
+    if (opt->extended_bsi_1 || opt->extended_bsi_2) {
+        if (s->bitstream_id > 8 && s->bitstream_id < 11) {
+            static int warn_once = 1;
+            if (warn_once) {
+                av_log(avctx, AV_LOG_WARNING, "alternate bitstream syntax is "
+                       "not compatible with reduced samplerates. writing of "
+                       "extended bitstream information will be disabled.\n");
+                warn_once = 0;
+            }
+        } else {
+            s->bitstream_id = 6;
+        }
+    }
+
+    return 0;
+}
+
+
 /**
  * Encode a single AC-3 frame.
  */
@@ -1489,6 +1900,12 @@ static int ac3_encode_frame(AVCodecContext *avctx, unsigned char *frame,
     const SampleType *samples = data;
     int ret;
 
+    if (s->options.allow_per_frame_metadata) {
+        ret = validate_metadata(avctx);
+        if (ret)
+            return ret;
+    }
+
     if (s->bit_alloc.sr_code == 1)
         adjust_frame_size(s);
 
@@ -1597,6 +2014,8 @@ static av_cold int set_channel_info(AC3EncodeContext *s, int channels,
     default:
         return AVERROR(EINVAL);
     }
+    s->has_center   = (s->channel_mode & 0x01) && s->channel_mode != AC3_CHMODE_MONO;
+    s->has_surround =  s->channel_mode & 0x04;
 
     s->channel_map  = ff_ac3_enc_channel_map[s->channel_mode][s->lfe_on];
     *channel_layout = ch_layout;
@@ -1635,6 +2054,7 @@ static av_cold int validate_options(AVCodecContext *avctx, AC3EncodeContext *s)
     s->sample_rate        = avctx->sample_rate;
     s->bit_alloc.sr_shift = i % 3;
     s->bit_alloc.sr_code  = i / 3;
+    s->bitstream_id       = 8 + s->bit_alloc.sr_shift;
 
     /* validate bit rate */
     for (i = 0; i < 19; i++) {
@@ -1669,6 +2089,10 @@ static av_cold int validate_options(AVCodecContext *avctx, AC3EncodeContext *s)
         return AVERROR(EINVAL);
     }
 
+    ret = validate_metadata(avctx);
+    if (ret)
+        return ret;
+
     return 0;
 }
 
@@ -1810,7 +2234,6 @@ static av_cold int ac3_encode_init(AVCodecContext *avctx)
     if (ret)
         return ret;
 
-    s->bitstream_id   = 8 + s->bit_alloc.sr_shift;
     s->bitstream_mode = avctx->audio_service_type;
     if (s->bitstream_mode == AV_AUDIO_SERVICE_TYPE_KARAOKE)
         s->bitstream_mode = 0x7;
@@ -1849,6 +2272,8 @@ static av_cold int ac3_encode_init(AVCodecContext *avctx)
     dsputil_init(&s->dsp, avctx);
     ff_ac3dsp_init(&s->ac3dsp, avctx->flags & CODEC_FLAG_BITEXACT);
 
+    dprint_options(avctx);
+
     return 0;
 init_fail:
     ac3_encode_close(avctx);
diff --git a/libavcodec/ac3enc_fixed.c b/libavcodec/ac3enc_fixed.c
index f682aa625f..e7942abe99 100644
--- a/libavcodec/ac3enc_fixed.c
+++ b/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,
 };
diff --git a/libavcodec/ac3enc_float.c b/libavcodec/ac3enc_float.c
index f5b01f7d6f..faed30da50 100644
--- a/libavcodec/ac3enc_float.c
+++ b/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,
 };
diff --git a/libavdevice/Makefile b/libavdevice/Makefile
index 472cb95f50..5cfc5e8ecc 100644
--- a/libavdevice/Makefile
+++ b/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
diff --git a/libavdevice/alldevices.c b/libavdevice/alldevices.c
index 0c000dcb86..a0c9b08c6f 100644
--- a/libavdevice/alldevices.c
+++ b/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);
diff --git a/libavdevice/sndio_common.c b/libavdevice/sndio_common.c
new file mode 100644
index 0000000000..60b7970051
--- /dev/null
+++ b/libavdevice/sndio_common.c
@@ -0,0 +1,120 @@
+/*
+ * 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;
+}
diff --git a/libavdevice/sndio_common.h b/libavdevice/sndio_common.h
new file mode 100644
index 0000000000..41c984ba79
--- /dev/null
+++ b/libavdevice/sndio_common.h
@@ -0,0 +1,46 @@
+/*
+ * 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 */
diff --git a/libavdevice/sndio_dec.c b/libavdevice/sndio_dec.c
new file mode 100644
index 0000000000..ff2adeb0af
--- /dev/null
+++ b/libavdevice/sndio_dec.c
@@ -0,0 +1,108 @@
+/*
+ * 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,
+};
diff --git a/libavdevice/sndio_enc.c b/libavdevice/sndio_enc.c
new file mode 100644
index 0000000000..6745ba4893
--- /dev/null
+++ b/libavdevice/sndio_enc.c
@@ -0,0 +1,95 @@
+/*
+ * 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,
+};