diff --git a/Changelog b/Changelog
index c4e21949791ff6c810573efe2d3e8ba07d16733e..c91faabfe68472603d553288286b2ecb64204653 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 54973e959ba3ca046df438014622dd81e1ac09f1..1750141a340c72514583cc0867f9c1559dc05ab6 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 cab98fb0bdfb3c5b198d64d4eecd7918f7a38f38..2f347f4fb1042c1859eb21e82abd096eefc8da69 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 1cd2dd63cbf8777b8ad9a2404eb57ec6b0cfa7df..5a8a8fa9b0509062aad8eabbccdf78e3a4a1bfbb 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 3c0acee984e30e63891e20168c4cc8701c41232b..fbb312363c96f282990178375b4d930548eb1fe1 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 1a8cce9a6d8dc42a3220e9149d8cc0c4b1ce298f..b4092c4be667369eb625a4b7f67773bf7cf5c1f8 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 c4365170f9ae5d5550069d82a25e8a6596f09b31..b1f09f28b1e82dae51dfb79e334fc251d4b76960 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 f41fd2da6143200b63940f2ee4d7235fabf2636b..debd63d56ab34aef2a6f7ee31eb5b281566a13de 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"
@@ -65,6 +66,36 @@
#endif
+/**
+ * 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.
*/
@@ -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 f682aa625f4e34e8ee0af04aac851d5b4e7003a8..e7942abe9985fe6d8a2ae886d40afb1cbd43bd7a 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 f5b01f7d6f81944c813d4dcbf7d73869d7e17bfb..faed30da5051e81a067340329cccb8c189f87426 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 472cb95f50c930bf9a2b5fa63304dd1db9727720..5cfc5e8ecc04472269e7beb43f18a488483518cc 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 0c000dcb864c59c9abfc7c4261744c3f1ab496ab..a0c9b08c6fbe9a84b3928d536a96303eb0de622f 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 0000000000000000000000000000000000000000..60b7970051a3ac0908e9013f1e8768884ae28310
--- /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 0000000000000000000000000000000000000000..41c984ba7972c55555281f38f0b0e177227d986c
--- /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 0000000000000000000000000000000000000000..ff2adeb0af449404918ba0654781ec8f1b3727dc
--- /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 0000000000000000000000000000000000000000..6745ba4893d1ec10fa454a219d66c469cb31bfd8
--- /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,
+};