diff options
author | Michael Niedermayer <michaelni@gmx.at> | 2011-03-28 01:50:36 +0200 |
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committer | Michael Niedermayer <michaelni@gmx.at> | 2011-03-28 01:50:36 +0200 |
commit | 25d8099beb4c7eb93539f9162af1336ef7130fed (patch) | |
tree | 4f668222da738409baf5ad8a1ee9d1c9f9004f96 | |
parent | 57e2ded4234356c26bf92b0a627ec383ea8d288f (diff) | |
parent | 991f3de1bb696a55f7604e4b7d53492299fe44b5 (diff) | |
download | ffmpeg-streaming-25d8099beb4c7eb93539f9162af1336ef7130fed.zip ffmpeg-streaming-25d8099beb4c7eb93539f9162af1336ef7130fed.tar.gz |
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>
-rw-r--r-- | Changelog | 1 | ||||
-rwxr-xr-x | configure | 6 | ||||
-rw-r--r-- | doc/encoders.texi | 336 | ||||
-rw-r--r-- | doc/indevs.texi | 17 | ||||
-rw-r--r-- | doc/outdevs.texi | 4 | ||||
-rw-r--r-- | libavcodec/ac3.h | 11 | ||||
-rw-r--r-- | libavcodec/ac3dec.c | 10 | ||||
-rw-r--r-- | libavcodec/ac3enc.c | 441 | ||||
-rw-r--r-- | libavcodec/ac3enc_fixed.c | 1 | ||||
-rw-r--r-- | libavcodec/ac3enc_float.c | 1 | ||||
-rw-r--r-- | libavdevice/Makefile | 3 | ||||
-rw-r--r-- | libavdevice/alldevices.c | 1 | ||||
-rw-r--r-- | libavdevice/sndio_common.c | 120 | ||||
-rw-r--r-- | libavdevice/sndio_common.h | 46 | ||||
-rw-r--r-- | libavdevice/sndio_dec.c | 108 | ||||
-rw-r--r-- | libavdevice/sndio_enc.c | 95 |
16 files changed, 1183 insertions, 18 deletions
@@ -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: @@ -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 cab98fb..2f347f4 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 1cd2dd6..5a8a8fa 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 3c0acee..fbb3123 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 1a8cce9..b4092c4 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 c436517..b1f09f2 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 f41fd2d..debd63d 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 f682aa6..e7942ab 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 f5b01f7..faed30d 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 472cb95..5cfc5e8 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 0c000dc..a0c9b08 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 0000000..60b7970 --- /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 0000000..41c984b --- /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 0000000..ff2adeb --- /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 0000000..6745ba4 --- /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, +}; |