- DXVA2-accelerated HEVC decoding
- AAC ELD 480 decoding
- Intel QSV-accelerated H.264 decoding
+ - DSS SP decoder
+version 2.5:
+- HEVC/H.265 RTP payload format (draft v6) packetizer
+- SUP/PGS subtitle demuxer
+- ffprobe -show_pixel_formats option
+- CAST128 symmetric block cipher, ECB mode
+- STL subtitle demuxer and decoder
+- libutvideo YUV 4:2:2 10bit support
+- XCB-based screen-grabber
+- UDP-Lite support (RFC 3828)
+- xBR scaling filter
+- AVFoundation screen capturing support
+- ffserver supports codec private options
+- creating DASH compatible fragmented MP4, MPEG-DASH segmenting muxer
+- WebP muxer with animated WebP support
+- zygoaudio decoding support
+- APNG demuxer
+- postproc visualization support
-version 11:
-- libx265 encoder
+
+version 2.4:
+- Icecast protocol
+- ported lenscorrection filter from frei0r filter
+- large optimizations in dctdnoiz to make it usable
+- ICY metadata are now requested by default with the HTTP protocol
+- support for using metadata in stream specifiers in fftools
+- LZMA compression support in TIFF decoder
+- H.261 RTP payload format (RFC 4587) depacketizer and experimental packetizer
+- HEVC/H.265 RTP payload format (draft v6) depacketizer
+- added codecview filter to visualize information exported by some codecs
+- Matroska 3D support thorugh side data
+- HTML generation using texi2html is deprecated in favor of makeinfo/texi2any
+- silenceremove filter
+
+
+version 2.3:
+- AC3 fixed-point decoding
- shuffleplanes filter
+- subfile protocol
+- Phantom Cine demuxer
- replaygain data export
+- VP7 video decoder
- Alias PIX image encoder and decoder
-- BRender PIX image decoder
-- Amazing Studio PAF playback support
-- XBM decoder
-- BMP standalone parser
-- OpenEXR image decoder
-- support encoding and decoding 4-channel SGI images
+- Improvements to the BRender PIX image decoder
+- Improvements to the XBM decoder
+- QTKit input device
+- improvements to OpenEXR image decoder
- support decoding 16-bit RLE SGI images
-- VP7 video decoder
-- LucasArts SMUSH SANM video decoder
-- LucasArts SMUSH VIMA audio decoder (ADPCM)
-- LucasArts SMUSH demuxer
-- MP2 encoding via TwoLAME
-- asettb filter
-- Silicon Graphics RLE 8-bit video decoder
-- Silicon Graphics Motion Video Compressor 1 & 2 decoder
-- Silicon Graphics Movie demuxer
+- GDI screen grabbing for Windows
+- alternative rendition support for HTTP Live Streaming
+- AVFoundation input device
+- Direct Stream Digital (DSD) decoder
+- Magic Lantern Video (MLV) demuxer
- On2 AVC (Audio for Video) decoder
-- support for decoding through DXVA2 in avconv
+- support for decoding through DXVA2 in ffmpeg
- libbs2b-based stereo-to-binaural audio filter
+- libx264 reference frames count limiting depending on level
- native Opus decoder
- display matrix export and rotation API
-- drop avserver, it was unmaintained for years and largely broken
-- Icecast protocol
-- request Icecast metadata by default
-- support for using metadata in stream specifiers in avtools
-- Matroska 3D support
+- WebVTT encoder
+- showcqt multimedia filter
+- zoompan filter
+- signalstats filter
+- hqx filter (hq2x, hq3x, hq4x)
+- flanger filter
+- Image format auto-detection
+- LRC demuxer and muxer
+- Samba protocol (via libsmbclient)
+- WebM DASH Manifest muxer
+- libfribidi support in drawtext
-version 10:
-- av_strnstr
-- support ID3v2 tags in ASF files
+version 2.2:
+
+- HNM version 4 demuxer and video decoder
+- Live HDS muxer
+- setsar/setdar filters now support variables in ratio expressions
+- elbg filter
+- string validation in ffprobe
+- support for decoding through VDPAU in ffmpeg (the -hwaccel option)
+- complete Voxware MetaSound decoder
+- remove mp3_header_compress bitstream filter
+- Windows resource files for shared libraries
+- aeval filter
+- stereoscopic 3d metadata handling
+- WebP encoding via libwebp
+- ATRAC3+ decoder
+- VP8 in Ogg demuxing
+- side & metadata support in NUT
+- framepack filter
+- XYZ12 rawvideo support in NUT
+- Exif metadata support in WebP decoder
+- OpenGL device
+- Use metadata_header_padding to control padding in ID3 tags (currently used in
+ MP3, AIFF, and OMA files), FLAC header, and the AVI "junk" block.
+- Mirillis FIC video decoder
+- Support DNx444
+- libx265 encoder
+- dejudder filter
+- Autodetect VDA like all other hardware accelerations
+- aliases and defaults for Ogg subtypes (opus, spx)
+
+
+version 2.1:
+
+- aecho filter
+- perspective filter ported from libmpcodecs
+- ffprobe -show_programs option
+- compand filter
+- RTMP seek support
+- when transcoding with ffmpeg (i.e. not streamcopying), -ss is now accurate
+ even when used as an input option. Previous behavior can be restored with
+ the -noaccurate_seek option.
+- ffmpeg -t option can now be used for inputs, to limit the duration of
+ data read from an input file
+- incomplete Voxware MetaSound decoder
+- read EXIF metadata from JPEG
+- DVB teletext decoder
+- phase filter ported from libmpcodecs
+- w3fdif filter
+- Opus support in Matroska
+- FFV1 version 1.3 is stable and no longer experimental
+- FFV1: YUVA(444,422,420) 9, 10 and 16 bit support
+- changed DTS stream id in lavf mpeg ps muxer from 0x8a to 0x88, to be
+ more consistent with other muxers.
+- adelay filter
+- pullup filter ported from libmpcodecs
+- ffprobe -read_intervals option
+- Lossless and alpha support for WebP decoder
+- Error Resilient AAC syntax (ER AAC LC) decoding
+- Low Delay AAC (ER AAC LD) decoding
+- mux chapters in ASF files
+- SFTP protocol (via libssh)
+- libx264: add ability to encode in YUVJ422P and YUVJ444P
+- Fraps: use BT.709 colorspace by default for yuv, as reference fraps decoder does
+- make decoding alpha optional for prores, ffv1 and vp6 by setting
+ the skip_alpha flag.
+- ladspa wrapper filter
+- native VP9 decoder
+- dpx parser
+- max_error_rate parameter in ffmpeg
+- PulseAudio output device
+- ReplayGain scanner
+- Enhanced Low Delay AAC (ER AAC ELD) decoding (no LD SBR support)
+- Linux framebuffer output device
+- HEVC decoder
+- raw HEVC, HEVC in MOV/MP4, HEVC in Matroska, HEVC in MPEG-TS demuxing
+- mergeplanes filter
+
+
+version 2.0:
+
+- curves filter
- reference-counting for AVFrame and AVPacket data
-- avconv now fails when input options are used for output file
+- ffmpeg now fails when input options are used for output file
or vice versa
-- avconv options -filter_script and -filter_complex_script, which allow a
+- support for Monkey's Audio versions from 3.93
+- perms and aperms filters
+- audio filtering support in ffplay
+- 10% faster aac encoding on x86 and MIPS
+- sine audio filter source
+- WebP demuxing and decoding support
+- ffmpeg options -filter_script and -filter_complex_script, which allow a
filtergraph description to be read from a file
+- OpenCL support
+- audio phaser filter
+- separatefields filter
+- libquvi demuxer
- uniform options syntax across all filters
+- telecine filter
- interlace filter
-- JPEG 2000 decoder
-- asetpts filter (same as setpts, but for audio)
+- smptehdbars source
+- inverse telecine filters (fieldmatch and decimate)
+- colorbalance filter
+- colorchannelmixer filter
+- The matroska demuxer can now output proper verbatim ASS packets. It will
+ become the default at the next libavformat major bump.
+- decent native animated GIF encoding
+- asetrate filter
+- interleave filter
+- timeline editing with filters
+- vidstabdetect and vidstabtransform filters for video stabilization using
+ the vid.stab library
+- astats filter
- trim and atrim filters
-- avconv -t and -ss (output-only) options are now sample-accurate when
+- ffmpeg -t and -ss (output-only) options are now sample-accurate when
transcoding audio
- Matroska muxer can now put the index at the beginning of the file.
-- avconv -deinterlace option removed, the yadif filter should be used instead
+- extractplanes filter
+- avectorscope filter
+- ADPCM DTK decoder
+- ADP demuxer
+- RSD demuxer
+- RedSpark demuxer
+- ADPCM IMA Radical decoder
+- zmq filters
+- DCT denoiser filter (dctdnoiz)
+- Wavelet denoiser filter ported from libmpcodecs as owdenoise (formerly "ow")
- Apple Intermediate Codec decoder
- Escape 130 video decoder
+- FTP protocol support
+- V4L2 output device
+- 3D LUT filter (lut3d)
+- SMPTE 302M audio encoder
- support for slice multithreading in libavfilter
+- Hald CLUT support (generation and filtering)
- VC-1 interlaced B-frame support
- support for WavPack muxing (raw and in Matroska)
+- XVideo output device
+- vignette filter
+- True Audio (TTA) encoder
- Go2Webinar decoder
+- mcdeint filter ported from libmpcodecs
+- sab filter ported from libmpcodecs
+- ffprobe -show_chapters option
- WavPack encoding through libwavpack
-- Added the -n parameter to avconv
-- RTMP seek support
-- when transcoding with avconv (i.e. not streamcopying), -ss is now accurate
- even when used as an input option. Previous behavior can be restored with
- the -noaccurate_seek option.
-- avconv -t option can now be used for inputs, to limit the duration of
- data read from an input file
-- Voxware MetaSound decoder
-- WebP decoder
-- Error Resilient AAC syntax (ER AAC LC) decoding
-- Low Delay AAC (ER AAC LD) decoding
-- mux chapters in ASF files
-- Opus in Ogg demuxing
-- Enhanced Low Delay AAC (ER AAC ELD) decoding (no LD SBR support)
-- F4V muxer
-- HNM version 4 demuxer and video decoder
-- HEVC decoder
-- raw HEVC, HEVC in MOV/MP4, HEVC in Matroska, HEVC in MPEG-TS demuxing
-- remove avplay -vismv option, which has not worked for a long time
-- Live HDS muxer
-- setsar/setdar filters now support variables in ratio expressions
-- dar variable in the scale filter now returns the actual DAR (i.e. a * sar)
-- VP9 decoder
-- support for decoding through VDPAU in avconv (the -hwaccel option)
-- remove mp3_header_(de)compress bitstream filters
-- stereoscopic 3d metadata handling
-- png standalone parser
-- WebP encoding via libwebp
-- ATRAC3+ decoder
-- framepack filter
-- Mirillis FIC video decoder
-- Support DNx444
-- compand audio filter
+- rotate filter
+- spp filter ported from libmpcodecs
+- libgme support
+- psnr filter
-version 9:
-- av_basename and av_dirname
-- adobe and limelight publisher authentication in RTMP
+version 1.2:
+
- VDPAU hardware acceleration through normal hwaccel
- SRTP support
+- Error diffusion dither in Swscale
+- Chained Ogg support
+- Theora Midstream reconfiguration support
+- EVRC decoder
+- audio fade filter
+- filtering audio with unknown channel layout
+- allpass, bass, bandpass, bandreject, biquad, equalizer, highpass, lowpass
+ and treble audio filter
+- improved showspectrum filter, with multichannel support and sox-like colors
+- histogram filter
+- tee muxer
+- il filter ported from libmpcodecs
+- support ID3v2 tags in ASF files
+- encrypted TTA stream decoding support
+- RF64 support in WAV muxer
+- noise filter ported from libmpcodecs
+- Subtitles character encoding conversion
+- blend filter
+- stereo3d filter ported from libmpcodecs
+
+version 1.1:
-version 9_beta3:
-- ashowinfo audio filter
+- stream disposition information printing in ffprobe
+- filter for loudness analysis following EBU R128
+- Opus encoder using libopus
+- ffprobe -select_streams option
+- Pinnacle TARGA CineWave YUV16 decoder
+- TAK demuxer, decoder and parser
+- DTS-HD demuxer
+- remove -same_quant, it hasn't worked for years
+- FFM2 support
+- X-Face image encoder and decoder
- 24-bit FLAC encoding
-- audio volume filter
-- deprecated the avconv -vol option. the volume filter is to be used instead.
- multi-channel ALAC encoding up to 7.1
-- TAK demuxer, parser, and decoder
-- adaptive frame-level multithreading for H.264
-
-
-version 9_beta2:
- metadata (INFO tag) support in WAV muxer
+- subtitles raw text decoder
- support for building DLLs using MSVC
-- remove avserver daemon mode
+- LVF demuxer
+- ffescape tool
+- metadata (info chunk) support in CAF muxer
+- field filter ported from libmpcodecs
+- AVR demuxer
+- geq filter ported from libmpcodecs
+- remove ffserver daemon mode
+- AST muxer/demuxer
+- new expansion syntax for drawtext
+- BRender PIX image decoder
+- ffprobe -show_entries option
+- ffprobe -sections option
+- ADPCM IMA Dialogic decoder
+- BRSTM demuxer
+- animated GIF decoder and demuxer
+- PVF demuxer
+- subtitles filter
+- IRCAM muxer/demuxer
+- Paris Audio File demuxer
+- Virtual concatenation demuxer
+- VobSub demuxer
+- JSON captions for TED talks decoding support
+- SOX Resampler support in libswresample
+- aselect filter
+- SGI RLE 8-bit / Silicon Graphics RLE 8-bit video decoder
+- Silicon Graphics Motion Video Compressor 1 & 2 decoder
+- Silicon Graphics Movie demuxer
+- apad filter
+- Resolution & pixel format change support with multithreading for H.264
+- documentation split into per-component manuals
+- pp (postproc) filter ported from MPlayer
+- NIST Sphere demuxer
+- MPL2, VPlayer, MPlayer, AQTitle, PJS and SubViewer v1 subtitles demuxers and decoders
+- Sony Wave64 muxer
+- adobe and limelight publisher authentication in RTMP
+- data: URI scheme
- support building on the Plan 9 operating system
-- ffv1: support version 1.3
+- kerndeint filter ported from MPlayer
+- histeq filter ported from VirtualDub
+- Megalux Frame demuxer
+- 012v decoder
+- Improved AVC Intra decoding support
-version 9_beta1:
+version 1.0:
-- XWD encoder and decoder
-- Support for fragmentation in the mov/mp4 muxer
-- ISMV (Smooth Streaming) muxer
-- CDXL demuxer and decoder
-- Apple ProRes encoder
-- Sun Rasterfile Encoder
-- remove libpostproc
-- ID3v2 attached pictures reading and writing
-- WMA Lossless decoder
-- XBM encoder
-- RealAudio Lossless decoder
-- ZeroCodec decoder
-- drop support for avconv without libavfilter
-- add libavresample audio conversion library
-- audio filters support in libavfilter and avconv
-- add fps filter
-- audio split filter
-- audio mix filter
-- avprobe output is now standard INI or JSON. The old format can still
- be used with -of old.
+- INI and flat output in ffprobe
+- Scene detection in libavfilter
- Indeo Audio decoder
- channelsplit audio filter
+- setnsamples audio filter
+- atempo filter
+- ffprobe -show_data option
- RTMPT protocol support
- iLBC encoding/decoding via libilbc
- Microsoft Screen 1 decoder
--- /dev/null
- * This file is part of Libav.
+ /*
+ * Digital Speech Standard - Standard Play mode (DSS SP) audio decoder.
+ * Copyright (C) 2014 Oleksij Rempel <linux@rempel-privat.de>
+ *
- * Libav is free software; you can redistribute it and/or
++ * This file is part of FFmpeg.
+ *
- * Libav is distributed in the hope that it will be useful,
++ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
- * License along with Libav; if not, write to the Free Software
++ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
++ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+ #include "libavutil/channel_layout.h"
+ #include "libavutil/common.h"
+ #include "libavutil/mem.h"
+ #include "libavutil/opt.h"
+
+ #include "avcodec.h"
+ #include "get_bits.h"
+ #include "internal.h"
+
+ #define SUBFRAMES 4
+ #define PULSE_MAX 8
+
+ #define DSS_SP_FRAME_SIZE 42
+ #define DSS_SP_SAMPLE_COUNT (66 * SUBFRAMES)
+ #define DSS_SP_FORMULA(a, b, c) ((((a) << 15) + (b) * (c)) + 0x4000) >> 15
+
+ typedef struct DssSpSubframe {
+ int16_t gain;
+ int32_t combined_pulse_pos;
+ int16_t pulse_pos[7];
+ int16_t pulse_val[7];
+ } DssSpSubframe;
+
+ typedef struct DssSpFrame {
+ int16_t filter_idx[14];
+ int16_t sf_adaptive_gain[SUBFRAMES];
+ int16_t pitch_lag[SUBFRAMES];
+ struct DssSpSubframe sf[SUBFRAMES];
+ } DssSpFrame;
+
+ typedef struct DssSpContext {
+ int32_t excitation[288 + 6];
+ int32_t history[187];
+ DssSpFrame fparam;
+ int32_t working_buffer[SUBFRAMES][72];
+ int32_t audio_buf[15];
+ int32_t err_buf1[15];
+ int32_t lpc_filter[14];
+ int32_t filter[15];
+ int32_t vector_buf[72];
+ int noise_state;
+ int32_t err_buf2[15];
+
+ int pulse_dec_mode;
+
+ DECLARE_ALIGNED(16, uint8_t, bits)[DSS_SP_FRAME_SIZE +
+ FF_INPUT_BUFFER_PADDING_SIZE];
+ } DssSpContext;
+
+ /*
+ * Used for the coding/decoding of the pulse positions for the MP-MLQ codebook.
+ */
+ static const uint32_t dss_sp_combinatorial_table[PULSE_MAX][72] = {
+ { 0, 0, 0, 0, 0, 0,
+ 0, 0, 0, 0, 0, 0,
+ 0, 0, 0, 0, 0, 0,
+ 0, 0, 0, 0, 0, 0,
+ 0, 0, 0, 0, 0, 0,
+ 0, 0, 0, 0, 0, 0,
+ 0, 0, 0, 0, 0, 0,
+ 0, 0, 0, 0, 0, 0,
+ 0, 0, 0, 0, 0, 0,
+ 0, 0, 0, 0, 0, 0,
+ 0, 0, 0, 0, 0, 0,
+ 0, 0, 0, 0, 0, 0 },
+ { 0, 1, 2, 3, 4, 5,
+ 6, 7, 8, 9, 10, 11,
+ 12, 13, 14, 15, 16, 17,
+ 18, 19, 20, 21, 22, 23,
+ 24, 25, 26, 27, 28, 29,
+ 30, 31, 32, 33, 34, 35,
+ 36, 37, 38, 39, 40, 41,
+ 42, 43, 44, 45, 46, 47,
+ 48, 49, 50, 51, 52, 53,
+ 54, 55, 56, 57, 58, 59,
+ 60, 61, 62, 63, 64, 65,
+ 66, 67, 68, 69, 70, 71 },
+ { 0, 0, 1, 3, 6, 10,
+ 15, 21, 28, 36, 45, 55,
+ 66, 78, 91, 105, 120, 136,
+ 153, 171, 190, 210, 231, 253,
+ 276, 300, 325, 351, 378, 406,
+ 435, 465, 496, 528, 561, 595,
+ 630, 666, 703, 741, 780, 820,
+ 861, 903, 946, 990, 1035, 1081,
+ 1128, 1176, 1225, 1275, 1326, 1378,
+ 1431, 1485, 1540, 1596, 1653, 1711,
+ 1770, 1830, 1891, 1953, 2016, 2080,
+ 2145, 2211, 2278, 2346, 2415, 2485 },
+ { 0, 0, 0, 1, 4, 10,
+ 20, 35, 56, 84, 120, 165,
+ 220, 286, 364, 455, 560, 680,
+ 816, 969, 1140, 1330, 1540, 1771,
+ 2024, 2300, 2600, 2925, 3276, 3654,
+ 4060, 4495, 4960, 5456, 5984, 6545,
+ 7140, 7770, 8436, 9139, 9880, 10660,
+ 11480, 12341, 13244, 14190, 15180, 16215,
+ 17296, 18424, 19600, 20825, 22100, 23426,
+ 24804, 26235, 27720, 29260, 30856, 32509,
+ 34220, 35990, 37820, 39711, 41664, 43680,
+ 45760, 47905, 50116, 52394, 54740, 57155 },
+ { 0, 0, 0, 0, 1, 5,
+ 15, 35, 70, 126, 210, 330,
+ 495, 715, 1001, 1365, 1820, 2380,
+ 3060, 3876, 4845, 5985, 7315, 8855,
+ 10626, 12650, 14950, 17550, 20475, 23751,
+ 27405, 31465, 35960, 40920, 46376, 52360,
+ 58905, 66045, 73815, 82251, 91390, 101270,
+ 111930, 123410, 135751, 148995, 163185, 178365,
+ 194580, 211876, 230300, 249900, 270725, 292825,
+ 316251, 341055, 367290, 395010, 424270, 455126,
+ 487635, 521855, 557845, 595665, 635376, 677040,
+ 720720, 766480, 814385, 864501, 916895, 971635 },
+ { 0, 0, 0, 0, 0, 1,
+ 6, 21, 56, 126, 252, 462,
+ 792, 1287, 2002, 3003, 4368, 6188,
+ 8568, 11628, 15504, 20349, 26334, 33649,
+ 42504, 53130, 65780, 80730, 98280, 118755,
+ 142506, 169911, 201376, 237336, 278256, 324632,
+ 376992, 435897, 501942, 575757, 658008, 749398,
+ 850668, 962598, 1086008, 1221759, 1370754, 1533939,
+ 1712304, 1906884, 2118760, 2349060, 2598960, 2869685,
+ 3162510, 3478761, 3819816, 4187106, 4582116, 5006386,
+ 5461512, 5949147, 6471002, 7028847, 7624512, 8259888,
+ 8936928, 9657648, 10424128, 11238513, 12103014, 13019909 },
+ { 0, 0, 0, 0, 0, 0,
+ 1, 7, 28, 84, 210, 462,
+ 924, 1716, 3003, 5005, 8008, 12376,
+ 18564, 27132, 38760, 54264, 74613, 100947,
+ 134596, 177100, 230230, 296010, 376740, 475020,
+ 593775, 736281, 906192, 1107568, 1344904, 1623160,
+ 1947792, 2324784, 2760681, 3262623, 3838380, 4496388,
+ 5245786, 6096454, 7059052, 8145060, 9366819, 10737573,
+ 12271512, 13983816, 15890700, 18009460, 20358520, 22957480,
+ 25827165, 28989675, 32468436, 36288252, 40475358, 45057474,
+ 50063860, 55525372, 61474519, 67945521, 74974368, 82598880,
+ 90858768, 99795696, 109453344, 119877472, 131115985, 143218999 },
+ { 0, 0, 0, 0, 0, 0,
+ 0, 1, 8, 36, 120, 330,
+ 792, 1716, 3432, 6435, 11440, 19448,
+ 31824, 50388, 77520, 116280, 170544, 245157,
+ 346104, 480700, 657800, 888030, 1184040, 1560780,
+ 2035800, 2629575, 3365856, 4272048, 5379616, 6724520,
+ 8347680, 10295472, 12620256, 15380937, 18643560, 22481940,
+ 26978328, 32224114, 38320568, 45379620, 53524680, 62891499,
+ 73629072, 85900584, 99884400, 115775100, 133784560, 154143080,
+ 177100560, 202927725, 231917400, 264385836, 300674088, 341149446,
+ 386206920, 436270780, 491796152, 553270671, 621216192, 696190560,
+ 778789440, 869648208, 969443904, 1078897248, 1198774720, 1329890705 },
+ };
+
+ static const int16_t dss_sp_filter_cb[14][32] = {
+ { -32653, -32587, -32515, -32438, -32341, -32216, -32062, -31881,
+ -31665, -31398, -31080, -30724, -30299, -29813, -29248, -28572,
+ -27674, -26439, -24666, -22466, -19433, -16133, -12218, -7783,
+ -2834, 1819, 6544, 11260, 16050, 20220, 24774, 28120 },
+
+ { -27503, -24509, -20644, -17496, -14187, -11277, -8420, -5595,
+ -3013, -624, 1711, 3880, 5844, 7774, 9739, 11592,
+ 13364, 14903, 16426, 17900, 19250, 20586, 21803, 23006,
+ 24142, 25249, 26275, 27300, 28359, 29249, 30118, 31183 },
+
+ { -27827, -24208, -20943, -17781, -14843, -11848, -9066, -6297,
+ -3660, -910, 1918, 5025, 8223, 11649, 15086, 18423,
+ 0, 0, 0, 0, 0, 0, 0, 0,
+ 0, 0, 0, 0, 0, 0, 0, 0 },
+
+ { -17128, -11975, -8270, -5123, -2296, 183, 2503, 4707,
+ 6798, 8945, 11045, 13239, 15528, 18248, 21115, 24785,
+ 0, 0, 0, 0, 0, 0, 0, 0,
+ 0, 0, 0, 0, 0, 0, 0, 0 },
+
+ { -21557, -17280, -14286, -11644, -9268, -7087, -4939, -2831,
+ -691, 1407, 3536, 5721, 8125, 10677, 13721, 17731,
+ 0, 0, 0, 0, 0, 0, 0, 0,
+ 0, 0, 0, 0, 0, 0, 0, 0 },
+
+ { -15030, -10377, -7034, -4327, -1900, 364, 2458, 4450,
+ 6422, 8374, 10374, 12486, 14714, 16997, 19626, 22954,
+ 0, 0, 0, 0, 0, 0, 0, 0,
+ 0, 0, 0, 0, 0, 0, 0, 0 },
+
+ { -16155, -12362, -9698, -7460, -5258, -3359, -1547, 219,
+ 1916, 3599, 5299, 6994, 8963, 11226, 13716, 16982,
+ 0, 0, 0, 0, 0, 0, 0, 0,
+ 0, 0, 0, 0, 0, 0, 0, 0 },
+
+ { -14742, -9848, -6921, -4648, -2769, -1065, 499, 2083,
+ 3633, 5219, 6857, 8580, 10410, 12672, 15561, 20101,
+ 0, 0, 0, 0, 0, 0, 0, 0,
+ 0, 0, 0, 0, 0, 0, 0, 0 },
+
+ { -11099, -7014, -3855, -1025, 1680, 4544, 7807, 11932,
+ 0, 0, 0, 0, 0, 0, 0, 0,
+ 0, 0, 0, 0, 0, 0, 0, 0,
+ 0, 0, 0, 0, 0, 0, 0, 0 },
+
+ { -9060, -4570, -1381, 1419, 4034, 6728, 9865, 14149,
+ 0, 0, 0, 0, 0, 0, 0, 0,
+ 0, 0, 0, 0, 0, 0, 0, 0,
+ 0, 0, 0, 0, 0, 0, 0, 0 },
+
+ { -12450, -7985, -4596, -1734, 961, 3629, 6865, 11142,
+ 0, 0, 0, 0, 0, 0, 0, 0,
+ 0, 0, 0, 0, 0, 0, 0, 0,
+ 0, 0, 0, 0, 0, 0, 0, 0 },
+
+ { -11831, -7404, -4010, -1096, 1606, 4291, 7386, 11482,
+ 0, 0, 0, 0, 0, 0, 0, 0,
+ 0, 0, 0, 0, 0, 0, 0, 0,
+ 0, 0, 0, 0, 0, 0, 0, 0 },
+
+ { -13404, -9250, -5995, -3312, -890, 1594, 4464, 8198,
+ 0, 0, 0, 0, 0, 0, 0, 0,
+ 0, 0, 0, 0, 0, 0, 0, 0,
+ 0, 0, 0, 0, 0, 0, 0, 0 },
+
+ { -11239, -7220, -4040, -1406, 971, 3321, 6006, 9697,
+ 0, 0, 0, 0, 0, 0, 0, 0,
+ 0, 0, 0, 0, 0, 0, 0, 0,
+ 0, 0, 0, 0, 0, 0, 0, 0 },
+ };
+
+ static const uint16_t dss_sp_fixed_cb_gain[64] = {
+ 0, 4, 8, 13, 17, 22, 26, 31,
+ 35, 40, 44, 48, 53, 58, 63, 69,
+ 76, 83, 91, 99, 109, 119, 130, 142,
+ 155, 170, 185, 203, 222, 242, 265, 290,
+ 317, 346, 378, 414, 452, 494, 540, 591,
+ 646, 706, 771, 843, 922, 1007, 1101, 1204,
+ 1316, 1438, 1572, 1719, 1879, 2053, 2244, 2453,
+ 2682, 2931, 3204, 3502, 3828, 4184, 4574, 5000,
+ };
+
+ static const int16_t dss_sp_pulse_val[8] = {
+ -31182, -22273, -13364, -4455, 4455, 13364, 22273, 31182
+ };
+
+ static const uint16_t binary_decreasing_array[] = {
+ 32767, 16384, 8192, 4096, 2048, 1024, 512, 256,
+ 128, 64, 32, 16, 8, 4, 2,
+ };
+
+ static const uint16_t dss_sp_unc_decreasing_array[] = {
+ 32767, 26214, 20972, 16777, 13422, 10737, 8590, 6872,
+ 5498, 4398, 3518, 2815, 2252, 1801, 1441,
+ };
+
+ static const uint16_t dss_sp_adaptive_gain[] = {
+ 102, 231, 360, 488, 617, 746, 875, 1004,
+ 1133, 1261, 1390, 1519, 1648, 1777, 1905, 2034,
+ 2163, 2292, 2421, 2550, 2678, 2807, 2936, 3065,
+ 3194, 3323, 3451, 3580, 3709, 3838, 3967, 4096,
+ };
+
+ static const int32_t dss_sp_sinc[67] = {
+ 262, 293, 323, 348, 356, 336, 269, 139,
+ -67, -358, -733, -1178, -1668, -2162, -2607, -2940,
+ -3090, -2986, -2562, -1760, -541, 1110, 3187, 5651,
+ 8435, 11446, 14568, 17670, 20611, 23251, 25460, 27125,
+ 28160, 28512, 28160,
+ 27125, 25460, 23251, 20611, 17670, 14568, 11446, 8435,
+ 5651, 3187, 1110, -541, -1760, -2562, -2986, -3090,
+ -2940, -2607, -2162, -1668, -1178, -733, -358, -67,
+ 139, 269, 336, 356, 348, 323, 293, 262,
+ };
+
+ static av_cold int dss_sp_decode_init(AVCodecContext *avctx)
+ {
+ DssSpContext *p = avctx->priv_data;
+ avctx->channel_layout = AV_CH_LAYOUT_MONO;
+ avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+ avctx->channels = 1;
+ avctx->sample_rate = 11025;
+
+ memset(p->history, 0, sizeof(p->history));
+ p->pulse_dec_mode = 1;
+
+ return 0;
+ }
+
+ static void dss_sp_unpack_coeffs(DssSpContext *p, const uint8_t *src)
+ {
+ GetBitContext gb;
+ DssSpFrame *fparam = &p->fparam;
+ int i;
+ int subframe_idx;
+ uint32_t combined_pitch;
+ uint32_t tmp;
+ uint32_t pitch_lag;
+
+ for (i = 0; i < DSS_SP_FRAME_SIZE; i += 2) {
+ p->bits[i] = src[i + 1];
+ p->bits[i + 1] = src[i];
+ }
+
+ init_get_bits(&gb, p->bits, DSS_SP_FRAME_SIZE * 8);
+
+ for (i = 0; i < 2; i++)
+ fparam->filter_idx[i] = get_bits(&gb, 5);
+ for (; i < 8; i++)
+ fparam->filter_idx[i] = get_bits(&gb, 4);
+ for (; i < 14; i++)
+ fparam->filter_idx[i] = get_bits(&gb, 3);
+
+ for (subframe_idx = 0; subframe_idx < 4; subframe_idx++) {
+ fparam->sf_adaptive_gain[subframe_idx] = get_bits(&gb, 5);
+
+ fparam->sf[subframe_idx].combined_pulse_pos = get_bits_long(&gb, 31);
+
+ fparam->sf[subframe_idx].gain = get_bits(&gb, 6);
+
+ for (i = 0; i < 7; i++)
+ fparam->sf[subframe_idx].pulse_val[i] = get_bits(&gb, 3);
+ }
+
+ for (subframe_idx = 0; subframe_idx < 4; subframe_idx++) {
+ unsigned int C72_binomials[PULSE_MAX] = {
+ 72, 2556, 59640, 1028790, 13991544, 156238908, 1473109704,
+ 3379081753
+ };
+ unsigned int combined_pulse_pos =
+ fparam->sf[subframe_idx].combined_pulse_pos;
+ int index = 6;
+
+ if (combined_pulse_pos < C72_binomials[PULSE_MAX - 1]) {
+ if (p->pulse_dec_mode) {
+ int pulse, pulse_idx;
+ pulse = PULSE_MAX - 1;
+ pulse_idx = 71;
+ combined_pulse_pos =
+ fparam->sf[subframe_idx].combined_pulse_pos;
+
+ /* this part seems to be close to g723.1 gen_fcb_excitation()
+ * RATE_6300 */
+
+ /* TODO: what is 7? size of subframe? */
+ for (i = 0; i < 7; i++) {
+ for (;
+ combined_pulse_pos <
+ dss_sp_combinatorial_table[pulse][pulse_idx];
+ --pulse_idx)
+ ;
+ combined_pulse_pos -=
+ dss_sp_combinatorial_table[pulse][pulse_idx];
+ pulse--;
+ fparam->sf[subframe_idx].pulse_pos[i] = pulse_idx;
+ }
+ }
+ } else {
+ p->pulse_dec_mode = 0;
+
+ /* why do we need this? */
+ fparam->sf[subframe_idx].pulse_pos[6] = 0;
+
+ for (i = 71; i >= 0; i--) {
+ if (C72_binomials[index] <= combined_pulse_pos) {
+ combined_pulse_pos -= C72_binomials[index];
+
+ fparam->sf[subframe_idx].pulse_pos[(index ^ 7) - 1] = i;
+
+ if (!index)
+ break;
+ --index;
+ }
+ --C72_binomials[0];
+ if (index) {
+ int a;
+ for (a = 0; a < index; a++)
+ C72_binomials[a + 1] -= C72_binomials[a];
+ }
+ }
+ }
+ }
+
+ combined_pitch = get_bits(&gb, 24);
+
+ fparam->pitch_lag[0] = (combined_pitch % 151) + 36;
+
+ combined_pitch /= 151;
+
+ for (i = 1; i < SUBFRAMES; i++) {
+ fparam->pitch_lag[i] = combined_pitch % 48;
+ combined_pitch /= 48;
+ }
+
+ pitch_lag = fparam->pitch_lag[0];
+ for (i = 1; i < SUBFRAMES; i++) {
+ if (pitch_lag > 162) {
+ fparam->pitch_lag[i] += 162 - 23;
+ } else {
+ tmp = pitch_lag - 23;
+ if (tmp < 36)
+ tmp = 36;
+ fparam->pitch_lag[i] += tmp;
+ }
+ pitch_lag = fparam->pitch_lag[i];
+ }
+ }
+
+ static void dss_sp_unpack_filter(DssSpContext *p)
+ {
+ int i;
+
+ for (i = 0; i < 14; i++)
+ p->lpc_filter[i] = dss_sp_filter_cb[i][p->fparam.filter_idx[i]];
+ }
+
+ static void dss_sp_convert_coeffs(int32_t *lpc_filter, int32_t *coeffs)
+ {
+ int a, a_plus, i;
+
+ coeffs[0] = 0x2000;
+ for (a = 0; a < 14; a++) {
+ a_plus = a + 1;
+ coeffs[a_plus] = lpc_filter[a] >> 2;
+ if (a_plus / 2 >= 1) {
+ for (i = 1; i <= a_plus / 2; i++) {
+ int coeff_1, coeff_2, tmp;
+
+ coeff_1 = coeffs[i];
+ coeff_2 = coeffs[a_plus - i];
+
+ tmp = DSS_SP_FORMULA(coeff_1, lpc_filter[a], coeff_2);
+ coeffs[i] = av_clip_int16(tmp);
+
+ tmp = DSS_SP_FORMULA(coeff_2, lpc_filter[a], coeff_1);
+ coeffs[a_plus - i] = av_clip_int16(tmp);
+ }
+ }
+ }
+ }
+
+ static void dss_sp_add_pulses(int32_t *vector_buf,
+ const struct DssSpSubframe *sf)
+ {
+ int i;
+
+ for (i = 0; i < 7; i++)
+ vector_buf[sf->pulse_pos[i]] += (dss_sp_fixed_cb_gain[sf->gain] *
+ dss_sp_pulse_val[sf->pulse_val[i]] +
+ 0x4000) >> 15;
+ }
+
+ static void dss_sp_gen_exc(int32_t *vector, int32_t *prev_exc,
+ int pitch_lag, int gain)
+ {
+ int i;
+
+ /* do we actually need this check? we can use just [a3 - i % a3]
+ * for both cases */
+ if (pitch_lag < 72)
+ for (i = 0; i < 72; i++)
+ vector[i] = prev_exc[pitch_lag - i % pitch_lag];
+ else
+ for (i = 0; i < 72; i++)
+ vector[i] = prev_exc[pitch_lag - i];
+
+ for (i = 0; i < 72; i++) {
+ int tmp = gain * vector[i] >> 11;
+ vector[i] = av_clip_int16(tmp);
+ }
+ }
+
+ static void dss_sp_scale_vector(int32_t *vec, int bits, int size)
+ {
+ int i;
+
+ if (bits < 0)
+ for (i = 0; i < size; i++)
+ vec[i] = vec[i] >> -bits;
+ else
+ for (i = 0; i < size; i++)
+ vec[i] = vec[i] << bits;
+ }
+
+ static void dss_sp_update_buf(int32_t *hist, int32_t *vector)
+ {
+ int i;
+
+ for (i = 114; i > 0; i--)
+ vector[i + 72] = vector[i];
+
+ for (i = 0; i < 72; i++)
+ vector[72 - i] = hist[i];
+ }
+
+ static void dss_sp_shift_sq_sub(const int32_t *filter_buf,
+ int32_t *error_buf, int32_t *dst)
+ {
+ int a;
+
+ for (a = 0; a < 72; a++) {
+ int i, tmp;
+
+ tmp = dst[a] * filter_buf[0];
+
+ for (i = 14; i > 0; i--)
+ tmp -= error_buf[i] * filter_buf[i];
+
+ for (i = 14; i > 0; i--)
+ error_buf[i] = error_buf[i - 1];
+
+ tmp = (tmp + 4096) >> 13;
+
+ error_buf[1] = tmp;
+
+ dst[a] = av_clip_int16(tmp);
+ }
+ }
+
+ static void dss_sp_shift_sq_add(const int32_t *filter_buf, int32_t *audio_buf,
+ int32_t *dst)
+ {
+ int a;
+
+ for (a = 0; a < 72; a++) {
+ int i, tmp = 0;
+
+ audio_buf[0] = dst[a];
+
+ for (i = 14; i >= 0; i--)
+ tmp += audio_buf[i] * filter_buf[i];
+
+ for (i = 14; i > 0; i--)
+ audio_buf[i] = audio_buf[i - 1];
+
+ tmp = (tmp + 4096) >> 13;
+
+ dst[a] = av_clip_int16(tmp);
+ }
+ }
+
+ static void dss_sp_vec_mult(const int32_t *src, int32_t *dst,
+ const int16_t *mult)
+ {
+ int i;
+
+ dst[0] = src[0];
+
+ for (i = 1; i < 15; i++)
+ dst[i] = (src[i] * mult[i] + 0x4000) >> 15;
+ }
+
+ static int dss_sp_get_normalize_bits(int32_t *vector_buf, int16_t size)
+ {
+ unsigned int val;
+ int max_val;
+ int i;
+
+ val = 1;
+ for (i = 0; i < size; i++)
+ val |= FFABS(vector_buf[i]);
+
+ for (max_val = 0; val <= 0x4000; ++max_val)
+ val *= 2;
+ return max_val;
+ }
+
+ static int dss_sp_vector_sum(DssSpContext *p, int size)
+ {
+ int i, sum = 0;
+ for (i = 0; i < size; i++)
+ sum += FFABS(p->vector_buf[i]);
+ return sum;
+ }
+
+ static void dss_sp_sf_synthesis(DssSpContext *p, int32_t lpc_filter,
+ int32_t *dst, int size)
+ {
+ int32_t tmp_buf[15];
+ int32_t noise[72];
+ int bias, vsum_2 = 0, vsum_1 = 0, v36, normalize_bits;
+ int i, tmp;
+
+ if (size > 0) {
+ vsum_1 = dss_sp_vector_sum(p, size);
+
+ if (vsum_1 > 0xFFFFF)
+ vsum_1 = 0xFFFFF;
+ }
+
+ normalize_bits = dss_sp_get_normalize_bits(p->vector_buf, size);
+
+ dss_sp_scale_vector(p->vector_buf, normalize_bits - 3, size);
+ dss_sp_scale_vector(p->audio_buf, normalize_bits, 15);
+ dss_sp_scale_vector(p->err_buf1, normalize_bits, 15);
+
+ v36 = p->err_buf1[1];
+
+ dss_sp_vec_mult(p->filter, tmp_buf, binary_decreasing_array);
+ dss_sp_shift_sq_add(tmp_buf, p->audio_buf, p->vector_buf);
+
+ dss_sp_vec_mult(p->filter, tmp_buf, dss_sp_unc_decreasing_array);
+ dss_sp_shift_sq_sub(tmp_buf, p->err_buf1, p->vector_buf);
+
+ /* lpc_filter can be negative */
+ lpc_filter = lpc_filter >> 1;
+ if (lpc_filter >= 0)
+ lpc_filter = 0;
+
+ if (size > 1) {
+ for (i = size - 1; i > 0; i--) {
+ tmp = DSS_SP_FORMULA(p->vector_buf[i], lpc_filter,
+ p->vector_buf[i - 1]);
+ p->vector_buf[i] = av_clip_int16(tmp);
+ }
+ }
+
+ tmp = DSS_SP_FORMULA(p->vector_buf[0], lpc_filter, v36);
+ p->vector_buf[0] = av_clip_int16(tmp);
+
+ dss_sp_scale_vector(p->vector_buf, -normalize_bits, size);
+ dss_sp_scale_vector(p->audio_buf, -normalize_bits, 15);
+ dss_sp_scale_vector(p->err_buf1, -normalize_bits, 15);
+
+ if (size > 0)
+ vsum_2 = dss_sp_vector_sum(p, size);
+
+ if (vsum_2 >= 0x40)
+ tmp = (vsum_1 << 11) / vsum_2;
+ else
+ tmp = 1;
+
+ bias = 409 * tmp >> 15 << 15;
+ tmp = (bias + 32358 * p->noise_state) >> 15;
+ noise[0] = av_clip_int16(tmp);
+
+ for (i = 1; i < size; i++) {
+ tmp = (bias + 32358 * noise[i - 1]) >> 15;
+ noise[i] = av_clip_int16(tmp);
+ }
+
+ p->noise_state = noise[size - 1];
+ for (i = 0; i < size; i++) {
+ tmp = (p->vector_buf[i] * noise[i]) >> 11;
+ dst[i] = av_clip_int16(tmp);
+ }
+ }
+
+ static void dss_sp_update_state(DssSpContext *p, int32_t *dst)
+ {
+ int i, offset = 6, counter = 0, a = 0;
+
+ for (i = 0; i < 6; i++)
+ p->excitation[i] = p->excitation[288 + i];
+
+ for (i = 0; i < 72 * SUBFRAMES; i++)
+ p->excitation[6 + i] = dst[i];
+
+ do {
+ int tmp = 0;
+
+ for (i = 0; i < 6; i++)
+ tmp += p->excitation[offset--] * dss_sp_sinc[a + i * 11];
+
+ offset += 7;
+
+ tmp >>= 15;
+ dst[counter] = av_clip_int16(tmp);
+
+ counter++;
+
+ a = (a + 1) % 11;
+ if (!a)
+ offset++;
+ } while (offset < FF_ARRAY_ELEMS(p->excitation));
+ }
+
+ static void dss_sp_32to16bit(int16_t *dst, int32_t *src, int size)
+ {
+ int i;
+
+ for (i = 0; i < size; i++)
+ dst[i] = av_clip_int16(src[i]);
+ }
+
+ static int dss_sp_decode_one_frame(DssSpContext *p,
+ int16_t *abuf_dst, const uint8_t *abuf_src)
+ {
+ int i, j;
+
+ dss_sp_unpack_coeffs(p, abuf_src);
+
+ dss_sp_unpack_filter(p);
+
+ dss_sp_convert_coeffs(p->lpc_filter, p->filter);
+
+ for (j = 0; j < SUBFRAMES; j++) {
+ dss_sp_gen_exc(p->vector_buf, p->history,
+ p->fparam.pitch_lag[j],
+ dss_sp_adaptive_gain[p->fparam.sf_adaptive_gain[j]]);
+
+ dss_sp_add_pulses(p->vector_buf, &p->fparam.sf[j]);
+
+ dss_sp_update_buf(p->vector_buf, p->history);
+
+ for (i = 0; i < 72; i++)
+ p->vector_buf[i] = p->history[72 - i];
+
+ dss_sp_shift_sq_sub(p->filter,
+ p->err_buf2, p->vector_buf);
+
+ dss_sp_sf_synthesis(p, p->lpc_filter[0],
+ &p->working_buffer[j][0], 72);
+ }
+
+ dss_sp_update_state(p, &p->working_buffer[0][0]);
+
+ dss_sp_32to16bit(abuf_dst,
+ &p->working_buffer[0][0], 264);
+ return 0;
+ }
+
+ static int dss_sp_decode_frame(AVCodecContext *avctx, void *data,
+ int *got_frame_ptr, AVPacket *avpkt)
+ {
+ DssSpContext *p = avctx->priv_data;
+ AVFrame *frame = data;
+ const uint8_t *buf = avpkt->data;
+ int buf_size = avpkt->size;
+
+ int16_t *out;
+ int ret;
+
+ if (buf_size < DSS_SP_FRAME_SIZE) {
+ if (buf_size)
+ av_log(avctx, AV_LOG_WARNING,
+ "Expected %d bytes, got %d - skipping packet.\n",
+ DSS_SP_FRAME_SIZE, buf_size);
+ *got_frame_ptr = 0;
+ return AVERROR_INVALIDDATA;
+ }
+
+ frame->nb_samples = DSS_SP_SAMPLE_COUNT;
+ if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
+ av_log(avctx, AV_LOG_ERROR, "get_buffer() failed.\n");
+ return ret;
+ }
+
+ out = (int16_t *)frame->data[0];
+
+ dss_sp_decode_one_frame(p, out, buf);
+
+ *got_frame_ptr = 1;
+
+ return DSS_SP_FRAME_SIZE;
+ }
+
+ AVCodec ff_dss_sp_decoder = {
+ .name = "DSS SP",
+ .long_name = NULL_IF_CONFIG_SMALL("Digital Speech Standard - Standard Play mode (DSS SP)"),
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = AV_CODEC_ID_DSS_SP,
+ .priv_data_size = sizeof(DssSpContext),
+ .init = dss_sp_decode_init,
+ .decode = dss_sp_decode_frame,
+ .capabilities = CODEC_CAP_DR1,
+ };
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