May 2016 - ffmpeg-devel - lists.ffmpeg.org

[PATCH] lavf/mpegtsenc: write metadata descriptor for timed ID3 packets
by Stefano Sabatini 06 Nov '16

06 Nov '16

This is required since some softwares are not able to correctly recognize the metadata. --- libavformat/mpegtsenc.c | 24 ++++++++++++++---------- 1 file changed, 14 insertions(+), 10 deletions(-) diff --git a/libavformat/mpegtsenc.c b/libavformat/mpegtsenc.c index 5f22032..4b9e71b 100644 --- a/libavformat/mpegtsenc.c +++ b/libavformat/mpegtsenc.c @@ -254,7 +254,7 @@ static void mpegts_write_pat(AVFormatContext *s) } /* NOTE: !str is accepted for an empty string */ -static void putstr8(uint8_t **q_ptr, const char *str) +static void putstr8(uint8_t **q_ptr, const char *str, int write_len) { uint8_t *q; int len; @@ -264,7 +264,8 @@ static void putstr8(uint8_t **q_ptr, const char *str) len = 0; else len = strlen(str); - *q++ = len; + if (write_len) + *q++ = len; memcpy(q, str, len); q += len; *q_ptr = q; @@ -626,12 +627,15 @@ static int mpegts_write_pmt(AVFormatContext *s, MpegTSService *service) *q++ = 'V'; *q++ = 'A'; } else if (st->codecpar->codec_id == AV_CODEC_ID_TIMED_ID3) { - *q++ = 0x5; /* MPEG-2 registration descriptor */ - *q++ = 4; - *q++ = 'I'; - *q++ = 'D'; - *q++ = '3'; - *q++ = ' '; + const char *tag = "ID3 "; + *q++ = 0x26; /* metadata descriptor */ + *q++ = 13; + put16(&q, 0xffff); /* metadata application format */ + putstr8(&q, tag, 0); + *q++ = 0xff; /* metadata format */ + putstr8(&q, tag, 0); + *q++ = 0; /* ID */ + *q++ = 0xF; } break; } @@ -676,8 +680,8 @@ static void mpegts_write_sdt(AVFormatContext *s) desc_len_ptr = q; q++; *q++ = ts->service_type; - putstr8(&q, service->provider_name); - putstr8(&q, service->name); + putstr8(&q, service->provider_name, 1); + putstr8(&q, service->name, 1); desc_len_ptr[0] = q - desc_len_ptr - 1; /* fill descriptor length */ -- 1.9.1

2 6

[PATCH] lavf/matroskaenc.c: use metadata key/value to set mastering metadata
by Neil Birkbeck 04 Nov '16

04 Nov '16

Use "master_display" key/value pair to specify mastering metadata in a similar formatting as accepted by libx265 (unless there is some other generic way to add side data to a stream from command line). Currently, the packet side data propagates from an input file to output file if it is transmuxed (mkv -> mkv). Perhaps we want to also use this same metadata key in matroskadec to also allow for the metadata to propagate during transcoding.

3 7

[PATCH] avcodec/qsv: fix coding options
by zeranoe＠gmail.com 31 Oct '16

31 Oct '16

From: Kyle Schwarz <zeranoe(a)gmail.com> --- libavcodec/qsvenc.c | 34 ++-------------------------------- 1 file changed, 2 insertions(+), 32 deletions(-) diff --git a/libavcodec/qsvenc.c b/libavcodec/qsvenc.c index 132cf47..b4821fc 100644 --- a/libavcodec/qsvenc.c +++ b/libavcodec/qsvenc.c @@ -132,9 +132,6 @@ static void dump_video_param(AVCodecContext *avctx, QSVEncContext *q, #if QSV_HAVE_CO2 mfxExtCodingOption2 *co2 = (mfxExtCodingOption2*)coding_opts[1]; #endif -#if QSV_HAVE_CO3 - mfxExtCodingOption3 *co3 = (mfxExtCodingOption3*)coding_opts[2]; -#endif av_log(avctx, AV_LOG_VERBOSE, "profile: %s; level: %"PRIu16"\n", print_profile(info->CodecProfile), info->CodecLevel); @@ -186,13 +183,6 @@ static void dump_video_param(AVCodecContext *avctx, QSVEncContext *q, info->ICQQuality, co2->LookAheadDepth); } #endif -#if QSV_HAVE_QVBR - else if (info->RateControlMethod == MFX_RATECONTROL_QVBR) { - av_log(avctx, AV_LOG_VERBOSE, "QVBRQuality: %"PRIu16"\n", - co3->QVBRQuality); - } -#endif - av_log(avctx, AV_LOG_VERBOSE, "NumSlice: %"PRIu16"; NumRefFrame: %"PRIu16"\n", info->NumSlice, info->NumRefFrame); av_log(avctx, AV_LOG_VERBOSE, "RateDistortionOpt: %s\n", @@ -567,31 +557,11 @@ static int qsv_retrieve_enc_params(AVCodecContext *avctx, QSVEncContext *q) .PPSBuffer = pps_buf, .PPSBufSize = sizeof(pps_buf) }; - mfxExtCodingOption co = { - .Header.BufferId = MFX_EXTBUFF_CODING_OPTION, - .Header.BufferSz = sizeof(co), - }; -#if QSV_HAVE_CO2 - mfxExtCodingOption2 co2 = { - .Header.BufferId = MFX_EXTBUFF_CODING_OPTION2, - .Header.BufferSz = sizeof(co2), - }; -#endif -#if QSV_HAVE_CO3 - mfxExtCodingOption3 co3 = { - .Header.BufferId = MFX_EXTBUFF_CODING_OPTION3, - .Header.BufferSz = sizeof(co3), - }; -#endif - mfxExtBuffer *ext_buffers[] = { (mfxExtBuffer*)&extradata, - (mfxExtBuffer*)&co, + (mfxExtBuffer*)&q->extco, #if QSV_HAVE_CO2 - (mfxExtBuffer*)&co2, -#endif -#if QSV_HAVE_CO3 - (mfxExtBuffer*)&co3, + (mfxExtBuffer*)&q->extco2, #endif }; -- 2.8.3.windows.1

3 4

[PATCH] lavf/img2dec: disable parsing if frame_size is specified
by Matthieu Bouron 19 Oct '16

19 Oct '16

From: Matthieu Bouron <matthieu.bouron(a)stupeflix.com> --- Hello, The following patch disable parsing if the frame_size option is specified. The main purpose here is to disable the use of parsers (which have a huge performance cost on embedded platforms) for single images when their size is known in advance. The patch sounds hackish to me though, but others might consider it OK (or not). The performance of the jpeg parser still need to be addressed at some point. Matthieu --- libavformat/img2dec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/libavformat/img2dec.c b/libavformat/img2dec.c index fe0e346..9aa6dd7 100644 --- a/libavformat/img2dec.c +++ b/libavformat/img2dec.c @@ -206,7 +206,7 @@ int ff_img_read_header(AVFormatContext *s1) s->is_pipe = 0; else { s->is_pipe = 1; - st->need_parsing = AVSTREAM_PARSE_FULL; + st->need_parsing = s->frame_size > 0 ? AVSTREAM_PARSE_NONE : AVSTREAM_PARSE_FULL; } if (s->ts_from_file == 2) { -- 2.7.2

2 2

[PATCH 0/3] simple_idct leftovers
by Christophe Gisquet 04 Sep '16

04 Sep '16

First patch addresses an issue raised by Donald. Second is one forgotten during the patchset iterations Third requires sample from ticket #4683, whose license isn't clear, all the more since it has a watermark indicating an unlicensed s/w. Having a 12bpp sample of that nature has been however very benefitial. Christophe Gisquet (3): x86: simple_idct10_template: use const simple_idct12: align C and x86 fate: add 12bpp sample libavcodec/simple_idct_template.c | 17 ++++------------ libavcodec/x86/constants.c | 28 -------------------------- libavcodec/x86/constants.h | 16 --------------- libavcodec/x86/idctdsp_init.c | 8 +++----- libavcodec/x86/proresdsp.asm | 13 ++++++++++++ libavcodec/x86/simple_idct10.asm | 33 +++++++++++++++++++++++++++---- libavcodec/x86/simple_idct10_template.asm | 13 ------------ tests/fate/image.mak | 6 ++++++ tests/ref/fate/jpg-12bpp | 2 ++ 9 files changed, 57 insertions(+), 79 deletions(-) create mode 100644 tests/ref/fate/jpg-12bpp -- 2.6.0

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[PATCH] lavf: add textdata virtual demuxer and demuxer
by Stefano Sabatini 30 Aug '16

30 Aug '16

This format is useful to inject custom user data into streams. --- doc/demuxers.texi | 40 +++++++++ doc/muxers.texi | 31 +++++++ libavformat/Makefile | 2 + libavformat/allformats.c | 1 + libavformat/fftextdatadec.c | 212 ++++++++++++++++++++++++++++++++++++++++++++ libavformat/fftextdataenc.c | 103 +++++++++++++++++++++ 6 files changed, 389 insertions(+) create mode 100644 libavformat/fftextdatadec.c create mode 100644 libavformat/fftextdataenc.c diff --git a/doc/demuxers.texi b/doc/demuxers.texi index e34f8b3..9fc58eb 100644 --- a/doc/demuxers.texi +++ b/doc/demuxers.texi @@ -254,6 +254,46 @@ This demuxer is used to demux FLV files and RTMP network streams. Allocate the streams according to the onMetaData array content. @end table +@anchor{fftextdata} +@section fftextdata, fftd + +FFmpeg text data demuxer. + +This special demuxer allows to read serialized data base64-encoded and +remux it. It is especially useful for injecting opaque data streams. + +The fftextdata bytestream consists of a sequence of packets. Each +packet starts with a timestamps expressed in a format recognized by +FFmpeg (see +@ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}) +followed by a sequence of spaces and the base64 encoded data for the +packet, terminated by ";". The data representation may contain +interleaved space characters (a space, a tab, or a newline) which are +ignored. + +At the moment a single stream can be represented by an fftextdata +bytestream. + +If an input filename is "fftextdata" or "fftd" then the file format is +recognized as fftextdata. + +@subsection Options +@table @option +@item codec_name +Set the codec name for the packets data. +@end table + +@subsection Examples + +@itemize +@item +Inject timed_id3 packed data stored into the data.fftd file into the +output file. +@example +ffmpeg -i input.mp4 -codec_name timed_id3 -f fftextdata -i data.fftd -y -map 0 -map 1 -c copy output.ts +@end example +@end itemize + @section libgme The Game Music Emu library is a collection of video game music file emulators. diff --git a/doc/muxers.texi b/doc/muxers.texi index c62d4b5..df5ec08 100644 --- a/doc/muxers.texi +++ b/doc/muxers.texi @@ -129,6 +129,37 @@ and the input video converted to MPEG-2 video, use the command: ffmpeg -i INPUT -c:a pcm_u8 -c:v mpeg2video -f crc - @end example +@section fftextdata, fftd + +FFmpeg text data muxer. + +The fftextdata bytestream consists of a sequence of packets. Each +packet starts with a timestamps expressed in a format recognized by +FFmpeg (see +@ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}) +followed by a sequence of spaces and the base64 encoded data for the +packet, terminated by ";". The data representation may contain +interleaved space characters (a space, a tab, or a newline) which are +ignored. + +At the moment only a single stream can be represented by an fftextdata +bytestream. + +This muxer can be used to reinject the stream (e.g. a data stream) in +a different output, or to provide serialized data of the encoded data. + +The output can then be read using the fftextdata demuxer. + +@subsection Examples + +@itemize +@item +Store a data stream to an output file: +@example +ffmpeg -i INPUT -codec copy -map 0 -an -vn data.fftd +@end example +@end itemize + @anchor{framecrc} @section framecrc diff --git a/libavformat/Makefile b/libavformat/Makefile index 742aff5..4effccd 100644 --- a/libavformat/Makefile +++ b/libavformat/Makefile @@ -162,6 +162,8 @@ OBJS-$(CONFIG_FFM_DEMUXER) += ffmdec.o OBJS-$(CONFIG_FFM_MUXER) += ffmenc.o OBJS-$(CONFIG_FFMETADATA_DEMUXER) += ffmetadec.o OBJS-$(CONFIG_FFMETADATA_MUXER) += ffmetaenc.o +OBJS-$(CONFIG_FFTEXTDATA_DEMUXER) += fftextdatadec.o +OBJS-$(CONFIG_FFTEXTDATA_MUXER) += fftextdataenc.o OBJS-$(CONFIG_FILMSTRIP_DEMUXER) += filmstripdec.o OBJS-$(CONFIG_FILMSTRIP_MUXER) += filmstripenc.o OBJS-$(CONFIG_FLAC_DEMUXER) += flacdec.o rawdec.o \ diff --git a/libavformat/allformats.c b/libavformat/allformats.c index e6ee8d6..7657f94 100644 --- a/libavformat/allformats.c +++ b/libavformat/allformats.c @@ -124,6 +124,7 @@ void av_register_all(void) REGISTER_MUXER (F4V, f4v); REGISTER_MUXDEMUX(FFM, ffm); REGISTER_MUXDEMUX(FFMETADATA, ffmetadata); + REGISTER_MUXDEMUX(FFTEXTDATA, fftextdata); REGISTER_MUXDEMUX(FILMSTRIP, filmstrip); REGISTER_MUXDEMUX(FLAC, flac); REGISTER_DEMUXER (FLIC, flic); diff --git a/libavformat/fftextdatadec.c b/libavformat/fftextdatadec.c new file mode 100644 index 0000000..9516559 --- /dev/null +++ b/libavformat/fftextdatadec.c @@ -0,0 +1,212 @@ +/* + * Copyright (c) 2016 Stefano Sabatini + * + * This file is part of FFmpeg. + * + * 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. + * + * 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 + */ + +/** + * @file + * timestamped data virtual demuxer + */ + +#include "libavutil/base64.h" +#include "libavutil/bprint.h" +#include "libavutil/opt.h" +#include "libavutil/parseutils.h" +#include "avformat.h" +#include "internal.h" + +typedef struct { + const AVClass *class; /**< Class for private options. */ + int nb_packets; + AVBPrint bp; + const char *codec_name; +} FFTextdataContext; + +av_cold static int fftextdata_read_close(AVFormatContext *avctx) +{ + FFTextdataContext *td = avctx->priv_data; + + av_bprint_finalize(&td->bp, NULL); + return 0; +} + +av_cold static int fftextdata_read_header(AVFormatContext *s) +{ + FFTextdataContext *td = s->priv_data; + AVStream *st; + const AVCodecDescriptor *cd; + + st = avformat_new_stream(s, NULL); + if (!st) + return AVERROR(ENOMEM); + + cd = avcodec_descriptor_get_by_name(td->codec_name); + if (!cd) { + av_log(s, AV_LOG_ERROR, "Impossible to find a codec with name '%s'\n", + td->codec_name); + return AVERROR(EINVAL); + } + + st->codecpar->codec_type = cd->type; + st->codecpar->codec_id = cd->id; + avpriv_set_pts_info(st, 64, 1, 1000000); + + av_bprint_init(&(td->bp), 0, 1); + td->nb_packets = 0; + + return 0; +} + +static inline int is_space(char c) +{ + return c == ' ' || c == '\t' || c == '\r' || c == '\n'; +} + +static int read_word(AVIOContext *avio, AVBPrint *bp) +{ + int c; + + av_bprint_clear(bp); + + /* skip spaces */ + do { + c = avio_r8(avio); + if (!c) + goto end; + } while (is_space(c)); + + /* read word */ + av_bprint_chars(bp, c, 1); + do { + c = avio_r8(avio); + if (!c) + goto end; + if (is_space(c)) { + avio_skip(avio, -1); + goto end; + } + av_bprint_chars(bp, c, 1); + } while (1); + +end: + return bp->len; +} + +static int read_data(AVIOContext *avio, AVBPrint *bp) +{ + int c; + + av_bprint_clear(bp); + + /* skip spaces */ + do { + c = avio_r8(avio); + if (!c) + goto end; + } while (is_space(c)); + + /* read data chunk */ + av_bprint_chars(bp, c, 1); + do { + c = avio_r8(avio); + if (!c || c == ';') + goto end; + if (is_space(c)) { + continue; + } + av_bprint_chars(bp, c, 1); + } while (1); + +end: + return bp->len; +} + +static int fftextdata_read_packet(AVFormatContext *s, AVPacket *pkt) +{ + FFTextdataContext *td = s->priv_data; + AVIOContext *avio = s->pb; + int ret; + AVBPrint *bp = &(td->bp); + + pkt->pos = avio_tell(avio); + + /* read PTS */ + ret = read_word(avio, bp); + if (ret == 0) + return AVERROR_EOF; + + ret = av_parse_time(&pkt->pts, bp->str, 1); + if (ret < 0) { + av_log(s, AV_LOG_ERROR, "Invalid time specification '%s' for data packet #%d\n", + bp->str, td->nb_packets); + return ret; + } + + ret = read_data(avio, bp); + if (ret == 0) { + av_log(s, AV_LOG_WARNING, "Incomplete packet #%d with no data at the end of the data stream\n", + td->nb_packets); + return AVERROR_EOF; + } + + pkt->size = AV_BASE64_DECODE_SIZE(ret); + pkt->data = av_malloc(pkt->size); + if (ret < 0) + return ret; + + ret = av_base64_decode(pkt->data, bp->str, pkt->size); + if (ret < 0) { + av_freep(&pkt->data); + return ret; + } + + pkt->size = ret; + pkt->flags |= AV_PKT_FLAG_KEY; + td->nb_packets++; + + return ret; +} + +#define OFFSET(x) offsetof(FFTextdataContext, x) + +#define D AV_OPT_FLAG_DECODING_PARAM + +#define OFFSET(x) offsetof(FFTextdataContext, x) + +static const AVOption options[] = { + { "codec_name", "set output codec name", OFFSET(codec_name), AV_OPT_TYPE_STRING, {.str = "bin_data"}, CHAR_MIN, CHAR_MAX, D }, + { NULL }, +}; + +static const AVClass fftextdata_class = { + .class_name = "fftexdata demuxer", + .item_name = av_default_item_name, + .option = options, + .version = LIBAVUTIL_VERSION_INT, +}; + +AVInputFormat ff_fftextdata_demuxer = { + .name = "fftextdata", + .long_name = NULL_IF_CONFIG_SMALL("Timestamped data virtual demuxer"), + .extensions = "fftextdata,fftd", + .priv_data_size = sizeof(FFTextdataContext), + .read_header = fftextdata_read_header, + .read_packet = fftextdata_read_packet, + .read_close = fftextdata_read_close, + .priv_class = &fftextdata_class, +}; diff --git a/libavformat/fftextdataenc.c b/libavformat/fftextdataenc.c new file mode 100644 index 0000000..6029ab0 --- /dev/null +++ b/libavformat/fftextdataenc.c @@ -0,0 +1,103 @@ +/* + * Copyright (c) 2016 Stefano Sabatini + * + * This file is part of FFmpeg. + * + * 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. + * + * 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 + */ + +/** + * @file + * timestamped data virtual muxer + */ + +#include "avformat.h" +#include "libavutil/base64.h" + +typedef struct { + uint8_t *buf; + size_t buf_size; +} FFTextdataContext; + +static int fftextdata_write_header(AVFormatContext *s) +{ + FFTextdataContext *td = s->priv_data; + + td->buf = NULL; + td->buf_size = 0; + + return 0; +} + +static int fftextdata_write_trailer(AVFormatContext *s) +{ + FFTextdataContext *td = s->priv_data; + + av_freep(&td->buf); + td->buf_size = 0; + + return 0; +} + +static int fftextdata_write_packet(AVFormatContext *s, AVPacket *pkt) +{ + FFTextdataContext *td = s->priv_data; + char ts[32]; + size_t encoded_data_size; + AVStream *st = s->streams[pkt->stream_index]; + int64_t pts = pkt->pts; + double secs; + int hours, mins; + + if (st->start_time != AV_NOPTS_VALUE) + pts += st->start_time; + + secs = (double)pkt->pts * av_q2d(st->time_base); + mins = (int)secs / 60; + secs = secs - mins * 60; + hours = mins / 60; + mins %= 60; + snprintf(ts, sizeof(ts), "%d:%02d:%09.6f", hours, mins, secs); + avio_put_str(s->pb, ts); + avio_skip(s->pb, -1); + avio_w8(s->pb, '\n'); + + encoded_data_size = AV_BASE64_SIZE(pkt->size); + if (encoded_data_size > td->buf_size) { + td->buf = av_realloc_f(td->buf, encoded_data_size, 1); + if (!td->buf) + return AVERROR(ENOMEM); + td->buf_size = encoded_data_size; + } + + av_base64_encode(td->buf, td->buf_size, pkt->data, pkt->size); + avio_put_str(s->pb, td->buf); + avio_skip(s->pb, -1); + + avio_put_str(s->pb, "\n;\n"); + avio_skip(s->pb, -1); + + return 0; +} + +AVOutputFormat ff_fftextdata_muxer = { + .name = "fftextdata", + .long_name = NULL_IF_CONFIG_SMALL("Timestamped data virtual muxer"), + .extensions = "fftextdata,fftd", + .priv_data_size = sizeof(FFTextdataContext), + .write_header = fftextdata_write_header, + .write_packet = fftextdata_write_packet, + .write_trailer = fftextdata_write_trailer, +}; -- 1.9.1

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[GSoC] Motion Interpolation
by Davinder Singh 30 Aug '16

30 Aug '16

There’s a lot of research done on Motion Estimation. Depending upon the intended application of the resultant motion vectors, the method used for motion estimation can be very different. Classification of Motion Estimation Methods: Direct Methods: In direct methods we calculate optical flow <https://en.wikipedia.org/wiki/Optical_flow> in the scene. - Phase Correlation - Block Matching - Spatio-Temporal Gradient - Optical flow: Uses optical flow equation to find motion in the scene. - Pel-recursive: Also compute optical flow, but in such a way that allow recursive computability on vector fields) Indirect Methods - Feature based Method: Find features in the frame, and used for estimation. Here are some papers on Frame Rate Up-Conversion (FRUC): Phase Correlation: This method relies on frequency-domain representation of data, calculated using fast Fourier transform. <https://en.wikipedia.org/wiki/Fast_Fourier_transform> Phase Correlation provides a correlation surface from the comparison of images. This enables the identification of motion on a pixel-by-pixel basis for correct processing of each motion type. Since phase correlation operates in the frequency rather than the spatial domain, it is able to zero in on details while ignoring such factors as noise and grain within the picture. In other words, the system is highly tolerant of the noise variations and rapid changes in luminance levels that are found in many types of content – resulting in high-quality performance on fades, objects moving in and out of shade, and light ashes. Papers: [1] "Disney Research » Phase-Based Frame Interpolation for Video." IEEE CVPR 2015 <https://www.disneyresearch.com/publication/phasebased/> [2] Yoo, DongGon et al. "Phase Correlated Bilateral Motion Estimation for Frame Rate Up-Conversion." The 23rd International Technical Conference on Circuits/Systems, Computers and Communications (ITC-CSCC Jul. 2008. <http://www.ieice.org/proceedings/ITC-CSCC2008/pdf/p385_G3-4.pdf> The video on paper [1] page demonstrate comparison between various methods. Optical Flow: http://www.cs.toronto.edu/~fleet/research/Papers/flowChapter05.pdf [3] Brox et al. "High accuracy optical flow estimation based on a theory for warping." Computer Vision - ECCV 2004: 25-36. < http://www.wisdom.weizmann.ac.il/~/vision/courses/2006_2/papers/optic_flow_… > Slowmovideo <http://slowmovideo.granjow.net/> open-source project is based on Optical flow equation. Algorithm we can implement is based on block matching method. Motion Compensated Frame Interpolation Paper: [4] Zhai et al. "A low complexity motion compensated frame interpolation method." IEEE ISCAS 2005: 4927-4930. <http://research.microsoft.com/pubs/69174/lowcomplexitymc.pdf> Block-based motion estimation and pixel-wise motion estimation are the two main categories of motion estimation methods. In general, pixel-wise motion estimation can attain accurate motion fields, but needs a substantial amount of computation. In contrast, block matching algorithms (BMA) can be efficiently implemented and provide good performance. Most MCFI algorithms utilize the block-matching algorithm (BMA) for motion estimation (ME). BMA is simple and easy to implement. It also generates a compactly represented motion field. However, unlike video compression, it is more important to find true motion trajectories in MCFI. The objective of MC in MCFI is not to minimize the energy of MC residual signals, but to reconstruct interpolated frames with better visual quality. The algorithm uses motion vectors which are embedded in bit-stream. If vectors exported by codec (using +export_mvs flag2) are used when available, computation of the motion vectors will be significantly reduced for realtime playback. Otherwise the mEstimate filter will generate MVs, and to make the process faster, same algorithms (used by x264 and x265) - Diamond, Hex, UMH, Star will be implemented in the filter. Other filter - mInterpolate will use the MVs in the frame side data to interpolate frames using various methods - OBMC (Overlapped block motion compensation), simple frame blending and frame duplication etc. However, MVs generated based on SAD or BAD might bring serious artifacts if they are used directly. So, the algorithm first examines the motion vectors and classify into two groups, one group with vectors which are considered to represent “true” motion, other having “bad” vectors, then carries out overlapped block bi-directional motion estimation on corresponding blocks having “bad” MVs. Finally, it utilizes motion vector post-processing and overlapped block motion compensation to generate interpolated frames and further reduce blocking artifacts. Details on each step are in the paper [4]. Paper 2: [5] Choi et al. "Motion-compensated frame interpolation using bilateral motion estimation and adaptive overlapped block motion compensation." Circuits and Systems for Video Technology, IEEE Transactions 2007: 407-416. <http://www.mku.edu.tr/getblogfile.php?keyid=793> Other Papers: Bai et al. "Visual-weighted motion compensation frame interpolation with motion vector refinement" Circuits and Systems (ISCAS), 2012 IEEE International Symposium 2012: 500-503. <http://www.icst.pku.edu.cn/course/icb/Pub%20Files/2012/bw_12-iscas.pdf> Park et al. "Motion compensated frame rate up-conversion using modified adaptive extended bilateral motion estimation" Journal of Automation and Control Engineering Vol 2.4 (2014). <http://www.joace.org/uploadfile/2014/0114/20140114120043876.pdf> Tsai et al. "Frame rate up-conversion using adaptive bilateral motion estimation" WSEAS International Conference. Proceedings. Mathematics and Computers in Science and Engineering: 2009. <http://www.wseas.us/e-library/conferences/2009/hangzhou/ACACOS/ACACOS31.pdf > Please share your thoughts on this. Meanwhile I'm implementing fast ME methods (dia, hex, star) in mEstimate and OBMC in mInterpolate. Regards, DSM_

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[PATCH] avutil/mem: remove av_realloc / av_malloc incompatibility warning
by Michael Niedermayer 23 Aug '16

23 Aug '16

memalign() is not guranteed to be compatible with free() or realloc() and for platforms in this category we have --enable-memalign-hack (which should be enabled automatically if such system is detected) Trying to somehow half support systems that can free() memalign memory but not reallocate it seems not worth the amount of work needed to keep 2 then incompatible allocation systems and ensure their seperation. That is unless this would affect a major platform on which we want to avoid the memalign hack code Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> --- libavutil/mem.h | 24 ------------------------ 1 file changed, 24 deletions(-) diff --git a/libavutil/mem.h b/libavutil/mem.h index 2a1e36d..761d3c6 100644 --- a/libavutil/mem.h +++ b/libavutil/mem.h @@ -107,12 +107,6 @@ av_alloc_size(1, 2) static inline void *av_malloc_array(size_t nmemb, size_t siz * reallocated. * @return Pointer to a newly-reallocated block or NULL if the block * cannot be reallocated or the function is used to free the memory block. - * @warning Pointers originating from the av_malloc() family of functions must - * not be passed to av_realloc(). The former can be implemented using - * memalign() (or other functions), and there is no guarantee that - * pointers from such functions can be passed to realloc() at all. - * The situation is undefined according to POSIX and may crash with - * some libc implementations. * @see av_fast_realloc() */ void *av_realloc(void *ptr, size_t size) av_alloc_size(2); @@ -137,12 +131,6 @@ void *av_realloc_f(void *ptr, size_t nelem, size_t elsize); * @param size Size in bytes for the memory block to be allocated or * reallocated * @return Zero on success, an AVERROR error code on failure. - * @warning Pointers originating from the av_malloc() family of functions must - * not be passed to av_reallocp(). The former can be implemented using - * memalign() (or other functions), and there is no guarantee that - * pointers from such functions can be passed to realloc() at all. - * The situation is undefined according to POSIX and may crash with - * some libc implementations. */ int av_reallocp(void *ptr, size_t size); @@ -156,12 +144,6 @@ int av_reallocp(void *ptr, size_t size); * @param size Size of the single element * @return Pointer to a newly-reallocated block or NULL if the block * cannot be reallocated or the function is used to free the memory block. - * @warning Pointers originating from the av_malloc() family of functions must - * not be passed to av_realloc(). The former can be implemented using - * memalign() (or other functions), and there is no guarantee that - * pointers from such functions can be passed to realloc() at all. - * The situation is undefined according to POSIX and may crash with - * some libc implementations. */ av_alloc_size(2, 3) void *av_realloc_array(void *ptr, size_t nmemb, size_t size); @@ -175,12 +157,6 @@ av_alloc_size(2, 3) void *av_realloc_array(void *ptr, size_t nmemb, size_t size) * @param nmemb Number of elements * @param size Size of the single element * @return Zero on success, an AVERROR error code on failure. - * @warning Pointers originating from the av_malloc() family of functions must - * not be passed to av_realloc(). The former can be implemented using - * memalign() (or other functions), and there is no guarantee that - * pointers from such functions can be passed to realloc() at all. - * The situation is undefined according to POSIX and may crash with - * some libc implementations. */ av_alloc_size(2, 3) int av_reallocp_array(void *ptr, size_t nmemb, size_t size); -- 1.7.9.5

5 8

[PATCH]ffmpeg: Do not set too large bits_per_raw_sample
by Carl Eugen Hoyos 17 Aug '16

17 Aug '16

Hi! Attached patch stops setting bits_per_raw_sample if it makes no sense as for example in the wmall24 -> pcm_s16 case: Stream #0:0: Audio: pcm_s16le, 96000 Hz, stereo, s16 (24 bit), 3072 kb/s Mostly tested with audio. Please comment, Carl Eugen

3 4

[PATCH] avcodec: Remove libfaac, the internal AAC encoder is better
by Kieran Kunhya 16 Aug '16

16 Aug '16

--- Changelog | 1 + configure | 6 -- doc/encoders.texi | 105 --------------------- doc/ffserver.conf | 2 +- doc/general.texi | 2 +- doc/muxers.texi | 4 +- doc/platform.texi | 2 +- libavcodec/Makefile | 1 - libavcodec/allcodecs.c | 1 - libavcodec/libfaac.c | 248 ------------------------------------------------- libavcodec/version.h | 2 +- 11 files changed, 7 insertions(+), 367 deletions(-) delete mode 100644 libavcodec/libfaac.c diff --git a/Changelog b/Changelog index b4a4dd7..7bcb0c4 100644 --- a/Changelog +++ b/Changelog @@ -22,6 +22,7 @@ version <next>: - musx demuxer - aix demuxer - remap filter +- libfaac removed version 3.0: - Common Encryption (CENC) MP4 encoding and decoding support diff --git a/configure b/configure index 94a66d8..32d710d 100755 --- a/configure +++ b/configure @@ -219,7 +219,6 @@ External library support: --enable-libcdio enable audio CD grabbing with libcdio [no] --enable-libdc1394 enable IIDC-1394 grabbing using libdc1394 and libraw1394 [no] - --enable-libfaac enable AAC encoding via libfaac [no] --enable-libfdk-aac enable AAC de/encoding via libfdk-aac [no] --enable-libflite enable flite (voice synthesis) support via libflite [no] --enable-libfreetype enable libfreetype, needed for drawtext filter [no] @@ -1467,7 +1466,6 @@ EXTERNAL_LIBRARY_LIST=" libcdio libcelt libdc1394 - libfaac libfdk_aac libflite libfontconfig @@ -2673,8 +2671,6 @@ pcm_mulaw_at_encoder_select="audio_frame_queue" chromaprint_muxer_deps="chromaprint" h264_videotoolbox_encoder_deps="videotoolbox_encoder pthreads" libcelt_decoder_deps="libcelt" -libfaac_encoder_deps="libfaac" -libfaac_encoder_select="audio_frame_queue" libfdk_aac_decoder_deps="libfdk_aac" libfdk_aac_encoder_deps="libfdk_aac" libfdk_aac_encoder_select="audio_frame_queue" @@ -4946,7 +4942,6 @@ die_license_disabled gpl libxvid die_license_disabled gpl x11grab die_license_disabled nonfree cuda -die_license_disabled nonfree libfaac die_license_disabled nonfree nvenc enabled gpl && die_license_disabled_gpl nonfree libfdk_aac enabled gpl && die_license_disabled_gpl nonfree openssl @@ -5534,7 +5529,6 @@ enabled libcelt && require libcelt celt/celt.h celt_decode -lcelt0 && { check_lib celt/celt.h celt_decoder_create_custom -lcelt0 || die "ERROR: libcelt must be installed and version must be >= 0.11.0."; } enabled libcaca && require_pkg_config caca caca.h caca_create_canvas -enabled libfaac && require2 libfaac "stdint.h faac.h" faacEncGetVersion -lfaac enabled libfdk_aac && { use_pkg_config fdk-aac "fdk-aac/aacenc_lib.h" aacEncOpen || { require libfdk_aac fdk-aac/aacenc_lib.h aacEncOpen -lfdk-aac && warn "using libfdk without pkg-config"; } } diff --git a/doc/encoders.texi b/doc/encoders.texi index f38cad3..5c09136 100644 --- a/doc/encoders.texi +++ b/doc/encoders.texi @@ -611,111 +611,6 @@ and slightly improves compression. @end table -@anchor{libfaac} -@section libfaac - -libfaac AAC (Advanced Audio Coding) encoder wrapper. - -This encoder is of much lower quality and is more unstable than any other AAC -encoders, so it's highly recommended to instead use other encoders, like -@ref{aacenc,,the native FFmpeg AAC encoder}. - -This encoder also requires the presence of the libfaac headers and library -during configuration. You need to explicitly configure the build with -@code{--enable-libfaac --enable-nonfree}. - -@subsection Options - -The following shared FFmpeg codec options are recognized. - -The following options are supported by the libfaac wrapper. The -@command{faac}-equivalent of the options are listed in parentheses. - -@table @option -@item b (@emph{-b}) -Set bit rate in bits/s for ABR (Average Bit Rate) mode. If the bit rate -is not explicitly specified, it is automatically set to a suitable -value depending on the selected profile. @command{faac} bitrate is -expressed in kilobits/s. - -Note that libfaac does not support CBR (Constant Bit Rate) but only -ABR (Average Bit Rate). - -If VBR mode is enabled this option is ignored. - -@item ar (@emph{-R}) -Set audio sampling rate (in Hz). - -@item ac (@emph{-c}) -Set the number of audio channels. - -@item cutoff (@emph{-C}) -Set cutoff frequency. If not specified (or explicitly set to 0) it -will use a value automatically computed by the library. Default value -is 0. - -@item profile -Set audio profile. - -The following profiles are recognized: -@table @samp -@item aac_main -Main AAC (Main) - -@item aac_low -Low Complexity AAC (LC) - -@item aac_ssr -Scalable Sample Rate (SSR) - -@item aac_ltp -Long Term Prediction (LTP) -@end table - -If not specified it is set to @samp{aac_low}. - -@item flags +qscale -Set constant quality VBR (Variable Bit Rate) mode. - -@item global_quality -Set quality in VBR mode as an integer number of lambda units. - -Only relevant when VBR mode is enabled with @code{flags +qscale}. The -value is converted to QP units by dividing it by @code{FF_QP2LAMBDA}, -and used to set the quality value used by libfaac. A reasonable range -for the option value in QP units is [10-500], the higher the value the -higher the quality. - -@item q (@emph{-q}) -Enable VBR mode when set to a non-negative value, and set constant -quality value as a double floating point value in QP units. - -The value sets the quality value used by libfaac. A reasonable range -for the option value is [10-500], the higher the value the higher the -quality. - -This option is valid only using the @command{ffmpeg} command-line -tool. For library interface users, use @option{global_quality}. -@end table - -@subsection Examples - -@itemize -@item -Use @command{ffmpeg} to convert an audio file to ABR 128 kbps AAC in an M4A (MP4) -container: -@example -ffmpeg -i input.wav -codec:a libfaac -b:a 128k -output.m4a -@end example - -@item -Use @command{ffmpeg} to convert an audio file to VBR AAC, using the -LTP AAC profile: -@example -ffmpeg -i input.wav -c:a libfaac -profile:a aac_ltp -q:a 100 output.m4a -@end example -@end itemize - @anchor{libfdk-aac-enc} @section libfdk_aac diff --git a/doc/ffserver.conf b/doc/ffserver.conf index 7a30fb6..e3f99bb 100644 --- a/doc/ffserver.conf +++ b/doc/ffserver.conf @@ -317,7 +317,7 @@ StartSendOnKey #AVPresetVideo baseline #AVOptionVideo flags +global_header # -#AudioCodec libfaac +#AudioCodec aac #AudioBitRate 32 #AudioChannels 2 #AudioSampleRate 22050 diff --git a/doc/general.texi b/doc/general.texi index 59ea4f4..fbcccdc 100644 --- a/doc/general.texi +++ b/doc/general.texi @@ -875,7 +875,7 @@ following image formats are supported: @item 8SVX exponential @tab @tab X @item 8SVX fibonacci @tab @tab X @item AAC @tab EX @tab X - @tab encoding supported through internal encoder and external libraries libfaac and libfdk-aac + @tab encoding supported through internal encoder and external library libfdk-aac @item AAC+ @tab E @tab IX @tab encoding supported through external library libfdk-aac @item AC-3 @tab IX @tab IX diff --git a/doc/muxers.texi b/doc/muxers.texi index 2aafbad..839909d 100644 --- a/doc/muxers.texi +++ b/doc/muxers.texi @@ -1290,9 +1290,9 @@ ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_fr @item Convert the @file{in.mkv} to TS segments using the @code{libx264} -and @code{libfaac} encoders: +and internal AAC encoders: @example -ffmpeg -i in.mkv -map 0 -codec:v libx264 -codec:a libfaac -f ssegment -segment_list out.list out%03d.ts +ffmpeg -i in.mkv -map 0 -codec:v libx264 -codec:a aac -f ssegment -segment_list out.list out%03d.ts @end example @item diff --git a/doc/platform.texi b/doc/platform.texi index f7ee456..65f3cb5 100644 --- a/doc/platform.texi +++ b/doc/platform.texi @@ -314,7 +314,7 @@ These library packages are only available from @uref{http://sourceware.org/cygwinports/, Cygwin Ports}: @example -yasm, libSDL-devel, libfaac-devel, libgsm-devel, libmp3lame-devel, +yasm, libSDL-devel, libgsm-devel, libmp3lame-devel, libschroedinger1.0-devel, speex-devel, libtheora-devel, libxvidcore-devel @end example diff --git a/libavcodec/Makefile b/libavcodec/Makefile index d4faf26..9b447fe 100644 --- a/libavcodec/Makefile +++ b/libavcodec/Makefile @@ -825,7 +825,6 @@ OBJS-$(CONFIG_ILBC_AT_ENCODER) += audiotoolboxenc.o OBJS-$(CONFIG_PCM_ALAW_AT_ENCODER) += audiotoolboxenc.o OBJS-$(CONFIG_PCM_MULAW_AT_ENCODER) += audiotoolboxenc.o OBJS-$(CONFIG_LIBCELT_DECODER) += libcelt_dec.o -OBJS-$(CONFIG_LIBFAAC_ENCODER) += libfaac.o OBJS-$(CONFIG_LIBFDK_AAC_DECODER) += libfdk-aacdec.o OBJS-$(CONFIG_LIBFDK_AAC_ENCODER) += libfdk-aacenc.o OBJS-$(CONFIG_LIBGSM_DECODER) += libgsmdec.o diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c index 44e7c3f..e4c8d37 100644 --- a/libavcodec/allcodecs.c +++ b/libavcodec/allcodecs.c @@ -581,7 +581,6 @@ void avcodec_register_all(void) REGISTER_DECODER(QDMC_AT, qdmc_at); REGISTER_DECODER(QDM2_AT, qdm2_at); REGISTER_DECODER(LIBCELT, libcelt); - REGISTER_ENCODER(LIBFAAC, libfaac); REGISTER_ENCDEC (LIBFDK_AAC, libfdk_aac); REGISTER_ENCDEC (LIBGSM, libgsm); REGISTER_ENCDEC (LIBGSM_MS, libgsm_ms); diff --git a/libavcodec/libfaac.c b/libavcodec/libfaac.c deleted file mode 100644 index 98b3ba8..0000000 --- a/libavcodec/libfaac.c +++ /dev/null @@ -1,248 +0,0 @@ -/* - * Interface to libfaac for aac encoding - * Copyright (c) 2002 Gildas Bazin <gbazin(a)netcourrier.com> - * - * This file is part of FFmpeg. - * - * 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. - * - * 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 - */ - -/** - * @file - * Interface to libfaac for aac encoding. - */ - -#include <faac.h> - -#include "libavutil/channel_layout.h" -#include "libavutil/common.h" -#include "avcodec.h" -#include "audio_frame_queue.h" -#include "internal.h" - - -/* libfaac has an encoder delay of 1024 samples */ -#define FAAC_DELAY_SAMPLES 1024 - -typedef struct FaacAudioContext { - faacEncHandle faac_handle; - AudioFrameQueue afq; -} FaacAudioContext; - -static av_cold int Faac_encode_close(AVCodecContext *avctx) -{ - FaacAudioContext *s = avctx->priv_data; - - av_freep(&avctx->extradata); - ff_af_queue_close(&s->afq); - - if (s->faac_handle) - faacEncClose(s->faac_handle); - - return 0; -} - -static const int channel_maps[][6] = { - { 2, 0, 1 }, //< C L R - { 2, 0, 1, 3 }, //< C L R Cs - { 2, 0, 1, 3, 4 }, //< C L R Ls Rs - { 2, 0, 1, 4, 5, 3 }, //< C L R Ls Rs LFE -}; - -static av_cold int Faac_encode_init(AVCodecContext *avctx) -{ - FaacAudioContext *s = avctx->priv_data; - faacEncConfigurationPtr faac_cfg; - unsigned long samples_input, max_bytes_output; - int ret; - - /* number of channels */ - if (avctx->channels < 1 || avctx->channels > 6) { - av_log(avctx, AV_LOG_ERROR, "encoding %d channel(s) is not allowed\n", avctx->channels); - ret = AVERROR(EINVAL); - goto error; - } - - s->faac_handle = faacEncOpen(avctx->sample_rate, - avctx->channels, - &samples_input, &max_bytes_output); - if (!s->faac_handle) { - av_log(avctx, AV_LOG_ERROR, "error in faacEncOpen()\n"); - ret = AVERROR_UNKNOWN; - goto error; - } - - /* check faac version */ - faac_cfg = faacEncGetCurrentConfiguration(s->faac_handle); - if (faac_cfg->version != FAAC_CFG_VERSION) { - av_log(avctx, AV_LOG_ERROR, "wrong libfaac version (compiled for: %d, using %d)\n", FAAC_CFG_VERSION, faac_cfg->version); - ret = AVERROR(EINVAL); - goto error; - } - - /* put the options in the configuration struct */ - switch(avctx->profile) { - case FF_PROFILE_AAC_MAIN: - faac_cfg->aacObjectType = MAIN; - break; - case FF_PROFILE_UNKNOWN: - case FF_PROFILE_AAC_LOW: - faac_cfg->aacObjectType = LOW; - break; - case FF_PROFILE_AAC_SSR: - faac_cfg->aacObjectType = SSR; - break; - case FF_PROFILE_AAC_LTP: - faac_cfg->aacObjectType = LTP; - break; - default: - av_log(avctx, AV_LOG_ERROR, "invalid AAC profile\n"); - ret = AVERROR(EINVAL); - goto error; - } - faac_cfg->mpegVersion = MPEG4; - faac_cfg->useTns = 0; - faac_cfg->allowMidside = 1; - faac_cfg->bitRate = avctx->bit_rate / avctx->channels; - faac_cfg->bandWidth = avctx->cutoff; - if(avctx->flags & AV_CODEC_FLAG_QSCALE) { - faac_cfg->bitRate = 0; - faac_cfg->quantqual = avctx->global_quality / FF_QP2LAMBDA; - } - faac_cfg->outputFormat = 1; - faac_cfg->inputFormat = FAAC_INPUT_16BIT; - if (avctx->channels > 2) - memcpy(faac_cfg->channel_map, channel_maps[avctx->channels-3], - avctx->channels * sizeof(int)); - - avctx->frame_size = samples_input / avctx->channels; - - /* Set decoder specific info */ - avctx->extradata_size = 0; - if (avctx->flags & AV_CODEC_FLAG_GLOBAL_HEADER) { - - unsigned char *buffer = NULL; - unsigned long decoder_specific_info_size; - - if (!faacEncGetDecoderSpecificInfo(s->faac_handle, &buffer, - &decoder_specific_info_size)) { - avctx->extradata = av_malloc(decoder_specific_info_size + AV_INPUT_BUFFER_PADDING_SIZE); - if (!avctx->extradata) { - ret = AVERROR(ENOMEM); - goto error; - } - avctx->extradata_size = decoder_specific_info_size; - memcpy(avctx->extradata, buffer, avctx->extradata_size); - faac_cfg->outputFormat = 0; - } - free(buffer); - } - - if (!faacEncSetConfiguration(s->faac_handle, faac_cfg)) { - int i; - for (i = avctx->bit_rate/1000; i ; i--) { - faac_cfg->bitRate = 1000*i / avctx->channels; - if (faacEncSetConfiguration(s->faac_handle, faac_cfg)) - break; - } - if (!i) { - av_log(avctx, AV_LOG_ERROR, "libfaac doesn't support this output format!\n"); - ret = AVERROR(EINVAL); - goto error; - } else { - avctx->bit_rate = 1000*i; - av_log(avctx, AV_LOG_WARNING, "libfaac doesn't support the specified bitrate, using %dkbit/s instead\n", i); - } - } - - avctx->initial_padding = FAAC_DELAY_SAMPLES; - ff_af_queue_init(avctx, &s->afq); - - return 0; -error: - Faac_encode_close(avctx); - return ret; -} - -static int Faac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, - const AVFrame *frame, int *got_packet_ptr) -{ - FaacAudioContext *s = avctx->priv_data; - int bytes_written, ret; - int num_samples = frame ? frame->nb_samples : 0; - void *samples = frame ? frame->data[0] : NULL; - - if ((ret = ff_alloc_packet2(avctx, avpkt, (7 + 768) * avctx->channels, 0)) < 0) - return ret; - - bytes_written = faacEncEncode(s->faac_handle, samples, - num_samples * avctx->channels, - avpkt->data, avpkt->size); - if (bytes_written < 0) { - av_log(avctx, AV_LOG_ERROR, "faacEncEncode() error\n"); - return bytes_written; - } - - /* add current frame to the queue */ - if (frame) { - if ((ret = ff_af_queue_add(&s->afq, frame)) < 0) - return ret; - } - - if (!bytes_written) - return 0; - - /* Get the next frame pts/duration */ - ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts, - &avpkt->duration); - - avpkt->size = bytes_written; - *got_packet_ptr = 1; - return 0; -} - -static const AVProfile profiles[] = { - { FF_PROFILE_AAC_MAIN, "Main" }, - { FF_PROFILE_AAC_LOW, "LC" }, - { FF_PROFILE_AAC_SSR, "SSR" }, - { FF_PROFILE_AAC_LTP, "LTP" }, - { FF_PROFILE_UNKNOWN }, -}; - -static const uint64_t faac_channel_layouts[] = { - AV_CH_LAYOUT_MONO, - AV_CH_LAYOUT_STEREO, - AV_CH_LAYOUT_SURROUND, - AV_CH_LAYOUT_4POINT0, - AV_CH_LAYOUT_5POINT0_BACK, - AV_CH_LAYOUT_5POINT1_BACK, - 0 -}; - -AVCodec ff_libfaac_encoder = { - .name = "libfaac", - .long_name = NULL_IF_CONFIG_SMALL("libfaac AAC (Advanced Audio Coding)"), - .type = AVMEDIA_TYPE_AUDIO, - .id = AV_CODEC_ID_AAC, - .priv_data_size = sizeof(FaacAudioContext), - .init = Faac_encode_init, - .encode2 = Faac_encode_frame, - .close = Faac_encode_close, - .capabilities = AV_CODEC_CAP_SMALL_LAST_FRAME | AV_CODEC_CAP_DELAY, - .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, - AV_SAMPLE_FMT_NONE }, - .profiles = NULL_IF_CONFIG_SMALL(profiles), - .channel_layouts = faac_channel_layouts, -}; diff --git a/libavcodec/version.h b/libavcodec/version.h index 1438e2e..c0f336b 100644 --- a/libavcodec/version.h +++ b/libavcodec/version.h @@ -28,7 +28,7 @@ #include "libavutil/version.h" #define LIBAVCODEC_VERSION_MAJOR 57 -#define LIBAVCODEC_VERSION_MINOR 34 +#define LIBAVCODEC_VERSION_MINOR 35 #define LIBAVCODEC_VERSION_MICRO 100 #define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \ -- 1.9.1

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