[FFmpeg-devel] [PATCH] AST muxer

Paul B Mahol onemda at gmail.com
Thu Nov 22 20:02:27 CET 2012


On 11/22/12, jamal <jamrial at gmail.com> wrote:
>
> ---
>  Changelog                |   2 +-
>  doc/general.texi         |   2 +-
>  libavformat/Makefile     |   1 +
>  libavformat/allformats.c |   2 +-
>  libavformat/astenc.c     | 202 +++++++++++++++++++++++++++++++++++++++++++++++
>  tests/fate/avformat.mak  |   1 +
>  tests/lavf-regression.sh |   4 +
>  tests/ref/lavf/ast       |   3 +
>  8 files changed, 214 insertions(+), 3 deletions(-)
>  create mode 100644 libavformat/astenc.c
>  create mode 100644 tests/ref/lavf/ast
>
> diff --git a/Changelog b/Changelog
> index 2ff7921..b719f12 100644
> --- a/Changelog
> +++ b/Changelog
> @@ -24,7 +24,7 @@ version <next>:
>  - AVR demuxer
>  - geq filter ported from libmpcodecs
>  - remove ffserver daemon mode
> -- AST demuxer
> +- AST muxer/demuxer
>  - new expansion syntax for drawtext
>  - BRender PIX image decoder
>
> diff --git a/doc/general.texi b/doc/general.texi
> index 4bc0b78..dd98857 100644
> --- a/doc/general.texi
> +++ b/doc/general.texi
> @@ -148,7 +148,7 @@ library:
>  @item Apple HTTP Live Streaming @tab   @tab X
>  @item Artworx Data Format       @tab   @tab X
>  @item ASF                       @tab X @tab X
> - at item AST                       @tab   @tab X
> + at item AST                       @tab X @tab X
>      @tab Used on the Nintendo Wii.
>  @item AVI                       @tab X @tab X
>  @item AVISynth                  @tab   @tab X
> diff --git a/libavformat/Makefile b/libavformat/Makefile
> index 136ada8..fbc5c9d 100644
> --- a/libavformat/Makefile
> +++ b/libavformat/Makefile
> @@ -51,6 +51,7 @@ OBJS-$(CONFIG_ASF_MUXER)                 += asfenc.o asf.o
>  OBJS-$(CONFIG_ASS_DEMUXER)               += assdec.o
>  OBJS-$(CONFIG_ASS_MUXER)                 += assenc.o
>  OBJS-$(CONFIG_AST_DEMUXER)               += ast.o
> +OBJS-$(CONFIG_AST_MUXER)                 += astenc.o
>  OBJS-$(CONFIG_AU_DEMUXER)                += au.o pcm.o
>  OBJS-$(CONFIG_AU_MUXER)                  += au.o
>  OBJS-$(CONFIG_AVI_DEMUXER)               += avidec.o
> diff --git a/libavformat/allformats.c b/libavformat/allformats.c
> index eaeb51a..b73eeae 100644
> --- a/libavformat/allformats.c
> +++ b/libavformat/allformats.c
> @@ -64,7 +64,7 @@ void av_register_all(void)
>      REGISTER_DEMUXER  (APE, ape);
>      REGISTER_MUXDEMUX (ASF, asf);
>      REGISTER_MUXDEMUX (ASS, ass);
> -    REGISTER_DEMUXER  (AST, ast);
> +    REGISTER_MUXDEMUX (AST, ast);
>      REGISTER_MUXER    (ASF_STREAM, asf_stream);
>      REGISTER_MUXDEMUX (AU, au);
>      REGISTER_MUXDEMUX (AVI, avi);
> diff --git a/libavformat/astenc.c b/libavformat/astenc.c
> new file mode 100644
> index 0000000..2121e0c
> --- /dev/null
> +++ b/libavformat/astenc.c
> @@ -0,0 +1,202 @@
> +/*
> + * AST muxer
> + * Copyright (c) 2012 James Almer
> + *
> + * 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
> + */
> +
> +#include "avformat.h"
> +#include "avio_internal.h"
> +#include "libavutil/opt.h"
> +
> +typedef struct ASTContext {

Name it something like ASTMuxContext, it is muxer after all.
> +    AVClass *class;
> +    int64_t size;
> +    int64_t samples;
> +    int64_t loopstart;
> +    int64_t loopend;
> +    int fbs;
> +} ASTContext;
> +
> +#define CHECK_LOOP(type)                                                         \
> +    if(ast->loop ## type) {                                                      \
> +        ast->loop ## type = ast->loop ## type * sample;                          \
> +        av_log(s, AV_LOG_DEBUG, "loop ## type: %"PRId64"\n", ast->loop ## type); \
> +        if(ast->loop ## type < 0 || ast->loop ## type > UINT_MAX) {              \
> +            av_log(s, AV_LOG_ERROR, "Invalid loop" #type " value\n");            \
> +            return -1;                                                           \
> +        }                                                                        \
> +    }
> +
> +static int ast_write_header(AVFormatContext *s)
> +{
> +    ASTContext *ast = s->priv_data;
> +    AVIOContext *pb = s->pb;
> +    AVCodecContext *enc = s->streams[0]->codec;

There should be check for number of streams, because ast supports only
one audio stream.
> +    int codec_id;
> +    double sample = (double)enc->sample_rate / 1000;
> +    av_log(s, AV_LOG_DEBUG, "(double) sample_rate / 1000: %f\n", sample);

This 2 lines are of dubious debug usability and should be removed.
> +
> +    CHECK_LOOP(start)
> +    CHECK_LOOP(end)

Is this really required? Bellow AVOptions should already take care of that.
> +
> +    if(ast->loopstart && ast->loopend && ast->loopstart >= ast->loopend) {
> +        av_log(s, AV_LOG_ERROR, "Loopend can't be less or equal to loopstart\n");
> +        return -1;

AVERROR(EINVAL);
> +    }
> +
> +    switch(enc->codec_id) {
> +    case AV_CODEC_ID_ADPCM_AFC:
> +        codec_id = 0;
> +        break;
> +    case AV_CODEC_ID_PCM_S16BE_PLANAR:
> +        codec_id = 1;
> +        break;
> +    default:
> +        av_log(s, AV_LOG_ERROR, "Unsupported codec\n");
> +        return AVERROR_INVALIDDATA;
> +    }

You could add table which could then be shared between muxer and demuxer.
(But first demuxer file should be renamed to astdec)
> +
> +    if(enc->channels != 2 && enc->channels != 4) {
> +        av_log(s, AV_LOG_ERROR, "Unsupported channel amount: %d\n", enc->channels);
> +        return AVERROR_INVALIDDATA;
> +    }

Being able to test on Wii that different channel count are supported
would be welcome.

> +
> +    ffio_wfourcc(pb, "STRM");
> +
> +    ast->size = avio_tell(pb);
> +    avio_wb32(pb, 0); /* File size minus header */
> +    avio_wb16(pb, codec_id);
> +    avio_wb16(pb, enc->bits_per_coded_sample);

Demuxer does not export this. And it is actually bits per sample for
output which is always 16 (and not input, where it is 4 for APC and 16 for PCM).

> +    avio_wb16(pb, enc->channels);
> +    avio_wb16(pb, 0xFFFF);
> +    avio_wb32(pb, enc->sample_rate);
> +
> +    ast->samples = avio_tell(pb);
> +    avio_wb32(pb, 0); /* Number of samples */
> +    avio_wb32(pb, 0); /* Loopstart */
> +    avio_wb32(pb, 0); /* Loopend */
> +    avio_wb32(pb, 0); /* Size of first block */
> +
> +    /* Unknown */
> +    avio_wb32(pb, 0);
> +    avio_wl32(pb, 0x7F);
> +    avio_wb64(pb, 0);
> +    avio_wb64(pb, 0);
> +    avio_wb32(pb, 0);
> +
> +    avio_flush(pb);
> +
> +    return 0;
> +}
> +
> +static int ast_write_packet(AVFormatContext *s, AVPacket *pkt)
> +{
> +    AVIOContext *pb = s->pb;
> +    ASTContext *ast = s->priv_data;
> +    AVCodecContext *enc = s->streams[0]->codec;
> +    int size = pkt->size / enc->channels;
> +
> +    if(enc->frame_number == 1)
> +        ast->fbs = size;
> +
> +    ffio_wfourcc(pb, "BLCK");
> +    avio_wb32(pb, size); /* Block size */
> +
> +    /* padding */
> +    avio_wb64(pb, 0);
> +    avio_wb64(pb, 0);
> +    avio_wb64(pb, 0);
> +
> +    avio_write(pb, pkt->data, pkt->size);
> +
> +    return 0;
> +}
> +
> +static int ast_write_trailer(AVFormatContext *s)
> +{
> +    AVIOContext *pb = s->pb;
> +    ASTContext *ast = s->priv_data;
> +    AVCodecContext *enc = s->streams[0]->codec;
> +    int64_t file_size = avio_tell(pb);
> +    int64_t samples = (file_size - 64 - (32 * enc->frame_number)) / enc->block_align;

Does this work for APC too? You could use -c copy and then duration of
output should match one of input.
> +
> +    av_log(s, AV_LOG_DEBUG, "total samples: %"PRId64"\n", samples);
> +
> +    if (s->pb->seekable) {
> +        /* File size minus header */
> +        avio_seek(pb, ast->size, SEEK_SET);
> +        avio_wb32(pb, file_size - 64);
> +
> +        /* Number of samples */
> +        avio_seek(pb, ast->samples, SEEK_SET);
> +        avio_wb32(pb, samples);
> +
> +        /* Loopstart if provided */
> +        if(ast->loopstart && ast->loopstart >= samples) {
> +            av_log(s, AV_LOG_WARNING, "Loopstart value is out of range and will be ignored\n");
> +            ast->loopstart = 0;
> +        }
> +        avio_wb32(pb, ast->loopstart);
> +
> +        /* Loopend if provided. Otherwise number of samples again */
> +        if(ast->loopend) {
> +            if(ast->loopend > samples) {
> +                av_log(s, AV_LOG_WARNING, "Loopend value is out of range and will be ignored\n");
> +                ast->loopend = samples;
> +            }
> +            avio_wb32(pb, ast->loopend);
> +        } else {
> +            avio_wb32(pb, samples);
> +        }
> +
> +        /* Size of first block */
> +        avio_seek(pb, ast->samples + 12, SEEK_SET);
> +        avio_wb32(pb, ast->fbs);
> +
> +        avio_seek(pb, file_size, SEEK_SET);

SEEK_END
> +        avio_flush(pb);
> +    }
> +    return 0;
> +}
> +
> +#define OFFSET(obj) offsetof(ASTContext, obj)
> +static const AVOption options[] = {
> +  { "loopstart", "Loopstart position in miliseconds.", OFFSET(loopstart), AV_OPT_TYPE_INT64, { .i64 = 0 }, 0, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM },
> +  { "loopend",   "Loopend position in miliseconds.",   OFFSET(loopend),   AV_OPT_TYPE_INT64, { .i64 = 0 }, 0, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM },
> +  { NULL },
> +};
> +
> +static const AVClass ast_muxer_class = {
> +    .class_name = "AST muxer",
> +    .item_name  = av_default_item_name,
> +    .option     = options,
> +    .version    = LIBAVUTIL_VERSION_INT,
> +};
> +
> +AVOutputFormat ff_ast_muxer = {
> +    .name              = "ast",
> +    .long_name         = NULL_IF_CONFIG_SMALL("AST (Audio Stream)"),
> +    .extensions        = "ast",
> +    .priv_data_size    = sizeof(ASTContext),
> +    .audio_codec       = AV_CODEC_ID_PCM_S16BE_PLANAR,
> +    .video_codec       = AV_CODEC_ID_NONE,
> +    .write_header      = ast_write_header,
> +    .write_packet      = ast_write_packet,
> +    .write_trailer     = ast_write_trailer,
> +    .priv_class        = &ast_muxer_class,
> +};
> diff --git a/tests/fate/avformat.mak b/tests/fate/avformat.mak
> index 77c6a2f..4c99707 100644
> --- a/tests/fate/avformat.mak
> +++ b/tests/fate/avformat.mak
> @@ -1,6 +1,7 @@
>  FATE_LAVF-$(call ENCDEC,  PCM_S16BE,             AIFF)               += aiff
>  FATE_LAVF-$(call ENCDEC,  PCM_ALAW,              PCM_ALAW)           += alaw
>  FATE_LAVF-$(call ENCDEC2, MSMPEG4V3,  MP2,       ASF)                += asf
> +FATE_LAVF-$(call ENCDEC,  PCM_S16BE_PLANAR,      AST)                += ast
>  FATE_LAVF-$(call ENCDEC,  PCM_S16BE,             AU)                 += au
>  FATE_LAVF-$(call ENCDEC2, MPEG4,      MP2,       AVI)                += avi
>  FATE_LAVF-$(call ENCDEC,  BMP,                   IMAGE2)             += bmp
> diff --git a/tests/lavf-regression.sh b/tests/lavf-regression.sh
> index 64ebc0a..473c235 100755
> --- a/tests/lavf-regression.sh
> +++ b/tests/lavf-regression.sh
> @@ -326,6 +326,10 @@ if [ -n "$do_caf" ] ; then
>  do_audio_only caf
>  fi
>
> +if [ -n "$do_ast" ] ; then
> +do_audio_only ast "-ac 2" "-loopstart 1 -loopend 10"
> +fi
> +
>  # pix_fmt conversions
>
>  if [ -n "$do_pixfmt" ] ; then
> diff --git a/tests/ref/lavf/ast b/tests/ref/lavf/ast
> new file mode 100644
> index 0000000..72a9824
> --- /dev/null
> +++ b/tests/ref/lavf/ast
> @@ -0,0 +1,3 @@
> +7fa8cd2dd7453428e71930a7c65f7b62 *./tests/data/lavf/lavf.ast
> +181696 ./tests/data/lavf/lavf.ast
> +./tests/data/lavf/lavf.ast CRC=0x7bd585ff
> --
> 1.8.0.msysgit.0
>

That's all (for now).


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