[FFmpeg-devel] [RFC] SVX8 stereo files and AVCODEC_MAX_AUDIO_FRAME_SIZE
Michael Niedermayer
michaelni at gmx.at
Tue May 17 04:15:31 CEST 2011
On Tue, May 17, 2011 at 01:07:33AM +0200, Stefano Sabatini wrote:
> On date Monday 2011-05-16 18:50:08 +0200, Michael Niedermayer encoded:
> > On Sun, May 15, 2011 at 09:23:11PM +0200, Stefano Sabatini wrote:
> > > On date Sunday 2011-05-15 19:59:21 +0200, Stefano Sabatini encoded:
> > > > On date Sunday 2011-05-15 16:42:03 +0200, Stefano Sabatini encoded:
> > > > > On date Saturday 2011-05-14 22:19:02 +0200, Michael Niedermayer encoded:
> > > > > > On Sat, May 14, 2011 at 11:07:24AM +0200, Stefano Sabatini wrote:
> > > > > [...]
> > > > > > the decoder can just copy the whole into a internal buffer and then
> > > > > > return whatever fake partial reads it likes
> > > > > > isnt pretty but its simple
> > > > >
> > > > > I'm reading the whole audio chunk into the first packet, which is sent
> > > > > to the decoder, and which decodes/interleaves the whole audio buffer
> > > > > and returns it in little frames.
> > > > >
> > > > > Check attached unfinished patches.
> > > > >
> > > > > Now my problems is that for 8SVX files audio data format is signed
> > > > > 8-bit, which I see is not directly supported as format (why?).
> > > > >
> > > > > I could convert samples in the decoder signed->unsigned, but really
> > > > > why signed 8-bit is not supported natively (and what would prevent
> > > > > from adding it to resample/convert routines?).
> > > >
> > > > Updated work in progress. int8 -> uint8 conversion is done in-codec,
> > > > the clipping in delta_decode() looks necessary for avoiding overflow
> > > > noise in 8svx_fib.iff (still can't understand why xine doesn't need
> > > > it).
> > > >
> > >
> > > > Not yet ready for commit (still some minor problems in the decode_frame
> > > > codepath), please comment on the overall design.
> > >
> > > Fixed.
> > >
> > > Patch updated.
> > > --
> > > FFmpeg = Fundamental Funny Multipurpose Powered Encoding/decoding God
> >
> > > libavcodec/8svx.c | 192 ++++++++++++++++++++++++++++++++++++++++---------
> > > libavcodec/allcodecs.c | 1
> > > libavcodec/avcodec.h | 1
> > > libavformat/iff.c | 40 +---------
> > > 4 files changed, 163 insertions(+), 71 deletions(-)
> > > 0ced7e8e1b767d93f03b902343c9db887daf7eea 0004-iff-8svx-re-desing-8SVX-demuxing-and-decoding-for-co.patch
> > > From 3b19d249a0c1ead558f80b0e8e4454ef236130bb Mon Sep 17 00:00:00 2001
> > > From: Stefano Sabatini <stefano.sabatini-lala at poste.it>
> > > Date: Sun, 15 May 2011 13:24:46 +0200
> > > Subject: [PATCH] iff/8svx: re-desing 8SVX demuxing and decoding for correctly handling stereo data
> > >
> > > Make the iff demuxer send the whole audio chunk to the decoder, and by
> > > moving interleaving from the iff demuxer to the decoder, and
> > > introducing an 8svx_raw decoder which performs interleaving.
> > >
> > > This is required for correctly handle stereo data, since samples is
> > > stored like:
> > > LLLLLL....RRRRRR
> > >
> > > that is all left samples are at the beginning of the chunk, all right
> > > samples at the end, so it is necessary to store and process the whole
> > > buffer in order to decode each frame, so the decoder needs all the
> > > data before starting to return interleaved data.
> > >
> > > Fix decoding of files 8svx_exp.iff and 8svx_fib.iff, fix trac issue #169.
> > > ---
> > > libavcodec/8svx.c | 192 +++++++++++++++++++++++++++++++++++++++---------
> > > libavcodec/allcodecs.c | 1 +
> > > libavcodec/avcodec.h | 1 +
> > > libavformat/iff.c | 40 +---------
> > > 4 files changed, 163 insertions(+), 71 deletions(-)
> > >
> > > diff --git a/libavcodec/8svx.c b/libavcodec/8svx.c
> > > index 4f95d90..1c5394f 100644
> > > --- a/libavcodec/8svx.c
> > > +++ b/libavcodec/8svx.c
> > > @@ -1,5 +1,6 @@
> > > /*
> > > * Copyright (C) 2008 Jaikrishnan Menon
> > > + * Copyright (C) 2011 Stefano Sabatini
> > > *
> > > * This file is part of FFmpeg.
> > > *
> > > @@ -38,44 +39,137 @@
> > >
> > > /** decoder context */
> > > typedef struct EightSvxContext {
> > > - int16_t fib_acc;
> > > - const int16_t *table;
> > > + const int8_t *table;
> > > +
> > > + /* buffer used to store the whole audio decoded/interleaved chunk,
> > > + * which is sent with the first packet */
> > > + uint8_t *samples;
> > > + size_t samples_size;
> > > + int samples_idx;
> > > } EightSvxContext;
> > >
> > > -static const int16_t fibonacci[16] = { -34<<8, -21<<8, -13<<8, -8<<8, -5<<8, -3<<8, -2<<8, -1<<8,
> > > - 0, 1<<8, 2<<8, 3<<8, 5<<8, 8<<8, 13<<8, 21<<8 };
> > > -static const int16_t exponential[16] = { -128<<8, -64<<8, -32<<8, -16<<8, -8<<8, -4<<8, -2<<8, -1<<8,
> > > - 0, 1<<8, 2<<8, 4<<8, 8<<8, 16<<8, 32<<8, 64<<8 };
> > > +static const int8_t fibonacci[16] = { -34, -21, -13, -8, -5, -3, -2, -1, 0, 1, 2, 3, 5, 8, 13, 21 };
> > > +static const int8_t exponential[16] = { -128, -64, -32, -16, -8, -4, -2, -1, 0, 1, 2, 4, 8, 16, 32, 64 };
> > > +
> > > +#define MAX_FRAME_SIZE 2048
> > > +
> > > +/**
> > > + * Interleave samples in buffer containing all left channel samples
> > > + * at the beginning, and right channel samples at the end.
> > > + * Each sample is assumed to be in signed 8-bit format.
> > > + *
> > > + * @param size the size in bytes of the dst and src buffer
> > > + */
> > > +static void interleave_stereo(uint8_t *dst, const uint8_t *src, int size)
> > > +{
> > > + uint8_t *dst_end = dst + size;
> > > + size = size>>1;
> > > +
> > > + while (dst < dst_end) {
> > > + *dst++ = *src;
> > > + *dst++ = *(src+size);
> > > + src++;
> > > + }
> > > +}
> > > +
> > > +/**
> > > + * Delta decode the compressed values in src, and put the resulting
> > > + * decoded n samples in dst.
> > > + *
> > > + * @param val starting value assumed by the delta sequence
> > > + * @param table delta sequence table
> > > + * @return size in bytes of the decoded data, must be src_size*2
> > > + */
> > > +static int delta_decode(int8_t *dst, const uint8_t *src, int src_size,
> > > + int8_t val, const int8_t *table)
> > > +{
> > > + int n = src_size;
> > > + int8_t *dst0 = dst;
> > > +
> > > + while (n--) {
> > > + uint8_t d = *src++;
> > > + val = av_clip(val + table[d & 0x0f], -127, 128);
> > > + *dst++ = val;
> > > + val = av_clip(val + table[d >> 4] , -127, 128);
> > > + *dst++ = val;
> > > + }
> > > +
> > > + return dst-dst0;
> > > +}
> > >
> > > static int eightsvx_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
> > > AVPacket *avpkt)
> > > {
> > > EightSvxContext *esc = avctx->priv_data;
> > > + int out_data_size, n, consumed;
> > > + uint8_t *src, *dst;
> > >
> > > + /* decode and interleave the first packet */
> > > + if (!esc->samples && avpkt) {
> > > + uint8_t *samples, *interleaved_samples;
> > >
> > > + esc->samples_size = avctx->codec->id == CODEC_ID_8SVX_RAW ?
> > > + avpkt->size : avctx->channels + (avpkt->size-avctx->channels) * 2;
> > > +
> > > + /* decompress */
> > > + if (avctx->codec->id == CODEC_ID_8SVX_FIB || avctx->codec->id == CODEC_ID_8SVX_EXP) {
> > > + const uint8_t *buf = avpkt->data;
> > > + int buf_size = avpkt->size;
> > > + int n = esc->samples_size;
> > >
> > > + if (!(samples = av_malloc(esc->samples_size)))
> > > + return AVERROR(ENOMEM);
> >
> > malloc() is not a terribly fast operation, maybe the buffer could be
> > reused.
>
> This buffer is indeed reused when possible (e.g. because there was no
> interleaving).
>
> What do you suggest to use in place of av_malloc()?
i was thinking of av_fast_malloc()
[...]
--
Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB
Republics decline into democracies and democracies degenerate into
despotisms. -- Aristotle
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