[FFmpeg-devel] [RFC] SVX8 stereo files and AVCODEC_MAX_AUDIO_FRAME_SIZE
Michael Niedermayer
michaelni at gmx.at
Mon May 16 18:50:08 CEST 2011
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.
>
> + /* the uncompressed starting value is contained in the first byte */
> + if (avctx->channels == 2) {
> + delta_decode(samples , buf+1, buf_size/2-1, buf[0], esc->table);
> + buf += buf_size/2;
> + delta_decode(samples+n/2-1, buf+1, buf_size/2-1, buf[0], esc->table);
> + } else
> + delta_decode(samples , buf+1, buf_size-1 , buf[0], esc->table);
> + } else
> + samples = avpkt->data;
> +
> + /* interleave */
> + if (avctx->channels == 2) {
> + if (!(interleaved_samples = av_malloc(esc->samples_size)))
> + return AVERROR(ENOMEM);
> + interleave_stereo(interleaved_samples, samples, esc->samples_size);
> + if (samples != avpkt->data)
> + av_freep(&samples);
> + } else
> + interleaved_samples = samples;
> +
> + /* copy data if necessary */
> + if (interleaved_samples == avpkt->data) {
> + if (!(interleaved_samples = av_malloc(esc->samples_size)))
> + return AVERROR(ENOMEM);
> + memcpy(interleaved_samples, samples, esc->samples_size);
> + }
> +
> + esc->samples = interleaved_samples;
> }
>
> + /* return frames with fixed size */
> + out_data_size = FFMIN(MAX_FRAME_SIZE, esc->samples_size - esc->samples_idx);
> + if (*data_size < out_data_size) {
> + av_log(avctx, AV_LOG_ERROR, "Provided buffer with size %d is too small.\n", *data_size);
> + return AVERROR(EINVAL);
> + }
> +
> + *data_size = out_data_size;
> + dst = data;
> + src = esc->samples + esc->samples_idx;
> + for (n = out_data_size; n > 0; n--)
> + *dst++ = *src++ + 128;
a ^0x80808080 could be used to add 128 to 4 at once
anyway, the patch is LGTM if tested, above can be done
later / seperately
[...]
--
Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB
Good people do not need laws to tell them to act responsibly, while bad
people will find a way around the laws. -- Plato
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