[FFmpeg-devel] [PATCH] iff/8svx: move decoding/deinterleaving in demuxer
Stefano Sabatini
stefano.sabatini-lala at poste.it
Sat May 28 13:42:52 CEST 2011
This is required for making possible to return audio data in packets
rather than return a huge packet with all the chunk data, which is
problematic for applications.
In particular ffplay cannot pause in the middle of a packet.
Also deprecate the 8SVX codec ids which are no longer required, and
somehow globally simplify the code.
Fix trac issue #215.
---
libavcodec/8svx.c | 225 ------------------------------------------------
libavcodec/Makefile | 3 -
libavcodec/allcodecs.c | 3 -
libavcodec/avcodec.h | 4 +
libavformat/iff.c | 169 +++++++++++++++++++++++++++++++-----
5 files changed, 150 insertions(+), 254 deletions(-)
delete mode 100644 libavcodec/8svx.c
diff --git a/libavcodec/8svx.c b/libavcodec/8svx.c
deleted file mode 100644
index 5d94e00..0000000
--- a/libavcodec/8svx.c
+++ /dev/null
@@ -1,225 +0,0 @@
-/*
- * Copyright (C) 2008 Jaikrishnan Menon
- * Copyright (C) 2011 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
- * 8svx audio decoder
- * supports: fibonacci delta encoding
- * : exponential encoding
- *
- * For more information about the 8SVX format:
- * http://netghost.narod.ru/gff/vendspec/iff/iff.txt
- * http://sox.sourceforge.net/AudioFormats-11.html
- * http://aminet.net/package/mus/misc/wavepak
- * http://amigan.1emu.net/reg/8SVX.txt
- *
- * Samples can be found here:
- * http://aminet.net/mods/smpl/
- */
-
-#include "avcodec.h"
-
-/** decoder context */
-typedef struct EightSvxContext {
- 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 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;
- uint8_t *src, *dst;
-
- /* decode and interleave the first packet */
- if (!esc->samples && avpkt) {
- uint8_t *deinterleaved_samples;
-
- esc->samples_size = avctx->codec->id == CODEC_ID_8SVX_RAW ?
- avpkt->size : avctx->channels + (avpkt->size-avctx->channels) * 2;
- if (!(esc->samples = av_malloc(esc->samples_size)))
- return AVERROR(ENOMEM);
-
- /* 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 (!(deinterleaved_samples = av_mallocz(n)))
- return AVERROR(ENOMEM);
-
- /* the uncompressed starting value is contained in the first byte */
- if (avctx->channels == 2) {
- delta_decode(deinterleaved_samples , buf+1, buf_size/2-1, buf[0], esc->table);
- buf += buf_size/2;
- delta_decode(deinterleaved_samples+n/2-1, buf+1, buf_size/2-1, buf[0], esc->table);
- } else
- delta_decode(deinterleaved_samples , buf+1, buf_size-1 , buf[0], esc->table);
- } else {
- deinterleaved_samples = avpkt->data;
- }
-
- if (avctx->channels == 2)
- interleave_stereo(esc->samples, deinterleaved_samples, esc->samples_size);
- else
- memcpy(esc->samples, deinterleaved_samples, esc->samples_size);
- }
-
- /* return single packed 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;
- esc->samples_idx += *data_size;
-
- return avctx->codec->id == CODEC_ID_8SVX_FIB || avctx->codec->id == CODEC_ID_8SVX_EXP ?
- (avctx->frame_number == 0)*2 + out_data_size / 2 :
- out_data_size;
-}
-
-static av_cold int eightsvx_decode_init(AVCodecContext *avctx)
-{
- EightSvxContext *esc = avctx->priv_data;
-
- if (avctx->channels > 2) {
- av_log(avctx, AV_LOG_ERROR, "8SVX does not support more than 2 channels\n");
- return AVERROR_INVALIDDATA;
- }
-
- switch (avctx->codec->id) {
- case CODEC_ID_8SVX_FIB: esc->table = fibonacci; break;
- case CODEC_ID_8SVX_EXP: esc->table = exponential; break;
- case CODEC_ID_8SVX_RAW: esc->table = NULL; break;
- default:
- av_log(avctx, AV_LOG_ERROR, "Invalid codec id %d.\n", avctx->codec->id);
- return AVERROR_INVALIDDATA;
- }
- avctx->sample_fmt = AV_SAMPLE_FMT_U8;
-
- return 0;
-}
-
-static av_cold int eightsvx_decode_close(AVCodecContext *avctx)
-{
- EightSvxContext *esc = avctx->priv_data;
-
- av_freep(&esc->samples);
- esc->samples_size = 0;
- esc->samples_idx = 0;
-
- return 0;
-}
-
-AVCodec ff_eightsvx_fib_decoder = {
- .name = "8svx_fib",
- .type = AVMEDIA_TYPE_AUDIO,
- .id = CODEC_ID_8SVX_FIB,
- .priv_data_size = sizeof (EightSvxContext),
- .init = eightsvx_decode_init,
- .decode = eightsvx_decode_frame,
- .close = eightsvx_decode_close,
- .long_name = NULL_IF_CONFIG_SMALL("8SVX fibonacci"),
-};
-
-AVCodec ff_eightsvx_exp_decoder = {
- .name = "8svx_exp",
- .type = AVMEDIA_TYPE_AUDIO,
- .id = CODEC_ID_8SVX_EXP,
- .priv_data_size = sizeof (EightSvxContext),
- .init = eightsvx_decode_init,
- .decode = eightsvx_decode_frame,
- .close = eightsvx_decode_close,
- .long_name = NULL_IF_CONFIG_SMALL("8SVX exponential"),
-};
-
-AVCodec ff_eightsvx_raw_decoder = {
- .name = "8svx_raw",
- .type = AVMEDIA_TYPE_AUDIO,
- .id = CODEC_ID_8SVX_RAW,
- .priv_data_size = sizeof(EightSvxContext),
- .init = eightsvx_decode_init,
- .decode = eightsvx_decode_frame,
- .close = eightsvx_decode_close,
- .long_name = NULL_IF_CONFIG_SMALL("8SVX rawaudio"),
-};
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index c6920ad..3698b18 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -137,9 +137,6 @@ OBJS-$(CONFIG_EATQI_DECODER) += eatqi.o eaidct.o mpeg12.o \
mpeg12data.o mpegvideo.o \
error_resilience.o
OBJS-$(CONFIG_EIGHTBPS_DECODER) += 8bps.o
-OBJS-$(CONFIG_EIGHTSVX_EXP_DECODER) += 8svx.o
-OBJS-$(CONFIG_EIGHTSVX_FIB_DECODER) += 8svx.o
-OBJS-$(CONFIG_EIGHTSVX_RAW_DECODER) += 8svx.o
OBJS-$(CONFIG_ESCAPE124_DECODER) += escape124.o
OBJS-$(CONFIG_FFV1_DECODER) += ffv1.o rangecoder.o
OBJS-$(CONFIG_FFV1_ENCODER) += ffv1.o rangecoder.o
diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c
index 2ad95bd..25d70ac 100644
--- a/libavcodec/allcodecs.c
+++ b/libavcodec/allcodecs.c
@@ -102,9 +102,6 @@ void avcodec_register_all(void)
REGISTER_DECODER (EATGV, eatgv);
REGISTER_DECODER (EATQI, eatqi);
REGISTER_DECODER (EIGHTBPS, eightbps);
- REGISTER_DECODER (EIGHTSVX_EXP, eightsvx_exp);
- REGISTER_DECODER (EIGHTSVX_FIB, eightsvx_fib);
- REGISTER_DECODER (EIGHTSVX_RAW, eightsvx_raw);
REGISTER_DECODER (ESCAPE124, escape124);
REGISTER_ENCDEC (FFV1, ffv1);
REGISTER_ENCDEC (FFVHUFF, ffvhuff);
diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h
index fdc86bb..2f362cb 100644
--- a/libavcodec/avcodec.h
+++ b/libavcodec/avcodec.h
@@ -166,8 +166,10 @@ enum CodecID {
CODEC_ID_INDEO5,
CODEC_ID_MIMIC,
CODEC_ID_RL2,
+#if LIBAVCODEC_VERSION_MAJOR < 54
CODEC_ID_8SVX_EXP,
CODEC_ID_8SVX_FIB,
+#endif
CODEC_ID_ESCAPE124,
CODEC_ID_DIRAC,
CODEC_ID_BFI,
@@ -204,7 +206,9 @@ enum CodecID {
CODEC_ID_PRORES,
CODEC_ID_JV,
CODEC_ID_DFA,
+#if LIBAVCODEC_VERSION_MAJOR < 54
CODEC_ID_8SVX_RAW,
+#endif
/* various PCM "codecs" */
CODEC_ID_PCM_S16LE= 0x10000,
diff --git a/libavformat/iff.c b/libavformat/iff.c
index 2dd1ef7..a3cf6cd 100644
--- a/libavformat/iff.c
+++ b/libavformat/iff.c
@@ -2,6 +2,7 @@
* Copyright (c) 2008 Jaikrishnan Menon <realityman at gmx.net>
* Copyright (c) 2010 Peter Ross <pross at xvid.org>
* Copyright (c) 2010 Sebastian Vater <cdgs.basty at googlemail.com>
+ * Copyright (C) 2011 Stefano Sabatini
*
* This file is part of FFmpeg.
*
@@ -92,8 +93,65 @@ typedef struct {
unsigned flags; ///< 1 for EHB, 0 is no extra half darkening
unsigned transparency; ///< transparency color index in palette
unsigned masking; ///< masking method used
+
+ /* for audio only*/
+ const int8_t *delta_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;
} IffDemuxContext;
+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_PACKET_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;
+}
+
/* Metadata string read */
static int get_metadata(AVFormatContext *s,
const char *const tag,
@@ -233,31 +291,24 @@ static int iff_read_header(AVFormatContext *s,
avio_seek(pb, iff->body_pos, SEEK_SET);
- switch(st->codec->codec_type) {
+ switch (st->codec->codec_type) {
case AVMEDIA_TYPE_AUDIO:
av_set_pts_info(st, 32, 1, st->codec->sample_rate);
- switch (iff->svx8_compression) {
- case COMP_NONE:
- st->codec->codec_id = CODEC_ID_8SVX_RAW;
- break;
- case COMP_FIB:
- st->codec->codec_id = CODEC_ID_8SVX_FIB;
- break;
- case COMP_EXP:
- st->codec->codec_id = CODEC_ID_8SVX_EXP;
- break;
- default:
+ if (iff->svx8_compression == COMP_FIB) {
+ iff->delta_table = fibonacci;
+ } else if (iff->svx8_compression == COMP_EXP) {
+ iff->delta_table = exponential;
+ } else if (iff->svx8_compression != COMP_NONE) {
av_log(s, AV_LOG_ERROR,
"Unknown SVX8 compression method '%d'\n", iff->svx8_compression);
- return -1;
+ return AVERROR_INVALIDDATA;
}
-
+ st->codec->codec_id = CODEC_ID_PCM_S8;
st->codec->bits_per_coded_sample = iff->svx8_compression == COMP_NONE ? 8 : 4;
st->codec->bit_rate = st->codec->channels * st->codec->sample_rate * st->codec->bits_per_coded_sample;
st->codec->block_align = st->codec->channels * st->codec->bits_per_coded_sample;
break;
-
case AVMEDIA_TYPE_VIDEO:
iff->bpp = st->codec->bits_per_coded_sample;
if ((screenmode & 0x800 /* Hold And Modify */) && iff->bpp <= 8) {
@@ -311,14 +362,75 @@ static int iff_read_packet(AVFormatContext *s,
AVStream *st = s->streams[0];
int ret;
- if(iff->sent_bytes >= iff->body_size)
- return AVERROR(EIO);
-
if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
- ret = av_get_packet(pb, pkt, iff->body_size);
+ size_t pkt_size;
+
+ /* slurp and process the whole audio data */
+ if (!iff->samples && pkt) {
+ uint8_t *deinterleaved_samples;
+ AVCodecContext *decctx = st->codec;
+ uint8_t *buf = av_malloc(iff->body_size);
+ if (!buf)
+ return AVERROR(ENOMEM);
+
+ if ((ret = avio_read(pb, buf, iff->body_size)) < 0) {
+ av_free(buf);
+ return ret;
+ }
+
+ /* read, decode, and interleave the whole data */
+ iff->samples_size = iff->svx8_compression == COMP_NONE ?
+ iff->body_size : decctx->channels + (iff->body_size - decctx->channels) * 2;
+
+ /* decompress */
+ if (iff->svx8_compression != COMP_NONE) {
+ uint8_t *buf_ptr = buf;
+ int buf_size = iff->body_size;
+ int n = iff->samples_size;
+
+ if (!(deinterleaved_samples = av_malloc(n))) {
+ av_free(buf);
+ return AVERROR(ENOMEM);
+ }
+
+ /* the uncompressed starting value is contained in the first byte */
+ if (decctx->channels == 2) {
+ delta_decode(deinterleaved_samples , buf_ptr+1, buf_size/2-1, buf_ptr[0], iff->delta_table);
+ buf_ptr += buf_size/2;
+ delta_decode(deinterleaved_samples+n/2-1, buf_ptr+1, buf_size/2-1, buf_ptr[0], iff->delta_table);
+ } else {
+ delta_decode(deinterleaved_samples , buf_ptr+1, buf_size-1 , buf_ptr[0], iff->delta_table);
+ }
+ av_freep(&buf);
+ } else
+ deinterleaved_samples = buf;
+
+ if (decctx->channels == 2) {
+ if (!(iff->samples = av_malloc(iff->samples_size))) {
+ av_free(deinterleaved_samples);
+ return AVERROR(ENOMEM);
+ }
+ interleave_stereo(iff->samples, deinterleaved_samples, iff->samples_size);
+ av_freep(&deinterleaved_samples);
+ } else
+ iff->samples = deinterleaved_samples;
+ }
+
+ pkt_size = FFMIN(MAX_PACKET_SIZE, iff->samples_size - iff->samples_idx);
+ ret = pkt_size;
+ if (iff->samples_idx >= iff->samples_size)
+ return AVERROR(EIO);
+
+ if (av_new_packet(pkt, pkt_size) < 0)
+ return AVERROR(ENOMEM);
+ memcpy(pkt->data, iff->samples + iff->samples_idx, pkt_size);
+ iff->samples_idx += pkt_size;
} else if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
uint8_t *buf;
+ if (iff->sent_bytes >= iff->body_size)
+ return AVERROR(EIO);
+
if (av_new_packet(pkt, iff->body_size + 2) < 0) {
return AVERROR(ENOMEM);
}
@@ -326,16 +438,26 @@ static int iff_read_packet(AVFormatContext *s,
buf = pkt->data;
bytestream_put_be16(&buf, 2);
ret = avio_read(pb, buf, iff->body_size);
+ if (iff->sent_bytes == 0)
+ pkt->flags |= AV_PKT_FLAG_KEY;
+ iff->sent_bytes = iff->body_size;
}
- if(iff->sent_bytes == 0)
- pkt->flags |= AV_PKT_FLAG_KEY;
- iff->sent_bytes = iff->body_size;
-
pkt->stream_index = 0;
return ret;
}
+static av_cold int iff_read_close(AVFormatContext *s)
+{
+ IffDemuxContext *iff = s->priv_data;
+
+ av_freep(&iff->samples);
+ iff->samples_size = 0;
+ iff->samples_idx = 0;
+
+ return 0;
+}
+
AVInputFormat ff_iff_demuxer = {
"IFF",
NULL_IF_CONFIG_SMALL("IFF format"),
@@ -343,4 +465,5 @@ AVInputFormat ff_iff_demuxer = {
iff_probe,
iff_read_header,
iff_read_packet,
+ iff_read_close,
};
--
1.7.2.3
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