[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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