libavcodec/8svx.c - third_party/ffmpeg - Git at Google

 /*
  * 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
  * @author Jaikrishnan Menon
  *
  * 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 "config_components.h"

 #include "libavutil/avassert.h"
 #include "libavutil/mem.h"
 #include "avcodec.h"
 #include "codec_internal.h"
 #include "decode.h"
 #include "libavutil/common.h"

 /** decoder context */
 typedef struct EightSvxContext {
     uint8_t fib_acc[2];
     const int8_t *table;

     /* buffer used to store the whole first packet.
        data is only sent as one large packet */
     uint8_t *data[2];
     int data_size;
     int data_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

 /**
  * Delta decode the compressed values in src, and put the resulting
  * decoded samples in dst.
  *
  * @param[in,out] state starting value. it is saved for use in the next call.
  * @param table delta sequence table
  */
 static void delta_decode(uint8_t *dst, const uint8_t *src, int src_size,
                          uint8_t *state, const int8_t *table)
 {
     uint8_t val = *state;

     while (src_size--) {
         uint8_t d = *src++;
         val = av_clip_uint8(val + table[d & 0xF]);
         *dst++ = val;
         val = av_clip_uint8(val + table[d >> 4]);
         *dst++ = val;
     }

     *state = val;
 }

 /** decode a frame */
 static int eightsvx_decode_frame(AVCodecContext *avctx, AVFrame *frame,
                                  int *got_frame_ptr, AVPacket *avpkt)
 {
     EightSvxContext *esc = avctx->priv_data;
     int channels         = avctx->ch_layout.nb_channels;
     int buf_size;
     int ch, ret;
     int hdr_size = 2;

     /* decode and interleave the first packet */
     if (!esc->data[0] && avpkt) {
         int chan_size = avpkt->size / channels - hdr_size;

         if (avpkt->size % channels) {
             av_log(avctx, AV_LOG_WARNING, "Packet with odd size, ignoring last byte\n");
         }
         if (avpkt->size < (hdr_size + 1) * channels) {
             av_log(avctx, AV_LOG_ERROR, "packet size is too small\n");
             return AVERROR_INVALIDDATA;
         }

         esc->fib_acc[0] = avpkt->data[1] + 128;
         if (channels == 2)
             esc->fib_acc[1] = avpkt->data[2+chan_size+1] + 128;

         esc->data_idx  = 0;
         esc->data_size = chan_size;
         if (!(esc->data[0] = av_malloc(chan_size)))
             return AVERROR(ENOMEM);
         if (channels == 2) {
             if (!(esc->data[1] = av_malloc(chan_size))) {
                 av_freep(&esc->data[0]);
                 return AVERROR(ENOMEM);
             }
         }
         memcpy(esc->data[0], &avpkt->data[hdr_size], chan_size);
         if (channels == 2)
             memcpy(esc->data[1], &avpkt->data[2*hdr_size+chan_size], chan_size);
     }
     if (!esc->data[0]) {
         av_log(avctx, AV_LOG_ERROR, "unexpected empty packet\n");
         return AVERROR_INVALIDDATA;
     }

     /* decode next piece of data from the buffer */
     buf_size = FFMIN(MAX_FRAME_SIZE, esc->data_size - esc->data_idx);
     if (buf_size <= 0) {
         *got_frame_ptr = 0;
         return avpkt->size;
     }

     /* get output buffer */
     frame->nb_samples = buf_size * 2;
     if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
         return ret;

     for (ch = 0; ch < channels; ch++) {
         delta_decode(frame->data[ch], &esc->data[ch][esc->data_idx],
                      buf_size, &esc->fib_acc[ch], esc->table);
     }

     esc->data_idx += buf_size;

     *got_frame_ptr = 1;

     return ((avctx->frame_num == 0) * hdr_size + buf_size) * channels;
 }

 static av_cold int eightsvx_decode_init(AVCodecContext *avctx)
 {
     EightSvxContext *esc = avctx->priv_data;

     if (avctx->ch_layout.nb_channels < 1 || avctx->ch_layout.nb_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 AV_CODEC_ID_8SVX_FIB: esc->table = fibonacci;    break;
     case AV_CODEC_ID_8SVX_EXP: esc->table = exponential;  break;
     default:
         av_assert1(0);
     }
     avctx->sample_fmt = AV_SAMPLE_FMT_U8P;

     return 0;
 }

 static av_cold int eightsvx_decode_close(AVCodecContext *avctx)
 {
     EightSvxContext *esc = avctx->priv_data;

     av_freep(&esc->data[0]);
     av_freep(&esc->data[1]);
     esc->data_size = 0;
     esc->data_idx = 0;

     return 0;
 }

 #if CONFIG_EIGHTSVX_FIB_DECODER
 const FFCodec ff_eightsvx_fib_decoder = {
   .p.name         = "8svx_fib",
   CODEC_LONG_NAME("8SVX fibonacci"),
   .p.type         = AVMEDIA_TYPE_AUDIO,
   .p.id           = AV_CODEC_ID_8SVX_FIB,
   .priv_data_size = sizeof (EightSvxContext),
   .init           = eightsvx_decode_init,
   FF_CODEC_DECODE_CB(eightsvx_decode_frame),
   .close          = eightsvx_decode_close,
   .p.capabilities = AV_CODEC_CAP_DR1,
   .p.sample_fmts  = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_U8P,
                                                     AV_SAMPLE_FMT_NONE },
 };
 #endif
 #if CONFIG_EIGHTSVX_EXP_DECODER
 const FFCodec ff_eightsvx_exp_decoder = {
   .p.name         = "8svx_exp",
   CODEC_LONG_NAME("8SVX exponential"),
   .p.type         = AVMEDIA_TYPE_AUDIO,
   .p.id           = AV_CODEC_ID_8SVX_EXP,
   .priv_data_size = sizeof (EightSvxContext),
   .init           = eightsvx_decode_init,
   FF_CODEC_DECODE_CB(eightsvx_decode_frame),
   .close          = eightsvx_decode_close,
   .p.capabilities = AV_CODEC_CAP_DR1,
   .p.sample_fmts  = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_U8P,
                                                     AV_SAMPLE_FMT_NONE },
 };
 #endif
	/*
	* 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
	* @author Jaikrishnan Menon
	*
	* 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 "config_components.h"

	#include "libavutil/avassert.h"
	#include "libavutil/mem.h"
	#include "avcodec.h"
	#include "codec_internal.h"
	#include "decode.h"
	#include "libavutil/common.h"

	/** decoder context */
	typedef struct EightSvxContext {
	uint8_t fib_acc[2];
	const int8_t *table;

	/* buffer used to store the whole first packet.
	data is only sent as one large packet */
	uint8_t *data[2];
	int data_size;
	int data_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

	/**
	* Delta decode the compressed values in src, and put the resulting
	* decoded samples in dst.
	*
	* @param[in,out] state starting value. it is saved for use in the next call.
	* @param table delta sequence table
	*/
	static void delta_decode(uint8_t dst, const uint8_t src, int src_size,
	uint8_t state, const int8_t table)
	{
	uint8_t val = *state;

	while (src_size--) {
	uint8_t d = *src++;
	val = av_clip_uint8(val + table[d & 0xF]);
	*dst++ = val;
	val = av_clip_uint8(val + table[d >> 4]);
	*dst++ = val;
	}

	*state = val;
	}

	/** decode a frame */
	static int eightsvx_decode_frame(AVCodecContext avctx, AVFrame frame,
	int got_frame_ptr, AVPacket avpkt)
	{
	EightSvxContext *esc = avctx->priv_data;
	int channels = avctx->ch_layout.nb_channels;
	int buf_size;
	int ch, ret;
	int hdr_size = 2;

	/* decode and interleave the first packet */
	if (!esc->data[0] && avpkt) {
	int chan_size = avpkt->size / channels - hdr_size;

	if (avpkt->size % channels) {
	av_log(avctx, AV_LOG_WARNING, "Packet with odd size, ignoring last byte\n");
	}
	if (avpkt->size < (hdr_size + 1) * channels) {
	av_log(avctx, AV_LOG_ERROR, "packet size is too small\n");
	return AVERROR_INVALIDDATA;
	}

	esc->fib_acc[0] = avpkt->data[1] + 128;
	if (channels == 2)
	esc->fib_acc[1] = avpkt->data[2+chan_size+1] + 128;

	esc->data_idx = 0;
	esc->data_size = chan_size;
	if (!(esc->data[0] = av_malloc(chan_size)))
	return AVERROR(ENOMEM);
	if (channels == 2) {
	if (!(esc->data[1] = av_malloc(chan_size))) {
	av_freep(&esc->data[0]);
	return AVERROR(ENOMEM);
	}
	}
	memcpy(esc->data[0], &avpkt->data[hdr_size], chan_size);
	if (channels == 2)
	memcpy(esc->data[1], &avpkt->data[2*hdr_size+chan_size], chan_size);
	}
	if (!esc->data[0]) {
	av_log(avctx, AV_LOG_ERROR, "unexpected empty packet\n");
	return AVERROR_INVALIDDATA;
	}

	/* decode next piece of data from the buffer */
	buf_size = FFMIN(MAX_FRAME_SIZE, esc->data_size - esc->data_idx);
	if (buf_size <= 0) {
	*got_frame_ptr = 0;
	return avpkt->size;
	}

	/* get output buffer */
	frame->nb_samples = buf_size * 2;
	if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
	return ret;

	for (ch = 0; ch < channels; ch++) {
	delta_decode(frame->data[ch], &esc->data[ch][esc->data_idx],
	buf_size, &esc->fib_acc[ch], esc->table);
	}

	esc->data_idx += buf_size;

	*got_frame_ptr = 1;

	return ((avctx->frame_num == 0) * hdr_size + buf_size) * channels;
	}

	static av_cold int eightsvx_decode_init(AVCodecContext *avctx)
	{
	EightSvxContext *esc = avctx->priv_data;

	if (avctx->ch_layout.nb_channels < 1 \|\| avctx->ch_layout.nb_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 AV_CODEC_ID_8SVX_FIB: esc->table = fibonacci; break;
	case AV_CODEC_ID_8SVX_EXP: esc->table = exponential; break;
	default:
	av_assert1(0);
	}
	avctx->sample_fmt = AV_SAMPLE_FMT_U8P;

	return 0;
	}

	static av_cold int eightsvx_decode_close(AVCodecContext *avctx)
	{
	EightSvxContext *esc = avctx->priv_data;

	av_freep(&esc->data[0]);
	av_freep(&esc->data[1]);
	esc->data_size = 0;
	esc->data_idx = 0;

	return 0;
	}

	#if CONFIG_EIGHTSVX_FIB_DECODER
	const FFCodec ff_eightsvx_fib_decoder = {
	.p.name = "8svx_fib",
	CODEC_LONG_NAME("8SVX fibonacci"),
	.p.type = AVMEDIA_TYPE_AUDIO,
	.p.id = AV_CODEC_ID_8SVX_FIB,
	.priv_data_size = sizeof (EightSvxContext),
	.init = eightsvx_decode_init,
	FF_CODEC_DECODE_CB(eightsvx_decode_frame),
	.close = eightsvx_decode_close,
	.p.capabilities = AV_CODEC_CAP_DR1,
	.p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_U8P,
	AV_SAMPLE_FMT_NONE },
	};
	#endif
	#if CONFIG_EIGHTSVX_EXP_DECODER
	const FFCodec ff_eightsvx_exp_decoder = {
	.p.name = "8svx_exp",
	CODEC_LONG_NAME("8SVX exponential"),
	.p.type = AVMEDIA_TYPE_AUDIO,
	.p.id = AV_CODEC_ID_8SVX_EXP,
	.priv_data_size = sizeof (EightSvxContext),
	.init = eightsvx_decode_init,
	FF_CODEC_DECODE_CB(eightsvx_decode_frame),
	.close = eightsvx_decode_close,
	.p.capabilities = AV_CODEC_CAP_DR1,
	.p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_U8P,
	AV_SAMPLE_FMT_NONE },
	};
	#endif