libavcodec/vmdaudio.c - play-games/ffmpeg - Git at Google


            
            /*

            
             * Sierra VMD audio decoder

            
             * Copyright (c) 2004 The FFmpeg Project

            
             *

            
             * 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

            
             * Sierra VMD audio decoder

            
             * by Vladimir "VAG" Gneushev (vagsoft at mail.ru)

            
             * for more information on the Sierra VMD format, visit:

            
             *   http://www.pcisys.net/~melanson/codecs/

            
             *

            
             * The audio decoder, expects each encoded data

            
             * chunk to be prepended with the appropriate 16-byte frame information

            
             * record from the VMD file. It does not require the 0x330-byte VMD file

            
             * header, but it does need the audio setup parameters passed in through

            
             * normal libavcodec API means.

            
             */


            #include <string.h>


            #include "libavutil/avassert.h"

            
            #include "libavutil/channel_layout.h"

            
            #include "libavutil/common.h"

            
            #include "libavutil/intreadwrite.h"


            #include "avcodec.h"

            
            #include "codec_internal.h"

            
            #include "decode.h"


            #define BLOCK_TYPE_AUDIO    1

            
            #define BLOCK_TYPE_INITIAL  2

            
            #define BLOCK_TYPE_SILENCE  3


            typedef struct VmdAudioContext {

            
                int out_bps;

            
                int chunk_size;

            
            } VmdAudioContext;


            static const uint16_t vmdaudio_table[128] = {

            
                0x000, 0x008, 0x010, 0x020, 0x030, 0x040, 0x050, 0x060, 0x070, 0x080,

            
                0x090, 0x0A0, 0x0B0, 0x0C0, 0x0D0, 0x0E0, 0x0F0, 0x100, 0x110, 0x120,

            
                0x130, 0x140, 0x150, 0x160, 0x170, 0x180, 0x190, 0x1A0, 0x1B0, 0x1C0,

            
                0x1D0, 0x1E0, 0x1F0, 0x200, 0x208, 0x210, 0x218, 0x220, 0x228, 0x230,

            
                0x238, 0x240, 0x248, 0x250, 0x258, 0x260, 0x268, 0x270, 0x278, 0x280,

            
                0x288, 0x290, 0x298, 0x2A0, 0x2A8, 0x2B0, 0x2B8, 0x2C0, 0x2C8, 0x2D0,

            
                0x2D8, 0x2E0, 0x2E8, 0x2F0, 0x2F8, 0x300, 0x308, 0x310, 0x318, 0x320,

            
                0x328, 0x330, 0x338, 0x340, 0x348, 0x350, 0x358, 0x360, 0x368, 0x370,

            
                0x378, 0x380, 0x388, 0x390, 0x398, 0x3A0, 0x3A8, 0x3B0, 0x3B8, 0x3C0,

            
                0x3C8, 0x3D0, 0x3D8, 0x3E0, 0x3E8, 0x3F0, 0x3F8, 0x400, 0x440, 0x480,

            
                0x4C0, 0x500, 0x540, 0x580, 0x5C0, 0x600, 0x640, 0x680, 0x6C0, 0x700,

            
                0x740, 0x780, 0x7C0, 0x800, 0x900, 0xA00, 0xB00, 0xC00, 0xD00, 0xE00,

            
                0xF00, 0x1000, 0x1400, 0x1800, 0x1C00, 0x2000, 0x3000, 0x4000

            
            };


            static av_cold int vmdaudio_decode_init(AVCodecContext *avctx)

            
            {

            
                VmdAudioContext *s = avctx->priv_data;

            
                int channels = avctx->ch_layout.nb_channels;


                if (channels < 1 || channels > 2) {

            
                    av_log(avctx, AV_LOG_ERROR, "invalid number of channels\n");

            
                    return AVERROR(EINVAL);

            
                }

            
                if (avctx->block_align < 1 || avctx->block_align % channels ||

            
                    avctx->block_align > INT_MAX - channels) {

            
                    av_log(avctx, AV_LOG_ERROR, "invalid block align\n");

            
                    return AVERROR(EINVAL);

            
                }


                av_channel_layout_uninit(&avctx->ch_layout);

            
                av_channel_layout_default(&avctx->ch_layout, channels);


                if (avctx->bits_per_coded_sample == 16)

            
                    avctx->sample_fmt = AV_SAMPLE_FMT_S16;

            
                else

            
                    avctx->sample_fmt = AV_SAMPLE_FMT_U8;

            
                s->out_bps = av_get_bytes_per_sample(avctx->sample_fmt);


                s->chunk_size = avctx->block_align + channels * (s->out_bps == 2);


                av_log(avctx, AV_LOG_DEBUG, "%d channels, %d bits/sample, "

            
                       "block align = %d, sample rate = %d\n",

            
                       channels, avctx->bits_per_coded_sample, avctx->block_align,

            
                       avctx->sample_rate);


                return 0;

            
            }


            static void decode_audio_s16(int16_t *out, const uint8_t *buf, int buf_size,

            
                                         int channels)

            
            {

            
                int ch;

            
                const uint8_t *buf_end = buf + buf_size;

            
                int predictor[2];

            
                int st = channels - 1;


                /* decode initial raw sample */

            
                for (ch = 0; ch < channels; ch++) {

            
                    predictor[ch] = (int16_t)AV_RL16(buf);

            
                    buf += 2;

            
                    *out++ = predictor[ch];

            
                }


                /* decode DPCM samples */

            
                ch = 0;

            
                while (buf < buf_end) {

            
                    uint8_t b = *buf++;

            
                    if (b & 0x80)

            
                        predictor[ch] -= vmdaudio_table[b & 0x7F];

            
                    else

            
                        predictor[ch] += vmdaudio_table[b];

            
                    predictor[ch] = av_clip_int16(predictor[ch]);

            
                    *out++ = predictor[ch];

            
                    ch ^= st;

            
                }

            
            }


            static int vmdaudio_decode_frame(AVCodecContext *avctx, AVFrame *frame,

            
                                             int *got_frame_ptr, AVPacket *avpkt)

            
            {

            
                const uint8_t *buf = avpkt->data;

            
                const uint8_t *buf_end;

            
                int buf_size = avpkt->size;

            
                VmdAudioContext *s = avctx->priv_data;

            
                int block_type, silent_chunks, audio_chunks;

            
                int ret;

            
                uint8_t *output_samples_u8;

            
                int16_t *output_samples_s16;

            
                int channels = avctx->ch_layout.nb_channels;


                if (buf_size < 16) {

            
                    av_log(avctx, AV_LOG_WARNING, "skipping small junk packet\n");

            
                    *got_frame_ptr = 0;

            
                    return buf_size;

            
                }


                block_type = buf[6];

            
                if (block_type < BLOCK_TYPE_AUDIO || block_type > BLOCK_TYPE_SILENCE) {

            
                    av_log(avctx, AV_LOG_ERROR, "unknown block type: %d\n", block_type);

            
                    return AVERROR(EINVAL);

            
                }

            
                buf      += 16;

            
                buf_size -= 16;


                /* get number of silent chunks */

            
                silent_chunks = 0;

            
                if (block_type == BLOCK_TYPE_INITIAL) {

            
                    uint32_t flags;

            
                    if (buf_size < 4) {

            
                        av_log(avctx, AV_LOG_ERROR, "packet is too small\n");

            
                        return AVERROR(EINVAL);

            
                    }

            
                    flags         = AV_RB32(buf);

            
                    silent_chunks = av_popcount(flags);

            
                    buf      += 4;

            
                    buf_size -= 4;

            
                } else if (block_type == BLOCK_TYPE_SILENCE) {

            
                    silent_chunks = 1;

            
                    buf_size = 0; // should already be zero but set it just to be sure

            
                }


                /* ensure output buffer is large enough */

            
                audio_chunks = buf_size / s->chunk_size;


                /* drop incomplete chunks */

            
                buf_size     = audio_chunks * s->chunk_size;


                if (silent_chunks + audio_chunks >= INT_MAX / avctx->block_align)

            
                    return AVERROR_INVALIDDATA;


                /* get output buffer */

            
                frame->nb_samples = ((silent_chunks + audio_chunks) * avctx->block_align) /

            
                                    avctx->ch_layout.nb_channels;

            
                if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)

            
                    return ret;

            
                output_samples_u8  =            frame->data[0];

            
                output_samples_s16 = (int16_t *)frame->data[0];


                /* decode silent chunks */

            
                if (silent_chunks > 0) {

            
                    int silent_size = avctx->block_align * silent_chunks;

            
                    av_assert0(avctx->block_align * silent_chunks <= frame->nb_samples * avctx->ch_layout.nb_channels);


                    if (s->out_bps == 2) {

            
                        memset(output_samples_s16, 0x00, silent_size * 2);

            
                        output_samples_s16 += silent_size;

            
                    } else {

            
                        memset(output_samples_u8,  0x80, silent_size);

            
                        output_samples_u8 += silent_size;

            
                    }

            
                }


                /* decode audio chunks */

            
                if (audio_chunks > 0) {

            
                    buf_end = buf + buf_size;

            
                    av_assert0((buf_size & (avctx->ch_layout.nb_channels > 1)) == 0);

            
                    while (buf_end - buf >= s->chunk_size) {

            
                        if (s->out_bps == 2) {

            
                            decode_audio_s16(output_samples_s16, buf, s->chunk_size, channels);

            
                            output_samples_s16 += avctx->block_align;

            
                        } else {

            
                            memcpy(output_samples_u8, buf, s->chunk_size);

            
                            output_samples_u8  += avctx->block_align;

            
                        }

            
                        buf += s->chunk_size;

            
                    }

            
                }


                *got_frame_ptr = 1;


                return avpkt->size;

            
            }


            const FFCodec ff_vmdaudio_decoder = {

            
                .p.name         = "vmdaudio",

            
                CODEC_LONG_NAME("Sierra VMD audio"),

            
                .p.type         = AVMEDIA_TYPE_AUDIO,

            
                .p.id           = AV_CODEC_ID_VMDAUDIO,

            
                .priv_data_size = sizeof(VmdAudioContext),

            
                .init           = vmdaudio_decode_init,

            
                FF_CODEC_DECODE_CB(vmdaudio_decode_frame),

            
                .p.capabilities = AV_CODEC_CAP_DR1,

            
            };
	/*
	* Sierra VMD audio decoder
	* Copyright (c) 2004 The FFmpeg Project
	*
	* 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
	* Sierra VMD audio decoder
	* by Vladimir "VAG" Gneushev (vagsoft at mail.ru)
	* for more information on the Sierra VMD format, visit:
	* http://www.pcisys.net/~melanson/codecs/
	*
	* The audio decoder, expects each encoded data
	* chunk to be prepended with the appropriate 16-byte frame information
	* record from the VMD file. It does not require the 0x330-byte VMD file
	* header, but it does need the audio setup parameters passed in through
	* normal libavcodec API means.
	*/

	#include <string.h>

	#include "libavutil/avassert.h"
	#include "libavutil/channel_layout.h"
	#include "libavutil/common.h"
	#include "libavutil/intreadwrite.h"

	#include "avcodec.h"
	#include "codec_internal.h"
	#include "decode.h"

	#define BLOCK_TYPE_AUDIO 1
	#define BLOCK_TYPE_INITIAL 2
	#define BLOCK_TYPE_SILENCE 3

	typedef struct VmdAudioContext {
	int out_bps;
	int chunk_size;
	} VmdAudioContext;

	static const uint16_t vmdaudio_table[128] = {
	0x000, 0x008, 0x010, 0x020, 0x030, 0x040, 0x050, 0x060, 0x070, 0x080,
	0x090, 0x0A0, 0x0B0, 0x0C0, 0x0D0, 0x0E0, 0x0F0, 0x100, 0x110, 0x120,
	0x130, 0x140, 0x150, 0x160, 0x170, 0x180, 0x190, 0x1A0, 0x1B0, 0x1C0,
	0x1D0, 0x1E0, 0x1F0, 0x200, 0x208, 0x210, 0x218, 0x220, 0x228, 0x230,
	0x238, 0x240, 0x248, 0x250, 0x258, 0x260, 0x268, 0x270, 0x278, 0x280,
	0x288, 0x290, 0x298, 0x2A0, 0x2A8, 0x2B0, 0x2B8, 0x2C0, 0x2C8, 0x2D0,
	0x2D8, 0x2E0, 0x2E8, 0x2F0, 0x2F8, 0x300, 0x308, 0x310, 0x318, 0x320,
	0x328, 0x330, 0x338, 0x340, 0x348, 0x350, 0x358, 0x360, 0x368, 0x370,
	0x378, 0x380, 0x388, 0x390, 0x398, 0x3A0, 0x3A8, 0x3B0, 0x3B8, 0x3C0,
	0x3C8, 0x3D0, 0x3D8, 0x3E0, 0x3E8, 0x3F0, 0x3F8, 0x400, 0x440, 0x480,
	0x4C0, 0x500, 0x540, 0x580, 0x5C0, 0x600, 0x640, 0x680, 0x6C0, 0x700,
	0x740, 0x780, 0x7C0, 0x800, 0x900, 0xA00, 0xB00, 0xC00, 0xD00, 0xE00,
	0xF00, 0x1000, 0x1400, 0x1800, 0x1C00, 0x2000, 0x3000, 0x4000
	};

	static av_cold int vmdaudio_decode_init(AVCodecContext *avctx)
	{
	VmdAudioContext *s = avctx->priv_data;
	int channels = avctx->ch_layout.nb_channels;

	if (channels < 1 \|\| channels > 2) {
	av_log(avctx, AV_LOG_ERROR, "invalid number of channels\n");
	return AVERROR(EINVAL);
	}
	if (avctx->block_align < 1 \|\| avctx->block_align % channels \|\|
	avctx->block_align > INT_MAX - channels) {
	av_log(avctx, AV_LOG_ERROR, "invalid block align\n");
	return AVERROR(EINVAL);
	}

	av_channel_layout_uninit(&avctx->ch_layout);
	av_channel_layout_default(&avctx->ch_layout, channels);

	if (avctx->bits_per_coded_sample == 16)
	avctx->sample_fmt = AV_SAMPLE_FMT_S16;
	else
	avctx->sample_fmt = AV_SAMPLE_FMT_U8;
	s->out_bps = av_get_bytes_per_sample(avctx->sample_fmt);

	s->chunk_size = avctx->block_align + channels * (s->out_bps == 2);

	av_log(avctx, AV_LOG_DEBUG, "%d channels, %d bits/sample, "
	"block align = %d, sample rate = %d\n",
	channels, avctx->bits_per_coded_sample, avctx->block_align,
	avctx->sample_rate);

	return 0;
	}

	static void decode_audio_s16(int16_t out, const uint8_t buf, int buf_size,
	int channels)
	{
	int ch;
	const uint8_t *buf_end = buf + buf_size;
	int predictor[2];
	int st = channels - 1;

	/* decode initial raw sample */
	for (ch = 0; ch < channels; ch++) {
	predictor[ch] = (int16_t)AV_RL16(buf);
	buf += 2;
	*out++ = predictor[ch];
	}

	/* decode DPCM samples */
	ch = 0;
	while (buf < buf_end) {
	uint8_t b = *buf++;
	if (b & 0x80)
	predictor[ch] -= vmdaudio_table[b & 0x7F];
	else
	predictor[ch] += vmdaudio_table[b];
	predictor[ch] = av_clip_int16(predictor[ch]);
	*out++ = predictor[ch];
	ch ^= st;
	}
	}

	static int vmdaudio_decode_frame(AVCodecContext avctx, AVFrame frame,
	int got_frame_ptr, AVPacket avpkt)
	{
	const uint8_t *buf = avpkt->data;
	const uint8_t *buf_end;
	int buf_size = avpkt->size;
	VmdAudioContext *s = avctx->priv_data;
	int block_type, silent_chunks, audio_chunks;
	int ret;
	uint8_t *output_samples_u8;
	int16_t *output_samples_s16;
	int channels = avctx->ch_layout.nb_channels;

	if (buf_size < 16) {
	av_log(avctx, AV_LOG_WARNING, "skipping small junk packet\n");
	*got_frame_ptr = 0;
	return buf_size;
	}

	block_type = buf[6];
	if (block_type < BLOCK_TYPE_AUDIO \|\| block_type > BLOCK_TYPE_SILENCE) {
	av_log(avctx, AV_LOG_ERROR, "unknown block type: %d\n", block_type);
	return AVERROR(EINVAL);
	}
	buf += 16;
	buf_size -= 16;

	/* get number of silent chunks */
	silent_chunks = 0;
	if (block_type == BLOCK_TYPE_INITIAL) {
	uint32_t flags;
	if (buf_size < 4) {
	av_log(avctx, AV_LOG_ERROR, "packet is too small\n");
	return AVERROR(EINVAL);
	}
	flags = AV_RB32(buf);
	silent_chunks = av_popcount(flags);
	buf += 4;
	buf_size -= 4;
	} else if (block_type == BLOCK_TYPE_SILENCE) {
	silent_chunks = 1;
	buf_size = 0; // should already be zero but set it just to be sure
	}

	/* ensure output buffer is large enough */
	audio_chunks = buf_size / s->chunk_size;

	/* drop incomplete chunks */
	buf_size = audio_chunks * s->chunk_size;

	if (silent_chunks + audio_chunks >= INT_MAX / avctx->block_align)
	return AVERROR_INVALIDDATA;

	/* get output buffer */
	frame->nb_samples = ((silent_chunks + audio_chunks) * avctx->block_align) /
	avctx->ch_layout.nb_channels;
	if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
	return ret;
	output_samples_u8 = frame->data[0];
	output_samples_s16 = (int16_t *)frame->data[0];

	/* decode silent chunks */
	if (silent_chunks > 0) {
	int silent_size = avctx->block_align * silent_chunks;
	av_assert0(avctx->block_align * silent_chunks <= frame->nb_samples * avctx->ch_layout.nb_channels);

	if (s->out_bps == 2) {
	memset(output_samples_s16, 0x00, silent_size * 2);
	output_samples_s16 += silent_size;
	} else {
	memset(output_samples_u8, 0x80, silent_size);
	output_samples_u8 += silent_size;
	}
	}

	/* decode audio chunks */
	if (audio_chunks > 0) {
	buf_end = buf + buf_size;
	av_assert0((buf_size & (avctx->ch_layout.nb_channels > 1)) == 0);
	while (buf_end - buf >= s->chunk_size) {
	if (s->out_bps == 2) {
	decode_audio_s16(output_samples_s16, buf, s->chunk_size, channels);
	output_samples_s16 += avctx->block_align;
	} else {
	memcpy(output_samples_u8, buf, s->chunk_size);
	output_samples_u8 += avctx->block_align;
	}
	buf += s->chunk_size;
	}
	}

	*got_frame_ptr = 1;

	return avpkt->size;
	}

	const FFCodec ff_vmdaudio_decoder = {
	.p.name = "vmdaudio",
	CODEC_LONG_NAME("Sierra VMD audio"),
	.p.type = AVMEDIA_TYPE_AUDIO,
	.p.id = AV_CODEC_ID_VMDAUDIO,
	.priv_data_size = sizeof(VmdAudioContext),
	.init = vmdaudio_decode_init,
	FF_CODEC_DECODE_CB(vmdaudio_decode_frame),
	.p.capabilities = AV_CODEC_CAP_DR1,
	};