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https://github.com/pineappleEA/pineapple-src.git
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241 lines
8.1 KiB
C
Executable File
241 lines
8.1 KiB
C
Executable File
/*
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* Sierra VMD audio decoder
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* Copyright (c) 2004 The FFmpeg Project
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* Sierra VMD audio decoder
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* by Vladimir "VAG" Gneushev (vagsoft at mail.ru)
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* for more information on the Sierra VMD format, visit:
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* http://www.pcisys.net/~melanson/codecs/
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*
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* The audio decoder, expects each encoded data
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* chunk to be prepended with the appropriate 16-byte frame information
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* record from the VMD file. It does not require the 0x330-byte VMD file
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* header, but it does need the audio setup parameters passed in through
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* normal libavcodec API means.
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*/
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#include <string.h>
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#include "libavutil/avassert.h"
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#include "libavutil/channel_layout.h"
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#include "libavutil/common.h"
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#include "libavutil/intreadwrite.h"
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#include "avcodec.h"
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#include "internal.h"
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#define BLOCK_TYPE_AUDIO 1
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#define BLOCK_TYPE_INITIAL 2
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#define BLOCK_TYPE_SILENCE 3
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typedef struct VmdAudioContext {
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int out_bps;
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int chunk_size;
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} VmdAudioContext;
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static const uint16_t vmdaudio_table[128] = {
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0x000, 0x008, 0x010, 0x020, 0x030, 0x040, 0x050, 0x060, 0x070, 0x080,
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0x090, 0x0A0, 0x0B0, 0x0C0, 0x0D0, 0x0E0, 0x0F0, 0x100, 0x110, 0x120,
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0x130, 0x140, 0x150, 0x160, 0x170, 0x180, 0x190, 0x1A0, 0x1B0, 0x1C0,
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0x1D0, 0x1E0, 0x1F0, 0x200, 0x208, 0x210, 0x218, 0x220, 0x228, 0x230,
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0x238, 0x240, 0x248, 0x250, 0x258, 0x260, 0x268, 0x270, 0x278, 0x280,
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0x288, 0x290, 0x298, 0x2A0, 0x2A8, 0x2B0, 0x2B8, 0x2C0, 0x2C8, 0x2D0,
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0x2D8, 0x2E0, 0x2E8, 0x2F0, 0x2F8, 0x300, 0x308, 0x310, 0x318, 0x320,
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0x328, 0x330, 0x338, 0x340, 0x348, 0x350, 0x358, 0x360, 0x368, 0x370,
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0x378, 0x380, 0x388, 0x390, 0x398, 0x3A0, 0x3A8, 0x3B0, 0x3B8, 0x3C0,
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0x3C8, 0x3D0, 0x3D8, 0x3E0, 0x3E8, 0x3F0, 0x3F8, 0x400, 0x440, 0x480,
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0x4C0, 0x500, 0x540, 0x580, 0x5C0, 0x600, 0x640, 0x680, 0x6C0, 0x700,
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0x740, 0x780, 0x7C0, 0x800, 0x900, 0xA00, 0xB00, 0xC00, 0xD00, 0xE00,
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0xF00, 0x1000, 0x1400, 0x1800, 0x1C00, 0x2000, 0x3000, 0x4000
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};
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static av_cold int vmdaudio_decode_init(AVCodecContext *avctx)
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{
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VmdAudioContext *s = avctx->priv_data;
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if (avctx->channels < 1 || avctx->channels > 2) {
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av_log(avctx, AV_LOG_ERROR, "invalid number of channels\n");
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return AVERROR(EINVAL);
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}
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if (avctx->block_align < 1 || avctx->block_align % avctx->channels ||
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avctx->block_align > INT_MAX - avctx->channels
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) {
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av_log(avctx, AV_LOG_ERROR, "invalid block align\n");
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return AVERROR(EINVAL);
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}
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avctx->channel_layout = avctx->channels == 1 ? AV_CH_LAYOUT_MONO :
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AV_CH_LAYOUT_STEREO;
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if (avctx->bits_per_coded_sample == 16)
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avctx->sample_fmt = AV_SAMPLE_FMT_S16;
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else
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avctx->sample_fmt = AV_SAMPLE_FMT_U8;
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s->out_bps = av_get_bytes_per_sample(avctx->sample_fmt);
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s->chunk_size = avctx->block_align + avctx->channels * (s->out_bps == 2);
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av_log(avctx, AV_LOG_DEBUG, "%d channels, %d bits/sample, "
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"block align = %d, sample rate = %d\n",
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avctx->channels, avctx->bits_per_coded_sample, avctx->block_align,
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avctx->sample_rate);
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return 0;
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}
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static void decode_audio_s16(int16_t *out, const uint8_t *buf, int buf_size,
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int channels)
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{
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int ch;
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const uint8_t *buf_end = buf + buf_size;
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int predictor[2];
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int st = channels - 1;
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/* decode initial raw sample */
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for (ch = 0; ch < channels; ch++) {
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predictor[ch] = (int16_t)AV_RL16(buf);
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buf += 2;
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*out++ = predictor[ch];
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}
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/* decode DPCM samples */
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ch = 0;
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while (buf < buf_end) {
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uint8_t b = *buf++;
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if (b & 0x80)
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predictor[ch] -= vmdaudio_table[b & 0x7F];
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else
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predictor[ch] += vmdaudio_table[b];
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predictor[ch] = av_clip_int16(predictor[ch]);
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*out++ = predictor[ch];
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ch ^= st;
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}
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}
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static int vmdaudio_decode_frame(AVCodecContext *avctx, void *data,
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int *got_frame_ptr, AVPacket *avpkt)
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{
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AVFrame *frame = data;
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const uint8_t *buf = avpkt->data;
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const uint8_t *buf_end;
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int buf_size = avpkt->size;
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VmdAudioContext *s = avctx->priv_data;
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int block_type, silent_chunks, audio_chunks;
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int ret;
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uint8_t *output_samples_u8;
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int16_t *output_samples_s16;
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if (buf_size < 16) {
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av_log(avctx, AV_LOG_WARNING, "skipping small junk packet\n");
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*got_frame_ptr = 0;
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return buf_size;
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}
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block_type = buf[6];
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if (block_type < BLOCK_TYPE_AUDIO || block_type > BLOCK_TYPE_SILENCE) {
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av_log(avctx, AV_LOG_ERROR, "unknown block type: %d\n", block_type);
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return AVERROR(EINVAL);
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}
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buf += 16;
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buf_size -= 16;
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/* get number of silent chunks */
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silent_chunks = 0;
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if (block_type == BLOCK_TYPE_INITIAL) {
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uint32_t flags;
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if (buf_size < 4) {
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av_log(avctx, AV_LOG_ERROR, "packet is too small\n");
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return AVERROR(EINVAL);
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}
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flags = AV_RB32(buf);
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silent_chunks = av_popcount(flags);
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buf += 4;
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buf_size -= 4;
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} else if (block_type == BLOCK_TYPE_SILENCE) {
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silent_chunks = 1;
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buf_size = 0; // should already be zero but set it just to be sure
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}
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/* ensure output buffer is large enough */
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audio_chunks = buf_size / s->chunk_size;
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/* drop incomplete chunks */
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buf_size = audio_chunks * s->chunk_size;
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if (silent_chunks + audio_chunks >= INT_MAX / avctx->block_align)
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return AVERROR_INVALIDDATA;
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/* get output buffer */
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frame->nb_samples = ((silent_chunks + audio_chunks) * avctx->block_align) /
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avctx->channels;
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if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
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return ret;
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output_samples_u8 = frame->data[0];
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output_samples_s16 = (int16_t *)frame->data[0];
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/* decode silent chunks */
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if (silent_chunks > 0) {
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int silent_size = avctx->block_align * silent_chunks;
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av_assert0(avctx->block_align * silent_chunks <= frame->nb_samples * avctx->channels);
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if (s->out_bps == 2) {
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memset(output_samples_s16, 0x00, silent_size * 2);
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output_samples_s16 += silent_size;
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} else {
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memset(output_samples_u8, 0x80, silent_size);
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output_samples_u8 += silent_size;
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}
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}
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/* decode audio chunks */
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if (audio_chunks > 0) {
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buf_end = buf + buf_size;
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av_assert0((buf_size & (avctx->channels > 1)) == 0);
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while (buf_end - buf >= s->chunk_size) {
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if (s->out_bps == 2) {
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decode_audio_s16(output_samples_s16, buf, s->chunk_size,
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avctx->channels);
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output_samples_s16 += avctx->block_align;
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} else {
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memcpy(output_samples_u8, buf, s->chunk_size);
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output_samples_u8 += avctx->block_align;
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}
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buf += s->chunk_size;
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}
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}
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*got_frame_ptr = 1;
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return avpkt->size;
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}
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AVCodec ff_vmdaudio_decoder = {
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.name = "vmdaudio",
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.long_name = NULL_IF_CONFIG_SMALL("Sierra VMD audio"),
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.type = AVMEDIA_TYPE_AUDIO,
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.id = AV_CODEC_ID_VMDAUDIO,
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.priv_data_size = sizeof(VmdAudioContext),
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.init = vmdaudio_decode_init,
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.decode = vmdaudio_decode_frame,
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.capabilities = AV_CODEC_CAP_DR1,
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};
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