libavcodec/vmdaudio.c
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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
  * License as published by the Free Software Foundation; either
  * version 2.1 of the License, or (at your option) any later version.
  *
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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
  * 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
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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
  */
 
 /**
  * @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>
 
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 #include "libavutil/avassert.h"
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 #include "libavutil/channel_layout.h"
 #include "libavutil/common.h"
 #include "libavutil/intreadwrite.h"
 
 #include "avcodec.h"
 #include "internal.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;
 
     if (avctx->channels < 1 || avctx->channels > 2) {
         av_log(avctx, AV_LOG_ERROR, "invalid number of channels\n");
         return AVERROR(EINVAL);
     }
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     if (avctx->block_align < 1 || avctx->block_align % avctx->channels) {
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         av_log(avctx, AV_LOG_ERROR, "invalid block align\n");
         return AVERROR(EINVAL);
     }
 
     avctx->channel_layout = avctx->channels == 1 ? AV_CH_LAYOUT_MONO :
                                                    AV_CH_LAYOUT_STEREO;
 
     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 + avctx->channels * (s->out_bps == 2);
 
     av_log(avctx, AV_LOG_DEBUG, "%d channels, %d bits/sample, "
            "block align = %d, sample rate = %d\n",
            avctx->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, void *data,
                                  int *got_frame_ptr, AVPacket *avpkt)
 {
     AVFrame *frame     = data;
     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;
 
     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;
 
     /* get output buffer */
     frame->nb_samples = ((silent_chunks + audio_chunks) * avctx->block_align) /
                         avctx->channels;
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     if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
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         return ret;
     output_samples_u8  =            frame->data[0];
     output_samples_s16 = (int16_t *)frame->data[0];
 
     /* decode silent chunks */
     if (silent_chunks > 0) {
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         int silent_size = avctx->block_align * silent_chunks;
         av_assert0(avctx->block_align * silent_chunks <= frame->nb_samples * avctx->channels);
 
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         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) {
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         buf_end = buf + buf_size;
         av_assert0((buf_size & (avctx->channels > 1)) == 0);
         while (buf_end - buf >= s->chunk_size) {
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             if (s->out_bps == 2) {
                 decode_audio_s16(output_samples_s16, buf, s->chunk_size,
                                  avctx->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;
 }
 
 AVCodec ff_vmdaudio_decoder = {
     .name           = "vmdaudio",
     .long_name      = NULL_IF_CONFIG_SMALL("Sierra VMD audio"),
     .type           = AVMEDIA_TYPE_AUDIO,
     .id             = AV_CODEC_ID_VMDAUDIO,
     .priv_data_size = sizeof(VmdAudioContext),
     .init           = vmdaudio_decode_init,
     .decode         = vmdaudio_decode_frame,
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     .capabilities   = AV_CODEC_CAP_DR1,
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 };