/* * 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 #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 "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; 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 == 1); 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, 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; 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", .p.long_name = NULL_IF_CONFIG_SMALL("Sierra VMD audio"), .p.type = AVMEDIA_TYPE_AUDIO, .p.id = AV_CODEC_ID_VMDAUDIO, .priv_data_size = sizeof(VmdAudioContext), .init = vmdaudio_decode_init, .decode = vmdaudio_decode_frame, .p.capabilities = AV_CODEC_CAP_DR1, .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE, };