| /* |
| * 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 "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); |
| } |
| if (avctx->block_align < 1 || avctx->block_align % avctx->channels) { |
| 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; |
| 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->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->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, |
| 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, |
| .capabilities = AV_CODEC_CAP_DR1, |
| }; |
