mirror of https://github.com/FFmpeg/FFmpeg.git
Nidhi Makhijani
11 years ago
committed by
Diego Biurrun
parent
3c650efb81
commit
246f869590
3 changed files with 238 additions and 219 deletions
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/*
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* Sierra VMD audio decoder |
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* |
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* This file is part of Libav. |
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* |
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* Libav 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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* Libav 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 Libav; 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/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) { |
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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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/* 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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av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); |
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return ret; |
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} |
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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 = FFMIN(avctx->block_align * silent_chunks, |
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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 & ~(avctx->channels > 1)); |
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while (buf + s->chunk_size <= buf_end) { |
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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 = CODEC_CAP_DR1, |
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}; |
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