From 246f869590b8c7313d26e1c2ef56db01f6fd2503 Mon Sep 17 00:00:00 2001 From: Nidhi Makhijani Date: Mon, 7 Jul 2014 09:41:04 +0530 Subject: vmd: Split audio and video decoder Signed-off-by: Diego Biurrun --- libavcodec/vmdaudio.c | 233 ++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 233 insertions(+) create mode 100644 libavcodec/vmdaudio.c (limited to 'libavcodec/vmdaudio.c') diff --git a/libavcodec/vmdaudio.c b/libavcodec/vmdaudio.c new file mode 100644 index 0000000000..66c5865f85 --- /dev/null +++ b/libavcodec/vmdaudio.c @@ -0,0 +1,233 @@ +/* + * Sierra VMD audio decoder + * + * This file is part of Libav. + * + * Libav 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. + * + * Libav 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 Libav; 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/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) { + 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) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + 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 = FFMIN(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 & ~(avctx->channels > 1)); + while (buf + s->chunk_size <= buf_end) { + 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 = CODEC_CAP_DR1, +}; -- cgit v1.2.3