/* * NellyMoser audio decoder * Copyright (c) 2007 a840bda5870ba11f19698ff6eb9581dfb0f95fa5, * 539459aeb7d425140b62a3ec7dbf6dc8e408a306, and * 520e17cd55896441042b14df2566a6eb610ed444 * Copyright (c) 2007 Loic Minier <lool at dooz.org> * Benjamin Larsson * * Permission is hereby granted, free of charge, to any person obtaining a * copy of this software and associated documentation files (the "Software"), * to deal in the Software without restriction, including without limitation * the rights to use, copy, modify, merge, publish, distribute, sublicense, * and/or sell copies of the Software, and to permit persons to whom the * Software is furnished to do so, subject to the following conditions: * * The above copyright notice and this permission notice shall be included in * all copies or substantial portions of the Software. * * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER * DEALINGS IN THE SOFTWARE. */ /** * @file * The 3 alphanumeric copyright notices are md5summed they are from the original * implementors. The original code is available from http://code.google.com/p/nelly2pcm/ */ #include "nellymoser.h" #include "libavutil/lfg.h" #include "libavutil/random_seed.h" #include "libavutil/audioconvert.h" #include "avcodec.h" #include "dsputil.h" #include "fft.h" #include "fmtconvert.h" #include "sinewin.h" #define ALT_BITSTREAM_READER_LE #include "get_bits.h" typedef struct NellyMoserDecodeContext { AVCodecContext* avctx; float *float_buf; DECLARE_ALIGNED(16, float, state)[NELLY_BUF_LEN]; AVLFG random_state; GetBitContext gb; float scale_bias; DSPContext dsp; FFTContext imdct_ctx; FmtConvertContext fmt_conv; DECLARE_ALIGNED(32, float, imdct_out)[NELLY_BUF_LEN * 2]; } NellyMoserDecodeContext; static void nelly_decode_block(NellyMoserDecodeContext *s, const unsigned char block[NELLY_BLOCK_LEN], float audio[NELLY_SAMPLES]) { int i,j; float buf[NELLY_FILL_LEN], pows[NELLY_FILL_LEN]; float *aptr, *bptr, *pptr, val, pval; int bits[NELLY_BUF_LEN]; unsigned char v; init_get_bits(&s->gb, block, NELLY_BLOCK_LEN * 8); bptr = buf; pptr = pows; val = ff_nelly_init_table[get_bits(&s->gb, 6)]; for (i=0 ; i<NELLY_BANDS ; i++) { if (i > 0) val += ff_nelly_delta_table[get_bits(&s->gb, 5)]; pval = -pow(2, val/2048) * s->scale_bias; for (j = 0; j < ff_nelly_band_sizes_table[i]; j++) { *bptr++ = val; *pptr++ = pval; } } ff_nelly_get_sample_bits(buf, bits); for (i = 0; i < 2; i++) { aptr = audio + i * NELLY_BUF_LEN; init_get_bits(&s->gb, block, NELLY_BLOCK_LEN * 8); skip_bits_long(&s->gb, NELLY_HEADER_BITS + i*NELLY_DETAIL_BITS); for (j = 0; j < NELLY_FILL_LEN; j++) { if (bits[j] <= 0) { aptr[j] = M_SQRT1_2*pows[j]; if (av_lfg_get(&s->random_state) & 1) aptr[j] *= -1.0; } else { v = get_bits(&s->gb, bits[j]); aptr[j] = ff_nelly_dequantization_table[(1<<bits[j])-1+v]*pows[j]; } } memset(&aptr[NELLY_FILL_LEN], 0, (NELLY_BUF_LEN - NELLY_FILL_LEN) * sizeof(float)); s->imdct_ctx.imdct_calc(&s->imdct_ctx, s->imdct_out, aptr); /* XXX: overlapping and windowing should be part of a more generic imdct function */ s->dsp.vector_fmul_reverse(s->state, s->state, ff_sine_128, NELLY_BUF_LEN); s->dsp.vector_fmul_add(aptr, s->imdct_out, ff_sine_128, s->state, NELLY_BUF_LEN); memcpy(s->state, s->imdct_out + NELLY_BUF_LEN, sizeof(float)*NELLY_BUF_LEN); } } static av_cold int decode_init(AVCodecContext * avctx) { NellyMoserDecodeContext *s = avctx->priv_data; s->avctx = avctx; av_lfg_init(&s->random_state, 0); ff_mdct_init(&s->imdct_ctx, 8, 1, 1.0); dsputil_init(&s->dsp, avctx); if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) { s->scale_bias = 1.0/(32768*8); avctx->sample_fmt = AV_SAMPLE_FMT_FLT; } else { s->scale_bias = 1.0/(1*8); avctx->sample_fmt = AV_SAMPLE_FMT_S16; ff_fmt_convert_init(&s->fmt_conv, avctx); s->float_buf = av_mallocz(NELLY_SAMPLES * sizeof(*s->float_buf)); if (!s->float_buf) { av_log(avctx, AV_LOG_ERROR, "error allocating float buffer\n"); return AVERROR(ENOMEM); } } /* Generate overlap window */ if (!ff_sine_128[127]) ff_init_ff_sine_windows(7); avctx->channel_layout = AV_CH_LAYOUT_MONO; return 0; } static int decode_tag(AVCodecContext * avctx, void *data, int *data_size, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; NellyMoserDecodeContext *s = avctx->priv_data; int blocks, i, block_size; int16_t *samples_s16 = data; float *samples_flt = data; block_size = NELLY_SAMPLES * av_get_bytes_per_sample(avctx->sample_fmt); blocks = buf_size / NELLY_BLOCK_LEN; if (blocks <= 0) { av_log(avctx, AV_LOG_ERROR, "Packet is too small\n"); return AVERROR_INVALIDDATA; } if (*data_size < blocks * block_size) { av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n"); return AVERROR(EINVAL); } if (buf_size % NELLY_BLOCK_LEN) { av_log(avctx, AV_LOG_WARNING, "Leftover bytes: %d.\n", buf_size % NELLY_BLOCK_LEN); } /* Normal numbers of blocks for sample rates: * 8000 Hz - 1 * 11025 Hz - 2 * 16000 Hz - 3 * 22050 Hz - 4 * 44100 Hz - 8 */ for (i=0 ; i<blocks ; i++) { if (avctx->sample_fmt == SAMPLE_FMT_FLT) { nelly_decode_block(s, buf, samples_flt); samples_flt += NELLY_SAMPLES; } else { nelly_decode_block(s, buf, s->float_buf); s->fmt_conv.float_to_int16(samples_s16, s->float_buf, NELLY_SAMPLES); samples_s16 += NELLY_SAMPLES; } buf += NELLY_BLOCK_LEN; } *data_size = blocks * block_size; return buf_size; } static av_cold int decode_end(AVCodecContext * avctx) { NellyMoserDecodeContext *s = avctx->priv_data; av_freep(&s->float_buf); ff_mdct_end(&s->imdct_ctx); return 0; } AVCodec ff_nellymoser_decoder = { .name = "nellymoser", .type = AVMEDIA_TYPE_AUDIO, .id = CODEC_ID_NELLYMOSER, .priv_data_size = sizeof(NellyMoserDecodeContext), .init = decode_init, .close = decode_end, .decode = decode_tag, .long_name = NULL_IF_CONFIG_SMALL("Nellymoser Asao"), .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE }, };