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https://github.com/FFmpeg/FFmpeg.git
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66f52d0c3d
It internally decodes as float and then converts to s16 by a call to float_to_int16(). The caller can do this just as well by using lavr.
207 lines
6.6 KiB
C
207 lines
6.6 KiB
C
/*
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* NellyMoser audio decoder
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* Copyright (c) 2007 a840bda5870ba11f19698ff6eb9581dfb0f95fa5,
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* 539459aeb7d425140b62a3ec7dbf6dc8e408a306, and
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* 520e17cd55896441042b14df2566a6eb610ed444
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* Copyright (c) 2007 Loic Minier <lool at dooz.org>
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* Benjamin Larsson
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*
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* Permission is hereby granted, free of charge, to any person obtaining a
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* copy of this software and associated documentation files (the "Software"),
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* to deal in the Software without restriction, including without limitation
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* the rights to use, copy, modify, merge, publish, distribute, sublicense,
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* and/or sell copies of the Software, and to permit persons to whom the
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* Software is furnished to do so, subject to the following conditions:
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*
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* The above copyright notice and this permission notice shall be included in
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* all copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
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* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
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* FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
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* DEALINGS IN THE SOFTWARE.
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*/
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/**
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* @file
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* The 3 alphanumeric copyright notices are md5summed they are from the original
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* implementors. The original code is available from http://code.google.com/p/nelly2pcm/
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*/
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#include "nellymoser.h"
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#include "libavutil/lfg.h"
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#include "libavutil/random_seed.h"
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#include "libavutil/audioconvert.h"
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#include "avcodec.h"
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#include "dsputil.h"
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#include "fft.h"
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#include "fmtconvert.h"
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#include "sinewin.h"
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#define BITSTREAM_READER_LE
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#include "get_bits.h"
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typedef struct NellyMoserDecodeContext {
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AVCodecContext* avctx;
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AVFrame frame;
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AVLFG random_state;
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GetBitContext gb;
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float scale_bias;
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DSPContext dsp;
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FFTContext imdct_ctx;
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DECLARE_ALIGNED(32, float, imdct_buf)[2][NELLY_BUF_LEN];
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float *imdct_out;
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float *imdct_prev;
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} NellyMoserDecodeContext;
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static void nelly_decode_block(NellyMoserDecodeContext *s,
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const unsigned char block[NELLY_BLOCK_LEN],
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float audio[NELLY_SAMPLES])
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{
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int i,j;
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float buf[NELLY_FILL_LEN], pows[NELLY_FILL_LEN];
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float *aptr, *bptr, *pptr, val, pval;
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int bits[NELLY_BUF_LEN];
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unsigned char v;
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init_get_bits(&s->gb, block, NELLY_BLOCK_LEN * 8);
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bptr = buf;
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pptr = pows;
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val = ff_nelly_init_table[get_bits(&s->gb, 6)];
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for (i=0 ; i<NELLY_BANDS ; i++) {
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if (i > 0)
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val += ff_nelly_delta_table[get_bits(&s->gb, 5)];
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pval = -pow(2, val/2048) * s->scale_bias;
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for (j = 0; j < ff_nelly_band_sizes_table[i]; j++) {
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*bptr++ = val;
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*pptr++ = pval;
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}
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}
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ff_nelly_get_sample_bits(buf, bits);
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for (i = 0; i < 2; i++) {
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aptr = audio + i * NELLY_BUF_LEN;
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init_get_bits(&s->gb, block, NELLY_BLOCK_LEN * 8);
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skip_bits_long(&s->gb, NELLY_HEADER_BITS + i*NELLY_DETAIL_BITS);
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for (j = 0; j < NELLY_FILL_LEN; j++) {
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if (bits[j] <= 0) {
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aptr[j] = M_SQRT1_2*pows[j];
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if (av_lfg_get(&s->random_state) & 1)
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aptr[j] *= -1.0;
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} else {
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v = get_bits(&s->gb, bits[j]);
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aptr[j] = ff_nelly_dequantization_table[(1<<bits[j])-1+v]*pows[j];
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}
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}
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memset(&aptr[NELLY_FILL_LEN], 0,
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(NELLY_BUF_LEN - NELLY_FILL_LEN) * sizeof(float));
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s->imdct_ctx.imdct_half(&s->imdct_ctx, s->imdct_out, aptr);
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s->dsp.vector_fmul_window(aptr, s->imdct_prev + NELLY_BUF_LEN/2, s->imdct_out, ff_sine_128, NELLY_BUF_LEN/2);
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FFSWAP(float *, s->imdct_out, s->imdct_prev);
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}
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}
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static av_cold int decode_init(AVCodecContext * avctx) {
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NellyMoserDecodeContext *s = avctx->priv_data;
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s->avctx = avctx;
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s->imdct_out = s->imdct_buf[0];
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s->imdct_prev = s->imdct_buf[1];
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av_lfg_init(&s->random_state, 0);
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ff_mdct_init(&s->imdct_ctx, 8, 1, 1.0);
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ff_dsputil_init(&s->dsp, avctx);
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s->scale_bias = 1.0/(32768*8);
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avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
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/* Generate overlap window */
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if (!ff_sine_128[127])
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ff_init_ff_sine_windows(7);
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avctx->channel_layout = AV_CH_LAYOUT_MONO;
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avcodec_get_frame_defaults(&s->frame);
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avctx->coded_frame = &s->frame;
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return 0;
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}
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static int decode_tag(AVCodecContext *avctx, void *data,
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int *got_frame_ptr, AVPacket *avpkt)
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{
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const uint8_t *buf = avpkt->data;
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int buf_size = avpkt->size;
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NellyMoserDecodeContext *s = avctx->priv_data;
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int blocks, i, ret;
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float *samples_flt;
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blocks = buf_size / NELLY_BLOCK_LEN;
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if (blocks <= 0) {
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av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
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return AVERROR_INVALIDDATA;
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}
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if (buf_size % NELLY_BLOCK_LEN) {
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av_log(avctx, AV_LOG_WARNING, "Leftover bytes: %d.\n",
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buf_size % NELLY_BLOCK_LEN);
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}
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/* Normal numbers of blocks for sample rates:
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* 8000 Hz - 1
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* 11025 Hz - 2
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* 16000 Hz - 3
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* 22050 Hz - 4
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* 44100 Hz - 8
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*/
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/* get output buffer */
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s->frame.nb_samples = NELLY_SAMPLES * blocks;
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if ((ret = avctx->get_buffer(avctx, &s->frame)) < 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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samples_flt = (float *)s->frame.data[0];
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for (i=0 ; i<blocks ; i++) {
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nelly_decode_block(s, buf, samples_flt);
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samples_flt += NELLY_SAMPLES;
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buf += NELLY_BLOCK_LEN;
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}
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*got_frame_ptr = 1;
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*(AVFrame *)data = s->frame;
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return buf_size;
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}
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static av_cold int decode_end(AVCodecContext * avctx) {
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NellyMoserDecodeContext *s = avctx->priv_data;
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ff_mdct_end(&s->imdct_ctx);
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return 0;
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}
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AVCodec ff_nellymoser_decoder = {
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.name = "nellymoser",
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.type = AVMEDIA_TYPE_AUDIO,
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.id = AV_CODEC_ID_NELLYMOSER,
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.priv_data_size = sizeof(NellyMoserDecodeContext),
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.init = decode_init,
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.close = decode_end,
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.decode = decode_tag,
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.capabilities = CODEC_CAP_DR1 | CODEC_CAP_PARAM_CHANGE,
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.long_name = NULL_IF_CONFIG_SMALL("Nellymoser Asao"),
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.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT,
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AV_SAMPLE_FMT_NONE },
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};
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