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https://github.com/FFmpeg/FFmpeg.git
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55c7bb0665
Commits957a593cd9
and11828b8885
made the flv demuxer export a certain flag as side data to be used by the nellymoser decoder for mid-stream sample rate changes. It used a custom side data type 'F' that was never officially documented. Yet since2215c39e94
(merged in commit52c522c720
) this information is exported via the properly documented AV_PKT_DATA_PARAM_CHANGE side data. The merge commit therefore stopped exporting the 'F' sidedata; yet the changes in the Nellymoser decoder (which are now dead code (and would become dangerous if lots of new side data types were added)) have not been removed. This commit does this. Reviewed-by: Anton Khirnov <anton@khirnov.net> Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
202 lines
6.5 KiB
C
202 lines
6.5 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 "libavutil/channel_layout.h"
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#include "libavutil/float_dsp.h"
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#include "libavutil/lfg.h"
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#include "libavutil/random_seed.h"
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#define BITSTREAM_READER_LE
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#include "avcodec.h"
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#include "fft.h"
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#include "get_bits.h"
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#include "internal.h"
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#include "nellymoser.h"
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#include "sinewin.h"
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typedef struct NellyMoserDecodeContext {
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AVCodecContext* avctx;
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AVLFG random_state;
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GetBitContext gb;
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float scale_bias;
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AVFloatDSPContext *fdsp;
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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 = -exp2(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->fdsp->vector_fmul_window(aptr, s->imdct_prev + NELLY_BUF_LEN / 2,
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s->imdct_out, ff_sine_128,
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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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s->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
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if (!s->fdsp)
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return AVERROR(ENOMEM);
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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->channels = 1;
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avctx->channel_layout = AV_CH_LAYOUT_MONO;
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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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AVFrame *frame = data;
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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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/* get output buffer */
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frame->nb_samples = NELLY_SAMPLES * blocks;
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if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
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return ret;
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samples_flt = (float *)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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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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av_freep(&s->fdsp);
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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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.long_name = NULL_IF_CONFIG_SMALL("Nellymoser Asao"),
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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 = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_PARAM_CHANGE,
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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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