* qatar/master: rtpdec_asf: Set the no_resync_search option for the chained asf demuxer asfdec: Add an option for not searching for the packet markers cosmetics: Clean up the tiffenc pix_fmts declaration to match the style of others cosmetics: Align codec declarations cosmetics: Convert mimic.c to utf-8 avconv: remove an unused function parameter. avconv: remove now pointless variables. avconv: drop support for building without libavfilter. nellymoserenc: fix crash due to memsetting the wrong area. libavformat: Only require first packet to be known for audio/video streams avplay: Don't try to scale timestamps if the tb isn't set Conflicts: Changelog configure ffmpeg.c libavcodec/aacenc.c libavcodec/bmpenc.c libavcodec/dnxhddec.c libavcodec/dnxhdenc.c libavcodec/ffv1.c libavcodec/flacenc.c libavcodec/fraps.c libavcodec/huffyuv.c libavcodec/libopenjpegdec.c libavcodec/mpeg12enc.c libavcodec/mpeg4videodec.c libavcodec/pamenc.c libavcodec/pgssubdec.c libavcodec/pngenc.c libavcodec/qtrleenc.c libavcodec/rawdec.c libavcodec/sgienc.c libavcodec/tiffenc.c libavcodec/v210dec.c libavcodec/wmv2dec.c libavformat/utils.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
		
			
				
	
	
		
			391 lines
		
	
	
		
			12 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
			
		
		
	
	
			391 lines
		
	
	
		
			12 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
| /*
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|  * AMR Audio decoder stub
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|  * Copyright (c) 2003 the ffmpeg project
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|  *
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|  * This file is part of FFmpeg.
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|  *
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|  * FFmpeg 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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|  * FFmpeg 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 FFmpeg; 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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| #include "avcodec.h"
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| #include "libavutil/avstring.h"
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| #include "libavutil/opt.h"
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| #include "audio_frame_queue.h"
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| #include "internal.h"
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| 
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| static void amr_decode_fix_avctx(AVCodecContext *avctx)
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| {
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|     const int is_amr_wb = 1 + (avctx->codec_id == CODEC_ID_AMR_WB);
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| 
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|     if (!avctx->sample_rate)
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|         avctx->sample_rate = 8000 * is_amr_wb;
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| 
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|     if (!avctx->channels)
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|         avctx->channels = 1;
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| 
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|     avctx->sample_fmt = AV_SAMPLE_FMT_S16;
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| }
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| 
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| #if CONFIG_LIBOPENCORE_AMRNB
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| 
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| #include <opencore-amrnb/interf_dec.h>
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| #include <opencore-amrnb/interf_enc.h>
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| 
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| /* Common code for fixed and float version*/
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| typedef struct AMR_bitrates {
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|     int       rate;
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|     enum Mode mode;
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| } AMR_bitrates;
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| 
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| /* Match desired bitrate */
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| static int get_bitrate_mode(int bitrate, void *log_ctx)
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| {
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|     /* make the correspondance between bitrate and mode */
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|     static const AMR_bitrates rates[] = {
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|         { 4750, MR475 }, { 5150, MR515 }, {  5900, MR59  }, {  6700, MR67  },
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|         { 7400, MR74 },  { 7950, MR795 }, { 10200, MR102 }, { 12200, MR122 }
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|     };
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|     int i, best = -1, min_diff = 0;
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|     char log_buf[200];
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| 
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|     for (i = 0; i < 8; i++) {
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|         if (rates[i].rate == bitrate)
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|             return rates[i].mode;
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|         if (best < 0 || abs(rates[i].rate - bitrate) < min_diff) {
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|             best     = i;
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|             min_diff = abs(rates[i].rate - bitrate);
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|         }
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|     }
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|     /* no bitrate matching exactly, log a warning */
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|     snprintf(log_buf, sizeof(log_buf), "bitrate not supported: use one of ");
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|     for (i = 0; i < 8; i++)
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|         av_strlcatf(log_buf, sizeof(log_buf), "%.2fk, ", rates[i].rate    / 1000.f);
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|     av_strlcatf(log_buf, sizeof(log_buf), "using %.2fk", rates[best].rate / 1000.f);
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|     av_log(log_ctx, AV_LOG_WARNING, "%s\n", log_buf);
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| 
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|     return best;
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| }
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| 
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| typedef struct AMRContext {
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|     AVClass *av_class;
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|     AVFrame frame;
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|     void *dec_state;
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|     void *enc_state;
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|     int   enc_bitrate;
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|     int   enc_mode;
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|     int   enc_dtx;
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|     int   enc_last_frame;
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|     AudioFrameQueue afq;
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| } AMRContext;
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| 
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| static const AVOption options[] = {
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|     { "dtx", "Allow DTX (generate comfort noise)", offsetof(AMRContext, enc_dtx), AV_OPT_TYPE_INT, { 0 }, 0, 1, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM },
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|     { NULL }
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| };
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| 
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| static const AVClass class = {
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|     "libopencore_amrnb", av_default_item_name, options, LIBAVUTIL_VERSION_INT
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| };
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| 
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| static av_cold int amr_nb_decode_init(AVCodecContext *avctx)
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| {
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|     AMRContext *s  = avctx->priv_data;
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| 
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|     s->dec_state   = Decoder_Interface_init();
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|     if (!s->dec_state) {
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|         av_log(avctx, AV_LOG_ERROR, "Decoder_Interface_init error\n");
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|         return -1;
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|     }
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| 
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|     amr_decode_fix_avctx(avctx);
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| 
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|     if (avctx->channels > 1) {
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|         av_log(avctx, AV_LOG_ERROR, "amr_nb: multichannel decoding not supported\n");
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|         return AVERROR(ENOSYS);
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|     }
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| 
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|     avcodec_get_frame_defaults(&s->frame);
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|     avctx->coded_frame = &s->frame;
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| 
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|     return 0;
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| }
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| 
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| static av_cold int amr_nb_decode_close(AVCodecContext *avctx)
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| {
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|     AMRContext *s = avctx->priv_data;
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| 
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|     Decoder_Interface_exit(s->dec_state);
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| 
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|     return 0;
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| }
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| 
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| static int amr_nb_decode_frame(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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|     AMRContext *s      = avctx->priv_data;
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|     static const uint8_t block_size[16] = { 12, 13, 15, 17, 19, 20, 26, 31, 5, 0, 0, 0, 0, 0, 0, 0 };
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|     enum Mode dec_mode;
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|     int packet_size, ret;
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| 
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|     av_dlog(avctx, "amr_decode_frame buf=%p buf_size=%d frame_count=%d!!\n",
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|             buf, buf_size, avctx->frame_number);
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| 
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|     /* get output buffer */
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|     s->frame.nb_samples = 160;
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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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| 
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|     dec_mode    = (buf[0] >> 3) & 0x000F;
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|     packet_size = block_size[dec_mode] + 1;
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| 
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|     if (packet_size > buf_size) {
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|         av_log(avctx, AV_LOG_ERROR, "amr frame too short (%u, should be %u)\n",
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|                buf_size, packet_size);
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|         return AVERROR_INVALIDDATA;
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|     }
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| 
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|     av_dlog(avctx, "packet_size=%d buf= 0x%X %X %X %X\n",
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|               packet_size, buf[0], buf[1], buf[2], buf[3]);
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|     /* call decoder */
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|     Decoder_Interface_Decode(s->dec_state, buf, (short *)s->frame.data[0], 0);
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| 
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|     *got_frame_ptr   = 1;
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|     *(AVFrame *)data = s->frame;
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| 
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|     return packet_size;
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| }
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| 
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| AVCodec ff_libopencore_amrnb_decoder = {
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|     .name           = "libopencore_amrnb",
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|     .type           = AVMEDIA_TYPE_AUDIO,
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|     .id             = CODEC_ID_AMR_NB,
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|     .priv_data_size = sizeof(AMRContext),
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|     .init           = amr_nb_decode_init,
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|     .close          = amr_nb_decode_close,
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|     .decode         = amr_nb_decode_frame,
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|     .capabilities   = CODEC_CAP_DR1,
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|     .long_name      = NULL_IF_CONFIG_SMALL("OpenCORE Adaptive Multi-Rate (AMR) Narrow-Band"),
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| };
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| 
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| static av_cold int amr_nb_encode_init(AVCodecContext *avctx)
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| {
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|     AMRContext *s = avctx->priv_data;
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| 
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|     if (avctx->sample_rate != 8000) {
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|         av_log(avctx, AV_LOG_ERROR, "Only 8000Hz sample rate supported\n");
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|         return AVERROR(ENOSYS);
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|     }
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| 
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|     if (avctx->channels != 1) {
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|         av_log(avctx, AV_LOG_ERROR, "Only mono supported\n");
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|         return AVERROR(ENOSYS);
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|     }
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| 
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|     avctx->frame_size  = 160;
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|     avctx->delay       =  50;
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|     ff_af_queue_init(avctx, &s->afq);
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| #if FF_API_OLD_ENCODE_AUDIO
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|     avctx->coded_frame = avcodec_alloc_frame();
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|     if (!avctx->coded_frame)
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|         return AVERROR(ENOMEM);
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| #endif
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| 
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|     s->enc_state = Encoder_Interface_init(s->enc_dtx);
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|     if (!s->enc_state) {
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|         av_log(avctx, AV_LOG_ERROR, "Encoder_Interface_init error\n");
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|         av_freep(&avctx->coded_frame);
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|         return -1;
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|     }
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| 
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|     s->enc_mode    = get_bitrate_mode(avctx->bit_rate, avctx);
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|     s->enc_bitrate = avctx->bit_rate;
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| 
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|     return 0;
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| }
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| 
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| static av_cold int amr_nb_encode_close(AVCodecContext *avctx)
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| {
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|     AMRContext *s = avctx->priv_data;
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| 
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|     Encoder_Interface_exit(s->enc_state);
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|     ff_af_queue_close(&s->afq);
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| #if FF_API_OLD_ENCODE_AUDIO
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|     av_freep(&avctx->coded_frame);
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| #endif
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|     return 0;
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| }
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| 
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| static int amr_nb_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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|                                const AVFrame *frame, int *got_packet_ptr)
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| {
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|     AMRContext *s = avctx->priv_data;
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|     int written, ret;
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|     int16_t *flush_buf = NULL;
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|     const int16_t *samples = frame ? (const int16_t *)frame->data[0] : NULL;
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| 
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|     if (s->enc_bitrate != avctx->bit_rate) {
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|         s->enc_mode    = get_bitrate_mode(avctx->bit_rate, avctx);
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|         s->enc_bitrate = avctx->bit_rate;
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|     }
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| 
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|     if ((ret = ff_alloc_packet2(avctx, avpkt, 32)))
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|         return ret;
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| 
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|     if (frame) {
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|         if (frame->nb_samples < avctx->frame_size) {
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|             flush_buf = av_mallocz(avctx->frame_size * sizeof(*flush_buf));
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|             if (!flush_buf)
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|                 return AVERROR(ENOMEM);
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|             memcpy(flush_buf, samples, frame->nb_samples * sizeof(*flush_buf));
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|             samples = flush_buf;
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|             if (frame->nb_samples < avctx->frame_size - avctx->delay)
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|                 s->enc_last_frame = -1;
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|         }
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|         if ((ret = ff_af_queue_add(&s->afq, frame) < 0)) {
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|             av_freep(&flush_buf);
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|             return ret;
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|         }
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|     } else {
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|         if (s->enc_last_frame < 0)
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|             return 0;
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|         flush_buf = av_mallocz(avctx->frame_size * sizeof(*flush_buf));
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|         if (!flush_buf)
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|             return AVERROR(ENOMEM);
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|         samples = flush_buf;
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|         s->enc_last_frame = -1;
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|     }
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| 
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|     written = Encoder_Interface_Encode(s->enc_state, s->enc_mode, samples,
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|                                        avpkt->data, 0);
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|     av_dlog(avctx, "amr_nb_encode_frame encoded %u bytes, bitrate %u, first byte was %#02x\n",
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|             written, s->enc_mode, frame[0]);
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| 
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|     /* Get the next frame pts/duration */
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|     ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
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|                        &avpkt->duration);
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| 
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|     avpkt->size = written;
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|     *got_packet_ptr = 1;
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|     av_freep(&flush_buf);
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|     return 0;
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| }
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| 
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| AVCodec ff_libopencore_amrnb_encoder = {
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|     .name           = "libopencore_amrnb",
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|     .type           = AVMEDIA_TYPE_AUDIO,
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|     .id             = CODEC_ID_AMR_NB,
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|     .priv_data_size = sizeof(AMRContext),
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|     .init           = amr_nb_encode_init,
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|     .encode2        = amr_nb_encode_frame,
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|     .close          = amr_nb_encode_close,
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|     .capabilities   = CODEC_CAP_DELAY | CODEC_CAP_SMALL_LAST_FRAME,
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|     .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
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|                                                      AV_SAMPLE_FMT_NONE },
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|     .long_name      = NULL_IF_CONFIG_SMALL("OpenCORE Adaptive Multi-Rate (AMR) Narrow-Band"),
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|     .priv_class     = &class,
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| };
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| 
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| #endif
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| 
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| /* -----------AMR wideband ------------*/
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| #if CONFIG_LIBOPENCORE_AMRWB
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| 
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| #include <opencore-amrwb/dec_if.h>
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| #include <opencore-amrwb/if_rom.h>
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| 
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| typedef struct AMRWBContext {
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|     AVFrame frame;
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|     void  *state;
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| } AMRWBContext;
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| 
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| static av_cold int amr_wb_decode_init(AVCodecContext *avctx)
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| {
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|     AMRWBContext *s = avctx->priv_data;
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| 
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|     s->state        = D_IF_init();
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| 
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|     amr_decode_fix_avctx(avctx);
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| 
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|     if (avctx->channels > 1) {
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|         av_log(avctx, AV_LOG_ERROR, "amr_wb: multichannel decoding not supported\n");
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|         return AVERROR(ENOSYS);
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|     }
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| 
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|     avcodec_get_frame_defaults(&s->frame);
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|     avctx->coded_frame = &s->frame;
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| 
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|     return 0;
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| }
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| 
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| static int amr_wb_decode_frame(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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|     AMRWBContext *s    = avctx->priv_data;
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|     int mode, ret;
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|     int packet_size;
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|     static const uint8_t block_size[16] = {18, 24, 33, 37, 41, 47, 51, 59, 61, 6, 6, 0, 0, 0, 1, 1};
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| 
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|     /* get output buffer */
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|     s->frame.nb_samples = 320;
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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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| 
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|     mode        = (buf[0] >> 3) & 0x000F;
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|     packet_size = block_size[mode];
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| 
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|     if (packet_size > buf_size) {
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|         av_log(avctx, AV_LOG_ERROR, "amr frame too short (%u, should be %u)\n",
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|                buf_size, packet_size + 1);
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|         return AVERROR_INVALIDDATA;
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|     }
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| 
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|     D_IF_decode(s->state, buf, (short *)s->frame.data[0], _good_frame);
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| 
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|     *got_frame_ptr   = 1;
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|     *(AVFrame *)data = s->frame;
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| 
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|     return packet_size;
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| }
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| 
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| static int amr_wb_decode_close(AVCodecContext *avctx)
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| {
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|     AMRWBContext *s = avctx->priv_data;
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| 
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|     D_IF_exit(s->state);
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|     return 0;
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| }
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| 
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| AVCodec ff_libopencore_amrwb_decoder = {
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|     .name           = "libopencore_amrwb",
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|     .type           = AVMEDIA_TYPE_AUDIO,
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|     .id             = CODEC_ID_AMR_WB,
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|     .priv_data_size = sizeof(AMRWBContext),
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|     .init           = amr_wb_decode_init,
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|     .close          = amr_wb_decode_close,
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|     .decode         = amr_wb_decode_frame,
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|     .capabilities   = CODEC_CAP_DR1,
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|     .long_name      = NULL_IF_CONFIG_SMALL("OpenCORE Adaptive Multi-Rate (AMR) Wide-Band"),
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| };
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| 
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| #endif /* CONFIG_LIBOPENCORE_AMRWB */
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