* qatar/master: (34 commits) mlp_parser: fix the channel mask value used for the top surround channel vorbisenc: check all allocations for failure roqaudioenc: return AVERROR codes instead of -1 roqaudioenc: set correct bit rate roqaudioenc: use AVCodecContext.frame_size correctly. roqaudioenc: remove unneeded sample_fmt check ra144enc: use int16_t* for input samples rather than void* ra144enc: set AVCodecContext.coded_frame ra144enc: remove unneeded sample_fmt check nellymoserenc: set AVCodecContext.coded_frame nellymoserenc: improve error checking in encode_init() nellymoserenc: return AVERROR codes instead of -1 libvorbis: improve error checking in oggvorbis_encode_init() mpegaudioenc: return AVERROR codes instead of -1 libfaac: improve error checking and handling in Faac_encode_init() avutil: add AVERROR_UNKNOWN check for coded_frame allocation failure in several audio encoders audio encoders: do not set coded_frame->key_frame. g722enc: check for trellis data allocation error libspeexenc: export encoder delay through AVCodecContext.delay ... Conflicts: doc/APIchanges libavcodec/avcodec.h libavcodec/fraps.c libavcodec/kgv1dec.c libavcodec/libfaac.c libavcodec/libgsm.c libavcodec/libvorbis.c libavcodec/mlp_parser.c libavcodec/roqaudioenc.c libavcodec/vorbisenc.c libavutil/avutil.h libavutil/error.c libavutil/error.h Merged-by: Michael Niedermayer <michaelni@gmx.at>
		
			
				
	
	
		
			323 lines
		
	
	
		
			12 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
			
		
		
	
	
			323 lines
		
	
	
		
			12 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
/*
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 * Copyright (C) 2009 Justin Ruggles
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 * Copyright (c) 2009 Xuggle Incorporated
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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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 * @file
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 * libspeex Speex audio encoder
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 *
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 * Usage Guide
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 * This explains the values that need to be set prior to initialization in
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 * order to control various encoding parameters.
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 *
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 * Channels
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 *     Speex only supports mono or stereo, so avctx->channels must be set to
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 *     1 or 2.
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 *
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 * Sample Rate / Encoding Mode
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 *     Speex has 3 modes, each of which uses a specific sample rate.
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 *         narrowband     :  8 kHz
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 *         wideband       : 16 kHz
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 *         ultra-wideband : 32 kHz
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 *     avctx->sample_rate must be set to one of these 3 values.  This will be
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 *     used to set the encoding mode.
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 *
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 * Rate Control
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 *     VBR mode is turned on by setting CODEC_FLAG_QSCALE in avctx->flags.
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 *     avctx->global_quality is used to set the encoding quality.
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 *     For CBR mode, avctx->bit_rate can be used to set the constant bitrate.
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 *     Alternatively, the 'cbr_quality' option can be set from 0 to 10 to set
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 *     a constant bitrate based on quality.
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 *     For ABR mode, set avctx->bit_rate and set the 'abr' option to 1.
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 *     Approx. Bitrate Range:
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 *         narrowband     : 2400 - 25600 bps
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 *         wideband       : 4000 - 43200 bps
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 *         ultra-wideband : 4400 - 45200 bps
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 *
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 * Complexity
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 *     Encoding complexity is controlled by setting avctx->compression_level.
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 *     The valid range is 0 to 10.  A higher setting gives generally better
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 *     quality at the expense of encoding speed.  This does not affect the
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 *     bit rate.
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 *
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 * Frames-per-Packet
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 *     The encoder defaults to using 1 frame-per-packet.  However, it is
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 *     sometimes desirable to use multiple frames-per-packet to reduce the
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 *     amount of container overhead.  This can be done by setting the
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 *     'frames_per_packet' option to a value 1 to 8.
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 */
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#include <speex/speex.h>
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#include <speex/speex_header.h>
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#include <speex/speex_stereo.h>
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#include "libavutil/opt.h"
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#include "avcodec.h"
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#include "internal.h"
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typedef struct {
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    AVClass *class;             ///< AVClass for private options
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    SpeexBits bits;             ///< libspeex bitwriter context
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    SpeexHeader header;         ///< libspeex header struct
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    void *enc_state;            ///< libspeex encoder state
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    int frames_per_packet;      ///< number of frames to encode in each packet
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    float vbr_quality;          ///< VBR quality 0.0 to 10.0
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    int cbr_quality;            ///< CBR quality 0 to 10
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    int abr;                    ///< flag to enable ABR
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    int pkt_frame_count;        ///< frame count for the current packet
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    int64_t next_pts;           ///< next pts, in sample_rate time base
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    int pkt_sample_count;       ///< sample count in the current packet
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} LibSpeexEncContext;
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static av_cold void print_enc_params(AVCodecContext *avctx,
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                                     LibSpeexEncContext *s)
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{
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    const char *mode_str = "unknown";
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    av_log(avctx, AV_LOG_DEBUG, "channels: %d\n", avctx->channels);
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    switch (s->header.mode) {
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    case SPEEX_MODEID_NB:  mode_str = "narrowband";     break;
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    case SPEEX_MODEID_WB:  mode_str = "wideband";       break;
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    case SPEEX_MODEID_UWB: mode_str = "ultra-wideband"; break;
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    }
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    av_log(avctx, AV_LOG_DEBUG, "mode: %s\n", mode_str);
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    if (s->header.vbr) {
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        av_log(avctx, AV_LOG_DEBUG, "rate control: VBR\n");
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        av_log(avctx, AV_LOG_DEBUG, "  quality: %f\n", s->vbr_quality);
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    } else if (s->abr) {
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        av_log(avctx, AV_LOG_DEBUG, "rate control: ABR\n");
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        av_log(avctx, AV_LOG_DEBUG, "  bitrate: %d bps\n", avctx->bit_rate);
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    } else {
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        av_log(avctx, AV_LOG_DEBUG, "rate control: CBR\n");
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        av_log(avctx, AV_LOG_DEBUG, "  bitrate: %d bps\n", avctx->bit_rate);
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    }
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    av_log(avctx, AV_LOG_DEBUG, "complexity: %d\n",
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           avctx->compression_level);
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    av_log(avctx, AV_LOG_DEBUG, "frame size: %d samples\n",
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           avctx->frame_size);
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    av_log(avctx, AV_LOG_DEBUG, "frames per packet: %d\n",
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           s->frames_per_packet);
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    av_log(avctx, AV_LOG_DEBUG, "packet size: %d\n",
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           avctx->frame_size * s->frames_per_packet);
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}
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static av_cold int encode_init(AVCodecContext *avctx)
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{
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    LibSpeexEncContext *s = avctx->priv_data;
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    const SpeexMode *mode;
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    uint8_t *header_data;
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    int header_size;
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    int32_t complexity;
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    /* channels */
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    if (avctx->channels < 1 || avctx->channels > 2) {
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        av_log(avctx, AV_LOG_ERROR, "Invalid channels (%d). Only stereo and "
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               "mono are supported\n", avctx->channels);
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        return AVERROR(EINVAL);
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    }
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    /* sample rate and encoding mode */
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    switch (avctx->sample_rate) {
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    case  8000: mode = &speex_nb_mode;  break;
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    case 16000: mode = &speex_wb_mode;  break;
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    case 32000: mode = &speex_uwb_mode; break;
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    default:
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        av_log(avctx, AV_LOG_ERROR, "Sample rate of %d Hz is not supported. "
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               "Resample to 8, 16, or 32 kHz.\n", avctx->sample_rate);
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        return AVERROR(EINVAL);
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    }
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    /* initialize libspeex */
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    s->enc_state = speex_encoder_init(mode);
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    if (!s->enc_state) {
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        av_log(avctx, AV_LOG_ERROR, "Error initializing libspeex\n");
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        return -1;
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    }
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    speex_init_header(&s->header, avctx->sample_rate, avctx->channels, mode);
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    /* rate control method and parameters */
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    if (avctx->flags & CODEC_FLAG_QSCALE) {
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        /* VBR */
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        s->header.vbr = 1;
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        speex_encoder_ctl(s->enc_state, SPEEX_SET_VBR, &s->header.vbr);
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        s->vbr_quality = av_clipf(avctx->global_quality / (float)FF_QP2LAMBDA,
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                                  0.0f, 10.0f);
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        speex_encoder_ctl(s->enc_state, SPEEX_SET_VBR_QUALITY, &s->vbr_quality);
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    } else {
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        s->header.bitrate = avctx->bit_rate;
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        if (avctx->bit_rate > 0) {
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            /* CBR or ABR by bitrate */
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            if (s->abr) {
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                speex_encoder_ctl(s->enc_state, SPEEX_SET_ABR,
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                                  &s->header.bitrate);
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                speex_encoder_ctl(s->enc_state, SPEEX_GET_ABR,
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                                  &s->header.bitrate);
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            } else {
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                speex_encoder_ctl(s->enc_state, SPEEX_SET_BITRATE,
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                                  &s->header.bitrate);
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                speex_encoder_ctl(s->enc_state, SPEEX_GET_BITRATE,
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                                  &s->header.bitrate);
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            }
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        } else {
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            /* CBR by quality */
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            speex_encoder_ctl(s->enc_state, SPEEX_SET_QUALITY,
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                              &s->cbr_quality);
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            speex_encoder_ctl(s->enc_state, SPEEX_GET_BITRATE,
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                              &s->header.bitrate);
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        }
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        /* stereo side information adds about 800 bps to the base bitrate */
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        /* TODO: this should be calculated exactly */
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        avctx->bit_rate = s->header.bitrate + (avctx->channels == 2 ? 800 : 0);
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    }
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    /* set encoding complexity */
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    if (avctx->compression_level > FF_COMPRESSION_DEFAULT) {
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        complexity = av_clip(avctx->compression_level, 0, 10);
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        speex_encoder_ctl(s->enc_state, SPEEX_SET_COMPLEXITY, &complexity);
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    }
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    speex_encoder_ctl(s->enc_state, SPEEX_GET_COMPLEXITY, &complexity);
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    avctx->compression_level = complexity;
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    /* set packet size */
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    avctx->frame_size = s->header.frame_size;
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    s->header.frames_per_packet = s->frames_per_packet;
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    /* set encoding delay */
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    speex_encoder_ctl(s->enc_state, SPEEX_GET_LOOKAHEAD, &avctx->delay);
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    /* create header packet bytes from header struct */
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    /* note: libspeex allocates the memory for header_data, which is freed
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             below with speex_header_free() */
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    header_data = speex_header_to_packet(&s->header, &header_size);
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    /* allocate extradata and coded_frame */
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    avctx->extradata   = av_malloc(header_size + FF_INPUT_BUFFER_PADDING_SIZE);
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    avctx->coded_frame = avcodec_alloc_frame();
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    if (!avctx->extradata || !avctx->coded_frame) {
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        speex_header_free(header_data);
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        speex_encoder_destroy(s->enc_state);
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        av_log(avctx, AV_LOG_ERROR, "memory allocation error\n");
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        return AVERROR(ENOMEM);
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    }
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    /* copy header packet to extradata */
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    memcpy(avctx->extradata, header_data, header_size);
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    avctx->extradata_size = header_size;
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    speex_header_free(header_data);
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    /* init libspeex bitwriter */
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    speex_bits_init(&s->bits);
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    print_enc_params(avctx, s);
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    return 0;
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}
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static int encode_frame(AVCodecContext *avctx, uint8_t *frame, int buf_size,
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                        void *data)
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{
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    LibSpeexEncContext *s = avctx->priv_data;
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    int16_t *samples      = data;
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    if (data) {
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        /* encode Speex frame */
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        if (avctx->channels == 2)
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            speex_encode_stereo_int(samples, s->header.frame_size, &s->bits);
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        speex_encode_int(s->enc_state, samples, &s->bits);
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        s->pkt_frame_count++;
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        s->pkt_sample_count += avctx->frame_size;
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    } else {
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        /* handle end-of-stream */
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        if (!s->pkt_frame_count)
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            return 0;
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        /* add extra terminator codes for unused frames in last packet */
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        while (s->pkt_frame_count < s->frames_per_packet) {
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            speex_bits_pack(&s->bits, 15, 5);
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            s->pkt_frame_count++;
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        }
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    }
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    /* write output if all frames for the packet have been encoded */
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    if (s->pkt_frame_count == s->frames_per_packet) {
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        s->pkt_frame_count = 0;
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        avctx->coded_frame->pts = ff_samples_to_time_base(avctx, s->next_pts -
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                                                          avctx->delay);
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        s->next_pts += s->pkt_sample_count;
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        s->pkt_sample_count = 0;
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        if (buf_size > speex_bits_nbytes(&s->bits)) {
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            int ret = speex_bits_write(&s->bits, frame, buf_size);
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            speex_bits_reset(&s->bits);
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            return ret;
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        } else {
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            av_log(avctx, AV_LOG_ERROR, "output buffer too small");
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            return AVERROR(EINVAL);
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        }
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    }
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    return 0;
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}
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static av_cold int encode_close(AVCodecContext *avctx)
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{
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    LibSpeexEncContext *s = avctx->priv_data;
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    speex_bits_destroy(&s->bits);
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    speex_encoder_destroy(s->enc_state);
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    av_freep(&avctx->coded_frame);
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    av_freep(&avctx->extradata);
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    return 0;
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}
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#define OFFSET(x) offsetof(LibSpeexEncContext, x)
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#define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
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static const AVOption options[] = {
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    { "abr",               "Use average bit rate",                      OFFSET(abr),               AV_OPT_TYPE_INT, { 0 }, 0,   1, AE },
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    { "cbr_quality",       "Set quality value (0 to 10) for CBR",       OFFSET(cbr_quality),       AV_OPT_TYPE_INT, { 8 }, 0,  10, AE },
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    { "frames_per_packet", "Number of frames to encode in each packet", OFFSET(frames_per_packet), AV_OPT_TYPE_INT, { 1 }, 1,   8, AE },
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    { NULL },
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};
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static const AVClass class = {
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    .class_name = "libspeex",
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    .item_name  = av_default_item_name,
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    .option     = options,
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    .version    = LIBAVUTIL_VERSION_INT,
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};
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static const AVCodecDefault defaults[] = {
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    { "b",                 "0" },
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    { "compression_level", "3" },
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    { NULL },
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};
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AVCodec ff_libspeex_encoder = {
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    .name           = "libspeex",
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    .type           = AVMEDIA_TYPE_AUDIO,
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    .id             = CODEC_ID_SPEEX,
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    .priv_data_size = sizeof(LibSpeexEncContext),
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    .init           = encode_init,
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    .encode         = encode_frame,
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    .close          = encode_close,
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    .capabilities   = CODEC_CAP_DELAY,
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    .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE },
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    .long_name      = NULL_IF_CONFIG_SMALL("libspeex Speex"),
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    .priv_class     = &class,
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    .defaults       = defaults,
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};
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