libavcodec/libmp3lame.c - manifest_repos/ffmpeg - Git at Google

 /*
  * Interface to libmp3lame for mp3 encoding
  * Copyright (c) 2002 Lennert Buytenhek <buytenh@gnu.org>
  *
  * This file is part of FFmpeg.
  *
  * FFmpeg is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Lesser General Public
  * License as published by the Free Software Foundation; either
  * version 2.1 of the License, or (at your option) any later version.
  *
  * FFmpeg is distributed in the hope that it will be useful,
  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  * Lesser General Public License for more details.
  *
  * You should have received a copy of the GNU Lesser General Public
  * License along with FFmpeg; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */

 /**
  * @file
  * Interface to libmp3lame for mp3 encoding.
  */

 #include <lame/lame.h>

 #include "libavutil/channel_layout.h"
 #include "libavutil/common.h"
 #include "libavutil/float_dsp.h"
 #include "libavutil/intreadwrite.h"
 #include "libavutil/log.h"
 #include "libavutil/opt.h"
 #include "avcodec.h"
 #include "audio_frame_queue.h"
 #include "internal.h"
 #include "mpegaudio.h"
 #include "mpegaudiodecheader.h"

 #define BUFFER_SIZE (7200 + 2 * MPA_FRAME_SIZE + MPA_FRAME_SIZE / 4+1000) // FIXME: Buffer size to small? Adding 1000 to make up for it.

 typedef struct LAMEContext {
     AVClass *class;
     AVCodecContext *avctx;
     lame_global_flags *gfp;
     uint8_t *buffer;
     int buffer_index;
     int buffer_size;
     int reservoir;
     int joint_stereo;
     int abr;
     int delay_sent;
     float *samples_flt[2];
     AudioFrameQueue afq;
     AVFloatDSPContext *fdsp;
 } LAMEContext;


 static int realloc_buffer(LAMEContext *s)
 {
     if (!s->buffer || s->buffer_size - s->buffer_index < BUFFER_SIZE) {
         int new_size = s->buffer_index + 2 * BUFFER_SIZE, err;

         ff_dlog(s->avctx, "resizing output buffer: %d -> %d\n", s->buffer_size,
                 new_size);
         if ((err = av_reallocp(&s->buffer, new_size)) < 0) {
             s->buffer_size = s->buffer_index = 0;
             return err;
         }
         s->buffer_size = new_size;
     }
     return 0;
 }

 static av_cold int mp3lame_encode_close(AVCodecContext *avctx)
 {
     LAMEContext *s = avctx->priv_data;

     av_freep(&s->samples_flt[0]);
     av_freep(&s->samples_flt[1]);
     av_freep(&s->buffer);
     av_freep(&s->fdsp);

     ff_af_queue_close(&s->afq);

     lame_close(s->gfp);
     return 0;
 }

 static av_cold int mp3lame_encode_init(AVCodecContext *avctx)
 {
     LAMEContext *s = avctx->priv_data;
     int ret;

     s->avctx = avctx;

     /* initialize LAME and get defaults */
     if (!(s->gfp = lame_init()))
         return AVERROR(ENOMEM);


     lame_set_num_channels(s->gfp, avctx->channels);
     lame_set_mode(s->gfp, avctx->channels > 1 ? s->joint_stereo ? JOINT_STEREO : STEREO : MONO);

     /* sample rate */
     lame_set_in_samplerate (s->gfp, avctx->sample_rate);
     lame_set_out_samplerate(s->gfp, avctx->sample_rate);

     /* algorithmic quality */
     if (avctx->compression_level != FF_COMPRESSION_DEFAULT)
         lame_set_quality(s->gfp, avctx->compression_level);

     /* rate control */
     if (avctx->flags & AV_CODEC_FLAG_QSCALE) { // VBR
         lame_set_VBR(s->gfp, vbr_default);
         lame_set_VBR_quality(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
     } else {
         if (avctx->bit_rate) {
             if (s->abr) {                   // ABR
                 lame_set_VBR(s->gfp, vbr_abr);
                 lame_set_VBR_mean_bitrate_kbps(s->gfp, avctx->bit_rate / 1000);
             } else                          // CBR
                 lame_set_brate(s->gfp, avctx->bit_rate / 1000);
         }
     }

     /* lowpass cutoff frequency */
     if (avctx->cutoff)
         lame_set_lowpassfreq(s->gfp, avctx->cutoff);

     /* do not get a Xing VBR header frame from LAME */
     lame_set_bWriteVbrTag(s->gfp,0);

     /* bit reservoir usage */
     lame_set_disable_reservoir(s->gfp, !s->reservoir);

     /* set specified parameters */
     if (lame_init_params(s->gfp) < 0) {
         ret = -1;
         goto error;
     }

     /* get encoder delay */
     avctx->initial_padding = lame_get_encoder_delay(s->gfp) + 528 + 1;
     ff_af_queue_init(avctx, &s->afq);

     avctx->frame_size  = lame_get_framesize(s->gfp);

     /* allocate float sample buffers */
     if (avctx->sample_fmt == AV_SAMPLE_FMT_FLTP) {
         int ch;
         for (ch = 0; ch < avctx->channels; ch++) {
             s->samples_flt[ch] = av_malloc_array(avctx->frame_size,
                                            sizeof(*s->samples_flt[ch]));
             if (!s->samples_flt[ch]) {
                 ret = AVERROR(ENOMEM);
                 goto error;
             }
         }
     }

     ret = realloc_buffer(s);
     if (ret < 0)
         goto error;

     s->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
     if (!s->fdsp) {
         ret = AVERROR(ENOMEM);
         goto error;
     }


     return 0;
 error:
     mp3lame_encode_close(avctx);
     return ret;
 }

 #define ENCODE_BUFFER(func, buf_type, buf_name) do {                        \
     lame_result = func(s->gfp,                                              \
                        (const buf_type *)buf_name[0],                       \
                        (const buf_type *)buf_name[1], frame->nb_samples,    \
                        s->buffer + s->buffer_index,                         \
                        s->buffer_size - s->buffer_index);                   \
 } while (0)

 static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
                                 const AVFrame *frame, int *got_packet_ptr)
 {
     LAMEContext *s = avctx->priv_data;
     MPADecodeHeader hdr;
     int len, ret, ch, discard_padding;
     int lame_result;
     uint32_t h;

     if (frame) {
         switch (avctx->sample_fmt) {
         case AV_SAMPLE_FMT_S16P:
             ENCODE_BUFFER(lame_encode_buffer, int16_t, frame->data);
             break;
         case AV_SAMPLE_FMT_S32P:
             ENCODE_BUFFER(lame_encode_buffer_int, int32_t, frame->data);
             break;
         case AV_SAMPLE_FMT_FLTP:
             if (frame->linesize[0] < 4 * FFALIGN(frame->nb_samples, 8)) {
                 av_log(avctx, AV_LOG_ERROR, "inadequate AVFrame plane padding\n");
                 return AVERROR(EINVAL);
             }
             for (ch = 0; ch < avctx->channels; ch++) {
                 s->fdsp->vector_fmul_scalar(s->samples_flt[ch],
                                            (const float *)frame->data[ch],
                                            32768.0f,
                                            FFALIGN(frame->nb_samples, 8));
             }
             ENCODE_BUFFER(lame_encode_buffer_float, float, s->samples_flt);
             break;
         default:
             return AVERROR_BUG;
         }
     } else if (!s->afq.frame_alloc) {
         lame_result = 0;
     } else {
         lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index,
                                         s->buffer_size - s->buffer_index);
     }
     if (lame_result < 0) {
         if (lame_result == -1) {
             av_log(avctx, AV_LOG_ERROR,
                    "lame: output buffer too small (buffer index: %d, free bytes: %d)\n",
                    s->buffer_index, s->buffer_size - s->buffer_index);
         }
         return -1;
     }
     s->buffer_index += lame_result;
     ret = realloc_buffer(s);
     if (ret < 0) {
         av_log(avctx, AV_LOG_ERROR, "error reallocating output buffer\n");
         return ret;
     }

     /* add current frame to the queue */
     if (frame) {
         if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
             return ret;
     }

     /* Move 1 frame from the LAME buffer to the output packet, if available.
        We have to parse the first frame header in the output buffer to
        determine the frame size. */
     if (s->buffer_index < 4)
         return 0;
     h = AV_RB32(s->buffer);

     ret = avpriv_mpegaudio_decode_header(&hdr, h);
     if (ret < 0) {
         av_log(avctx, AV_LOG_ERROR, "Invalid mp3 header at start of buffer\n");
         return AVERROR_BUG;
     } else if (ret) {
         av_log(avctx, AV_LOG_ERROR, "free format output not supported\n");
         return -1;
     }
     len = hdr.frame_size;
     ff_dlog(avctx, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len,
             s->buffer_index);
     if (len <= s->buffer_index) {
         if ((ret = ff_alloc_packet2(avctx, avpkt, len, 0)) < 0)
             return ret;
         memcpy(avpkt->data, s->buffer, len);
         s->buffer_index -= len;
         memmove(s->buffer, s->buffer + len, s->buffer_index);

         /* Get the next frame pts/duration */
         ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
                            &avpkt->duration);

         discard_padding = avctx->frame_size - avpkt->duration;
         // Check if subtraction resulted in an overflow
         if ((discard_padding < avctx->frame_size) != (avpkt->duration > 0)) {
             av_log(avctx, AV_LOG_ERROR, "discard padding overflow\n");
             av_packet_unref(avpkt);
             return AVERROR(EINVAL);
         }
         if ((!s->delay_sent && avctx->initial_padding > 0) || discard_padding > 0) {
             uint8_t* side_data = av_packet_new_side_data(avpkt,
                                                          AV_PKT_DATA_SKIP_SAMPLES,
                                                          10);
             if(!side_data) {
                 av_packet_unref(avpkt);
                 return AVERROR(ENOMEM);
             }
             if (!s->delay_sent) {
                 AV_WL32(side_data, avctx->initial_padding);
                 s->delay_sent = 1;
             }
             AV_WL32(side_data + 4, discard_padding);
         }

         avpkt->size = len;
         *got_packet_ptr = 1;
     }
     return 0;
 }

 #define OFFSET(x) offsetof(LAMEContext, x)
 #define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
 static const AVOption options[] = {
     { "reservoir",    "use bit reservoir", OFFSET(reservoir),    AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, AE },
     { "joint_stereo", "use joint stereo",  OFFSET(joint_stereo), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, AE },
     { "abr",          "use ABR",           OFFSET(abr),          AV_OPT_TYPE_BOOL, { .i64 = 0 }, 0, 1, AE },
     { NULL },
 };

 static const AVClass libmp3lame_class = {
     .class_name = "libmp3lame encoder",
     .item_name  = av_default_item_name,
     .option     = options,
     .version    = LIBAVUTIL_VERSION_INT,
 };

 static const AVCodecDefault libmp3lame_defaults[] = {
     { "b",          "0" },
     { NULL },
 };

 static const int libmp3lame_sample_rates[] = {
     44100, 48000,  32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
 };

 AVCodec ff_libmp3lame_encoder = {
     .name                  = "libmp3lame",
     .long_name             = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
     .type                  = AVMEDIA_TYPE_AUDIO,
     .id                    = AV_CODEC_ID_MP3,
     .priv_data_size        = sizeof(LAMEContext),
     .init                  = mp3lame_encode_init,
     .encode2               = mp3lame_encode_frame,
     .close                 = mp3lame_encode_close,
     .capabilities          = AV_CODEC_CAP_DELAY | AV_CODEC_CAP_SMALL_LAST_FRAME,
     .sample_fmts           = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P,
                                                              AV_SAMPLE_FMT_FLTP,
                                                              AV_SAMPLE_FMT_S16P,
                                                              AV_SAMPLE_FMT_NONE },
     .supported_samplerates = libmp3lame_sample_rates,
     .channel_layouts       = (const uint64_t[]) { AV_CH_LAYOUT_MONO,
                                                   AV_CH_LAYOUT_STEREO,
                                                   0 },
     .priv_class            = &libmp3lame_class,
     .defaults              = libmp3lame_defaults,
     .wrapper_name          = "libmp3lame",
 };
	/*
	* Interface to libmp3lame for mp3 encoding
	* Copyright (c) 2002 Lennert Buytenhek <buytenh@gnu.org>
	*
	* This file is part of FFmpeg.
	*
	* FFmpeg is free software; you can redistribute it and/or
	* modify it under the terms of the GNU Lesser General Public
	* License as published by the Free Software Foundation; either
	* version 2.1 of the License, or (at your option) any later version.
	*
	* FFmpeg is distributed in the hope that it will be useful,
	* but WITHOUT ANY WARRANTY; without even the implied warranty of
	* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
	* Lesser General Public License for more details.
	*
	* You should have received a copy of the GNU Lesser General Public
	* License along with FFmpeg; if not, write to the Free Software
	* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
	*/

	/**
	* @file
	* Interface to libmp3lame for mp3 encoding.
	*/

	#include <lame/lame.h>

	#include "libavutil/channel_layout.h"
	#include "libavutil/common.h"
	#include "libavutil/float_dsp.h"
	#include "libavutil/intreadwrite.h"
	#include "libavutil/log.h"
	#include "libavutil/opt.h"
	#include "avcodec.h"
	#include "audio_frame_queue.h"
	#include "internal.h"
	#include "mpegaudio.h"
	#include "mpegaudiodecheader.h"

	#define BUFFER_SIZE (7200 + 2 * MPA_FRAME_SIZE + MPA_FRAME_SIZE / 4+1000) // FIXME: Buffer size to small? Adding 1000 to make up for it.

	typedef struct LAMEContext {
	AVClass *class;
	AVCodecContext *avctx;
	lame_global_flags *gfp;
	uint8_t *buffer;
	int buffer_index;
	int buffer_size;
	int reservoir;
	int joint_stereo;
	int abr;
	int delay_sent;
	float *samples_flt[2];
	AudioFrameQueue afq;
	AVFloatDSPContext *fdsp;
	} LAMEContext;


	static int realloc_buffer(LAMEContext *s)
	{
	if (!s->buffer \|\| s->buffer_size - s->buffer_index < BUFFER_SIZE) {
	int new_size = s->buffer_index + 2 * BUFFER_SIZE, err;

	ff_dlog(s->avctx, "resizing output buffer: %d -> %d\n", s->buffer_size,
	new_size);
	if ((err = av_reallocp(&s->buffer, new_size)) < 0) {
	s->buffer_size = s->buffer_index = 0;
	return err;
	}
	s->buffer_size = new_size;
	}
	return 0;
	}

	static av_cold int mp3lame_encode_close(AVCodecContext *avctx)
	{
	LAMEContext *s = avctx->priv_data;

	av_freep(&s->samples_flt[0]);
	av_freep(&s->samples_flt[1]);
	av_freep(&s->buffer);
	av_freep(&s->fdsp);

	ff_af_queue_close(&s->afq);

	lame_close(s->gfp);
	return 0;
	}

	static av_cold int mp3lame_encode_init(AVCodecContext *avctx)
	{
	LAMEContext *s = avctx->priv_data;
	int ret;

	s->avctx = avctx;

	/* initialize LAME and get defaults */
	if (!(s->gfp = lame_init()))
	return AVERROR(ENOMEM);


	lame_set_num_channels(s->gfp, avctx->channels);
	lame_set_mode(s->gfp, avctx->channels > 1 ? s->joint_stereo ? JOINT_STEREO : STEREO : MONO);

	/* sample rate */
	lame_set_in_samplerate (s->gfp, avctx->sample_rate);
	lame_set_out_samplerate(s->gfp, avctx->sample_rate);

	/* algorithmic quality */
	if (avctx->compression_level != FF_COMPRESSION_DEFAULT)
	lame_set_quality(s->gfp, avctx->compression_level);

	/* rate control */
	if (avctx->flags & AV_CODEC_FLAG_QSCALE) { // VBR
	lame_set_VBR(s->gfp, vbr_default);
	lame_set_VBR_quality(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
	} else {
	if (avctx->bit_rate) {
	if (s->abr) { // ABR
	lame_set_VBR(s->gfp, vbr_abr);
	lame_set_VBR_mean_bitrate_kbps(s->gfp, avctx->bit_rate / 1000);
	} else // CBR
	lame_set_brate(s->gfp, avctx->bit_rate / 1000);
	}
	}

	/* lowpass cutoff frequency */
	if (avctx->cutoff)
	lame_set_lowpassfreq(s->gfp, avctx->cutoff);

	/* do not get a Xing VBR header frame from LAME */
	lame_set_bWriteVbrTag(s->gfp,0);

	/* bit reservoir usage */
	lame_set_disable_reservoir(s->gfp, !s->reservoir);

	/* set specified parameters */
	if (lame_init_params(s->gfp) < 0) {
	ret = -1;
	goto error;
	}

	/* get encoder delay */
	avctx->initial_padding = lame_get_encoder_delay(s->gfp) + 528 + 1;
	ff_af_queue_init(avctx, &s->afq);

	avctx->frame_size = lame_get_framesize(s->gfp);

	/* allocate float sample buffers */
	if (avctx->sample_fmt == AV_SAMPLE_FMT_FLTP) {
	int ch;
	for (ch = 0; ch < avctx->channels; ch++) {
	s->samples_flt[ch] = av_malloc_array(avctx->frame_size,
	sizeof(*s->samples_flt[ch]));
	if (!s->samples_flt[ch]) {
	ret = AVERROR(ENOMEM);
	goto error;
	}
	}
	}

	ret = realloc_buffer(s);
	if (ret < 0)
	goto error;

	s->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
	if (!s->fdsp) {
	ret = AVERROR(ENOMEM);
	goto error;
	}


	return 0;
	error:
	mp3lame_encode_close(avctx);
	return ret;
	}

	#define ENCODE_BUFFER(func, buf_type, buf_name) do { \
	lame_result = func(s->gfp, \
	(const buf_type *)buf_name[0], \
	(const buf_type *)buf_name[1], frame->nb_samples, \
	s->buffer + s->buffer_index, \
	s->buffer_size - s->buffer_index); \
	} while (0)

	static int mp3lame_encode_frame(AVCodecContext avctx, AVPacket avpkt,
	const AVFrame frame, int got_packet_ptr)
	{
	LAMEContext *s = avctx->priv_data;
	MPADecodeHeader hdr;
	int len, ret, ch, discard_padding;
	int lame_result;
	uint32_t h;

	if (frame) {
	switch (avctx->sample_fmt) {
	case AV_SAMPLE_FMT_S16P:
	ENCODE_BUFFER(lame_encode_buffer, int16_t, frame->data);
	break;
	case AV_SAMPLE_FMT_S32P:
	ENCODE_BUFFER(lame_encode_buffer_int, int32_t, frame->data);
	break;
	case AV_SAMPLE_FMT_FLTP:
	if (frame->linesize[0] < 4 * FFALIGN(frame->nb_samples, 8)) {
	av_log(avctx, AV_LOG_ERROR, "inadequate AVFrame plane padding\n");
	return AVERROR(EINVAL);
	}
	for (ch = 0; ch < avctx->channels; ch++) {
	s->fdsp->vector_fmul_scalar(s->samples_flt[ch],
	(const float *)frame->data[ch],
	32768.0f,
	FFALIGN(frame->nb_samples, 8));
	}
	ENCODE_BUFFER(lame_encode_buffer_float, float, s->samples_flt);
	break;
	default:
	return AVERROR_BUG;
	}
	} else if (!s->afq.frame_alloc) {
	lame_result = 0;
	} else {
	lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index,
	s->buffer_size - s->buffer_index);
	}
	if (lame_result < 0) {
	if (lame_result == -1) {
	av_log(avctx, AV_LOG_ERROR,
	"lame: output buffer too small (buffer index: %d, free bytes: %d)\n",
	s->buffer_index, s->buffer_size - s->buffer_index);
	}
	return -1;
	}
	s->buffer_index += lame_result;
	ret = realloc_buffer(s);
	if (ret < 0) {
	av_log(avctx, AV_LOG_ERROR, "error reallocating output buffer\n");
	return ret;
	}

	/* add current frame to the queue */
	if (frame) {
	if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
	return ret;
	}

	/* Move 1 frame from the LAME buffer to the output packet, if available.
	We have to parse the first frame header in the output buffer to
	determine the frame size. */
	if (s->buffer_index < 4)
	return 0;
	h = AV_RB32(s->buffer);

	ret = avpriv_mpegaudio_decode_header(&hdr, h);
	if (ret < 0) {
	av_log(avctx, AV_LOG_ERROR, "Invalid mp3 header at start of buffer\n");
	return AVERROR_BUG;
	} else if (ret) {
	av_log(avctx, AV_LOG_ERROR, "free format output not supported\n");
	return -1;
	}
	len = hdr.frame_size;
	ff_dlog(avctx, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len,
	s->buffer_index);
	if (len <= s->buffer_index) {
	if ((ret = ff_alloc_packet2(avctx, avpkt, len, 0)) < 0)
	return ret;
	memcpy(avpkt->data, s->buffer, len);
	s->buffer_index -= len;
	memmove(s->buffer, s->buffer + len, s->buffer_index);

	/* Get the next frame pts/duration */
	ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
	&avpkt->duration);

	discard_padding = avctx->frame_size - avpkt->duration;
	// Check if subtraction resulted in an overflow
	if ((discard_padding < avctx->frame_size) != (avpkt->duration > 0)) {
	av_log(avctx, AV_LOG_ERROR, "discard padding overflow\n");
	av_packet_unref(avpkt);
	return AVERROR(EINVAL);
	}
	if ((!s->delay_sent && avctx->initial_padding > 0) \|\| discard_padding > 0) {
	uint8_t* side_data = av_packet_new_side_data(avpkt,
	AV_PKT_DATA_SKIP_SAMPLES,
	10);
	if(!side_data) {
	av_packet_unref(avpkt);
	return AVERROR(ENOMEM);
	}
	if (!s->delay_sent) {
	AV_WL32(side_data, avctx->initial_padding);
	s->delay_sent = 1;
	}
	AV_WL32(side_data + 4, discard_padding);
	}

	avpkt->size = len;
	*got_packet_ptr = 1;
	}
	return 0;
	}

	#define OFFSET(x) offsetof(LAMEContext, x)
	#define AE AV_OPT_FLAG_AUDIO_PARAM \| AV_OPT_FLAG_ENCODING_PARAM
	static const AVOption options[] = {
	{ "reservoir", "use bit reservoir", OFFSET(reservoir), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, AE },
	{ "joint_stereo", "use joint stereo", OFFSET(joint_stereo), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, AE },
	{ "abr", "use ABR", OFFSET(abr), AV_OPT_TYPE_BOOL, { .i64 = 0 }, 0, 1, AE },
	{ NULL },
	};

	static const AVClass libmp3lame_class = {
	.class_name = "libmp3lame encoder",
	.item_name = av_default_item_name,
	.option = options,
	.version = LIBAVUTIL_VERSION_INT,
	};

	static const AVCodecDefault libmp3lame_defaults[] = {
	{ "b", "0" },
	{ NULL },
	};

	static const int libmp3lame_sample_rates[] = {
	44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
	};

	AVCodec ff_libmp3lame_encoder = {
	.name = "libmp3lame",
	.long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
	.type = AVMEDIA_TYPE_AUDIO,
	.id = AV_CODEC_ID_MP3,
	.priv_data_size = sizeof(LAMEContext),
	.init = mp3lame_encode_init,
	.encode2 = mp3lame_encode_frame,
	.close = mp3lame_encode_close,
	.capabilities = AV_CODEC_CAP_DELAY \| AV_CODEC_CAP_SMALL_LAST_FRAME,
	.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P,
	AV_SAMPLE_FMT_FLTP,
	AV_SAMPLE_FMT_S16P,
	AV_SAMPLE_FMT_NONE },
	.supported_samplerates = libmp3lame_sample_rates,
	.channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_MONO,
	AV_CH_LAYOUT_STEREO,
	0 },
	.priv_class = &libmp3lame_class,
	.defaults = libmp3lame_defaults,
	.wrapper_name = "libmp3lame",
	};