libavcodec/libmp3lame.c - third_party/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/audioconvert.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[BUFFER_SIZE];
     int buffer_index;
     int reservoir;
     void *planar_samples[2];
     AudioFrameQueue afq;
 } LAMEContext;


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

 #if FF_API_OLD_ENCODE_AUDIO
     av_freep(&avctx->coded_frame);
 #endif
     av_freep(&s->planar_samples[0]);
     av_freep(&s->planar_samples[1]);

     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()) == NULL)
         return AVERROR(ENOMEM);


     lame_set_num_channels(s->gfp, avctx->channels);
     lame_set_mode(s->gfp, avctx->channels > 1 ? JOINT_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, 5);
     else
         lame_set_quality(s->gfp, avctx->compression_level);

     /* rate control */
     if (avctx->flags & CODEC_FLAG_QSCALE) {
         lame_set_VBR(s->gfp, vbr_default);
         lame_set_VBR_quality(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
     } else {
         if (avctx->bit_rate)
             lame_set_brate(s->gfp, avctx->bit_rate / 1000);
     }

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

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

 #if FF_API_OLD_ENCODE_AUDIO
     avctx->coded_frame = avcodec_alloc_frame();
     if (!avctx->coded_frame) {
         ret = AVERROR(ENOMEM);
         goto error;
     }
 #endif

     /* sample format */
     if (avctx->sample_fmt == AV_SAMPLE_FMT_S32 ||
         avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {
         int ch;
         for (ch = 0; ch < avctx->channels; ch++) {
             s->planar_samples[ch] = av_malloc(avctx->frame_size *
                                               av_get_bytes_per_sample(avctx->sample_fmt));
             if (!s->planar_samples[ch]) {
                 ret = AVERROR(ENOMEM);
                 goto error;
             }
         }
     }

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

 #define DEINTERLEAVE(type, scale) do {                  \
     int ch, i;                                          \
     for (ch = 0; ch < s->avctx->channels; ch++) {       \
         const type *input = samples;                    \
         type      *output = s->planar_samples[ch];      \
         input += ch;                                    \
         for (i = 0; i < nb_samples; i++) {              \
             output[i] = *input * scale;                 \
             input += s->avctx->channels;                \
         }                                               \
     }                                                   \
 } while (0)

 static int encode_frame_int16(LAMEContext *s, void *samples, int nb_samples)
 {
     if (s->avctx->channels > 1) {
         return lame_encode_buffer_interleaved(s->gfp, samples,
                                               nb_samples,
                                               s->buffer + s->buffer_index,
                                               BUFFER_SIZE - s->buffer_index);
     } else {
         return lame_encode_buffer(s->gfp, samples, NULL, nb_samples,
                                   s->buffer + s->buffer_index,
                                   BUFFER_SIZE - s->buffer_index);
     }
 }

 static int encode_frame_int32(LAMEContext *s, void *samples, int nb_samples)
 {
     DEINTERLEAVE(int32_t, 1);

     return lame_encode_buffer_int(s->gfp,
                                   s->planar_samples[0], s->planar_samples[1],
                                   nb_samples,
                                   s->buffer + s->buffer_index,
                                   BUFFER_SIZE - s->buffer_index);
 }

 static int encode_frame_float(LAMEContext *s, void *samples, int nb_samples)
 {
     DEINTERLEAVE(float, 32768.0f);

     return lame_encode_buffer_float(s->gfp,
                                     s->planar_samples[0], s->planar_samples[1],
                                     nb_samples,
                                     s->buffer + s->buffer_index,
                                     BUFFER_SIZE - s->buffer_index);
 }

 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;
     int lame_result;

     if (frame) {
         switch (avctx->sample_fmt) {
         case AV_SAMPLE_FMT_S16:
             lame_result = encode_frame_int16(s, frame->data[0], frame->nb_samples);
             break;
         case AV_SAMPLE_FMT_S32:
             lame_result = encode_frame_int32(s, frame->data[0], frame->nb_samples);
             break;
         case AV_SAMPLE_FMT_FLT:
             lame_result = encode_frame_float(s, frame->data[0], frame->nb_samples);
             break;
         default:
             return AVERROR_BUG;
         }
     } else {
         lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index,
                                         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, BUFFER_SIZE - s->buffer_index);
         }
         return -1;
     }
     s->buffer_index += lame_result;

     /* 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;
     if (avpriv_mpegaudio_decode_header(&hdr, AV_RB32(s->buffer))) {
         av_log(avctx, AV_LOG_ERROR, "free format output not supported\n");
         return -1;
     }
     len = hdr.frame_size;
     av_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)))
             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);

         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_INT, { 1 }, 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",
     .type                  = AVMEDIA_TYPE_AUDIO,
     .id                    = CODEC_ID_MP3,
     .priv_data_size        = sizeof(LAMEContext),
     .init                  = mp3lame_encode_init,
     .encode2               = mp3lame_encode_frame,
     .close                 = mp3lame_encode_close,
     .capabilities          = CODEC_CAP_DELAY | CODEC_CAP_SMALL_LAST_FRAME,
     .sample_fmts           = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32,
                                                              AV_SAMPLE_FMT_FLT,
                                                              AV_SAMPLE_FMT_S16,
                                                              AV_SAMPLE_FMT_NONE },
     .supported_samplerates = libmp3lame_sample_rates,
     .channel_layouts       = (const uint64_t[]) { AV_CH_LAYOUT_MONO,
                                                   AV_CH_LAYOUT_STEREO,
                                                   0},
     .long_name             = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
     .priv_class            = &libmp3lame_class,
     .defaults              = libmp3lame_defaults,
 };
	/*
	* 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/audioconvert.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[BUFFER_SIZE];
	int buffer_index;
	int reservoir;
	void *planar_samples[2];
	AudioFrameQueue afq;
	} LAMEContext;


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

	#if FF_API_OLD_ENCODE_AUDIO
	av_freep(&avctx->coded_frame);
	#endif
	av_freep(&s->planar_samples[0]);
	av_freep(&s->planar_samples[1]);

	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()) == NULL)
	return AVERROR(ENOMEM);


	lame_set_num_channels(s->gfp, avctx->channels);
	lame_set_mode(s->gfp, avctx->channels > 1 ? JOINT_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, 5);
	else
	lame_set_quality(s->gfp, avctx->compression_level);

	/* rate control */
	if (avctx->flags & CODEC_FLAG_QSCALE) {
	lame_set_VBR(s->gfp, vbr_default);
	lame_set_VBR_quality(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
	} else {
	if (avctx->bit_rate)
	lame_set_brate(s->gfp, avctx->bit_rate / 1000);
	}

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

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

	#if FF_API_OLD_ENCODE_AUDIO
	avctx->coded_frame = avcodec_alloc_frame();
	if (!avctx->coded_frame) {
	ret = AVERROR(ENOMEM);
	goto error;
	}
	#endif

	/* sample format */
	if (avctx->sample_fmt == AV_SAMPLE_FMT_S32 \|\|
	avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {
	int ch;
	for (ch = 0; ch < avctx->channels; ch++) {
	s->planar_samples[ch] = av_malloc(avctx->frame_size *
	av_get_bytes_per_sample(avctx->sample_fmt));
	if (!s->planar_samples[ch]) {
	ret = AVERROR(ENOMEM);
	goto error;
	}
	}
	}

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

	#define DEINTERLEAVE(type, scale) do { \
	int ch, i; \
	for (ch = 0; ch < s->avctx->channels; ch++) { \
	const type *input = samples; \
	type *output = s->planar_samples[ch]; \
	input += ch; \
	for (i = 0; i < nb_samples; i++) { \
	output[i] = input scale; \
	input += s->avctx->channels; \
	} \
	} \
	} while (0)

	static int encode_frame_int16(LAMEContext s, void samples, int nb_samples)
	{
	if (s->avctx->channels > 1) {
	return lame_encode_buffer_interleaved(s->gfp, samples,
	nb_samples,
	s->buffer + s->buffer_index,
	BUFFER_SIZE - s->buffer_index);
	} else {
	return lame_encode_buffer(s->gfp, samples, NULL, nb_samples,
	s->buffer + s->buffer_index,
	BUFFER_SIZE - s->buffer_index);
	}
	}

	static int encode_frame_int32(LAMEContext s, void samples, int nb_samples)
	{
	DEINTERLEAVE(int32_t, 1);

	return lame_encode_buffer_int(s->gfp,
	s->planar_samples[0], s->planar_samples[1],
	nb_samples,
	s->buffer + s->buffer_index,
	BUFFER_SIZE - s->buffer_index);
	}

	static int encode_frame_float(LAMEContext s, void samples, int nb_samples)
	{
	DEINTERLEAVE(float, 32768.0f);

	return lame_encode_buffer_float(s->gfp,
	s->planar_samples[0], s->planar_samples[1],
	nb_samples,
	s->buffer + s->buffer_index,
	BUFFER_SIZE - s->buffer_index);
	}

	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;
	int lame_result;

	if (frame) {
	switch (avctx->sample_fmt) {
	case AV_SAMPLE_FMT_S16:
	lame_result = encode_frame_int16(s, frame->data[0], frame->nb_samples);
	break;
	case AV_SAMPLE_FMT_S32:
	lame_result = encode_frame_int32(s, frame->data[0], frame->nb_samples);
	break;
	case AV_SAMPLE_FMT_FLT:
	lame_result = encode_frame_float(s, frame->data[0], frame->nb_samples);
	break;
	default:
	return AVERROR_BUG;
	}
	} else {
	lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index,
	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, BUFFER_SIZE - s->buffer_index);
	}
	return -1;
	}
	s->buffer_index += lame_result;

	/* 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;
	if (avpriv_mpegaudio_decode_header(&hdr, AV_RB32(s->buffer))) {
	av_log(avctx, AV_LOG_ERROR, "free format output not supported\n");
	return -1;
	}
	len = hdr.frame_size;
	av_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)))
	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);

	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_INT, { 1 }, 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",
	.type = AVMEDIA_TYPE_AUDIO,
	.id = CODEC_ID_MP3,
	.priv_data_size = sizeof(LAMEContext),
	.init = mp3lame_encode_init,
	.encode2 = mp3lame_encode_frame,
	.close = mp3lame_encode_close,
	.capabilities = CODEC_CAP_DELAY \| CODEC_CAP_SMALL_LAST_FRAME,
	.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32,
	AV_SAMPLE_FMT_FLT,
	AV_SAMPLE_FMT_S16,
	AV_SAMPLE_FMT_NONE },
	.supported_samplerates = libmp3lame_sample_rates,
	.channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_MONO,
	AV_CH_LAYOUT_STEREO,
	0},
	.long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
	.priv_class = &libmp3lame_class,
	.defaults = libmp3lame_defaults,
	};