libavformat/latmenc.c - manifest_repos/ffmpeg - Git at Google

 /*
  * LATM/LOAS muxer
  * Copyright (c) 2011 Kieran Kunhya <kieran@kunhya.com>
  *
  * 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
  */

 #include "libavcodec/get_bits.h"
 #include "libavcodec/put_bits.h"
 #include "libavcodec/avcodec.h"
 #include "libavcodec/mpeg4audio.h"
 #include "libavutil/opt.h"
 #include "avformat.h"
 #include "internal.h"
 #include "rawenc.h"

 #define MAX_EXTRADATA_SIZE 1024

 typedef struct LATMContext {
     AVClass *av_class;
     int off;
     int channel_conf;
     int object_type;
     int counter;
     int mod;
     uint8_t buffer[0x1fff + MAX_EXTRADATA_SIZE + 1024];
 } LATMContext;

 static const AVOption options[] = {
     {"smc-interval", "StreamMuxConfig interval.",
      offsetof(LATMContext, mod), AV_OPT_TYPE_INT, {.i64 = 0x0014}, 0x0001, 0xffff, AV_OPT_FLAG_ENCODING_PARAM},
     {NULL},
 };

 static const AVClass latm_muxer_class = {
     .class_name = "LATM/LOAS muxer",
     .item_name  = av_default_item_name,
     .option     = options,
     .version    = LIBAVUTIL_VERSION_INT,
 };

 static int latm_decode_extradata(AVFormatContext *s, uint8_t *buf, int size)
 {
     LATMContext *ctx = s->priv_data;
     MPEG4AudioConfig m4ac;

     if (size > MAX_EXTRADATA_SIZE) {
         av_log(s, AV_LOG_ERROR, "Extradata is larger than currently supported.\n");
         return AVERROR_INVALIDDATA;
     }
     ctx->off = avpriv_mpeg4audio_get_config2(&m4ac, buf, size, 1, s);
     if (ctx->off < 0)
         return ctx->off;

     if (ctx->object_type == AOT_ALS && (ctx->off & 7)) {
         // as long as avpriv_mpeg4audio_get_config works correctly this is impossible
         av_log(s, AV_LOG_ERROR, "BUG: ALS offset is not byte-aligned\n");
         return AVERROR_INVALIDDATA;
     }
     /* FIXME: are any formats not allowed in LATM? */

     if (m4ac.object_type > AOT_SBR && m4ac.object_type != AOT_ALS) {
         av_log(s, AV_LOG_ERROR, "Muxing MPEG-4 AOT %d in LATM is not supported\n", m4ac.object_type);
         return AVERROR_INVALIDDATA;
     }
     ctx->channel_conf = m4ac.chan_config;
     ctx->object_type  = m4ac.object_type;

     return 0;
 }

 static int latm_write_header(AVFormatContext *s)
 {
     AVCodecParameters *par = s->streams[0]->codecpar;

     if (par->codec_id == AV_CODEC_ID_AAC_LATM)
         return 0;
     if (par->codec_id != AV_CODEC_ID_AAC && par->codec_id != AV_CODEC_ID_MP4ALS) {
         av_log(s, AV_LOG_ERROR, "Only AAC, LATM and ALS are supported\n");
         return AVERROR(EINVAL);
     }

     if (par->extradata_size > 0 &&
         latm_decode_extradata(s, par->extradata, par->extradata_size) < 0)
         return AVERROR_INVALIDDATA;

     return 0;
 }

 static void copy_bits(PutBitContext *pb, const uint8_t *src, int length)
 {
     int words = length >> 4;
     int bits  = length & 15;
     int i;
     for (i = 0; i < words; i++)
         put_bits(pb, 16, AV_RB16(src + 2 * i));
     if (bits)
         put_bits(pb, bits, AV_RB16(src + 2 * words) >> (16 - bits));
 }

 static void latm_write_frame_header(AVFormatContext *s, PutBitContext *bs)
 {
     LATMContext *ctx = s->priv_data;
     AVCodecParameters *par = s->streams[0]->codecpar;
     int header_size;

     /* AudioMuxElement */
     put_bits(bs, 1, !!ctx->counter);

     if (!ctx->counter) {
         /* StreamMuxConfig */
         put_bits(bs, 1, 0); /* audioMuxVersion */
         put_bits(bs, 1, 1); /* allStreamsSameTimeFraming */
         put_bits(bs, 6, 0); /* numSubFrames */
         put_bits(bs, 4, 0); /* numProgram */
         put_bits(bs, 3, 0); /* numLayer */

         /* AudioSpecificConfig */
         if (ctx->object_type == AOT_ALS) {
             header_size = (par->extradata_size - (ctx->off >> 3)) * 8;
             copy_bits(bs, &par->extradata[ctx->off >> 3], header_size);
         } else {
             // + 3 assumes not scalable and dependsOnCoreCoder == 0,
             // see decode_ga_specific_config in libavcodec/aacdec.c
             copy_bits(bs, par->extradata, ctx->off + 3);

             if (!ctx->channel_conf) {
                 GetBitContext gb;
                 int ret = init_get_bits8(&gb, par->extradata, par->extradata_size);
                 av_assert0(ret >= 0); // extradata size has been checked already, so this should not fail
                 skip_bits_long(&gb, ctx->off + 3);
                 ff_copy_pce_data(bs, &gb);
             }
         }

         put_bits(bs, 3, 0); /* frameLengthType */
         put_bits(bs, 8, 0xff); /* latmBufferFullness */

         put_bits(bs, 1, 0); /* otherDataPresent */
         put_bits(bs, 1, 0); /* crcCheckPresent */
     }

     ctx->counter++;
     ctx->counter %= ctx->mod;
 }

 static int latm_write_packet(AVFormatContext *s, AVPacket *pkt)
 {
     LATMContext *ctx = s->priv_data;
     AVCodecParameters *par = s->streams[0]->codecpar;
     AVIOContext *pb = s->pb;
     PutBitContext bs;
     int i, len;
     uint8_t loas_header[] = "\x56\xe0\x00";

     if (par->codec_id == AV_CODEC_ID_AAC_LATM)
         return ff_raw_write_packet(s, pkt);

     if (!par->extradata) {
         if(pkt->size > 2 && pkt->data[0] == 0x56 && (pkt->data[1] >> 4) == 0xe &&
             (AV_RB16(pkt->data + 1) & 0x1FFF) + 3 == pkt->size)
             return ff_raw_write_packet(s, pkt);
         else {
             uint8_t *side_data;
             int side_data_size, ret;

             side_data = av_packet_get_side_data(pkt, AV_PKT_DATA_NEW_EXTRADATA,
                                                 &side_data_size);
             if (side_data_size) {
                 if (latm_decode_extradata(s, side_data, side_data_size) < 0)
                     return AVERROR_INVALIDDATA;
                 ret = ff_alloc_extradata(par, side_data_size);
                 if (ret < 0)
                     return ret;
                 memcpy(par->extradata, side_data, side_data_size);
             } else
                 return AVERROR_INVALIDDATA;
         }
     }

     if (pkt->size > 0x1fff)
         goto too_large;

     init_put_bits(&bs, ctx->buffer, pkt->size+1024+MAX_EXTRADATA_SIZE);

     latm_write_frame_header(s, &bs);

     /* PayloadLengthInfo() */
     for (i = 0; i <= pkt->size-255; i+=255)
         put_bits(&bs, 8, 255);

     put_bits(&bs, 8, pkt->size-i);

     /* The LATM payload is written unaligned */

     /* PayloadMux() */
     if (pkt->size && (pkt->data[0] & 0xe1) == 0x81) {
         // Convert byte-aligned DSE to non-aligned.
         // Due to the input format encoding we know that
         // it is naturally byte-aligned in the input stream,
         // so there are no padding bits to account for.
         // To avoid having to add padding bits and rearrange
         // the whole stream we just remove the byte-align flag.
         // This allows us to remux our FATE AAC samples into latm
         // files that are still playable with minimal effort.
         put_bits(&bs, 8, pkt->data[0] & 0xfe);
         copy_bits(&bs, pkt->data + 1, 8*pkt->size - 8);
     } else
         copy_bits(&bs, pkt->data, 8*pkt->size);

     flush_put_bits(&bs);

     len = put_bits_count(&bs) >> 3;

     if (len > 0x1fff)
         goto too_large;

     loas_header[1] |= (len >> 8) & 0x1f;
     loas_header[2] |= len & 0xff;

     avio_write(pb, loas_header, 3);
     avio_write(pb, ctx->buffer, len);

     return 0;

 too_large:
     av_log(s, AV_LOG_ERROR, "LATM packet size larger than maximum size 0x1fff\n");
     return AVERROR_INVALIDDATA;
 }

 static int latm_check_bitstream(struct AVFormatContext *s, const AVPacket *pkt)
 {
     int ret = 1;
     AVStream *st = s->streams[pkt->stream_index];

     if (st->codecpar->codec_id == AV_CODEC_ID_AAC) {
         if (pkt->size > 2 && (AV_RB16(pkt->data) & 0xfff0) == 0xfff0)
             ret = ff_stream_add_bitstream_filter(st, "aac_adtstoasc", NULL);
     }

     return ret;
 }

 AVOutputFormat ff_latm_muxer = {
     .name           = "latm",
     .long_name      = NULL_IF_CONFIG_SMALL("LOAS/LATM"),
     .mime_type      = "audio/MP4A-LATM",
     .extensions     = "latm,loas",
     .priv_data_size = sizeof(LATMContext),
     .audio_codec    = AV_CODEC_ID_AAC,
     .video_codec    = AV_CODEC_ID_NONE,
     .write_header   = latm_write_header,
     .write_packet   = latm_write_packet,
     .priv_class     = &latm_muxer_class,
     .check_bitstream= latm_check_bitstream,
     .flags          = AVFMT_NOTIMESTAMPS,
 };
	/*
	* LATM/LOAS muxer
	* Copyright (c) 2011 Kieran Kunhya <kieran@kunhya.com>
	*
	* 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
	*/

	#include "libavcodec/get_bits.h"
	#include "libavcodec/put_bits.h"
	#include "libavcodec/avcodec.h"
	#include "libavcodec/mpeg4audio.h"
	#include "libavutil/opt.h"
	#include "avformat.h"
	#include "internal.h"
	#include "rawenc.h"

	#define MAX_EXTRADATA_SIZE 1024

	typedef struct LATMContext {
	AVClass *av_class;
	int off;
	int channel_conf;
	int object_type;
	int counter;
	int mod;
	uint8_t buffer[0x1fff + MAX_EXTRADATA_SIZE + 1024];
	} LATMContext;

	static const AVOption options[] = {
	{"smc-interval", "StreamMuxConfig interval.",
	offsetof(LATMContext, mod), AV_OPT_TYPE_INT, {.i64 = 0x0014}, 0x0001, 0xffff, AV_OPT_FLAG_ENCODING_PARAM},
	{NULL},
	};

	static const AVClass latm_muxer_class = {
	.class_name = "LATM/LOAS muxer",
	.item_name = av_default_item_name,
	.option = options,
	.version = LIBAVUTIL_VERSION_INT,
	};

	static int latm_decode_extradata(AVFormatContext s, uint8_t buf, int size)
	{
	LATMContext *ctx = s->priv_data;
	MPEG4AudioConfig m4ac;

	if (size > MAX_EXTRADATA_SIZE) {
	av_log(s, AV_LOG_ERROR, "Extradata is larger than currently supported.\n");
	return AVERROR_INVALIDDATA;
	}
	ctx->off = avpriv_mpeg4audio_get_config2(&m4ac, buf, size, 1, s);
	if (ctx->off < 0)
	return ctx->off;

	if (ctx->object_type == AOT_ALS && (ctx->off & 7)) {
	// as long as avpriv_mpeg4audio_get_config works correctly this is impossible
	av_log(s, AV_LOG_ERROR, "BUG: ALS offset is not byte-aligned\n");
	return AVERROR_INVALIDDATA;
	}
	/* FIXME: are any formats not allowed in LATM? */

	if (m4ac.object_type > AOT_SBR && m4ac.object_type != AOT_ALS) {
	av_log(s, AV_LOG_ERROR, "Muxing MPEG-4 AOT %d in LATM is not supported\n", m4ac.object_type);
	return AVERROR_INVALIDDATA;
	}
	ctx->channel_conf = m4ac.chan_config;
	ctx->object_type = m4ac.object_type;

	return 0;
	}

	static int latm_write_header(AVFormatContext *s)
	{
	AVCodecParameters *par = s->streams[0]->codecpar;

	if (par->codec_id == AV_CODEC_ID_AAC_LATM)
	return 0;
	if (par->codec_id != AV_CODEC_ID_AAC && par->codec_id != AV_CODEC_ID_MP4ALS) {
	av_log(s, AV_LOG_ERROR, "Only AAC, LATM and ALS are supported\n");
	return AVERROR(EINVAL);
	}

	if (par->extradata_size > 0 &&
	latm_decode_extradata(s, par->extradata, par->extradata_size) < 0)
	return AVERROR_INVALIDDATA;

	return 0;
	}

	static void copy_bits(PutBitContext pb, const uint8_t src, int length)
	{
	int words = length >> 4;
	int bits = length & 15;
	int i;
	for (i = 0; i < words; i++)
	put_bits(pb, 16, AV_RB16(src + 2 * i));
	if (bits)
	put_bits(pb, bits, AV_RB16(src + 2 * words) >> (16 - bits));
	}

	static void latm_write_frame_header(AVFormatContext s, PutBitContext bs)
	{
	LATMContext *ctx = s->priv_data;
	AVCodecParameters *par = s->streams[0]->codecpar;
	int header_size;

	/* AudioMuxElement */
	put_bits(bs, 1, !!ctx->counter);

	if (!ctx->counter) {
	/* StreamMuxConfig */
	put_bits(bs, 1, 0); /* audioMuxVersion */
	put_bits(bs, 1, 1); /* allStreamsSameTimeFraming */
	put_bits(bs, 6, 0); /* numSubFrames */
	put_bits(bs, 4, 0); /* numProgram */
	put_bits(bs, 3, 0); /* numLayer */

	/* AudioSpecificConfig */
	if (ctx->object_type == AOT_ALS) {
	header_size = (par->extradata_size - (ctx->off >> 3)) * 8;
	copy_bits(bs, &par->extradata[ctx->off >> 3], header_size);
	} else {
	// + 3 assumes not scalable and dependsOnCoreCoder == 0,
	// see decode_ga_specific_config in libavcodec/aacdec.c
	copy_bits(bs, par->extradata, ctx->off + 3);

	if (!ctx->channel_conf) {
	GetBitContext gb;
	int ret = init_get_bits8(&gb, par->extradata, par->extradata_size);
	av_assert0(ret >= 0); // extradata size has been checked already, so this should not fail
	skip_bits_long(&gb, ctx->off + 3);
	ff_copy_pce_data(bs, &gb);
	}
	}

	put_bits(bs, 3, 0); /* frameLengthType */
	put_bits(bs, 8, 0xff); /* latmBufferFullness */

	put_bits(bs, 1, 0); /* otherDataPresent */
	put_bits(bs, 1, 0); /* crcCheckPresent */
	}

	ctx->counter++;
	ctx->counter %= ctx->mod;
	}

	static int latm_write_packet(AVFormatContext s, AVPacket pkt)
	{
	LATMContext *ctx = s->priv_data;
	AVCodecParameters *par = s->streams[0]->codecpar;
	AVIOContext *pb = s->pb;
	PutBitContext bs;
	int i, len;
	uint8_t loas_header[] = "\x56\xe0\x00";

	if (par->codec_id == AV_CODEC_ID_AAC_LATM)
	return ff_raw_write_packet(s, pkt);

	if (!par->extradata) {
	if(pkt->size > 2 && pkt->data[0] == 0x56 && (pkt->data[1] >> 4) == 0xe &&
	(AV_RB16(pkt->data + 1) & 0x1FFF) + 3 == pkt->size)
	return ff_raw_write_packet(s, pkt);
	else {
	uint8_t *side_data;
	int side_data_size, ret;

	side_data = av_packet_get_side_data(pkt, AV_PKT_DATA_NEW_EXTRADATA,
	&side_data_size);
	if (side_data_size) {
	if (latm_decode_extradata(s, side_data, side_data_size) < 0)
	return AVERROR_INVALIDDATA;
	ret = ff_alloc_extradata(par, side_data_size);
	if (ret < 0)
	return ret;
	memcpy(par->extradata, side_data, side_data_size);
	} else
	return AVERROR_INVALIDDATA;
	}
	}

	if (pkt->size > 0x1fff)
	goto too_large;

	init_put_bits(&bs, ctx->buffer, pkt->size+1024+MAX_EXTRADATA_SIZE);

	latm_write_frame_header(s, &bs);

	/* PayloadLengthInfo() */
	for (i = 0; i <= pkt->size-255; i+=255)
	put_bits(&bs, 8, 255);

	put_bits(&bs, 8, pkt->size-i);

	/* The LATM payload is written unaligned */

	/* PayloadMux() */
	if (pkt->size && (pkt->data[0] & 0xe1) == 0x81) {
	// Convert byte-aligned DSE to non-aligned.
	// Due to the input format encoding we know that
	// it is naturally byte-aligned in the input stream,
	// so there are no padding bits to account for.
	// To avoid having to add padding bits and rearrange
	// the whole stream we just remove the byte-align flag.
	// This allows us to remux our FATE AAC samples into latm
	// files that are still playable with minimal effort.
	put_bits(&bs, 8, pkt->data[0] & 0xfe);
	copy_bits(&bs, pkt->data + 1, 8*pkt->size - 8);
	} else
	copy_bits(&bs, pkt->data, 8*pkt->size);

	flush_put_bits(&bs);

	len = put_bits_count(&bs) >> 3;

	if (len > 0x1fff)
	goto too_large;

	loas_header[1] \|= (len >> 8) & 0x1f;
	loas_header[2] \|= len & 0xff;

	avio_write(pb, loas_header, 3);
	avio_write(pb, ctx->buffer, len);

	return 0;

	too_large:
	av_log(s, AV_LOG_ERROR, "LATM packet size larger than maximum size 0x1fff\n");
	return AVERROR_INVALIDDATA;
	}

	static int latm_check_bitstream(struct AVFormatContext s, const AVPacket pkt)
	{
	int ret = 1;
	AVStream *st = s->streams[pkt->stream_index];

	if (st->codecpar->codec_id == AV_CODEC_ID_AAC) {
	if (pkt->size > 2 && (AV_RB16(pkt->data) & 0xfff0) == 0xfff0)
	ret = ff_stream_add_bitstream_filter(st, "aac_adtstoasc", NULL);
	}

	return ret;
	}

	AVOutputFormat ff_latm_muxer = {
	.name = "latm",
	.long_name = NULL_IF_CONFIG_SMALL("LOAS/LATM"),
	.mime_type = "audio/MP4A-LATM",
	.extensions = "latm,loas",
	.priv_data_size = sizeof(LATMContext),
	.audio_codec = AV_CODEC_ID_AAC,
	.video_codec = AV_CODEC_ID_NONE,
	.write_header = latm_write_header,
	.write_packet = latm_write_packet,
	.priv_class = &latm_muxer_class,
	.check_bitstream= latm_check_bitstream,
	.flags = AVFMT_NOTIMESTAMPS,
	};