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https://github.com/FFmpeg/FFmpeg.git
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Add single stream LATM/LOAS decoder
The decoder is just a wrapper around the AAC decoder.
based on patch by Paul Kendall { paul <ät> kcbbs gen nz }
Originally committed as revision 25642 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Janne Grunau
parent
6c003e6de8
commit
136e19e1cf
@ -50,6 +50,7 @@ version <next>:
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- transpose filter added
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- ffmpeg -force_key_frames option added
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- demuxer for receiving raw rtp:// URLs without an SDP description
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- single stream LATM/LOAS decoder
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version 0.6:
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1
configure
vendored
1
configure
vendored
@ -1188,6 +1188,7 @@ rdft_select="fft"
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# decoders / encoders / hardware accelerators
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aac_decoder_select="mdct rdft"
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aac_encoder_select="mdct"
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aac_latm_decoder_select="aac_decoder aac_latm_parser"
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ac3_decoder_select="mdct ac3_parser"
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alac_encoder_select="lpc"
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amrnb_decoder_select="lsp"
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@ -576,6 +576,7 @@ OBJS-$(CONFIG_H264_PARSER) += h264_parser.o h264.o \
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h264_loopfilter.o h264_cabac.o \
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h264_cavlc.o h264_ps.o \
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mpegvideo.o error_resilience.o
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OBJS-$(CONFIG_AAC_LATM_PARSER) += latm_parser.o
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OBJS-$(CONFIG_MJPEG_PARSER) += mjpeg_parser.o
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OBJS-$(CONFIG_MLP_PARSER) += mlp_parser.o mlp.o
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OBJS-$(CONFIG_MPEG4VIDEO_PARSER) += mpeg4video_parser.o h263.o \
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@ -3,6 +3,10 @@
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* Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
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* Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
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*
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* AAC LATM decoder
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* Copyright (c) 2008-2010 Paul Kendall <paul@kcbbs.gen.nz>
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* Copyright (c) 2010 Janne Grunau <janne-ffmpeg@jannau.net>
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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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@ -2098,6 +2102,261 @@ static av_cold int aac_decode_close(AVCodecContext *avctx)
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return 0;
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}
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#define LOAS_SYNC_WORD 0x2b7 ///< 11 bits LOAS sync word
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struct LATMContext {
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AACContext aac_ctx; ///< containing AACContext
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int initialized; ///< initilized after a valid extradata was seen
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// parser data
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int audio_mux_version_A; ///< LATM syntax version
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int frame_length_type; ///< 0/1 variable/fixed frame length
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int frame_length; ///< frame length for fixed frame length
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};
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static inline uint32_t latm_get_value(GetBitContext *b)
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{
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int length = get_bits(b, 2);
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return get_bits_long(b, (length+1)*8);
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}
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static int latm_decode_audio_specific_config(struct LATMContext *latmctx,
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GetBitContext *gb)
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{
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AVCodecContext *avctx = latmctx->aac_ctx.avctx;
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MPEG4AudioConfig m4ac;
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int config_start_bit = get_bits_count(gb);
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int bits_consumed, esize;
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if (config_start_bit % 8) {
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av_log_missing_feature(latmctx->aac_ctx.avctx, "audio specific "
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"config not byte aligned.\n", 1);
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return AVERROR_INVALIDDATA;
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} else {
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bits_consumed =
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decode_audio_specific_config(NULL, avctx, &m4ac,
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gb->buffer + (config_start_bit / 8),
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get_bits_left(gb) / 8);
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if (bits_consumed < 0)
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return AVERROR_INVALIDDATA;
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esize = (bits_consumed+7) / 8;
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if (avctx->extradata_size <= esize) {
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av_free(avctx->extradata);
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avctx->extradata = av_malloc(esize + FF_INPUT_BUFFER_PADDING_SIZE);
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if (!avctx->extradata)
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return AVERROR(ENOMEM);
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}
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avctx->extradata_size = esize;
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memcpy(avctx->extradata, gb->buffer + (config_start_bit/8), esize);
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memset(avctx->extradata+esize, 0, FF_INPUT_BUFFER_PADDING_SIZE);
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}
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return bits_consumed;
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}
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static int read_stream_mux_config(struct LATMContext *latmctx,
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GetBitContext *gb)
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{
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int ret, audio_mux_version = get_bits(gb, 1);
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latmctx->audio_mux_version_A = 0;
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if (audio_mux_version)
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latmctx->audio_mux_version_A = get_bits(gb, 1);
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if (!latmctx->audio_mux_version_A) {
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if (audio_mux_version)
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latm_get_value(gb); // taraFullness
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skip_bits(gb, 1); // allStreamSameTimeFraming
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skip_bits(gb, 6); // numSubFrames
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// numPrograms
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if (get_bits(gb, 4)) { // numPrograms
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av_log_missing_feature(latmctx->aac_ctx.avctx,
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"multiple programs are not supported\n", 1);
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return AVERROR_PATCHWELCOME;
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}
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// for each program (which there is only on in DVB)
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// for each layer (which there is only on in DVB)
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if (get_bits(gb, 3)) { // numLayer
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av_log_missing_feature(latmctx->aac_ctx.avctx,
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"multiple layers are not supported\n", 1);
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return AVERROR_PATCHWELCOME;
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}
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// for all but first stream: use_same_config = get_bits(gb, 1);
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if (!audio_mux_version) {
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if ((ret = latm_decode_audio_specific_config(latmctx, gb)) < 0)
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return ret;
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} else {
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int ascLen = latm_get_value(gb);
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if ((ret = latm_decode_audio_specific_config(latmctx, gb)) < 0)
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return ret;
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ascLen -= ret;
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skip_bits_long(gb, ascLen);
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}
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latmctx->frame_length_type = get_bits(gb, 3);
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switch (latmctx->frame_length_type) {
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case 0:
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skip_bits(gb, 8); // latmBufferFullness
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break;
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case 1:
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latmctx->frame_length = get_bits(gb, 9);
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break;
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case 3:
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case 4:
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case 5:
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skip_bits(gb, 6); // CELP frame length table index
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break;
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case 6:
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case 7:
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skip_bits(gb, 1); // HVXC frame length table index
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break;
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}
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if (get_bits(gb, 1)) { // other data
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if (audio_mux_version) {
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latm_get_value(gb); // other_data_bits
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} else {
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int esc;
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do {
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esc = get_bits(gb, 1);
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skip_bits(gb, 8);
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} while (esc);
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}
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}
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if (get_bits(gb, 1)) // crc present
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skip_bits(gb, 8); // config_crc
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}
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return 0;
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}
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static int read_payload_length_info(struct LATMContext *ctx, GetBitContext *gb)
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{
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uint8_t tmp;
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if (ctx->frame_length_type == 0) {
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int mux_slot_length = 0;
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do {
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tmp = get_bits(gb, 8);
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mux_slot_length += tmp;
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} while (tmp == 255);
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return mux_slot_length;
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} else if (ctx->frame_length_type == 1) {
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return ctx->frame_length;
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} else if (ctx->frame_length_type == 3 ||
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ctx->frame_length_type == 5 ||
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ctx->frame_length_type == 7) {
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skip_bits(gb, 2); // mux_slot_length_coded
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}
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return 0;
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}
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static int read_audio_mux_element(struct LATMContext *latmctx,
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GetBitContext *gb)
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{
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int err;
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uint8_t use_same_mux = get_bits(gb, 1);
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if (!use_same_mux) {
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if ((err = read_stream_mux_config(latmctx, gb)) < 0)
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return err;
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} else if (!latmctx->aac_ctx.avctx->extradata) {
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av_log(latmctx->aac_ctx.avctx, AV_LOG_DEBUG,
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"no decoder config found\n");
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return AVERROR(EAGAIN);
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}
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if (latmctx->audio_mux_version_A == 0) {
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int mux_slot_length_bytes = read_payload_length_info(latmctx, gb);
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if (mux_slot_length_bytes * 8 > get_bits_left(gb)) {
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av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR, "incomplete frame\n");
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return AVERROR_INVALIDDATA;
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} else if (mux_slot_length_bytes * 8 + 256 < get_bits_left(gb)) {
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av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
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"frame length mismatch %d << %d\n",
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mux_slot_length_bytes * 8, get_bits_left(gb));
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return AVERROR_INVALIDDATA;
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}
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}
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return 0;
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}
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static int latm_decode_frame(AVCodecContext *avctx, void *out, int *out_size,
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AVPacket *avpkt)
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{
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struct LATMContext *latmctx = avctx->priv_data;
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int muxlength, err;
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GetBitContext gb;
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if (avpkt->size == 0)
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return 0;
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init_get_bits(&gb, avpkt->data, avpkt->size * 8);
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// check for LOAS sync word
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if (get_bits(&gb, 11) != LOAS_SYNC_WORD)
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return AVERROR_INVALIDDATA;
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muxlength = get_bits(&gb, 13);
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// not enough data, the parser should have sorted this
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if (muxlength+3 > avpkt->size)
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return AVERROR_INVALIDDATA;
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if ((err = read_audio_mux_element(latmctx, &gb)) < 0)
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return err;
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if (!latmctx->initialized) {
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if (!avctx->extradata) {
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*out_size = 0;
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return avpkt->size;
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} else {
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if ((err = aac_decode_init(avctx)) < 0)
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return err;
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latmctx->initialized = 1;
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}
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}
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if (show_bits(&gb, 12) == 0xfff) {
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av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
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"ADTS header detected, probably as result of configuration "
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"misparsing\n");
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return AVERROR_INVALIDDATA;
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}
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if ((err = aac_decode_frame_int(avctx, out, out_size, &gb)) < 0)
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return err;
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return muxlength;
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}
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av_cold static int latm_decode_init(AVCodecContext *avctx)
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{
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struct LATMContext *latmctx = avctx->priv_data;
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int ret;
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ret = aac_decode_init(avctx);
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if (avctx->extradata_size > 0) {
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latmctx->initialized = !ret;
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} else {
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latmctx->initialized = 0;
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}
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return ret;
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}
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AVCodec aac_decoder = {
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"aac",
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AVMEDIA_TYPE_AUDIO,
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@ -2113,3 +2372,23 @@ AVCodec aac_decoder = {
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},
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.channel_layouts = aac_channel_layout,
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};
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/*
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Note: This decoder filter is intended to decode LATM streams transferred
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in MPEG transport streams which only contain one program.
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To do a more complex LATM demuxing a separate LATM demuxer should be used.
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*/
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AVCodec aac_latm_decoder = {
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.name = "aac_latm",
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.type = CODEC_TYPE_AUDIO,
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.id = CODEC_ID_AAC_LATM,
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.priv_data_size = sizeof(struct LATMContext),
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.init = latm_decode_init,
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.close = aac_decode_close,
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.decode = latm_decode_frame,
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.long_name = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Codec LATM syntax)"),
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.sample_fmts = (const enum SampleFormat[]) {
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SAMPLE_FMT_S16,SAMPLE_FMT_NONE
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},
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.channel_layouts = aac_channel_layout,
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};
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@ -220,6 +220,7 @@ void avcodec_register_all(void)
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/* audio codecs */
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REGISTER_ENCDEC (AAC, aac);
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REGISTER_DECODER (AAC_LATM, aac_latm);
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REGISTER_ENCDEC (AC3, ac3);
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REGISTER_ENCDEC (ALAC, alac);
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REGISTER_DECODER (ALS, als);
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@ -366,6 +367,7 @@ void avcodec_register_all(void)
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/* parsers */
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REGISTER_PARSER (AAC, aac);
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REGISTER_PARSER (AAC_LATM, aac_latm);
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REGISTER_PARSER (AC3, ac3);
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REGISTER_PARSER (CAVSVIDEO, cavsvideo);
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REGISTER_PARSER (DCA, dca);
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@ -31,7 +31,7 @@
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#include "libavutil/cpu.h"
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#define LIBAVCODEC_VERSION_MAJOR 52
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#define LIBAVCODEC_VERSION_MINOR 93
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#define LIBAVCODEC_VERSION_MINOR 94
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#define LIBAVCODEC_VERSION_MICRO 0
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#define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
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@ -376,6 +376,7 @@ enum CodecID {
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CODEC_ID_ATRAC1,
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CODEC_ID_BINKAUDIO_RDFT,
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CODEC_ID_BINKAUDIO_DCT,
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CODEC_ID_AAC_LATM,
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/* subtitle codecs */
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CODEC_ID_DVD_SUBTITLE= 0x17000,
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119
libavcodec/latm_parser.c
Normal file
119
libavcodec/latm_parser.c
Normal file
@ -0,0 +1,119 @@
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/*
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* copyright (c) 2008 Paul Kendall <paul@kcbbs.gen.nz>
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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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* AAC LATM parser
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*/
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#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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#include <math.h>
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#include <sys/types.h>
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#include "parser.h"
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#define LATM_HEADER 0x56e000 // 0x2b7 (11 bits)
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#define LATM_MASK 0xFFE000 // top 11 bits
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#define LATM_SIZE_MASK 0x001FFF // bottom 13 bits
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typedef struct LATMParseContext{
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ParseContext pc;
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int count;
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} LATMParseContext;
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/**
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* finds the end of the current frame in the bitstream.
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* @return the position of the first byte of the next frame, or -1
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*/
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static int latm_find_frame_end(AVCodecParserContext *s1, const uint8_t *buf,
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int buf_size)
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{
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LATMParseContext *s = s1->priv_data;
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ParseContext *pc = &s->pc;
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int pic_found, i;
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uint32_t state;
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pic_found = pc->frame_start_found;
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state = pc->state;
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i = 0;
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if (!pic_found) {
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for (i = 0; i < buf_size; i++) {
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state = (state<<8) | buf[i];
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if ((state & LATM_MASK) == LATM_HEADER) {
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i++;
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s->count = -i;
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pic_found = 1;
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break;
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}
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}
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}
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if (pic_found) {
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/* EOF considered as end of frame */
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if (buf_size == 0)
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return 0;
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if ((state & LATM_SIZE_MASK) - s->count <= buf_size) {
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pc->frame_start_found = 0;
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pc->state = -1;
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return (state & LATM_SIZE_MASK) - s->count;
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}
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}
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s->count += buf_size;
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pc->frame_start_found = pic_found;
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pc->state = state;
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return END_NOT_FOUND;
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}
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static int latm_parse(AVCodecParserContext *s1, AVCodecContext *avctx,
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const uint8_t **poutbuf, int *poutbuf_size,
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const uint8_t *buf, int buf_size)
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{
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LATMParseContext *s = s1->priv_data;
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ParseContext *pc = &s->pc;
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int next;
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|
||||
if (s1->flags & PARSER_FLAG_COMPLETE_FRAMES) {
|
||||
next = buf_size;
|
||||
} else {
|
||||
next = latm_find_frame_end(s1, buf, buf_size);
|
||||
|
||||
if (ff_combine_frame(pc, next, &buf, &buf_size) < 0) {
|
||||
*poutbuf = NULL;
|
||||
*poutbuf_size = 0;
|
||||
return buf_size;
|
||||
}
|
||||
}
|
||||
*poutbuf = buf;
|
||||
*poutbuf_size = buf_size;
|
||||
return next;
|
||||
}
|
||||
|
||||
AVCodecParser aac_latm_parser = {
|
||||
{ CODEC_ID_AAC_LATM },
|
||||
sizeof(LATMParseContext),
|
||||
NULL,
|
||||
latm_parse,
|
||||
ff_parse_close
|
||||
};
|
||||
Loading…
Reference in New Issue
Block a user