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

5 files changed, 403 insertions, 1 deletions

diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index 385ae02..5abc495 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -576,6 +576,7 @@ OBJS-$(CONFIG_H264_PARSER)             += h264_parser.o h264.o            \
                                           h264_loopfilter.o h264_cabac.o \
                                           h264_cavlc.o h264_ps.o \
                                           mpegvideo.o error_resilience.o
+OBJS-$(CONFIG_AAC_LATM_PARSER)         += latm_parser.o
 OBJS-$(CONFIG_MJPEG_PARSER)            += mjpeg_parser.o
 OBJS-$(CONFIG_MLP_PARSER)              += mlp_parser.o mlp.o
 OBJS-$(CONFIG_MPEG4VIDEO_PARSER)       += mpeg4video_parser.o h263.o \
diff --git a/libavcodec/aacdec.c b/libavcodec/aacdec.c
index 7c23bdf..f3073c6 100644
--- a/libavcodec/aacdec.c
+++ b/libavcodec/aacdec.c
@@ -3,6 +3,10 @@
  * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
  * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
  *
+ * AAC LATM decoder
+ * Copyright (c) 2008-2010 Paul Kendall <paul@kcbbs.gen.nz>
+ * Copyright (c) 2010      Janne Grunau <janne-ffmpeg@jannau.net>
+ *
  * This file is part of FFmpeg.
  *
  * FFmpeg is free software; you can redistribute it and/or
@@ -2098,6 +2102,261 @@ static av_cold int aac_decode_close(AVCodecContext *avctx)
     return 0;
 }
 
+
+#define LOAS_SYNC_WORD   0x2b7       ///< 11 bits LOAS sync word
+
+struct LATMContext {
+    AACContext      aac_ctx;             ///< containing AACContext
+    int             initialized;         ///< initilized after a valid extradata was seen
+
+    // parser data
+    int             audio_mux_version_A; ///< LATM syntax version
+    int             frame_length_type;   ///< 0/1 variable/fixed frame length
+    int             frame_length;        ///< frame length for fixed frame length
+};
+
+static inline uint32_t latm_get_value(GetBitContext *b)
+{
+    int length = get_bits(b, 2);
+
+    return get_bits_long(b, (length+1)*8);
+}
+
+static int latm_decode_audio_specific_config(struct LATMContext *latmctx,
+                                             GetBitContext *gb)
+{
+    AVCodecContext *avctx = latmctx->aac_ctx.avctx;
+    MPEG4AudioConfig m4ac;
+    int  config_start_bit = get_bits_count(gb);
+    int     bits_consumed, esize;
+
+    if (config_start_bit % 8) {
+        av_log_missing_feature(latmctx->aac_ctx.avctx, "audio specific "
+                               "config not byte aligned.\n", 1);
+        return AVERROR_INVALIDDATA;
+    } else {
+        bits_consumed =
+            decode_audio_specific_config(NULL, avctx, &m4ac,
+                                         gb->buffer + (config_start_bit / 8),
+                                         get_bits_left(gb) / 8);
+
+        if (bits_consumed < 0)
+            return AVERROR_INVALIDDATA;
+
+        esize = (bits_consumed+7) / 8;
+
+        if (avctx->extradata_size <= esize) {
+            av_free(avctx->extradata);
+            avctx->extradata = av_malloc(esize + FF_INPUT_BUFFER_PADDING_SIZE);
+            if (!avctx->extradata)
+                return AVERROR(ENOMEM);
+        }
+
+        avctx->extradata_size = esize;
+        memcpy(avctx->extradata, gb->buffer + (config_start_bit/8), esize);
+        memset(avctx->extradata+esize, 0, FF_INPUT_BUFFER_PADDING_SIZE);
+    }
+
+    return bits_consumed;
+}
+
+static int read_stream_mux_config(struct LATMContext *latmctx,
+                                  GetBitContext *gb)
+{
+    int ret, audio_mux_version = get_bits(gb, 1);
+
+    latmctx->audio_mux_version_A = 0;
+    if (audio_mux_version)
+        latmctx->audio_mux_version_A = get_bits(gb, 1);
+
+    if (!latmctx->audio_mux_version_A) {
+
+        if (audio_mux_version)
+            latm_get_value(gb);                 // taraFullness
+
+        skip_bits(gb, 1);                       // allStreamSameTimeFraming
+        skip_bits(gb, 6);                       // numSubFrames
+        // numPrograms
+        if (get_bits(gb, 4)) {                  // numPrograms
+            av_log_missing_feature(latmctx->aac_ctx.avctx,
+                                   "multiple programs are not supported\n", 1);
+            return AVERROR_PATCHWELCOME;
+        }
+
+        // for each program (which there is only on in DVB)
+
+        // for each layer (which there is only on in DVB)
+        if (get_bits(gb, 3)) {                   // numLayer
+            av_log_missing_feature(latmctx->aac_ctx.avctx,
+                                   "multiple layers are not supported\n", 1);
+            return AVERROR_PATCHWELCOME;
+        }
+
+        // for all but first stream: use_same_config = get_bits(gb, 1);
+        if (!audio_mux_version) {
+            if ((ret = latm_decode_audio_specific_config(latmctx, gb)) < 0)
+                return ret;
+        } else {
+            int ascLen = latm_get_value(gb);
+            if ((ret = latm_decode_audio_specific_config(latmctx, gb)) < 0)
+                return ret;
+            ascLen -= ret;
+            skip_bits_long(gb, ascLen);
+        }
+
+        latmctx->frame_length_type = get_bits(gb, 3);
+        switch (latmctx->frame_length_type) {
+        case 0:
+            skip_bits(gb, 8);       // latmBufferFullness
+            break;
+        case 1:
+            latmctx->frame_length = get_bits(gb, 9);
+            break;
+        case 3:
+        case 4:
+        case 5:
+            skip_bits(gb, 6);       // CELP frame length table index
+            break;
+        case 6:
+        case 7:
+            skip_bits(gb, 1);       // HVXC frame length table index
+            break;
+        }
+
+        if (get_bits(gb, 1)) {                  // other data
+            if (audio_mux_version) {
+                latm_get_value(gb);             // other_data_bits
+            } else {
+                int esc;
+                do {
+                    esc = get_bits(gb, 1);
+                    skip_bits(gb, 8);
+                } while (esc);
+            }
+        }
+
+        if (get_bits(gb, 1))                     // crc present
+            skip_bits(gb, 8);                    // config_crc
+    }
+
+    return 0;
+}
+
+static int read_payload_length_info(struct LATMContext *ctx, GetBitContext *gb)
+{
+    uint8_t tmp;
+
+    if (ctx->frame_length_type == 0) {
+        int mux_slot_length = 0;
+        do {
+            tmp = get_bits(gb, 8);
+            mux_slot_length += tmp;
+        } while (tmp == 255);
+        return mux_slot_length;
+    } else if (ctx->frame_length_type == 1) {
+        return ctx->frame_length;
+    } else if (ctx->frame_length_type == 3 ||
+               ctx->frame_length_type == 5 ||
+               ctx->frame_length_type == 7) {
+        skip_bits(gb, 2);          // mux_slot_length_coded
+    }
+    return 0;
+}
+
+static int read_audio_mux_element(struct LATMContext *latmctx,
+                                  GetBitContext *gb)
+{
+    int err;
+    uint8_t use_same_mux = get_bits(gb, 1);
+    if (!use_same_mux) {
+        if ((err = read_stream_mux_config(latmctx, gb)) < 0)
+            return err;
+    } else if (!latmctx->aac_ctx.avctx->extradata) {
+        av_log(latmctx->aac_ctx.avctx, AV_LOG_DEBUG,
+               "no decoder config found\n");
+        return AVERROR(EAGAIN);
+    }
+    if (latmctx->audio_mux_version_A == 0) {
+        int mux_slot_length_bytes = read_payload_length_info(latmctx, gb);
+        if (mux_slot_length_bytes * 8 > get_bits_left(gb)) {
+            av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR, "incomplete frame\n");
+            return AVERROR_INVALIDDATA;
+        } else if (mux_slot_length_bytes * 8 + 256 < get_bits_left(gb)) {
+            av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
+                   "frame length mismatch %d << %d\n",
+                   mux_slot_length_bytes * 8, get_bits_left(gb));
+            return AVERROR_INVALIDDATA;
+        }
+    }
+    return 0;
+}
+
+
+static int latm_decode_frame(AVCodecContext *avctx, void *out, int *out_size,
+                             AVPacket *avpkt)
+{
+    struct LATMContext *latmctx = avctx->priv_data;
+    int                 muxlength, err;
+    GetBitContext       gb;
+
+    if (avpkt->size == 0)
+        return 0;
+
+    init_get_bits(&gb, avpkt->data, avpkt->size * 8);
+
+    // check for LOAS sync word
+    if (get_bits(&gb, 11) != LOAS_SYNC_WORD)
+        return AVERROR_INVALIDDATA;
+
+    muxlength = get_bits(&gb, 13);
+    // not enough data, the parser should have sorted this
+    if (muxlength+3 > avpkt->size)
+        return AVERROR_INVALIDDATA;
+
+    if ((err = read_audio_mux_element(latmctx, &gb)) < 0)
+        return err;
+
+    if (!latmctx->initialized) {
+        if (!avctx->extradata) {
+            *out_size = 0;
+            return avpkt->size;
+        } else {
+            if ((err = aac_decode_init(avctx)) < 0)
+                return err;
+            latmctx->initialized = 1;
+        }
+    }
+
+    if (show_bits(&gb, 12) == 0xfff) {
+        av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
+               "ADTS header detected, probably as result of configuration "
+               "misparsing\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    if ((err = aac_decode_frame_int(avctx, out, out_size, &gb)) < 0)
+        return err;
+
+    return muxlength;
+}
+
+av_cold static int latm_decode_init(AVCodecContext *avctx)
+{
+    struct LATMContext *latmctx = avctx->priv_data;
+    int ret;
+
+    ret = aac_decode_init(avctx);
+
+    if (avctx->extradata_size > 0) {
+        latmctx->initialized = !ret;
+    } else {
+        latmctx->initialized = 0;
+    }
+
+    return ret;
+}
+
+
 AVCodec aac_decoder = {
     "aac",
     AVMEDIA_TYPE_AUDIO,
@@ -2113,3 +2372,23 @@ AVCodec aac_decoder = {
     },
     .channel_layouts = aac_channel_layout,
 };
+
+/*
+    Note: This decoder filter is intended to decode LATM streams transferred
+    in MPEG transport streams which only contain one program.
+    To do a more complex LATM demuxing a separate LATM demuxer should be used.
+*/
+AVCodec aac_latm_decoder = {
+    .name = "aac_latm",
+    .type = CODEC_TYPE_AUDIO,
+    .id   = CODEC_ID_AAC_LATM,
+    .priv_data_size = sizeof(struct LATMContext),
+    .init   = latm_decode_init,
+    .close  = aac_decode_close,
+    .decode = latm_decode_frame,
+    .long_name = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Codec LATM syntax)"),
+    .sample_fmts = (const enum SampleFormat[]) {
+        SAMPLE_FMT_S16,SAMPLE_FMT_NONE
+    },
+    .channel_layouts = aac_channel_layout,
+};
diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c
index 89614ab..c3e4647 100644
--- a/libavcodec/allcodecs.c
+++ b/libavcodec/allcodecs.c
@@ -220,6 +220,7 @@ void avcodec_register_all(void)
 
     /* audio codecs */
     REGISTER_ENCDEC  (AAC, aac);
+    REGISTER_DECODER (AAC_LATM, aac_latm);
     REGISTER_ENCDEC  (AC3, ac3);
     REGISTER_ENCDEC  (ALAC, alac);
     REGISTER_DECODER (ALS, als);
@@ -366,6 +367,7 @@ void avcodec_register_all(void)
 
     /* parsers */
     REGISTER_PARSER  (AAC, aac);
+    REGISTER_PARSER  (AAC_LATM, aac_latm);
     REGISTER_PARSER  (AC3, ac3);
     REGISTER_PARSER  (CAVSVIDEO, cavsvideo);
     REGISTER_PARSER  (DCA, dca);
diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h
index 705259e..96d7987 100644
--- a/libavcodec/avcodec.h
+++ b/libavcodec/avcodec.h
@@ -31,7 +31,7 @@
 #include "libavutil/cpu.h"
 
 #define LIBAVCODEC_VERSION_MAJOR 52
-#define LIBAVCODEC_VERSION_MINOR 93
+#define LIBAVCODEC_VERSION_MINOR 94
 #define LIBAVCODEC_VERSION_MICRO  0
 
 #define LIBAVCODEC_VERSION_INT  AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
@@ -376,6 +376,7 @@ enum CodecID {
     CODEC_ID_ATRAC1,
     CODEC_ID_BINKAUDIO_RDFT,
     CODEC_ID_BINKAUDIO_DCT,
+    CODEC_ID_AAC_LATM,
 
     /* subtitle codecs */
     CODEC_ID_DVD_SUBTITLE= 0x17000,
diff --git a/libavcodec/latm_parser.c b/libavcodec/latm_parser.c
new file mode 100644
index 0000000..b8b67ef
--- /dev/null
+++ b/libavcodec/latm_parser.c
@@ -0,0 +1,119 @@
+/*
+ * copyright (c) 2008 Paul Kendall <paul@kcbbs.gen.nz>
+ *
+ * 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
+ * AAC LATM parser
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <math.h>
+#include <sys/types.h>
+
+#include "parser.h"
+
+#define LATM_HEADER     0x56e000        // 0x2b7 (11 bits)
+#define LATM_MASK       0xFFE000        // top 11 bits
+#define LATM_SIZE_MASK  0x001FFF        // bottom 13 bits
+
+typedef struct LATMParseContext{
+    ParseContext pc;
+    int count;
+} LATMParseContext;
+
+/**
+ * finds the end of the current frame in the bitstream.
+ * @return the position of the first byte of the next frame, or -1
+ */
+static int latm_find_frame_end(AVCodecParserContext *s1, const uint8_t *buf,
+                               int buf_size)
+{
+    LATMParseContext *s = s1->priv_data;
+    ParseContext *pc    = &s->pc;
+    int pic_found, i;
+    uint32_t state;
+
+    pic_found = pc->frame_start_found;
+    state     = pc->state;
+
+    i = 0;
+    if (!pic_found) {
+        for (i = 0; i < buf_size; i++) {
+            state = (state<<8) | buf[i];
+            if ((state & LATM_MASK) == LATM_HEADER) {
+                i++;
+                s->count  = -i;
+                pic_found = 1;
+                break;
+            }
+        }
+    }
+
+    if (pic_found) {
+        /* EOF considered as end of frame */
+        if (buf_size == 0)
+            return 0;
+        if ((state & LATM_SIZE_MASK) - s->count <= buf_size) {
+            pc->frame_start_found = 0;
+            pc->state             = -1;
+            return (state & LATM_SIZE_MASK) - s->count;
+        }
+    }
+
+    s->count             += buf_size;
+    pc->frame_start_found = pic_found;
+    pc->state             = state;
+
+    return END_NOT_FOUND;
+}
+
+static int latm_parse(AVCodecParserContext *s1, AVCodecContext *avctx,
+                      const uint8_t **poutbuf, int *poutbuf_size,
+                      const uint8_t *buf, int buf_size)
+{
+    LATMParseContext *s = s1->priv_data;
+    ParseContext *pc    = &s->pc;
+    int next;
+
+    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
+};