Support ADTS AAC files in the ffaac decoder (limited to streams containing one

raw_data_block() per ADTS frame)

Patch by Alex Converse ( alex converse gmail com) based on a patch by Robert
Swain ( robert swain gmail com )

Originally committed as revision 16485 to svn://svn.ffmpeg.org/ffmpeg/trunk

3 files changed, 153 insertions, 35 deletions

diff --git a/libavcodec/aac.c b/libavcodec/aac.c
index e815041618..545f1254dd 100644
--- a/libavcodec/aac.c
+++ b/libavcodec/aac.c
@@ -86,6 +86,7 @@
 #include "aactab.h"
 #include "aacdectab.h"
 #include "mpeg4audio.h"
+#include "aac_parser.h"
 
 #include <assert.h>
 #include <errno.h>
@@ -384,12 +385,24 @@ static av_cold int aac_decode_init(AVCodecContext * avccontext) {
 
     ac->avccontext = avccontext;
 
-    if (avccontext->extradata_size <= 0 ||
-        decode_audio_specific_config(ac, avccontext->extradata, avccontext->extradata_size))
+    if (avccontext->extradata_size > 0) {
+        if(decode_audio_specific_config(ac, avccontext->extradata, avccontext->extradata_size))
+            return -1;
+        avccontext->sample_rate = ac->m4ac.sample_rate;
+    } else if (avccontext->channels > 0) {
+        enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
+        memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
+        if(set_default_channel_config(ac, new_che_pos, avccontext->channels - (avccontext->channels == 8)))
+            return -1;
+        if(output_configure(ac, ac->che_pos, new_che_pos))
+            return -1;
+        ac->m4ac.sample_rate = avccontext->sample_rate;
+    } else {
+        ff_log_missing_feature(ac->avccontext, "Implicit channel configuration is", 0);
         return -1;
+    }
 
     avccontext->sample_fmt  = SAMPLE_FMT_S16;
-    avccontext->sample_rate = ac->m4ac.sample_rate;
     avccontext->frame_size  = 1024;
 
     AAC_INIT_VLC_STATIC( 0, 144);
@@ -1506,6 +1519,29 @@ static void spectral_to_sample(AACContext * ac) {
     }
 }
 
+static int parse_adts_frame_header(AACContext * ac, GetBitContext * gb) {
+
+    int size;
+    AACADTSHeaderInfo hdr_info;
+
+    size = ff_aac_parse_header(gb, &hdr_info);
+    if (size > 0) {
+        if (hdr_info.chan_config)
+            ac->m4ac.chan_config = hdr_info.chan_config;
+        ac->m4ac.sample_rate     = hdr_info.sample_rate;
+        ac->m4ac.sampling_index  = hdr_info.sampling_index;
+        ac->m4ac.object_type     = hdr_info.object_type;
+    }
+    if (hdr_info.num_aac_frames == 1) {
+        if (!hdr_info.crc_absent)
+            skip_bits(gb, 16);
+    } else {
+        ff_log_missing_feature(ac->avccontext, "More than one AAC RDB per ADTS frame is", 0);
+        return -1;
+    }
+    return size;
+}
+
 static int aac_decode_frame(AVCodecContext * avccontext, void * data, int * data_size, const uint8_t * buf, int buf_size) {
     AACContext * ac = avccontext->priv_data;
     GetBitContext gb;
@@ -1514,6 +1550,13 @@ static int aac_decode_frame(AVCodecContext * avccontext, void * data, int * data
 
     init_get_bits(&gb, buf, buf_size*8);
 
+    if (show_bits(&gb, 12) == 0xfff) {
+        if ((err = parse_adts_frame_header(ac, &gb)) < 0) {
+            av_log(avccontext, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
+            return -1;
+        }
+    }
+
     // parse
     while ((elem_type = get_bits(&gb, 3)) != TYPE_END) {
         elem_id = get_bits(&gb, 4);
diff --git a/libavcodec/aac_parser.c b/libavcodec/aac_parser.c
index 3ff14163da..e38b5ecf01 100644
--- a/libavcodec/aac_parser.c
+++ b/libavcodec/aac_parser.c
@@ -22,16 +22,64 @@
 
 #include "parser.h"
 #include "aac_ac3_parser.h"
+#include "aac_parser.h"
 #include "bitstream.h"
 #include "mpeg4audio.h"
 
 #define AAC_HEADER_SIZE 7
 
+int ff_aac_parse_header(GetBitContext *gbc, AACADTSHeaderInfo *hdr)
+{
+    int size, rdb, ch, sr;
+    int aot, crc_abs;
+
+    if(get_bits(gbc, 12) != 0xfff)
+        return AAC_AC3_PARSE_ERROR_SYNC;
+
+    skip_bits1(gbc);             /* id */
+    skip_bits(gbc, 2);           /* layer */
+    crc_abs = get_bits1(gbc);    /* protection_absent */
+    aot     = get_bits(gbc, 2);  /* profile_objecttype */
+    sr      = get_bits(gbc, 4);  /* sample_frequency_index */
+    if(!ff_mpeg4audio_sample_rates[sr])
+        return AAC_AC3_PARSE_ERROR_SAMPLE_RATE;
+    skip_bits1(gbc);             /* private_bit */
+    ch      = get_bits(gbc, 3);  /* channel_configuration */
+
+    if(!ff_mpeg4audio_channels[ch])
+        return AAC_AC3_PARSE_ERROR_CHANNEL_CFG;
+
+    skip_bits1(gbc);             /* original/copy */
+    skip_bits1(gbc);             /* home */
+
+    /* adts_variable_header */
+    skip_bits1(gbc);             /* copyright_identification_bit */
+    skip_bits1(gbc);             /* copyright_identification_start */
+    size    = get_bits(gbc, 13); /* aac_frame_length */
+    if(size < AAC_HEADER_SIZE)
+        return AAC_AC3_PARSE_ERROR_FRAME_SIZE;
+
+    skip_bits(gbc, 11);          /* adts_buffer_fullness */
+    rdb = get_bits(gbc, 2);      /* number_of_raw_data_blocks_in_frame */
+
+    hdr->object_type    = aot;
+    hdr->chan_config    = ch;
+    hdr->crc_absent     = crc_abs;
+    hdr->num_aac_frames = rdb + 1;
+    hdr->sampling_index = sr;
+    hdr->sample_rate    = ff_mpeg4audio_sample_rates[sr];
+    hdr->samples        = (rdb + 1) * 1024;
+    hdr->bit_rate       = size * 8 * hdr->sample_rate / hdr->samples;
+
+    return size;
+}
+
 static int aac_sync(uint64_t state, AACAC3ParseContext *hdr_info,
         int *need_next_header, int *new_frame_start)
 {
     GetBitContext bits;
-    int size, rdb, ch, sr;
+    AACADTSHeaderInfo hdr;
+    int size;
     union {
         uint64_t u64;
         uint8_t  u8[8];
@@ -40,40 +88,14 @@ static int aac_sync(uint64_t state, AACAC3ParseContext *hdr_info,
     tmp.u64 = be2me_64(state);
     init_get_bits(&bits, tmp.u8+8-AAC_HEADER_SIZE, AAC_HEADER_SIZE * 8);
 
-    if(get_bits(&bits, 12) != 0xfff)
-        return 0;
-
-    skip_bits1(&bits);          /* id */
-    skip_bits(&bits, 2);        /* layer */
-    skip_bits1(&bits);          /* protection_absent */
-    skip_bits(&bits, 2);        /* profile_objecttype */
-    sr = get_bits(&bits, 4);    /* sample_frequency_index */
-    if(!ff_mpeg4audio_sample_rates[sr])
-        return 0;
-    skip_bits1(&bits);          /* private_bit */
-    ch = get_bits(&bits, 3);    /* channel_configuration */
-    if(!ff_mpeg4audio_channels[ch])
+    if ((size = ff_aac_parse_header(&bits, &hdr)) < 0)
         return 0;
-    skip_bits1(&bits);          /* original/copy */
-    skip_bits1(&bits);          /* home */
-
-    /* adts_variable_header */
-    skip_bits1(&bits);          /* copyright_identification_bit */
-    skip_bits1(&bits);          /* copyright_identification_start */
-    size = get_bits(&bits, 13); /* aac_frame_length */
-    if(size < AAC_HEADER_SIZE)
-        return 0;
-
-    skip_bits(&bits, 11);       /* adts_buffer_fullness */
-    rdb = get_bits(&bits, 2);   /* number_of_raw_data_blocks_in_frame */
-
-    hdr_info->channels = ff_mpeg4audio_channels[ch];
-    hdr_info->sample_rate = ff_mpeg4audio_sample_rates[sr];
-    hdr_info->samples = (rdb + 1) * 1024;
-    hdr_info->bit_rate = size * 8 * hdr_info->sample_rate / hdr_info->samples;
-
     *need_next_header = 0;
     *new_frame_start  = 1;
+    hdr_info->sample_rate = hdr.sample_rate;
+    hdr_info->channels    = ff_mpeg4audio_channels[hdr.chan_config];
+    hdr_info->samples     = hdr.samples;
+    hdr_info->bit_rate    = hdr.bit_rate;
     return size;
 }
 
diff --git a/libavcodec/aac_parser.h b/libavcodec/aac_parser.h
new file mode 100644
index 0000000000..efc46785cd
--- /dev/null
+++ b/libavcodec/aac_parser.h
@@ -0,0 +1,53 @@
+/*
+ * AAC parser prototypes
+ * Copyright (c) 2003 Fabrice Bellard
+ * Copyright (c) 2003 Michael Niedermayer
+ *
+ * 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
+ */
+
+#ifndef AVCODEC_AAC_PARSER_H
+#define AVCODEC_AAC_PARSER_H
+
+#include <stdint.h>
+#include "aac_ac3_parser.h"
+#include "bitstream.h"
+
+typedef struct {
+    uint32_t sample_rate;
+    uint32_t samples;
+    uint32_t bit_rate;
+    uint8_t  crc_absent;
+    uint8_t  object_type;
+    uint8_t  sampling_index;
+    uint8_t  chan_config;
+    uint8_t  num_aac_frames;
+} AACADTSHeaderInfo;
+
+/**
+ * Parses AAC frame header.
+ * Parses the ADTS frame header to the end of the variable header, which is
+ * the first 54 bits.
+ * @param gbc[in] BitContext containing the first 54 bits of the frame.
+ * @param hdr[out] Pointer to struct where header info is written.
+ * @return Returns 0 on success, -1 if there is a sync word mismatch,
+ * -2 if the version element is invalid, -3 if the sample rate
+ * element is invalid, or -4 if the bit rate element is invalid.
+ */
+int ff_aac_parse_header(GetBitContext *gbc, AACADTSHeaderInfo *hdr);
+
+#endif /* AVCODEC_AAC_PARSER_H */