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Diffstat (limited to 'libavcodec/aacdec.c')
-rw-r--r--libavcodec/aacdec.c116
1 files changed, 82 insertions, 34 deletions
diff --git a/libavcodec/aacdec.c b/libavcodec/aacdec.c
index 958c9d2f97..909ecb173f 100644
--- a/libavcodec/aacdec.c
+++ b/libavcodec/aacdec.c
@@ -7,20 +7,20 @@
* Copyright (c) 2008-2010 Paul Kendall <paul@kcbbs.gen.nz>
* Copyright (c) 2010 Janne Grunau <janne-libav@jannau.net>
*
- * This file is part of Libav.
+ * This file is part of FFmpeg.
*
- * Libav is free software; you can redistribute it and/or
+ * 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.
*
- * Libav is distributed in the hope that it will be useful,
+ * 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 Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
@@ -112,7 +112,7 @@
static VLC vlc_scalefactors;
static VLC vlc_spectral[11];
-static const char overread_err[] = "Input buffer exhausted before END element found\n";
+#define overread_err "Input buffer exhausted before END element found\n"
static int count_channels(uint8_t (*layout)[3], int tags)
{
@@ -129,7 +129,7 @@ static int count_channels(uint8_t (*layout)[3], int tags)
/**
* Check for the channel element in the current channel position configuration.
* If it exists, make sure the appropriate element is allocated and map the
- * channel order to match the internal Libav channel layout.
+ * channel order to match the internal FFmpeg channel layout.
*
* @param che_pos current channel position configuration
* @param type channel element type
@@ -149,6 +149,10 @@ static av_cold int che_configure(AACContext *ac,
ff_aac_sbr_ctx_init(ac, &ac->che[type][id]->sbr);
}
if (type != TYPE_CCE) {
+ if (*channels >= MAX_CHANNELS - (type == TYPE_CPE || (type == TYPE_SCE && ac->oc[1].m4ac.ps == 1))) {
+ av_log(ac->avctx, AV_LOG_ERROR, "Too many channels\n");
+ return AVERROR_INVALIDDATA;
+ }
ac->output_data[(*channels)++] = ac->che[type][id]->ch[0].ret;
if (type == TYPE_CPE ||
(type == TYPE_SCE && ac->oc[1].m4ac.ps == 1)) {
@@ -366,9 +370,11 @@ static void push_output_configuration(AACContext *ac) {
*/
static void pop_output_configuration(AACContext *ac) {
if (ac->oc[1].status != OC_LOCKED) {
- ac->oc[1] = ac->oc[0];
- ac->avctx->channels = ac->oc[1].channels;
- ac->avctx->channel_layout = ac->oc[1].channel_layout;
+ if (ac->oc[0].status == OC_LOCKED) {
+ ac->oc[1] = ac->oc[0];
+ ac->avctx->channels = ac->oc[1].channels;
+ ac->avctx->channel_layout = ac->oc[1].channel_layout;
+ }
}
}
@@ -413,13 +419,31 @@ static int output_configure(AACContext *ac,
}
memcpy(ac->tag_che_map, ac->che, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0]));
- avctx->channel_layout = ac->oc[1].channel_layout = layout;
+ if (layout) avctx->channel_layout = layout;
+ ac->oc[1].channel_layout = layout;
avctx->channels = ac->oc[1].channels = channels;
ac->oc[1].status = oc_type;
return 0;
}
+static void flush(AVCodecContext *avctx)
+{
+ AACContext *ac= avctx->priv_data;
+ int type, i, j;
+
+ for (type = 3; type >= 0; type--) {
+ for (i = 0; i < MAX_ELEM_ID; i++) {
+ ChannelElement *che = ac->che[type][i];
+ if (che) {
+ for (j = 0; j <= 1; j++) {
+ memset(che->ch[j].saved, 0, sizeof(che->ch[j].saved));
+ }
+ }
+ }
+ }
+}
+
/**
* Set up channel positions based on a default channel configuration
* as specified in table 1.17.
@@ -453,6 +477,8 @@ static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
int layout_map_tags;
push_output_configuration(ac);
+ av_log(ac->avctx, AV_LOG_DEBUG, "mono with CPE\n");
+
if (set_default_channel_config(ac->avctx, layout_map, &layout_map_tags,
2) < 0)
return NULL;
@@ -469,6 +495,8 @@ static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
int layout_map_tags;
push_output_configuration(ac);
+ av_log(ac->avctx, AV_LOG_DEBUG, "stereo with SCE\n");
+
if (set_default_channel_config(ac->avctx, layout_map, &layout_map_tags,
1) < 0)
return NULL;
@@ -547,6 +575,8 @@ static void decode_channel_map(uint8_t layout_map[][3],
case AAC_CHANNEL_LFE:
syn_ele = TYPE_LFE;
break;
+ default:
+ av_assert0(0);
}
layout_map[0][0] = syn_ele;
layout_map[0][1] = get_bits(gb, 4);
@@ -589,6 +619,10 @@ static int decode_pce(AVCodecContext *avctx, MPEG4AudioConfig *m4ac,
if (get_bits1(gb))
skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround
+ if (get_bits_left(gb) < 4 * (num_front + num_side + num_back + num_lfe + num_assoc_data + num_cc)) {
+ av_log(avctx, AV_LOG_ERROR, "decode_pce: " overread_err);
+ return -1;
+ }
decode_channel_map(layout_map , AAC_CHANNEL_FRONT, gb, num_front);
tags = num_front;
decode_channel_map(layout_map + tags, AAC_CHANNEL_SIDE, gb, num_side);
@@ -608,7 +642,7 @@ static int decode_pce(AVCodecContext *avctx, MPEG4AudioConfig *m4ac,
/* comment field, first byte is length */
comment_len = get_bits(gb, 8) * 8;
if (get_bits_left(gb) < comment_len) {
- av_log(avctx, AV_LOG_ERROR, overread_err);
+ av_log(avctx, AV_LOG_ERROR, "decode_pce: " overread_err);
return -1;
}
skip_bits_long(gb, comment_len);
@@ -706,9 +740,9 @@ static int decode_audio_specific_config(AACContext *ac,
GetBitContext gb;
int i;
- av_dlog(avctx, "extradata size %d\n", avctx->extradata_size);
- for (i = 0; i < avctx->extradata_size; i++)
- av_dlog(avctx, "%02x ", avctx->extradata[i]);
+ av_dlog(avctx, "audio specific config size %d\n", bit_size >> 3);
+ for (i = 0; i < bit_size >> 3; i++)
+ av_dlog(avctx, "%02x ", data[i]);
av_dlog(avctx, "\n");
init_get_bits(&gb, data, bit_size);
@@ -808,6 +842,14 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
ac->avctx = avctx;
ac->oc[1].m4ac.sample_rate = avctx->sample_rate;
+ if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
+ avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
+ output_scale_factor = 1.0 / 32768.0;
+ } else {
+ avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+ output_scale_factor = 1.0;
+ }
+
if (avctx->extradata_size > 0) {
if (decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
avctx->extradata,
@@ -843,14 +885,6 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
}
}
- if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
- avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
- output_scale_factor = 1.0 / 32768.0;
- } else {
- avctx->sample_fmt = AV_SAMPLE_FMT_S16;
- output_scale_factor = 1.0;
- }
-
AAC_INIT_VLC_STATIC( 0, 304);
AAC_INIT_VLC_STATIC( 1, 270);
AAC_INIT_VLC_STATIC( 2, 550);
@@ -908,7 +942,7 @@ static int skip_data_stream_element(AACContext *ac, GetBitContext *gb)
align_get_bits(gb);
if (get_bits_left(gb) < 8 * count) {
- av_log(ac->avctx, AV_LOG_ERROR, overread_err);
+ av_log(ac->avctx, AV_LOG_ERROR, "skip_data_stream_element: "overread_err);
return -1;
}
skip_bits_long(gb, 8 * count);
@@ -989,11 +1023,11 @@ static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
if (ics->predictor_present) {
if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN) {
if (decode_prediction(ac, ics, gb)) {
- return AVERROR_INVALIDDATA;
+ goto fail;
}
} else if (ac->oc[1].m4ac.object_type == AOT_AAC_LC) {
av_log(ac->avctx, AV_LOG_ERROR, "Prediction is not allowed in AAC-LC.\n");
- return AVERROR_INVALIDDATA;
+ goto fail;
} else {
if ((ics->ltp.present = get_bits(gb, 1)))
decode_ltp(ac, &ics->ltp, gb, ics->max_sfb);
@@ -1005,10 +1039,13 @@ static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
av_log(ac->avctx, AV_LOG_ERROR,
"Number of scalefactor bands in group (%d) exceeds limit (%d).\n",
ics->max_sfb, ics->num_swb);
- return AVERROR_INVALIDDATA;
+ goto fail;
}
return 0;
+fail:
+ ics->max_sfb = 0;
+ return AVERROR_INVALIDDATA;
}
/**
@@ -1039,7 +1076,7 @@ static int decode_band_types(AACContext *ac, enum BandType band_type[120],
sect_len_incr = get_bits(gb, bits);
sect_end += sect_len_incr;
if (get_bits_left(gb) < 0) {
- av_log(ac->avctx, AV_LOG_ERROR, overread_err);
+ av_log(ac->avctx, AV_LOG_ERROR, "decode_band_types: "overread_err);
return -1;
}
if (sect_end > ics->max_sfb) {
@@ -2322,9 +2359,11 @@ static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
size = avpriv_aac_parse_header(gb, &hdr_info);
if (size > 0) {
- if (hdr_info.num_aac_frames != 1) {
+ if (!ac->warned_num_aac_frames && hdr_info.num_aac_frames != 1) {
+ // This is 2 for "VLB " audio in NSV files.
+ // See samples/nsv/vlb_audio.
av_log_missing_feature(ac->avctx, "More than one AAC RDB per ADTS frame is", 0);
- return -1;
+ ac->warned_num_aac_frames = 1;
}
push_output_configuration(ac);
if (hdr_info.chan_config) {
@@ -2431,6 +2470,8 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
pop_output_configuration(ac);
} else {
err = output_configure(ac, layout_map, tags, 0, OC_TRIAL_PCE);
+ if (!err)
+ ac->oc[1].m4ac.chan_config = 0;
pce_found = 1;
}
break;
@@ -2440,7 +2481,7 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
if (elem_id == 15)
elem_id += get_bits(gb, 8) - 1;
if (get_bits_left(gb) < 8 * elem_id) {
- av_log(avctx, AV_LOG_ERROR, overread_err);
+ av_log(avctx, AV_LOG_ERROR, "TYPE_FIL: "overread_err);
err = -1;
goto fail;
}
@@ -2521,7 +2562,7 @@ static int aac_decode_frame(AVCodecContext *avctx, void *data,
AV_PKT_DATA_NEW_EXTRADATA,
&new_extradata_size);
- if (new_extradata) {
+ if (new_extradata && 0) {
av_free(avctx->extradata);
avctx->extradata = av_mallocz(new_extradata_size +
FF_INPUT_BUFFER_PADDING_SIZE);
@@ -2575,7 +2616,7 @@ static av_cold int aac_decode_close(AVCodecContext *avctx)
struct LATMContext {
AACContext aac_ctx; ///< containing AACContext
- int initialized; ///< initilized after a valid extradata was seen
+ int initialized; ///< initialized after a valid extradata was seen
// parser data
int audio_mux_version_A; ///< LATM syntax version
@@ -2620,10 +2661,15 @@ static int latm_decode_audio_specific_config(struct LATMContext *latmctx,
if (bits_consumed < 0)
return AVERROR_INVALIDDATA;
- if (ac->oc[1].m4ac.sample_rate != m4ac.sample_rate ||
+ if (!latmctx->initialized ||
+ ac->oc[1].m4ac.sample_rate != m4ac.sample_rate ||
ac->oc[1].m4ac.chan_config != m4ac.chan_config) {
- av_log(avctx, AV_LOG_INFO, "audio config changed\n");
+ if(latmctx->initialized) {
+ av_log(avctx, AV_LOG_INFO, "audio config changed\n");
+ } else {
+ av_log(avctx, AV_LOG_INFO, "initializing latmctx\n");
+ }
latmctx->initialized = 0;
esize = (bits_consumed+7) / 8;
@@ -2852,6 +2898,7 @@ AVCodec ff_aac_decoder = {
},
.capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
.channel_layouts = aac_channel_layout,
+ .flush = flush,
};
/*
@@ -2873,4 +2920,5 @@ AVCodec ff_aac_latm_decoder = {
},
.capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
.channel_layouts = aac_channel_layout,
+ .flush = flush,
};
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-12 20:28:25 +0000