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-rw-r--r--libavcodec/aacdec.c206
1 files changed, 167 insertions, 39 deletions
diff --git a/libavcodec/aacdec.c b/libavcodec/aacdec.c
index b7caecb39b..44869c5e7d 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)) {
@@ -413,13 +417,31 @@ 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 +475,8 @@ 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 +493,8 @@ 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 +573,8 @@ 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 +617,10 @@ 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 +640,7 @@ 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 +738,9 @@ 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);
@@ -749,7 +781,7 @@ static int decode_audio_specific_config(AACContext *ac,
*
* @return Returns a 32-bit pseudorandom integer
*/
-static av_always_inline int lcg_random(int previous_val)
+static av_always_inline int lcg_random(unsigned previous_val)
{
return previous_val * 1664525 + 1013904223;
}
@@ -808,6 +840,14 @@ 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 +883,6 @@ 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 +940,7 @@ 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 +1021,11 @@ 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 +1037,13 @@ 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 +1074,7 @@ 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) {
@@ -1207,7 +1242,7 @@ static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb,
for (idx = 0; idx < cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb; idx++)
cpe->ms_mask[idx] = get_bits1(gb);
} else if (ms_present == 2) {
- memset(cpe->ms_mask, 1, cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb * sizeof(cpe->ms_mask[0]));
+ memset(cpe->ms_mask, 1, sizeof(cpe->ms_mask[0]) * cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb);
}
}
@@ -1912,6 +1947,32 @@ static int decode_dynamic_range(DynamicRangeControl *che_drc,
return n;
}
+static int decode_fill(AACContext *ac, GetBitContext *gb, int len) {
+ uint8_t buf[256];
+ int i, major, minor;
+
+ if (len < 13+7*8)
+ goto unknown;
+
+ get_bits(gb, 13); len -= 13;
+
+ for(i=0; i+1<sizeof(buf) && len>=8; i++, len-=8)
+ buf[i] = get_bits(gb, 8);
+
+ buf[i] = 0;
+ if (ac->avctx->debug & FF_DEBUG_PICT_INFO)
+ av_log(ac->avctx, AV_LOG_DEBUG, "FILL:%s\n", buf);
+
+ if (sscanf(buf, "libfaac %d.%d", &major, &minor) == 2){
+ ac->avctx->internal->skip_samples = 1024;
+ }
+
+unknown:
+ skip_bits_long(gb, len);
+
+ return 0;
+}
+
/**
* Decode extension data (incomplete); reference: table 4.51.
*
@@ -1953,6 +2014,8 @@ static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt,
res = decode_dynamic_range(&ac->che_drc, gb, cnt);
break;
case EXT_FILL:
+ decode_fill(ac, gb, 8 * cnt - 4);
+ break;
case EXT_FILL_DATA:
case EXT_DATA_ELEMENT:
default:
@@ -2322,9 +2385,11 @@ 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", 0);
- return AVERROR_PATCHWELCOME;
+ ac->warned_num_aac_frames = 1;
}
push_output_configuration(ac);
if (hdr_info.chan_config) {
@@ -2338,6 +2403,21 @@ static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
return -7;
} else {
ac->oc[1].m4ac.chan_config = 0;
+ /**
+ * dual mono frames in Japanese DTV can have chan_config 0
+ * WITHOUT specifying PCE.
+ * thus, set dual mono as default.
+ */
+ if (ac->enable_jp_dmono && ac->oc[0].status == OC_NONE) {
+ layout_map_tags = 2;
+ layout_map[0][0] = layout_map[1][0] = TYPE_SCE;
+ layout_map[0][2] = layout_map[1][2] = AAC_CHANNEL_FRONT;
+ layout_map[0][1] = 0;
+ layout_map[1][1] = 1;
+ if (output_configure(ac, layout_map, layout_map_tags,
+ 0, OC_TRIAL_FRAME))
+ return -7;
+ }
}
ac->oc[1].m4ac.sample_rate = hdr_info.sample_rate;
ac->oc[1].m4ac.sampling_index = hdr_info.sampling_index;
@@ -2355,13 +2435,15 @@ static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
}
static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
- int *got_frame_ptr, GetBitContext *gb)
+ int *got_frame_ptr, GetBitContext *gb, AVPacket *avpkt)
{
AACContext *ac = avctx->priv_data;
ChannelElement *che = NULL, *che_prev = NULL;
enum RawDataBlockType elem_type, elem_type_prev = TYPE_END;
int err, elem_id;
int samples = 0, multiplier, audio_found = 0, pce_found = 0;
+ int is_dmono, sce_count = 0;
+ float *tmp = NULL;
if (show_bits(gb, 12) == 0xfff) {
if (parse_adts_frame_header(ac, gb) < 0) {
@@ -2396,6 +2478,7 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
case TYPE_SCE:
err = decode_ics(ac, &che->ch[0], gb, 0, 0);
audio_found = 1;
+ sce_count++;
break;
case TYPE_CPE:
@@ -2431,6 +2514,8 @@ 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 +2525,7 @@ 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;
}
@@ -2472,6 +2557,20 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
multiplier = (ac->oc[1].m4ac.sbr == 1) ? ac->oc[1].m4ac.ext_sample_rate > ac->oc[1].m4ac.sample_rate : 0;
samples <<= multiplier;
+ /* for dual-mono audio (SCE + SCE) */
+ is_dmono = ac->enable_jp_dmono && sce_count == 2 &&
+ ac->oc[1].channel_layout == (AV_CH_FRONT_LEFT | AV_CH_FRONT_RIGHT);
+
+ if (is_dmono) {
+ if (ac->dmono_mode == 0) {
+ tmp = ac->output_data[1];
+ ac->output_data[1] = ac->output_data[0];
+ } else if (ac->dmono_mode == 1) {
+ tmp = ac->output_data[0];
+ ac->output_data[0] = ac->output_data[1];
+ }
+ }
+
if (samples) {
/* get output buffer */
ac->frame.nb_samples = samples;
@@ -2494,12 +2593,25 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
}
*got_frame_ptr = !!samples;
+ if (is_dmono) {
+ if (ac->dmono_mode == 0)
+ ac->output_data[1] = tmp;
+ else if (ac->dmono_mode == 1)
+ ac->output_data[0] = tmp;
+ }
+
if (ac->oc[1].status && audio_found) {
avctx->sample_rate = ac->oc[1].m4ac.sample_rate << multiplier;
avctx->frame_size = samples;
ac->oc[1].status = OC_LOCKED;
}
+ if (multiplier) {
+ int side_size;
+ uint32_t *side = av_packet_get_side_data(avpkt, AV_PKT_DATA_SKIP_SAMPLES, &side_size);
+ if (side && side_size>=4)
+ AV_WL32(side, 2*AV_RL32(side));
+ }
return 0;
fail:
pop_output_configuration(ac);
@@ -2520,8 +2632,12 @@ static int aac_decode_frame(AVCodecContext *avctx, void *data,
const uint8_t *new_extradata = av_packet_get_side_data(avpkt,
AV_PKT_DATA_NEW_EXTRADATA,
&new_extradata_size);
+ int jp_dualmono_size;
+ const uint8_t *jp_dualmono = av_packet_get_side_data(avpkt,
+ AV_PKT_DATA_JP_DUALMONO,
+ &jp_dualmono_size);
- if (new_extradata) {
+ if (new_extradata && 0) {
av_free(avctx->extradata);
avctx->extradata = av_mallocz(new_extradata_size +
FF_INPUT_BUFFER_PADDING_SIZE);
@@ -2538,9 +2654,14 @@ static int aac_decode_frame(AVCodecContext *avctx, void *data,
}
}
+ ac->enable_jp_dmono = !!jp_dualmono;
+ ac->dmono_mode = 0;
+ if (jp_dualmono && jp_dualmono_size > 0)
+ ac->dmono_mode = *jp_dualmono;
+
init_get_bits(&gb, buf, buf_size * 8);
- if ((err = aac_decode_frame_int(avctx, data, got_frame_ptr, &gb)) < 0)
+ if ((err = aac_decode_frame_int(avctx, data, got_frame_ptr, &gb, avpkt)) < 0)
return err;
buf_consumed = (get_bits_count(&gb) + 7) >> 3;
@@ -2575,7 +2696,7 @@ 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 +2741,15 @@ 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;
@@ -2667,9 +2793,9 @@ static int read_stream_mux_config(struct LATMContext *latmctx,
return AVERROR_PATCHWELCOME;
}
- // for each program (which there is only on in DVB)
+ // for each program (which there is only one in DVB)
- // for each layer (which there is only on in DVB)
+ // for each layer (which there is only one in DVB)
if (get_bits(gb, 3)) { // numLayer
av_log_missing_feature(latmctx->aac_ctx.avctx,
"Multiple layers", 1);
@@ -2790,7 +2916,7 @@ static int latm_decode_frame(AVCodecContext *avctx, void *out,
return AVERROR_INVALIDDATA;
muxlength = get_bits(&gb, 13) + 3;
- // not enough data, the parser should have sorted this
+ // not enough data, the parser should have sorted this out
if (muxlength > avpkt->size)
return AVERROR_INVALIDDATA;
@@ -2820,7 +2946,7 @@ static int latm_decode_frame(AVCodecContext *avctx, void *out,
return AVERROR_INVALIDDATA;
}
- if ((err = aac_decode_frame_int(avctx, out, got_frame_ptr, &gb)) < 0)
+ if ((err = aac_decode_frame_int(avctx, out, got_frame_ptr, &gb, avpkt)) < 0)
return err;
return muxlength;
@@ -2852,6 +2978,7 @@ AVCodec ff_aac_decoder = {
},
.capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
.channel_layouts = aac_channel_layout,
+ .flush = flush,
};
/*
@@ -2873,4 +3000,5 @@ 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 2024-07-10 03:07:25 +0000