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-rw-r--r--libavcodec/aacdec.c269
1 files changed, 221 insertions, 48 deletions
diff --git a/libavcodec/aacdec.c b/libavcodec/aacdec.c
index 3219ec6185..f60060adb5 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
*/
@@ -80,6 +80,7 @@
*/
#include "libavutil/float_dsp.h"
+#include "libavutil/opt.h"
#include "avcodec.h"
#include "internal.h"
#include "get_bits.h"
@@ -106,12 +107,18 @@
#if ARCH_ARM
# include "arm/aac.h"
+#elif ARCH_MIPS
+# include "mips/aacdec_mips.h"
#endif
static VLC vlc_scalefactors;
static VLC vlc_spectral[11];
-static const char overread_err[] = "Input buffer exhausted before END element found\n";
+static int output_configure(AACContext *ac,
+ uint8_t layout_map[MAX_ELEM_ID*4][3], int tags,
+ enum OCStatus oc_type, int get_new_frame);
+
+#define overread_err "Input buffer exhausted before END element found\n"
static int count_channels(uint8_t (*layout)[3], int tags)
{
@@ -128,7 +135,7 @@ 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
@@ -150,6 +157,10 @@ 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_element[(*channels)++] = &ac->che[type][id]->ch[0];
if (type == TYPE_CPE ||
(type == TYPE_SCE && ac->oc[1].m4ac.ps == 1)) {
@@ -183,10 +194,8 @@ static int frame_configure_elements(AVCodecContext *avctx)
/* get output buffer */
av_frame_unref(ac->frame);
ac->frame->nb_samples = 2048;
- if ((ret = ff_get_buffer(avctx, ac->frame, 0)) < 0) {
- av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+ if ((ret = ff_get_buffer(avctx, ac->frame, 0)) < 0)
return ret;
- }
/* map output channel pointers to AVFrame data */
for (ch = 0; ch < avctx->channels; ch++) {
@@ -403,6 +412,8 @@ static void pop_output_configuration(AACContext *ac) {
ac->oc[1] = ac->oc[0];
ac->avctx->channels = ac->oc[1].channels;
ac->avctx->channel_layout = ac->oc[1].channel_layout;
+ output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags,
+ ac->oc[1].status, 0);
}
}
@@ -447,7 +458,8 @@ 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;
@@ -459,6 +471,23 @@ static int output_configure(AACContext *ac,
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.
@@ -492,6 +521,8 @@ 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;
@@ -508,6 +539,8 @@ 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;
@@ -586,6 +619,8 @@ 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);
@@ -628,6 +663,10 @@ 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);
@@ -647,7 +686,7 @@ 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);
@@ -743,13 +782,15 @@ static int decode_audio_specific_config(AACContext *ac,
{
GetBitContext gb;
int i;
+ int ret;
- 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);
+ if ((ret = init_get_bits(&gb, data, bit_size)) < 0)
+ return ret;
if ((i = avpriv_mpeg4audio_get_config(m4ac, data, bit_size, sync_extension)) < 0)
return -1;
@@ -787,7 +828,7 @@ 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)
{
union { unsigned u; int s; } v = { previous_val * 1664525u + 1013904223 };
return v.s;
@@ -839,6 +880,8 @@ static void reset_predictor_group(PredictorState *ps, int group_num)
ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), sizeof(ff_aac_spectral_codes[num][0]), \
size);
+static void aacdec_init(AACContext *ac);
+
static av_cold int aac_decode_init(AVCodecContext *avctx)
{
AACContext *ac = avctx->priv_data;
@@ -846,6 +889,8 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
ac->avctx = avctx;
ac->oc[1].m4ac.sample_rate = avctx->sample_rate;
+ aacdec_init(ac);
+
avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
if (avctx->extradata_size > 0) {
@@ -883,6 +928,11 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
}
}
+ if (avctx->channels > MAX_CHANNELS) {
+ av_log(avctx, AV_LOG_ERROR, "Too many channels\n");
+ return AVERROR_INVALIDDATA;
+ }
+
AAC_INIT_VLC_STATIC( 0, 304);
AAC_INIT_VLC_STATIC( 1, 270);
AAC_INIT_VLC_STATIC( 2, 550);
@@ -936,7 +986,7 @@ 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);
@@ -1017,11 +1067,11 @@ 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(&ics->ltp, gb, ics->max_sfb);
@@ -1033,10 +1083,13 @@ 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;
}
/**
@@ -1067,7 +1120,7 @@ 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) {
@@ -1235,7 +1288,7 @@ 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);
}
}
@@ -1938,6 +1991,32 @@ 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.
*
@@ -1979,6 +2058,8 @@ static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt,
res = decode_dynamic_range(&ac->che_drc, gb);
break;
case EXT_FILL:
+ decode_fill(ac, gb, 8 * cnt - 4);
+ break;
case EXT_FILL_DATA:
case EXT_DATA_ELEMENT:
default:
@@ -2001,7 +2082,7 @@ static void apply_tns(float coef[1024], TemporalNoiseShaping *tns,
int w, filt, m, i;
int bottom, top, order, start, end, size, inc;
float lpc[TNS_MAX_ORDER];
- float tmp[TNS_MAX_ORDER + 1];
+ float tmp[TNS_MAX_ORDER+1];
for (w = 0; w < ics->num_windows; w++) {
bottom = ics->num_swb;
@@ -2093,10 +2174,10 @@ static void apply_ltp(AACContext *ac, SingleChannelElement *sce)
predTime[i] = sce->ltp_state[i + 2048 - ltp->lag] * ltp->coef;
memset(&predTime[i], 0, (2048 - i) * sizeof(float));
- windowing_and_mdct_ltp(ac, predFreq, predTime, &sce->ics);
+ ac->windowing_and_mdct_ltp(ac, predFreq, predTime, &sce->ics);
if (sce->tns.present)
- apply_tns(predFreq, &sce->tns, &sce->ics, 0);
+ ac->apply_tns(predFreq, &sce->tns, &sce->ics, 0);
for (sfb = 0; sfb < FFMIN(sce->ics.max_sfb, MAX_LTP_LONG_SFB); sfb++)
if (ltp->used[sfb])
@@ -2308,25 +2389,25 @@ static void spectral_to_sample(AACContext *ac)
if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
if (che->ch[0].ics.predictor_present) {
if (che->ch[0].ics.ltp.present)
- apply_ltp(ac, &che->ch[0]);
+ ac->apply_ltp(ac, &che->ch[0]);
if (che->ch[1].ics.ltp.present && type == TYPE_CPE)
- apply_ltp(ac, &che->ch[1]);
+ ac->apply_ltp(ac, &che->ch[1]);
}
}
if (che->ch[0].tns.present)
- apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1);
+ ac->apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1);
if (che->ch[1].tns.present)
- apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1);
+ ac->apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1);
if (type <= TYPE_CPE)
apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, apply_dependent_coupling);
if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) {
- imdct_and_windowing(ac, &che->ch[0]);
+ ac->imdct_and_windowing(ac, &che->ch[0]);
if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
- update_ltp(ac, &che->ch[0]);
+ ac->update_ltp(ac, &che->ch[0]);
if (type == TYPE_CPE) {
- imdct_and_windowing(ac, &che->ch[1]);
+ ac->imdct_and_windowing(ac, &che->ch[1]);
if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
- update_ltp(ac, &che->ch[1]);
+ ac->update_ltp(ac, &che->ch[1]);
}
if (ac->oc[1].m4ac.sbr > 0) {
ff_sbr_apply(ac, &che->sbr, type, che->ch[0].ret, che->ch[1].ret);
@@ -2348,10 +2429,12 @@ 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.
avpriv_report_missing_feature(ac->avctx,
"More than one AAC RDB per ADTS frame");
- return AVERROR_PATCHWELCOME;
+ ac->warned_num_aac_frames = 1;
}
push_output_configuration(ac);
if (hdr_info.chan_config) {
@@ -2364,6 +2447,21 @@ 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->dmono_mode && 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,
+ OC_TRIAL_FRAME, 0))
+ return -7;
+ }
}
ac->oc[1].m4ac.sample_rate = hdr_info.sample_rate;
ac->oc[1].m4ac.sampling_index = hdr_info.sampling_index;
@@ -2381,13 +2479,14 @@ 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;
ac->frame = data;
@@ -2429,6 +2528,7 @@ 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:
@@ -2461,9 +2561,10 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
if (pce_found) {
av_log(avctx, AV_LOG_ERROR,
"Not evaluating a further program_config_element as this construct is dubious at best.\n");
- pop_output_configuration(ac);
} else {
err = output_configure(ac, layout_map, tags, OC_TRIAL_PCE, 1);
+ if (!err)
+ ac->oc[1].m4ac.chan_config = 0;
pce_found = 1;
}
break;
@@ -2473,7 +2574,7 @@ 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;
}
@@ -2504,17 +2605,33 @@ 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->dmono_mode && sce_count == 2 &&
+ ac->oc[1].channel_layout == (AV_CH_FRONT_LEFT | AV_CH_FRONT_RIGHT);
if (samples)
ac->frame->nb_samples = samples;
*got_frame_ptr = !!samples;
+ if (is_dmono) {
+ if (ac->dmono_mode == 1)
+ ((AVFrame *)data)->data[1] =((AVFrame *)data)->data[0];
+ else if (ac->dmono_mode == 2)
+ ((AVFrame *)data)->data[0] =((AVFrame *)data)->data[1];
+ }
+
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;
+ const uint8_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);
@@ -2535,8 +2652,12 @@ 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);
@@ -2553,9 +2674,18 @@ static int aac_decode_frame(AVCodecContext *avctx, void *data,
}
}
+ ac->dmono_mode = 0;
+ if (jp_dualmono && jp_dualmono_size > 0)
+ ac->dmono_mode = 1 + *jp_dualmono;
+ if (ac->force_dmono_mode >= 0)
+ ac->dmono_mode = ac->force_dmono_mode;
+
+ if (INT_MAX / 8 <= buf_size)
+ return AVERROR_INVALIDDATA;
+
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;
@@ -2590,7 +2720,7 @@ 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
@@ -2635,10 +2765,15 @@ 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_DEBUG, "initializing latmctx\n");
+ }
latmctx->initialized = 0;
esize = (bits_consumed+7) / 8;
@@ -2681,9 +2816,9 @@ 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
avpriv_request_sample(latmctx->aac_ctx.avctx, "Multiple layers");
return AVERROR_PATCHWELCOME;
@@ -2796,14 +2931,15 @@ static int latm_decode_frame(AVCodecContext *avctx, void *out,
int muxlength, err;
GetBitContext gb;
- init_get_bits(&gb, avpkt->data, avpkt->size * 8);
+ if ((err = init_get_bits8(&gb, avpkt->data, avpkt->size)) < 0)
+ return err;
// check for LOAS sync word
if (get_bits(&gb, 11) != LOAS_SYNC_WORD)
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;
@@ -2833,7 +2969,7 @@ 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;
@@ -2850,6 +2986,40 @@ static av_cold int latm_decode_init(AVCodecContext *avctx)
return ret;
}
+static void aacdec_init(AACContext *c)
+{
+ c->imdct_and_windowing = imdct_and_windowing;
+ c->apply_ltp = apply_ltp;
+ c->apply_tns = apply_tns;
+ c->windowing_and_mdct_ltp = windowing_and_mdct_ltp;
+ c->update_ltp = update_ltp;
+
+ if(ARCH_MIPS)
+ ff_aacdec_init_mips(c);
+}
+/**
+ * AVOptions for Japanese DTV specific extensions (ADTS only)
+ */
+#define AACDEC_FLAGS AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
+static const AVOption options[] = {
+ {"dual_mono_mode", "Select the channel to decode for dual mono",
+ offsetof(AACContext, force_dmono_mode), AV_OPT_TYPE_INT, {.i64=-1}, -1, 2,
+ AACDEC_FLAGS, "dual_mono_mode"},
+
+ {"auto", "autoselection", 0, AV_OPT_TYPE_CONST, {.i64=-1}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
+ {"main", "Select Main/Left channel", 0, AV_OPT_TYPE_CONST, {.i64= 1}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
+ {"sub" , "Select Sub/Right channel", 0, AV_OPT_TYPE_CONST, {.i64= 2}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
+ {"both", "Select both channels", 0, AV_OPT_TYPE_CONST, {.i64= 0}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
+
+ {NULL},
+};
+
+static const AVClass aac_decoder_class = {
+ .class_name = "AAC decoder",
+ .item_name = av_default_item_name,
+ .option = options,
+ .version = LIBAVUTIL_VERSION_INT,
+};
AVCodec ff_aac_decoder = {
.name = "aac",
@@ -2865,6 +3035,8 @@ AVCodec ff_aac_decoder = {
},
.capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
.channel_layouts = aac_channel_layout,
+ .flush = flush,
+ .priv_class = &aac_decoder_class,
};
/*
@@ -2886,4 +3058,5 @@ 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 2024-07-09 19:22:08 +0000