*
* AAC LATM decoder
* Copyright (c) 2008-2010 Paul Kendall <paul@kcbbs.gen.nz>
- * Copyright (c) 2010 Janne Grunau <janne-ffmpeg@jannau.net>
+ * Copyright (c) 2010 Janne Grunau <janne-libav@jannau.net>
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav 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,
+ * Libav 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
+ * License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
* Y filterbank - standard
* N (code in SoC repo) filterbank - Scalable Sample Rate
* Y Temporal Noise Shaping
- * N (code in SoC repo) Long Term Prediction
+ * Y Long Term Prediction
* Y intensity stereo
* Y channel coupling
* Y frequency domain prediction
Parametric Stereo.
*/
-
+#include "libavutil/float_dsp.h"
#include "avcodec.h"
#include "internal.h"
#include "get_bits.h"
#include "dsputil.h"
#include "fft.h"
+#include "fmtconvert.h"
#include "lpc.h"
+#include "kbdwin.h"
+#include "sinewin.h"
#include "aac.h"
#include "aactab.h"
#include "aacsbr.h"
#include "mpeg4audio.h"
#include "aacadtsdec.h"
+#include "libavutil/intfloat.h"
#include <assert.h>
#include <errno.h>
# include "arm/aac.h"
#endif
-union float754 {
- float f;
- uint32_t i;
-};
-
static VLC vlc_scalefactors;
static VLC vlc_spectral[11];
static const char overread_err[] = "Input buffer exhausted before END element found\n";
-static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
+static int count_channels(uint8_t (*layout)[3], int tags)
{
- // For PCE based channel configurations map the channels solely based on tags.
- if (!ac->m4ac.chan_config) {
- return ac->tag_che_map[type][elem_id];
- }
- // For indexed channel configurations map the channels solely based on position.
- switch (ac->m4ac.chan_config) {
- case 7:
- if (ac->tags_mapped == 3 && type == TYPE_CPE) {
- ac->tags_mapped++;
- return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][2];
- }
- case 6:
- /* Some streams incorrectly code 5.1 audio as SCE[0] CPE[0] CPE[1] SCE[1]
- instead of SCE[0] CPE[0] CPE[1] LFE[0]. If we seem to have
- encountered such a stream, transfer the LFE[0] element to the SCE[1]'s mapping */
- if (ac->tags_mapped == tags_per_config[ac->m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
- ac->tags_mapped++;
- return ac->tag_che_map[type][elem_id] = ac->che[TYPE_LFE][0];
- }
- case 5:
- if (ac->tags_mapped == 2 && type == TYPE_CPE) {
- ac->tags_mapped++;
- return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][1];
- }
- case 4:
- if (ac->tags_mapped == 2 && ac->m4ac.chan_config == 4 && type == TYPE_SCE) {
- ac->tags_mapped++;
- return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
- }
- case 3:
- case 2:
- if (ac->tags_mapped == (ac->m4ac.chan_config != 2) && type == TYPE_CPE) {
- ac->tags_mapped++;
- return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0];
- } else if (ac->m4ac.chan_config == 2) {
- return NULL;
- }
- case 1:
- if (!ac->tags_mapped && type == TYPE_SCE) {
- ac->tags_mapped++;
- return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][0];
- }
- default:
- return NULL;
+ int i, sum = 0;
+ for (i = 0; i < tags; i++) {
+ int syn_ele = layout[i][0];
+ int pos = layout[i][2];
+ sum += (1 + (syn_ele == TYPE_CPE)) *
+ (pos != AAC_CHANNEL_OFF && pos != AAC_CHANNEL_CC);
}
+ return sum;
}
/**
* 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 FFmpeg channel layout.
+ * channel order to match the internal Libav channel layout.
*
* @param che_pos current channel position configuration
* @param type channel element type
* @return Returns error status. 0 - OK, !0 - error
*/
static av_cold int che_configure(AACContext *ac,
- enum ChannelPosition che_pos[4][MAX_ELEM_ID],
- int type, int id,
- int *channels)
+ enum ChannelPosition che_pos,
+ int type, int id, int *channels)
{
- if (che_pos[type][id]) {
- if (!ac->che[type][id] && !(ac->che[type][id] = av_mallocz(sizeof(ChannelElement))))
- return AVERROR(ENOMEM);
- ff_aac_sbr_ctx_init(&ac->che[type][id]->sbr);
+ if (che_pos) {
+ if (!ac->che[type][id]) {
+ if (!(ac->che[type][id] = av_mallocz(sizeof(ChannelElement))))
+ return AVERROR(ENOMEM);
+ ff_aac_sbr_ctx_init(ac, &ac->che[type][id]->sbr);
+ }
if (type != TYPE_CCE) {
ac->output_data[(*channels)++] = ac->che[type][id]->ch[0].ret;
if (type == TYPE_CPE ||
- (type == TYPE_SCE && ac->m4ac.ps == 1)) {
+ (type == TYPE_SCE && ac->oc[1].m4ac.ps == 1)) {
ac->output_data[(*channels)++] = ac->che[type][id]->ch[1].ret;
}
}
return 0;
}
+struct elem_to_channel {
+ uint64_t av_position;
+ uint8_t syn_ele;
+ uint8_t elem_id;
+ uint8_t aac_position;
+};
+
+static int assign_pair(struct elem_to_channel e2c_vec[MAX_ELEM_ID],
+ uint8_t (*layout_map)[3], int offset, uint64_t left,
+ uint64_t right, int pos)
+{
+ if (layout_map[offset][0] == TYPE_CPE) {
+ e2c_vec[offset] = (struct elem_to_channel) {
+ .av_position = left | right, .syn_ele = TYPE_CPE,
+ .elem_id = layout_map[offset ][1], .aac_position = pos };
+ return 1;
+ } else {
+ e2c_vec[offset] = (struct elem_to_channel) {
+ .av_position = left, .syn_ele = TYPE_SCE,
+ .elem_id = layout_map[offset ][1], .aac_position = pos };
+ e2c_vec[offset + 1] = (struct elem_to_channel) {
+ .av_position = right, .syn_ele = TYPE_SCE,
+ .elem_id = layout_map[offset + 1][1], .aac_position = pos };
+ return 2;
+ }
+}
+
+static int count_paired_channels(uint8_t (*layout_map)[3], int tags, int pos, int *current) {
+ int num_pos_channels = 0;
+ int first_cpe = 0;
+ int sce_parity = 0;
+ int i;
+ for (i = *current; i < tags; i++) {
+ if (layout_map[i][2] != pos)
+ break;
+ if (layout_map[i][0] == TYPE_CPE) {
+ if (sce_parity) {
+ if (pos == AAC_CHANNEL_FRONT && !first_cpe) {
+ sce_parity = 0;
+ } else {
+ return -1;
+ }
+ }
+ num_pos_channels += 2;
+ first_cpe = 1;
+ } else {
+ num_pos_channels++;
+ sce_parity ^= 1;
+ }
+ }
+ if (sce_parity &&
+ ((pos == AAC_CHANNEL_FRONT && first_cpe) || pos == AAC_CHANNEL_SIDE))
+ return -1;
+ *current = i;
+ return num_pos_channels;
+}
+
+static uint64_t sniff_channel_order(uint8_t (*layout_map)[3], int tags)
+{
+ int i, n, total_non_cc_elements;
+ struct elem_to_channel e2c_vec[4*MAX_ELEM_ID] = {{ 0 }};
+ int num_front_channels, num_side_channels, num_back_channels;
+ uint64_t layout;
+
+ if (FF_ARRAY_ELEMS(e2c_vec) < tags)
+ return 0;
+
+ i = 0;
+ num_front_channels =
+ count_paired_channels(layout_map, tags, AAC_CHANNEL_FRONT, &i);
+ if (num_front_channels < 0)
+ return 0;
+ num_side_channels =
+ count_paired_channels(layout_map, tags, AAC_CHANNEL_SIDE, &i);
+ if (num_side_channels < 0)
+ return 0;
+ num_back_channels =
+ count_paired_channels(layout_map, tags, AAC_CHANNEL_BACK, &i);
+ if (num_back_channels < 0)
+ return 0;
+
+ i = 0;
+ if (num_front_channels & 1) {
+ e2c_vec[i] = (struct elem_to_channel) {
+ .av_position = AV_CH_FRONT_CENTER, .syn_ele = TYPE_SCE,
+ .elem_id = layout_map[i][1], .aac_position = AAC_CHANNEL_FRONT };
+ i++;
+ num_front_channels--;
+ }
+ if (num_front_channels >= 4) {
+ i += assign_pair(e2c_vec, layout_map, i,
+ AV_CH_FRONT_LEFT_OF_CENTER,
+ AV_CH_FRONT_RIGHT_OF_CENTER,
+ AAC_CHANNEL_FRONT);
+ num_front_channels -= 2;
+ }
+ if (num_front_channels >= 2) {
+ i += assign_pair(e2c_vec, layout_map, i,
+ AV_CH_FRONT_LEFT,
+ AV_CH_FRONT_RIGHT,
+ AAC_CHANNEL_FRONT);
+ num_front_channels -= 2;
+ }
+ while (num_front_channels >= 2) {
+ i += assign_pair(e2c_vec, layout_map, i,
+ UINT64_MAX,
+ UINT64_MAX,
+ AAC_CHANNEL_FRONT);
+ num_front_channels -= 2;
+ }
+
+ if (num_side_channels >= 2) {
+ i += assign_pair(e2c_vec, layout_map, i,
+ AV_CH_SIDE_LEFT,
+ AV_CH_SIDE_RIGHT,
+ AAC_CHANNEL_FRONT);
+ num_side_channels -= 2;
+ }
+ while (num_side_channels >= 2) {
+ i += assign_pair(e2c_vec, layout_map, i,
+ UINT64_MAX,
+ UINT64_MAX,
+ AAC_CHANNEL_SIDE);
+ num_side_channels -= 2;
+ }
+
+ while (num_back_channels >= 4) {
+ i += assign_pair(e2c_vec, layout_map, i,
+ UINT64_MAX,
+ UINT64_MAX,
+ AAC_CHANNEL_BACK);
+ num_back_channels -= 2;
+ }
+ if (num_back_channels >= 2) {
+ i += assign_pair(e2c_vec, layout_map, i,
+ AV_CH_BACK_LEFT,
+ AV_CH_BACK_RIGHT,
+ AAC_CHANNEL_BACK);
+ num_back_channels -= 2;
+ }
+ if (num_back_channels) {
+ e2c_vec[i] = (struct elem_to_channel) {
+ .av_position = AV_CH_BACK_CENTER, .syn_ele = TYPE_SCE,
+ .elem_id = layout_map[i][1], .aac_position = AAC_CHANNEL_BACK };
+ i++;
+ num_back_channels--;
+ }
+
+ if (i < tags && layout_map[i][2] == AAC_CHANNEL_LFE) {
+ e2c_vec[i] = (struct elem_to_channel) {
+ .av_position = AV_CH_LOW_FREQUENCY, .syn_ele = TYPE_LFE,
+ .elem_id = layout_map[i][1], .aac_position = AAC_CHANNEL_LFE };
+ i++;
+ }
+ while (i < tags && layout_map[i][2] == AAC_CHANNEL_LFE) {
+ e2c_vec[i] = (struct elem_to_channel) {
+ .av_position = UINT64_MAX, .syn_ele = TYPE_LFE,
+ .elem_id = layout_map[i][1], .aac_position = AAC_CHANNEL_LFE };
+ i++;
+ }
+
+ // Must choose a stable sort
+ total_non_cc_elements = n = i;
+ do {
+ int next_n = 0;
+ for (i = 1; i < n; i++) {
+ if (e2c_vec[i-1].av_position > e2c_vec[i].av_position) {
+ FFSWAP(struct elem_to_channel, e2c_vec[i-1], e2c_vec[i]);
+ next_n = i;
+ }
+ }
+ n = next_n;
+ } while (n > 0);
+
+ layout = 0;
+ for (i = 0; i < total_non_cc_elements; i++) {
+ layout_map[i][0] = e2c_vec[i].syn_ele;
+ layout_map[i][1] = e2c_vec[i].elem_id;
+ layout_map[i][2] = e2c_vec[i].aac_position;
+ if (e2c_vec[i].av_position != UINT64_MAX) {
+ layout |= e2c_vec[i].av_position;
+ }
+ }
+
+ return layout;
+}
+
+/**
+ * Save current output configuration if and only if it has been locked.
+ */
+static void push_output_configuration(AACContext *ac) {
+ if (ac->oc[1].status == OC_LOCKED) {
+ ac->oc[0] = ac->oc[1];
+ }
+ ac->oc[1].status = OC_NONE;
+}
+
+/**
+ * Restore the previous output configuration if and only if the current
+ * configuration is unlocked.
+ */
+static void pop_output_configuration(AACContext *ac) {
+ if (ac->oc[1].status != OC_LOCKED && ac->oc[0].status != OC_NONE) {
+ ac->oc[1] = ac->oc[0];
+ ac->avctx->channels = ac->oc[1].channels;
+ ac->avctx->channel_layout = ac->oc[1].channel_layout;
+ }
+}
+
/**
* Configure output channel order based on the current program configuration element.
*
- * @param che_pos current channel position configuration
- * @param new_che_pos New channel position configuration - we only do something if it differs from the current one.
- *
* @return Returns error status. 0 - OK, !0 - error
*/
-static av_cold int output_configure(AACContext *ac,
- enum ChannelPosition che_pos[4][MAX_ELEM_ID],
- enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
- int channel_config, enum OCStatus oc_type)
+static int output_configure(AACContext *ac,
+ uint8_t layout_map[MAX_ELEM_ID*4][3], int tags,
+ enum OCStatus oc_type)
{
AVCodecContext *avctx = ac->avctx;
- int i, type, channels = 0, ret;
+ int i, channels = 0, ret;
+ uint64_t layout = 0;
- if (new_che_pos != che_pos)
- memcpy(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
+ if (ac->oc[1].layout_map != layout_map) {
+ memcpy(ac->oc[1].layout_map, layout_map, tags * sizeof(layout_map[0]));
+ ac->oc[1].layout_map_tags = tags;
+ }
- if (channel_config) {
- for (i = 0; i < tags_per_config[channel_config]; i++) {
- if ((ret = che_configure(ac, che_pos,
- aac_channel_layout_map[channel_config - 1][i][0],
- aac_channel_layout_map[channel_config - 1][i][1],
- &channels)))
- return ret;
+ // Try to sniff a reasonable channel order, otherwise output the
+ // channels in the order the PCE declared them.
+ if (avctx->request_channel_layout != AV_CH_LAYOUT_NATIVE)
+ layout = sniff_channel_order(layout_map, tags);
+ for (i = 0; i < tags; i++) {
+ int type = layout_map[i][0];
+ int id = layout_map[i][1];
+ int position = layout_map[i][2];
+ // Allocate or free elements depending on if they are in the
+ // current program configuration.
+ ret = che_configure(ac, position, type, id, &channels);
+ if (ret < 0)
+ return ret;
+ }
+ if (ac->oc[1].m4ac.ps == 1 && channels == 2) {
+ if (layout == AV_CH_FRONT_CENTER) {
+ layout = AV_CH_FRONT_LEFT|AV_CH_FRONT_RIGHT;
+ } else {
+ layout = 0;
}
+ }
- memset(ac->tag_che_map, 0, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0]));
-
- avctx->channel_layout = aac_channel_layout[channel_config - 1];
- } else {
- /* Allocate or free elements depending on if they are in the
- * current program configuration.
- *
- * Set up default 1:1 output mapping.
- *
- * For a 5.1 stream the output order will be:
- * [ Center ] [ Front Left ] [ Front Right ] [ LFE ] [ Surround Left ] [ Surround Right ]
- */
-
- for (i = 0; i < MAX_ELEM_ID; i++) {
- for (type = 0; type < 4; type++) {
- if ((ret = che_configure(ac, che_pos, type, i, &channels)))
- return ret;
- }
- }
+ 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;
+ avctx->channels = ac->oc[1].channels = channels;
+ ac->oc[1].status = oc_type;
- memcpy(ac->tag_che_map, ac->che, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0]));
+ return 0;
+}
- avctx->channel_layout = 0;
+/**
+ * Set up channel positions based on a default channel configuration
+ * as specified in table 1.17.
+ *
+ * @return Returns error status. 0 - OK, !0 - error
+ */
+static int set_default_channel_config(AVCodecContext *avctx,
+ uint8_t (*layout_map)[3],
+ int *tags,
+ int channel_config)
+{
+ if (channel_config < 1 || channel_config > 7) {
+ av_log(avctx, AV_LOG_ERROR, "invalid default channel configuration (%d)\n",
+ channel_config);
+ return -1;
}
+ *tags = tags_per_config[channel_config];
+ memcpy(layout_map, aac_channel_layout_map[channel_config-1], *tags * sizeof(*layout_map));
+ return 0;
+}
- avctx->channels = channels;
+static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
+{
+ // For PCE based channel configurations map the channels solely based on tags.
+ if (!ac->oc[1].m4ac.chan_config) {
+ return ac->tag_che_map[type][elem_id];
+ }
+ // Allow single CPE stereo files to be signalled with mono configuration.
+ if (!ac->tags_mapped && type == TYPE_CPE && ac->oc[1].m4ac.chan_config == 1) {
+ uint8_t layout_map[MAX_ELEM_ID*4][3];
+ int layout_map_tags;
+ push_output_configuration(ac);
+
+ if (set_default_channel_config(ac->avctx, layout_map, &layout_map_tags,
+ 2) < 0)
+ return NULL;
+ if (output_configure(ac, layout_map, layout_map_tags,
+ OC_TRIAL_FRAME) < 0)
+ return NULL;
- ac->output_configured = oc_type;
+ ac->oc[1].m4ac.chan_config = 2;
+ ac->oc[1].m4ac.ps = 0;
+ }
+ // And vice-versa
+ if (!ac->tags_mapped && type == TYPE_SCE && ac->oc[1].m4ac.chan_config == 2) {
+ uint8_t layout_map[MAX_ELEM_ID*4][3];
+ int layout_map_tags;
+ push_output_configuration(ac);
+
+ if (set_default_channel_config(ac->avctx, layout_map, &layout_map_tags,
+ 1) < 0)
+ return NULL;
+ if (output_configure(ac, layout_map, layout_map_tags,
+ OC_TRIAL_FRAME) < 0)
+ return NULL;
- return 0;
+ ac->oc[1].m4ac.chan_config = 1;
+ if (ac->oc[1].m4ac.sbr)
+ ac->oc[1].m4ac.ps = -1;
+ }
+ // For indexed channel configurations map the channels solely based on position.
+ switch (ac->oc[1].m4ac.chan_config) {
+ case 7:
+ if (ac->tags_mapped == 3 && type == TYPE_CPE) {
+ ac->tags_mapped++;
+ return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][2];
+ }
+ case 6:
+ /* Some streams incorrectly code 5.1 audio as SCE[0] CPE[0] CPE[1] SCE[1]
+ instead of SCE[0] CPE[0] CPE[1] LFE[0]. If we seem to have
+ encountered such a stream, transfer the LFE[0] element to the SCE[1]'s mapping */
+ if (ac->tags_mapped == tags_per_config[ac->oc[1].m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
+ ac->tags_mapped++;
+ return ac->tag_che_map[type][elem_id] = ac->che[TYPE_LFE][0];
+ }
+ case 5:
+ if (ac->tags_mapped == 2 && type == TYPE_CPE) {
+ ac->tags_mapped++;
+ return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][1];
+ }
+ case 4:
+ if (ac->tags_mapped == 2 && ac->oc[1].m4ac.chan_config == 4 && type == TYPE_SCE) {
+ ac->tags_mapped++;
+ return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
+ }
+ case 3:
+ case 2:
+ if (ac->tags_mapped == (ac->oc[1].m4ac.chan_config != 2) && type == TYPE_CPE) {
+ ac->tags_mapped++;
+ return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0];
+ } else if (ac->oc[1].m4ac.chan_config == 2) {
+ return NULL;
+ }
+ case 1:
+ if (!ac->tags_mapped && type == TYPE_SCE) {
+ ac->tags_mapped++;
+ return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][0];
+ }
+ default:
+ return NULL;
+ }
}
/**
* Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit.
*
- * @param cpe_map Stereo (Channel Pair Element) map, NULL if stereo bit is not present.
- * @param sce_map mono (Single Channel Element) map
* @param type speaker type/position for these channels
*/
-static void decode_channel_map(enum ChannelPosition *cpe_map,
- enum ChannelPosition *sce_map,
+static void decode_channel_map(uint8_t layout_map[][3],
enum ChannelPosition type,
GetBitContext *gb, int n)
{
while (n--) {
- enum ChannelPosition *map = cpe_map && get_bits1(gb) ? cpe_map : sce_map; // stereo or mono map
- map[get_bits(gb, 4)] = type;
+ enum RawDataBlockType syn_ele;
+ switch (type) {
+ case AAC_CHANNEL_FRONT:
+ case AAC_CHANNEL_BACK:
+ case AAC_CHANNEL_SIDE:
+ syn_ele = get_bits1(gb);
+ break;
+ case AAC_CHANNEL_CC:
+ skip_bits1(gb);
+ syn_ele = TYPE_CCE;
+ break;
+ case AAC_CHANNEL_LFE:
+ syn_ele = TYPE_LFE;
+ break;
+ }
+ layout_map[0][0] = syn_ele;
+ layout_map[0][1] = get_bits(gb, 4);
+ layout_map[0][2] = type;
+ layout_map++;
}
}
/**
* Decode program configuration element; reference: table 4.2.
*
- * @param new_che_pos New channel position configuration - we only do something if it differs from the current one.
- *
* @return Returns error status. 0 - OK, !0 - error
*/
static int decode_pce(AVCodecContext *avctx, MPEG4AudioConfig *m4ac,
- enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
+ uint8_t (*layout_map)[3],
GetBitContext *gb)
{
int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc, sampling_index;
int comment_len;
+ int tags;
skip_bits(gb, 2); // object_type
if (get_bits1(gb))
skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround
- decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_FRONT, gb, num_front);
- decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_SIDE, gb, num_side );
- decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_BACK, gb, num_back );
- decode_channel_map(NULL, new_che_pos[TYPE_LFE], AAC_CHANNEL_LFE, gb, num_lfe );
+ 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);
+ tags += num_side;
+ decode_channel_map(layout_map + tags, AAC_CHANNEL_BACK, gb, num_back);
+ tags += num_back;
+ decode_channel_map(layout_map + tags, AAC_CHANNEL_LFE, gb, num_lfe);
+ tags += num_lfe;
skip_bits_long(gb, 4 * num_assoc_data);
- decode_channel_map(new_che_pos[TYPE_CCE], new_che_pos[TYPE_CCE], AAC_CHANNEL_CC, gb, num_cc );
+ decode_channel_map(layout_map + tags, AAC_CHANNEL_CC, gb, num_cc);
+ tags += num_cc;
align_get_bits(gb);
return -1;
}
skip_bits_long(gb, comment_len);
- return 0;
-}
-
-/**
- * Set up channel positions based on a default channel configuration
- * as specified in table 1.17.
- *
- * @param new_che_pos New channel position configuration - we only do something if it differs from the current one.
- *
- * @return Returns error status. 0 - OK, !0 - error
- */
-static av_cold int set_default_channel_config(AVCodecContext *avctx,
- enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
- int channel_config)
-{
- if (channel_config < 1 || channel_config > 7) {
- av_log(avctx, AV_LOG_ERROR, "invalid default channel configuration (%d)\n",
- channel_config);
- return -1;
- }
-
- /* default channel configurations:
- *
- * 1ch : front center (mono)
- * 2ch : L + R (stereo)
- * 3ch : front center + L + R
- * 4ch : front center + L + R + back center
- * 5ch : front center + L + R + back stereo
- * 6ch : front center + L + R + back stereo + LFE
- * 7ch : front center + L + R + outer front left + outer front right + back stereo + LFE
- */
-
- if (channel_config != 2)
- new_che_pos[TYPE_SCE][0] = AAC_CHANNEL_FRONT; // front center (or mono)
- if (channel_config > 1)
- new_che_pos[TYPE_CPE][0] = AAC_CHANNEL_FRONT; // L + R (or stereo)
- if (channel_config == 4)
- new_che_pos[TYPE_SCE][1] = AAC_CHANNEL_BACK; // back center
- if (channel_config > 4)
- new_che_pos[TYPE_CPE][(channel_config == 7) + 1]
- = AAC_CHANNEL_BACK; // back stereo
- if (channel_config > 5)
- new_che_pos[TYPE_LFE][0] = AAC_CHANNEL_LFE; // LFE
- if (channel_config == 7)
- new_che_pos[TYPE_CPE][1] = AAC_CHANNEL_FRONT; // outer front left + outer front right
-
- return 0;
+ return tags;
}
/**
MPEG4AudioConfig *m4ac,
int channel_config)
{
- enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
int extension_flag, ret;
+ uint8_t layout_map[MAX_ELEM_ID*4][3];
+ int tags = 0;
if (get_bits1(gb)) { // frameLengthFlag
- av_log_missing_feature(avctx, "960/120 MDCT window is", 1);
- return -1;
+ av_log_missing_feature(avctx, "960/120 MDCT window", 1);
+ return AVERROR_PATCHWELCOME;
}
if (get_bits1(gb)) // dependsOnCoreCoder
m4ac->object_type == AOT_ER_AAC_SCALABLE)
skip_bits(gb, 3); // layerNr
- memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
if (channel_config == 0) {
skip_bits(gb, 4); // element_instance_tag
- if ((ret = decode_pce(avctx, m4ac, new_che_pos, gb)))
- return ret;
+ tags = decode_pce(avctx, m4ac, layout_map, gb);
+ if (tags < 0)
+ return tags;
} else {
- if ((ret = set_default_channel_config(avctx, new_che_pos, channel_config)))
+ if ((ret = set_default_channel_config(avctx, layout_map, &tags, channel_config)))
return ret;
}
- if (ac && (ret = output_configure(ac, ac->che_pos, new_che_pos, channel_config, OC_GLOBAL_HDR)))
+
+ if (count_channels(layout_map, tags) > 1) {
+ m4ac->ps = 0;
+ } else if (m4ac->sbr == 1 && m4ac->ps == -1)
+ m4ac->ps = 1;
+
+ if (ac && (ret = output_configure(ac, layout_map, tags, OC_GLOBAL_HDR)))
return ret;
if (extension_flag) {
* @param ac pointer to AACContext, may be null
* @param avctx pointer to AVCCodecContext, used for logging
* @param m4ac pointer to MPEG4AudioConfig, used for parsing
- * @param data pointer to AVCodecContext extradata
- * @param data_size size of AVCCodecContext extradata
+ * @param data pointer to buffer holding an audio specific config
+ * @param bit_size size of audio specific config or data in bits
+ * @param sync_extension look for an appended sync extension
*
* @return Returns error status or number of consumed bits. <0 - error
*/
static int decode_audio_specific_config(AACContext *ac,
AVCodecContext *avctx,
MPEG4AudioConfig *m4ac,
- const uint8_t *data, int data_size)
+ const uint8_t *data, int bit_size,
+ int sync_extension)
{
GetBitContext gb;
int i;
- init_get_bits(&gb, data, data_size * 8);
+ 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, "\n");
+
+ init_get_bits(&gb, data, bit_size);
- if ((i = ff_mpeg4audio_get_config(m4ac, data, data_size)) < 0)
+ if ((i = avpriv_mpeg4audio_get_config(m4ac, data, bit_size, sync_extension)) < 0)
return -1;
if (m4ac->sampling_index > 12) {
av_log(avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", m4ac->sampling_index);
return -1;
}
- if (m4ac->sbr == 1 && m4ac->ps == -1)
- m4ac->ps = 1;
skip_bits_long(&gb, i);
switch (m4ac->object_type) {
case AOT_AAC_MAIN:
case AOT_AAC_LC:
+ case AOT_AAC_LTP:
if (decode_ga_specific_config(ac, avctx, &gb, m4ac, m4ac->chan_config))
return -1;
break;
return -1;
}
+ av_dlog(avctx, "AOT %d chan config %d sampling index %d (%d) SBR %d PS %d\n",
+ m4ac->object_type, m4ac->chan_config, m4ac->sampling_index,
+ m4ac->sample_rate, m4ac->sbr, m4ac->ps);
+
return get_bits_count(&gb);
}
reset_predict_state(&ps[i]);
}
+static int sample_rate_idx (int rate)
+{
+ if (92017 <= rate) return 0;
+ else if (75132 <= rate) return 1;
+ else if (55426 <= rate) return 2;
+ else if (46009 <= rate) return 3;
+ else if (37566 <= rate) return 4;
+ else if (27713 <= rate) return 5;
+ else if (23004 <= rate) return 6;
+ else if (18783 <= rate) return 7;
+ else if (13856 <= rate) return 8;
+ else if (11502 <= rate) return 9;
+ else if (9391 <= rate) return 10;
+ else return 11;
+}
+
static void reset_predictor_group(PredictorState *ps, int group_num)
{
int i;
static av_cold int aac_decode_init(AVCodecContext *avctx)
{
AACContext *ac = avctx->priv_data;
+ float output_scale_factor;
ac->avctx = avctx;
- ac->m4ac.sample_rate = avctx->sample_rate;
+ ac->oc[1].m4ac.sample_rate = avctx->sample_rate;
if (avctx->extradata_size > 0) {
- if (decode_audio_specific_config(ac, ac->avctx, &ac->m4ac,
+ if (decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
avctx->extradata,
- avctx->extradata_size) < 0)
+ avctx->extradata_size*8, 1) < 0)
return -1;
+ } else {
+ int sr, i;
+ uint8_t layout_map[MAX_ELEM_ID*4][3];
+ int layout_map_tags;
+
+ sr = sample_rate_idx(avctx->sample_rate);
+ ac->oc[1].m4ac.sampling_index = sr;
+ ac->oc[1].m4ac.channels = avctx->channels;
+ ac->oc[1].m4ac.sbr = -1;
+ ac->oc[1].m4ac.ps = -1;
+
+ for (i = 0; i < FF_ARRAY_ELEMS(ff_mpeg4audio_channels); i++)
+ if (ff_mpeg4audio_channels[i] == avctx->channels)
+ break;
+ if (i == FF_ARRAY_ELEMS(ff_mpeg4audio_channels)) {
+ i = 0;
+ }
+ ac->oc[1].m4ac.chan_config = i;
+
+ if (ac->oc[1].m4ac.chan_config) {
+ int ret = set_default_channel_config(avctx, layout_map,
+ &layout_map_tags, ac->oc[1].m4ac.chan_config);
+ if (!ret)
+ output_configure(ac, layout_map, layout_map_tags,
+ OC_GLOBAL_HDR);
+ else if (avctx->err_recognition & AV_EF_EXPLODE)
+ return AVERROR_INVALIDDATA;
+ }
}
- avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+ 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);
ff_aac_sbr_init();
- dsputil_init(&ac->dsp, avctx);
+ ff_dsputil_init(&ac->dsp, avctx);
+ ff_fmt_convert_init(&ac->fmt_conv, avctx);
+ avpriv_float_dsp_init(&ac->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
ac->random_state = 0x1f2e3d4c;
- // -1024 - Compensate wrong IMDCT method.
- // 32768 - Required to scale values to the correct range for the bias method
- // for float to int16 conversion.
-
- if (ac->dsp.float_to_int16_interleave == ff_float_to_int16_interleave_c) {
- ac->add_bias = 385.0f;
- ac->sf_scale = 1. / (-1024. * 32768.);
- ac->sf_offset = 0;
- } else {
- ac->add_bias = 0.0f;
- ac->sf_scale = 1. / -1024.;
- ac->sf_offset = 60;
- }
-
ff_aac_tableinit();
INIT_VLC_STATIC(&vlc_scalefactors,7,FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]),
352);
- ff_mdct_init(&ac->mdct, 11, 1, 1.0);
- ff_mdct_init(&ac->mdct_small, 8, 1, 1.0);
+ ff_mdct_init(&ac->mdct, 11, 1, output_scale_factor/1024.0);
+ ff_mdct_init(&ac->mdct_small, 8, 1, output_scale_factor/128.0);
+ ff_mdct_init(&ac->mdct_ltp, 11, 0, -2.0/output_scale_factor);
// window initialization
ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
cbrt_tableinit();
+ avcodec_get_frame_defaults(&ac->frame);
+ avctx->coded_frame = &ac->frame;
+
return 0;
}
return -1;
}
}
- for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->m4ac.sampling_index]); sfb++) {
+ for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->oc[1].m4ac.sampling_index]); sfb++) {
ics->prediction_used[sfb] = get_bits1(gb);
}
return 0;
}
+/**
+ * Decode Long Term Prediction data; reference: table 4.xx.
+ */
+static void decode_ltp(LongTermPrediction *ltp,
+ GetBitContext *gb, uint8_t max_sfb)
+{
+ int sfb;
+
+ ltp->lag = get_bits(gb, 11);
+ ltp->coef = ltp_coef[get_bits(gb, 3)];
+ for (sfb = 0; sfb < FFMIN(max_sfb, MAX_LTP_LONG_SFB); sfb++)
+ ltp->used[sfb] = get_bits1(gb);
+}
+
/**
* Decode Individual Channel Stream info; reference: table 4.6.
- *
- * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information.
*/
static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
- GetBitContext *gb, int common_window)
+ GetBitContext *gb)
{
if (get_bits1(gb)) {
av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n");
- memset(ics, 0, sizeof(IndividualChannelStream));
- return -1;
+ return AVERROR_INVALIDDATA;
}
ics->window_sequence[1] = ics->window_sequence[0];
ics->window_sequence[0] = get_bits(gb, 2);
}
}
ics->num_windows = 8;
- ics->swb_offset = ff_swb_offset_128[ac->m4ac.sampling_index];
- ics->num_swb = ff_aac_num_swb_128[ac->m4ac.sampling_index];
- ics->tns_max_bands = ff_tns_max_bands_128[ac->m4ac.sampling_index];
+ ics->swb_offset = ff_swb_offset_128[ac->oc[1].m4ac.sampling_index];
+ ics->num_swb = ff_aac_num_swb_128[ac->oc[1].m4ac.sampling_index];
+ ics->tns_max_bands = ff_tns_max_bands_128[ac->oc[1].m4ac.sampling_index];
ics->predictor_present = 0;
} else {
ics->max_sfb = get_bits(gb, 6);
ics->num_windows = 1;
- ics->swb_offset = ff_swb_offset_1024[ac->m4ac.sampling_index];
- ics->num_swb = ff_aac_num_swb_1024[ac->m4ac.sampling_index];
- ics->tns_max_bands = ff_tns_max_bands_1024[ac->m4ac.sampling_index];
+ ics->swb_offset = ff_swb_offset_1024[ac->oc[1].m4ac.sampling_index];
+ ics->num_swb = ff_aac_num_swb_1024[ac->oc[1].m4ac.sampling_index];
+ ics->tns_max_bands = ff_tns_max_bands_1024[ac->oc[1].m4ac.sampling_index];
ics->predictor_present = get_bits1(gb);
ics->predictor_reset_group = 0;
if (ics->predictor_present) {
- if (ac->m4ac.object_type == AOT_AAC_MAIN) {
+ if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN) {
if (decode_prediction(ac, ics, gb)) {
- memset(ics, 0, sizeof(IndividualChannelStream));
- return -1;
+ return AVERROR_INVALIDDATA;
}
- } else if (ac->m4ac.object_type == AOT_AAC_LC) {
+ } 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");
- memset(ics, 0, sizeof(IndividualChannelStream));
- return -1;
+ return AVERROR_INVALIDDATA;
} else {
- av_log_missing_feature(ac->avctx, "Predictor bit set but LTP is", 1);
- memset(ics, 0, sizeof(IndividualChannelStream));
- return -1;
+ if ((ics->ltp.present = get_bits(gb, 1)))
+ decode_ltp(&ics->ltp, gb, ics->max_sfb);
}
}
}
av_log(ac->avctx, AV_LOG_ERROR,
"Number of scalefactor bands in group (%d) exceeds limit (%d).\n",
ics->max_sfb, ics->num_swb);
- memset(ics, 0, sizeof(IndividualChannelStream));
- return -1;
+ return AVERROR_INVALIDDATA;
}
return 0;
av_log(ac->avctx, AV_LOG_ERROR, "invalid band type\n");
return -1;
}
- while ((sect_len_incr = get_bits(gb, bits)) == (1 << bits) - 1)
+ do {
+ sect_len_incr = get_bits(gb, bits);
sect_end += sect_len_incr;
- sect_end += sect_len_incr;
- if (get_bits_left(gb) < 0) {
- av_log(ac->avctx, AV_LOG_ERROR, overread_err);
- return -1;
- }
- if (sect_end > ics->max_sfb) {
- av_log(ac->avctx, AV_LOG_ERROR,
- "Number of bands (%d) exceeds limit (%d).\n",
- sect_end, ics->max_sfb);
- return -1;
- }
+ if (get_bits_left(gb) < 0) {
+ av_log(ac->avctx, AV_LOG_ERROR, overread_err);
+ return -1;
+ }
+ if (sect_end > ics->max_sfb) {
+ av_log(ac->avctx, AV_LOG_ERROR,
+ "Number of bands (%d) exceeds limit (%d).\n",
+ sect_end, ics->max_sfb);
+ return -1;
+ }
+ } while (sect_len_incr == (1 << bits) - 1);
for (; k < sect_end; k++) {
band_type [idx] = sect_band_type;
band_type_run_end[idx++] = sect_end;
enum BandType band_type[120],
int band_type_run_end[120])
{
- const int sf_offset = ac->sf_offset + (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE ? 12 : 0);
int g, i, idx = 0;
- int offset[3] = { global_gain, global_gain - 90, 100 };
+ int offset[3] = { global_gain, global_gain - 90, 0 };
+ int clipped_offset;
int noise_flag = 1;
- static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" };
for (g = 0; g < ics->num_window_groups; g++) {
for (i = 0; i < ics->max_sfb;) {
int run_end = band_type_run_end[idx];
} else if ((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) {
for (; i < run_end; i++, idx++) {
offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
- if (offset[2] > 255U) {
- av_log(ac->avctx, AV_LOG_ERROR,
- "%s (%d) out of range.\n", sf_str[2], offset[2]);
- return -1;
+ clipped_offset = av_clip(offset[2], -155, 100);
+ if (offset[2] != clipped_offset) {
+ av_log_ask_for_sample(ac->avctx, "Intensity stereo "
+ "position clipped (%d -> %d).\nIf you heard an "
+ "audible artifact, there may be a bug in the "
+ "decoder. ", offset[2], clipped_offset);
}
- sf[idx] = ff_aac_pow2sf_tab[-offset[2] + 300];
+ sf[idx] = ff_aac_pow2sf_tab[-clipped_offset + POW_SF2_ZERO];
}
} else if (band_type[idx] == NOISE_BT) {
for (; i < run_end; i++, idx++) {
offset[1] += get_bits(gb, 9) - 256;
else
offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
- if (offset[1] > 255U) {
- av_log(ac->avctx, AV_LOG_ERROR,
- "%s (%d) out of range.\n", sf_str[1], offset[1]);
- return -1;
+ clipped_offset = av_clip(offset[1], -100, 155);
+ if (offset[1] != clipped_offset) {
+ av_log_ask_for_sample(ac->avctx, "Noise gain clipped "
+ "(%d -> %d).\nIf you heard an audible "
+ "artifact, there may be a bug in the decoder. ",
+ offset[1], clipped_offset);
}
- sf[idx] = -ff_aac_pow2sf_tab[offset[1] + sf_offset + 100];
+ sf[idx] = -ff_aac_pow2sf_tab[clipped_offset + POW_SF2_ZERO];
}
} else {
for (; i < run_end; i++, idx++) {
offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
if (offset[0] > 255U) {
av_log(ac->avctx, AV_LOG_ERROR,
- "%s (%d) out of range.\n", sf_str[0], offset[0]);
+ "Scalefactor (%d) out of range.\n", offset[0]);
return -1;
}
- sf[idx] = -ff_aac_pow2sf_tab[ offset[0] + sf_offset];
+ sf[idx] = -ff_aac_pow2sf_tab[offset[0] - 100 + POW_SF2_ZERO];
}
}
}
{
int w, filt, i, coef_len, coef_res, coef_compress;
const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE;
- const int tns_max_order = is8 ? 7 : ac->m4ac.object_type == AOT_AAC_MAIN ? 20 : 12;
+ const int tns_max_order = is8 ? 7 : ac->oc[1].m4ac.object_type == AOT_AAC_MAIN ? 20 : 12;
for (w = 0; w < ics->num_windows; w++) {
if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) {
coef_res = get_bits1(gb);
static inline float *VMUL2S(float *dst, const float *v, unsigned idx,
unsigned sign, const float *scale)
{
- union float754 s0, s1;
+ union av_intfloat32 s0, s1;
s0.f = s1.f = *scale;
s0.i ^= sign >> 1 << 31;
unsigned sign, const float *scale)
{
unsigned nz = idx >> 12;
- union float754 s = { .f = *scale };
- union float754 t;
+ union av_intfloat32 s = { .f = *scale };
+ union av_intfloat32 t;
- t.i = s.i ^ (sign & 1<<31);
+ t.i = s.i ^ (sign & 1U<<31);
*dst++ = v[idx & 3] * t.f;
sign <<= nz & 1; nz >>= 1;
- t.i = s.i ^ (sign & 1<<31);
+ t.i = s.i ^ (sign & 1U<<31);
*dst++ = v[idx>>2 & 3] * t.f;
sign <<= nz & 1; nz >>= 1;
- t.i = s.i ^ (sign & 1<<31);
+ t.i = s.i ^ (sign & 1U<<31);
*dst++ = v[idx>>4 & 3] * t.f;
- sign <<= nz & 1; nz >>= 1;
- t.i = s.i ^ (sign & 1<<31);
+ sign <<= nz & 1;
+ t.i = s.i ^ (sign & 1U<<31);
*dst++ = v[idx>>6 & 3] * t.f;
return dst;
GET_VLC(code, re, gb, vlc_tab, 8, 2);
cb_idx = cb_vector_idx[code];
nnz = cb_idx >> 8 & 15;
- bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
+ bits = nnz ? GET_CACHE(re, gb) : 0;
LAST_SKIP_BITS(re, gb, nnz);
cf = VMUL4S(cf, vq, cb_idx, bits, sf + idx);
} while (len -= 4);
GET_VLC(code, re, gb, vlc_tab, 8, 2);
cb_idx = cb_vector_idx[code];
nnz = cb_idx >> 8 & 15;
- sign = SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12);
+ sign = nnz ? SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12) : 0;
LAST_SKIP_BITS(re, gb, nnz);
cf = VMUL2S(cf, vq, cb_idx, sign, sf + idx);
} while (len -= 2);
b += 4;
n = (1 << b) + SHOW_UBITS(re, gb, b);
LAST_SKIP_BITS(re, gb, b);
- *icf++ = cbrt_tab[n] | (bits & 1<<31);
+ *icf++ = cbrt_tab[n] | (bits & 1U<<31);
bits <<= 1;
} else {
unsigned v = ((const uint32_t*)vq)[cb_idx & 15];
- *icf++ = (bits & 1<<31) | v;
+ *icf++ = (bits & 1U<<31) | v;
bits <<= !!v;
}
cb_idx >>= 4;
static av_always_inline float flt16_round(float pf)
{
- union float754 tmp;
+ union av_intfloat32 tmp;
tmp.f = pf;
tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U;
return tmp.f;
static av_always_inline float flt16_even(float pf)
{
- union float754 tmp;
+ union av_intfloat32 tmp;
tmp.f = pf;
tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U;
return tmp.f;
static av_always_inline float flt16_trunc(float pf)
{
- union float754 pun;
+ union av_intfloat32 pun;
pun.f = pf;
pun.i &= 0xFFFF0000U;
return pun.f;
}
static av_always_inline void predict(PredictorState *ps, float *coef,
- float sf_scale, float inv_sf_scale,
- int output_enable)
+ int output_enable)
{
const float a = 0.953125; // 61.0 / 64
const float alpha = 0.90625; // 29.0 / 32
pv = flt16_round(k1 * r0 + k2 * r1);
if (output_enable)
- *coef += pv * sf_scale;
+ *coef += pv;
- e0 = *coef * inv_sf_scale;
+ e0 = *coef;
e1 = e0 - k1 * r0;
ps->cor1 = flt16_trunc(alpha * cor1 + r1 * e1);
static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
{
int sfb, k;
- float sf_scale = ac->sf_scale, inv_sf_scale = 1 / ac->sf_scale;
if (!sce->ics.predictor_initialized) {
reset_all_predictors(sce->predictor_state);
}
if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
- for (sfb = 0; sfb < ff_aac_pred_sfb_max[ac->m4ac.sampling_index]; sfb++) {
+ for (sfb = 0; sfb < ff_aac_pred_sfb_max[ac->oc[1].m4ac.sampling_index]; sfb++) {
for (k = sce->ics.swb_offset[sfb]; k < sce->ics.swb_offset[sfb + 1]; k++) {
predict(&sce->predictor_state[k], &sce->coeffs[k],
- sf_scale, inv_sf_scale,
sce->ics.predictor_present && sce->ics.prediction_used[sfb]);
}
}
global_gain = get_bits(gb, 8);
if (!common_window && !scale_flag) {
- if (decode_ics_info(ac, ics, gb, 0) < 0)
- return -1;
+ if (decode_ics_info(ac, ics, gb) < 0)
+ return AVERROR_INVALIDDATA;
}
if (decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics) < 0)
return -1;
if (get_bits1(gb)) {
av_log_missing_feature(ac->avctx, "SSR", 1);
- return -1;
+ return AVERROR_PATCHWELCOME;
}
}
if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present, &pulse, ics, sce->band_type) < 0)
return -1;
- if (ac->m4ac.object_type == AOT_AAC_MAIN && !common_window)
+ if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN && !common_window)
apply_prediction(ac, sce);
return 0;
* [1] mask is decoded from bitstream; [2] mask is all 1s;
* [3] reserved for scalable AAC
*/
-static void apply_intensity_stereo(ChannelElement *cpe, int ms_present)
+static void apply_intensity_stereo(AACContext *ac, ChannelElement *cpe, int ms_present)
{
const IndividualChannelStream *ics = &cpe->ch[1].ics;
SingleChannelElement *sce1 = &cpe->ch[1];
float *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs;
const uint16_t *offsets = ics->swb_offset;
- int g, group, i, k, idx = 0;
+ int g, group, i, idx = 0;
int c;
float scale;
for (g = 0; g < ics->num_window_groups; g++) {
c *= 1 - 2 * cpe->ms_mask[idx];
scale = c * sce1->sf[idx];
for (group = 0; group < ics->group_len[g]; group++)
- for (k = offsets[i]; k < offsets[i + 1]; k++)
- coef1[group * 128 + k] = scale * coef0[group * 128 + k];
+ ac->dsp.vector_fmul_scalar(coef1 + group * 128 + offsets[i],
+ coef0 + group * 128 + offsets[i],
+ scale,
+ offsets[i + 1] - offsets[i]);
}
} else {
int bt_run_end = sce1->band_type_run_end[idx];
common_window = get_bits1(gb);
if (common_window) {
- if (decode_ics_info(ac, &cpe->ch[0].ics, gb, 1))
- return -1;
+ if (decode_ics_info(ac, &cpe->ch[0].ics, gb))
+ return AVERROR_INVALIDDATA;
i = cpe->ch[1].ics.use_kb_window[0];
cpe->ch[1].ics = cpe->ch[0].ics;
cpe->ch[1].ics.use_kb_window[1] = i;
+ if (cpe->ch[1].ics.predictor_present && (ac->oc[1].m4ac.object_type != AOT_AAC_MAIN))
+ if ((cpe->ch[1].ics.ltp.present = get_bits(gb, 1)))
+ decode_ltp(&cpe->ch[1].ics.ltp, gb, cpe->ch[1].ics.max_sfb);
ms_present = get_bits(gb, 2);
if (ms_present == 3) {
av_log(ac->avctx, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
if (common_window) {
if (ms_present)
apply_mid_side_stereo(ac, cpe);
- if (ac->m4ac.object_type == AOT_AAC_MAIN) {
+ if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN) {
apply_prediction(ac, &cpe->ch[0]);
apply_prediction(ac, &cpe->ch[1]);
}
}
- apply_intensity_stereo(cpe, ms_present);
+ apply_intensity_stereo(ac, cpe, ms_present);
return 0;
}
/**
* Decode dynamic range information; reference: table 4.52.
*
- * @param cnt length of TYPE_FIL syntactic element in bytes
- *
* @return Returns number of bytes consumed.
*/
static int decode_dynamic_range(DynamicRangeControl *che_drc,
- GetBitContext *gb, int cnt)
+ GetBitContext *gb)
{
int n = 1;
int drc_num_bands = 1;
if (!che) {
av_log(ac->avctx, AV_LOG_ERROR, "SBR was found before the first channel element.\n");
return res;
- } else if (!ac->m4ac.sbr) {
+ } else if (!ac->oc[1].m4ac.sbr) {
av_log(ac->avctx, AV_LOG_ERROR, "SBR signaled to be not-present but was found in the bitstream.\n");
skip_bits_long(gb, 8 * cnt - 4);
return res;
- } else if (ac->m4ac.sbr == -1 && ac->output_configured == OC_LOCKED) {
+ } else if (ac->oc[1].m4ac.sbr == -1 && ac->oc[1].status == OC_LOCKED) {
av_log(ac->avctx, AV_LOG_ERROR, "Implicit SBR was found with a first occurrence after the first frame.\n");
skip_bits_long(gb, 8 * cnt - 4);
return res;
- } else if (ac->m4ac.ps == -1 && ac->output_configured < OC_LOCKED && ac->avctx->channels == 1) {
- ac->m4ac.sbr = 1;
- ac->m4ac.ps = 1;
- output_configure(ac, ac->che_pos, ac->che_pos, ac->m4ac.chan_config, ac->output_configured);
+ } else if (ac->oc[1].m4ac.ps == -1 && ac->oc[1].status < OC_LOCKED && ac->avctx->channels == 1) {
+ ac->oc[1].m4ac.sbr = 1;
+ ac->oc[1].m4ac.ps = 1;
+ output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags,
+ ac->oc[1].status);
} else {
- ac->m4ac.sbr = 1;
+ ac->oc[1].m4ac.sbr = 1;
}
res = ff_decode_sbr_extension(ac, &che->sbr, gb, crc_flag, cnt, elem_type);
break;
case EXT_DYNAMIC_RANGE:
- res = decode_dynamic_range(&ac->che_drc, gb, cnt);
+ res = decode_dynamic_range(&ac->che_drc, gb);
break;
case EXT_FILL:
case EXT_FILL_DATA:
int w, filt, m, i;
int bottom, top, order, start, end, size, inc;
float lpc[TNS_MAX_ORDER];
+ float tmp[TNS_MAX_ORDER];
for (w = 0; w < ics->num_windows; w++) {
bottom = ics->num_swb;
}
start += w * 128;
- // ar filter
- for (m = 0; m < size; m++, start += inc)
- for (i = 1; i <= FFMIN(m, order); i++)
- coef[start] -= coef[start - i * inc] * lpc[i - 1];
+ if (decode) {
+ // ar filter
+ for (m = 0; m < size; m++, start += inc)
+ for (i = 1; i <= FFMIN(m, order); i++)
+ coef[start] -= coef[start - i * inc] * lpc[i - 1];
+ } else {
+ // ma filter
+ for (m = 0; m < size; m++, start += inc) {
+ tmp[0] = coef[start];
+ for (i = 1; i <= FFMIN(m, order); i++)
+ coef[start] += tmp[i] * lpc[i - 1];
+ for (i = order; i > 0; i--)
+ tmp[i] = tmp[i - 1];
+ }
+ }
}
}
}
+/**
+ * Apply windowing and MDCT to obtain the spectral
+ * coefficient from the predicted sample by LTP.
+ */
+static void windowing_and_mdct_ltp(AACContext *ac, float *out,
+ float *in, IndividualChannelStream *ics)
+{
+ const float *lwindow = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
+ const float *swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
+ const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
+ const float *swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
+
+ if (ics->window_sequence[0] != LONG_STOP_SEQUENCE) {
+ ac->fdsp.vector_fmul(in, in, lwindow_prev, 1024);
+ } else {
+ memset(in, 0, 448 * sizeof(float));
+ ac->fdsp.vector_fmul(in + 448, in + 448, swindow_prev, 128);
+ }
+ if (ics->window_sequence[0] != LONG_START_SEQUENCE) {
+ ac->dsp.vector_fmul_reverse(in + 1024, in + 1024, lwindow, 1024);
+ } else {
+ ac->dsp.vector_fmul_reverse(in + 1024 + 448, in + 1024 + 448, swindow, 128);
+ memset(in + 1024 + 576, 0, 448 * sizeof(float));
+ }
+ ac->mdct_ltp.mdct_calc(&ac->mdct_ltp, out, in);
+}
+
+/**
+ * Apply the long term prediction
+ */
+static void apply_ltp(AACContext *ac, SingleChannelElement *sce)
+{
+ const LongTermPrediction *ltp = &sce->ics.ltp;
+ const uint16_t *offsets = sce->ics.swb_offset;
+ int i, sfb;
+
+ if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
+ float *predTime = sce->ret;
+ float *predFreq = ac->buf_mdct;
+ int16_t num_samples = 2048;
+
+ if (ltp->lag < 1024)
+ num_samples = ltp->lag + 1024;
+ for (i = 0; i < num_samples; i++)
+ 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);
+
+ if (sce->tns.present)
+ 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])
+ for (i = offsets[sfb]; i < offsets[sfb + 1]; i++)
+ sce->coeffs[i] += predFreq[i];
+ }
+}
+
+/**
+ * Update the LTP buffer for next frame
+ */
+static void update_ltp(AACContext *ac, SingleChannelElement *sce)
+{
+ IndividualChannelStream *ics = &sce->ics;
+ float *saved = sce->saved;
+ float *saved_ltp = sce->coeffs;
+ const float *lwindow = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
+ const float *swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
+ int i;
+
+ if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
+ memcpy(saved_ltp, saved, 512 * sizeof(float));
+ memset(saved_ltp + 576, 0, 448 * sizeof(float));
+ ac->dsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64);
+ for (i = 0; i < 64; i++)
+ saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * swindow[63 - i];
+ } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
+ memcpy(saved_ltp, ac->buf_mdct + 512, 448 * sizeof(float));
+ memset(saved_ltp + 576, 0, 448 * sizeof(float));
+ ac->dsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64);
+ for (i = 0; i < 64; i++)
+ saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * swindow[63 - i];
+ } else { // LONG_STOP or ONLY_LONG
+ ac->dsp.vector_fmul_reverse(saved_ltp, ac->buf_mdct + 512, &lwindow[512], 512);
+ for (i = 0; i < 512; i++)
+ saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * lwindow[511 - i];
+ }
+
+ memcpy(sce->ltp_state, sce->ltp_state+1024, 1024 * sizeof(*sce->ltp_state));
+ memcpy(sce->ltp_state+1024, sce->ret, 1024 * sizeof(*sce->ltp_state));
+ memcpy(sce->ltp_state+2048, saved_ltp, 1024 * sizeof(*sce->ltp_state));
+}
+
/**
* Conduct IMDCT and windowing.
*/
-static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce, float bias)
+static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
{
IndividualChannelStream *ics = &sce->ics;
float *in = sce->coeffs;
// imdct
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
for (i = 0; i < 1024; i += 128)
- ff_imdct_half(&ac->mdct_small, buf + i, in + i);
+ ac->mdct_small.imdct_half(&ac->mdct_small, buf + i, in + i);
} else
- ff_imdct_half(&ac->mdct, buf, in);
+ ac->mdct.imdct_half(&ac->mdct, buf, in);
/* window overlapping
* NOTE: To simplify the overlapping code, all 'meaningless' short to long
*/
if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) &&
(ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) {
- ac->dsp.vector_fmul_window( out, saved, buf, lwindow_prev, bias, 512);
+ ac->dsp.vector_fmul_window( out, saved, buf, lwindow_prev, 512);
} else {
- for (i = 0; i < 448; i++)
- out[i] = saved[i] + bias;
+ memcpy( out, saved, 448 * sizeof(float));
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
- ac->dsp.vector_fmul_window(out + 448 + 0*128, saved + 448, buf + 0*128, swindow_prev, bias, 64);
- ac->dsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow, bias, 64);
- ac->dsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow, bias, 64);
- ac->dsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow, bias, 64);
- ac->dsp.vector_fmul_window(temp, buf + 3*128 + 64, buf + 4*128, swindow, bias, 64);
+ ac->dsp.vector_fmul_window(out + 448 + 0*128, saved + 448, buf + 0*128, swindow_prev, 64);
+ ac->dsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow, 64);
+ ac->dsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow, 64);
+ ac->dsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow, 64);
+ ac->dsp.vector_fmul_window(temp, buf + 3*128 + 64, buf + 4*128, swindow, 64);
memcpy( out + 448 + 4*128, temp, 64 * sizeof(float));
} else {
- ac->dsp.vector_fmul_window(out + 448, saved + 448, buf, swindow_prev, bias, 64);
- for (i = 576; i < 1024; i++)
- out[i] = buf[i-512] + bias;
+ ac->dsp.vector_fmul_window(out + 448, saved + 448, buf, swindow_prev, 64);
+ memcpy( out + 576, buf + 64, 448 * sizeof(float));
}
}
// buffer update
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
- for (i = 0; i < 64; i++)
- saved[i] = temp[64 + i] - bias;
- ac->dsp.vector_fmul_window(saved + 64, buf + 4*128 + 64, buf + 5*128, swindow, 0, 64);
- ac->dsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 0, 64);
- ac->dsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 0, 64);
+ memcpy( saved, temp + 64, 64 * sizeof(float));
+ ac->dsp.vector_fmul_window(saved + 64, buf + 4*128 + 64, buf + 5*128, swindow, 64);
+ ac->dsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 64);
+ ac->dsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 64);
memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(float));
} else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
memcpy( saved, buf + 512, 448 * sizeof(float));
float *dest = target->coeffs;
const float *src = cce->ch[0].coeffs;
int g, i, group, k, idx = 0;
- if (ac->m4ac.object_type == AOT_AAC_LTP) {
+ if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
av_log(ac->avctx, AV_LOG_ERROR,
"Dependent coupling is not supported together with LTP\n");
return;
{
int i;
const float gain = cce->coup.gain[index][0];
- const float bias = ac->add_bias;
const float *src = cce->ch[0].ret;
float *dest = target->ret;
- const int len = 1024 << (ac->m4ac.sbr == 1);
+ const int len = 1024 << (ac->oc[1].m4ac.sbr == 1);
for (i = 0; i < len; i++)
- dest[i] += gain * (src[i] - bias);
+ dest[i] += gain * src[i];
}
/**
static void spectral_to_sample(AACContext *ac)
{
int i, type;
- float imdct_bias = (ac->m4ac.sbr <= 0) ? ac->add_bias : 0.0f;
for (type = 3; type >= 0; type--) {
for (i = 0; i < MAX_ELEM_ID; i++) {
ChannelElement *che = ac->che[type][i];
if (che) {
if (type <= TYPE_CPE)
apply_channel_coupling(ac, che, type, i, BEFORE_TNS, apply_dependent_coupling);
+ 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]);
+ if (che->ch[1].ics.ltp.present && type == TYPE_CPE)
+ 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);
if (che->ch[1].tns.present)
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], imdct_bias);
+ imdct_and_windowing(ac, &che->ch[0]);
+ if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
+ update_ltp(ac, &che->ch[0]);
if (type == TYPE_CPE) {
- imdct_and_windowing(ac, &che->ch[1], imdct_bias);
+ imdct_and_windowing(ac, &che->ch[1]);
+ if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
+ update_ltp(ac, &che->ch[1]);
}
- if (ac->m4ac.sbr > 0) {
+ if (ac->oc[1].m4ac.sbr > 0) {
ff_sbr_apply(ac, &che->sbr, type, che->ch[0].ret, che->ch[1].ret);
}
}
{
int size;
AACADTSHeaderInfo hdr_info;
+ uint8_t layout_map[MAX_ELEM_ID*4][3];
+ int layout_map_tags;
- size = ff_aac_parse_header(gb, &hdr_info);
+ size = avpriv_aac_parse_header(gb, &hdr_info);
if (size > 0) {
- if (ac->output_configured != OC_LOCKED && hdr_info.chan_config) {
- enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
- memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
- ac->m4ac.chan_config = hdr_info.chan_config;
- if (set_default_channel_config(ac->avctx, new_che_pos, hdr_info.chan_config))
+ if (hdr_info.num_aac_frames != 1) {
+ av_log_missing_feature(ac->avctx, "More than one AAC RDB per ADTS frame", 0);
+ return AVERROR_PATCHWELCOME;
+ }
+ push_output_configuration(ac);
+ if (hdr_info.chan_config) {
+ ac->oc[1].m4ac.chan_config = hdr_info.chan_config;
+ if (set_default_channel_config(ac->avctx, layout_map,
+ &layout_map_tags, hdr_info.chan_config))
return -7;
- if (output_configure(ac, ac->che_pos, new_che_pos, hdr_info.chan_config, OC_TRIAL_FRAME))
+ if (output_configure(ac, layout_map, layout_map_tags,
+ FFMAX(ac->oc[1].status, OC_TRIAL_FRAME)))
return -7;
- } else if (ac->output_configured != OC_LOCKED) {
- ac->output_configured = OC_NONE;
- }
- if (ac->output_configured != OC_LOCKED) {
- ac->m4ac.sbr = -1;
- ac->m4ac.ps = -1;
- }
- 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 (!ac->avctx->sample_rate)
- ac->avctx->sample_rate = hdr_info.sample_rate;
- if (hdr_info.num_aac_frames == 1) {
- if (!hdr_info.crc_absent)
- skip_bits(gb, 16);
} else {
- av_log_missing_feature(ac->avctx, "More than one AAC RDB per ADTS frame is", 0);
- return -1;
+ ac->oc[1].m4ac.chan_config = 0;
+ }
+ ac->oc[1].m4ac.sample_rate = hdr_info.sample_rate;
+ ac->oc[1].m4ac.sampling_index = hdr_info.sampling_index;
+ ac->oc[1].m4ac.object_type = hdr_info.object_type;
+ if (ac->oc[0].status != OC_LOCKED ||
+ ac->oc[0].m4ac.chan_config != hdr_info.chan_config ||
+ ac->oc[0].m4ac.sample_rate != hdr_info.sample_rate) {
+ ac->oc[1].m4ac.sbr = -1;
+ ac->oc[1].m4ac.ps = -1;
}
+ if (!hdr_info.crc_absent)
+ skip_bits(gb, 16);
}
return size;
}
static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
- int *data_size, GetBitContext *gb)
+ int *got_frame_ptr, GetBitContext *gb)
{
AACContext *ac = avctx->priv_data;
ChannelElement *che = NULL, *che_prev = NULL;
enum RawDataBlockType elem_type, elem_type_prev = TYPE_END;
- int err, elem_id, data_size_tmp;
- int samples = 0, multiplier;
+ int err, elem_id;
+ int samples = 0, multiplier, audio_found = 0, pce_found = 0;
if (show_bits(gb, 12) == 0xfff) {
if (parse_adts_frame_header(ac, gb) < 0) {
av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
- return -1;
+ err = -1;
+ goto fail;
}
- if (ac->m4ac.sampling_index > 12) {
- av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
- return -1;
+ if (ac->oc[1].m4ac.sampling_index > 12) {
+ av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->oc[1].m4ac.sampling_index);
+ err = -1;
+ goto fail;
}
}
if (!(che=get_che(ac, elem_type, elem_id))) {
av_log(ac->avctx, AV_LOG_ERROR, "channel element %d.%d is not allocated\n",
elem_type, elem_id);
- return -1;
+ err = -1;
+ goto fail;
}
samples = 1024;
}
case TYPE_SCE:
err = decode_ics(ac, &che->ch[0], gb, 0, 0);
+ audio_found = 1;
break;
case TYPE_CPE:
err = decode_cpe(ac, gb, che);
+ audio_found = 1;
break;
case TYPE_CCE:
case TYPE_LFE:
err = decode_ics(ac, &che->ch[0], gb, 0, 0);
+ audio_found = 1;
break;
case TYPE_DSE:
break;
case TYPE_PCE: {
- 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 ((err = decode_pce(avctx, &ac->m4ac, new_che_pos, gb)))
+ uint8_t layout_map[MAX_ELEM_ID*4][3];
+ int tags;
+ push_output_configuration(ac);
+ tags = decode_pce(avctx, &ac->oc[1].m4ac, layout_map, gb);
+ if (tags < 0) {
+ err = tags;
break;
- if (ac->output_configured > OC_TRIAL_PCE)
+ }
+ if (pce_found) {
av_log(avctx, AV_LOG_ERROR,
"Not evaluating a further program_config_element as this construct is dubious at best.\n");
- else
- err = output_configure(ac, ac->che_pos, new_che_pos, 0, OC_TRIAL_PCE);
+ pop_output_configuration(ac);
+ } else {
+ err = output_configure(ac, layout_map, tags, OC_TRIAL_PCE);
+ pce_found = 1;
+ }
break;
}
elem_id += get_bits(gb, 8) - 1;
if (get_bits_left(gb) < 8 * elem_id) {
av_log(avctx, AV_LOG_ERROR, overread_err);
- return -1;
+ err = -1;
+ goto fail;
}
while (elem_id > 0)
elem_id -= decode_extension_payload(ac, gb, elem_id, che_prev, elem_type_prev);
elem_type_prev = elem_type;
if (err)
- return err;
+ goto fail;
if (get_bits_left(gb) < 3) {
av_log(avctx, AV_LOG_ERROR, overread_err);
- return -1;
+ err = -1;
+ goto fail;
}
}
spectral_to_sample(ac);
- multiplier = (ac->m4ac.sbr == 1) ? ac->m4ac.ext_sample_rate > ac->m4ac.sample_rate : 0;
+ multiplier = (ac->oc[1].m4ac.sbr == 1) ? ac->oc[1].m4ac.ext_sample_rate > ac->oc[1].m4ac.sample_rate : 0;
samples <<= multiplier;
- if (ac->output_configured < OC_LOCKED) {
- avctx->sample_rate = ac->m4ac.sample_rate << multiplier;
- avctx->frame_size = samples;
- }
- data_size_tmp = samples * avctx->channels * sizeof(int16_t);
- if (*data_size < data_size_tmp) {
- av_log(avctx, AV_LOG_ERROR,
- "Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n",
- *data_size, data_size_tmp);
- return -1;
- }
- *data_size = data_size_tmp;
+ if (samples) {
+ /* get output buffer */
+ ac->frame.nb_samples = samples;
+ if ((err = avctx->get_buffer(avctx, &ac->frame)) < 0) {
+ av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+ err = -1;
+ goto fail;
+ }
- if (samples)
- ac->dsp.float_to_int16_interleave(data, (const float **)ac->output_data, samples, avctx->channels);
+ if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT)
+ ac->fmt_conv.float_interleave((float *)ac->frame.data[0],
+ (const float **)ac->output_data,
+ samples, avctx->channels);
+ else
+ ac->fmt_conv.float_to_int16_interleave((int16_t *)ac->frame.data[0],
+ (const float **)ac->output_data,
+ samples, avctx->channels);
- if (ac->output_configured)
- ac->output_configured = OC_LOCKED;
+ *(AVFrame *)data = ac->frame;
+ }
+ *got_frame_ptr = !!samples;
+
+ 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;
+ }
return 0;
+fail:
+ pop_output_configuration(ac);
+ return err;
}
static int aac_decode_frame(AVCodecContext *avctx, void *data,
- int *data_size, AVPacket *avpkt)
+ int *got_frame_ptr, AVPacket *avpkt)
{
+ AACContext *ac = avctx->priv_data;
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
GetBitContext gb;
int buf_consumed;
int buf_offset;
int err;
+ int new_extradata_size;
+ const uint8_t *new_extradata = av_packet_get_side_data(avpkt,
+ AV_PKT_DATA_NEW_EXTRADATA,
+ &new_extradata_size);
+
+ if (new_extradata) {
+ av_free(avctx->extradata);
+ avctx->extradata = av_mallocz(new_extradata_size +
+ FF_INPUT_BUFFER_PADDING_SIZE);
+ if (!avctx->extradata)
+ return AVERROR(ENOMEM);
+ avctx->extradata_size = new_extradata_size;
+ memcpy(avctx->extradata, new_extradata, new_extradata_size);
+ push_output_configuration(ac);
+ if (decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
+ avctx->extradata,
+ avctx->extradata_size*8, 1) < 0) {
+ pop_output_configuration(ac);
+ return AVERROR_INVALIDDATA;
+ }
+ }
init_get_bits(&gb, buf, buf_size * 8);
- if ((err = aac_decode_frame_int(avctx, data, data_size, &gb)) < 0)
+ if ((err = aac_decode_frame_int(avctx, data, got_frame_ptr, &gb)) < 0)
return err;
buf_consumed = (get_bits_count(&gb) + 7) >> 3;
ff_mdct_end(&ac->mdct);
ff_mdct_end(&ac->mdct_small);
+ ff_mdct_end(&ac->mdct_ltp);
return 0;
}
}
static int latm_decode_audio_specific_config(struct LATMContext *latmctx,
- GetBitContext *gb)
+ GetBitContext *gb, int asclen)
{
- AVCodecContext *avctx = latmctx->aac_ctx.avctx;
- MPEG4AudioConfig m4ac;
- int config_start_bit = get_bits_count(gb);
- int bits_consumed, esize;
+ AACContext *ac = &latmctx->aac_ctx;
+ AVCodecContext *avctx = ac->avctx;
+ MPEG4AudioConfig m4ac = { 0 };
+ int config_start_bit = get_bits_count(gb);
+ int sync_extension = 0;
+ int bits_consumed, esize;
+
+ if (asclen) {
+ sync_extension = 1;
+ asclen = FFMIN(asclen, get_bits_left(gb));
+ } else
+ asclen = get_bits_left(gb);
if (config_start_bit % 8) {
- av_log_missing_feature(latmctx->aac_ctx.avctx, "audio specific "
- "config not byte aligned.\n", 1);
+ av_log_missing_feature(latmctx->aac_ctx.avctx,
+ "Non-byte-aligned audio-specific config", 1);
+ return AVERROR_PATCHWELCOME;
+ }
+ if (asclen <= 0)
return AVERROR_INVALIDDATA;
- } else {
- bits_consumed =
- decode_audio_specific_config(NULL, avctx, &m4ac,
+ bits_consumed = decode_audio_specific_config(NULL, avctx, &m4ac,
gb->buffer + (config_start_bit / 8),
- get_bits_left(gb) / 8);
+ asclen, sync_extension);
- if (bits_consumed < 0)
- return AVERROR_INVALIDDATA;
+ if (bits_consumed < 0)
+ return AVERROR_INVALIDDATA;
+
+ if (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");
+ latmctx->initialized = 0;
esize = (bits_consumed+7) / 8;
- if (avctx->extradata_size <= esize) {
+ if (avctx->extradata_size < esize) {
av_free(avctx->extradata);
avctx->extradata = av_malloc(esize + FF_INPUT_BUFFER_PADDING_SIZE);
if (!avctx->extradata)
avctx->extradata_size = esize;
memcpy(avctx->extradata, gb->buffer + (config_start_bit/8), esize);
memset(avctx->extradata+esize, 0, FF_INPUT_BUFFER_PADDING_SIZE);
-
- skip_bits_long(gb, bits_consumed);
}
+ skip_bits_long(gb, bits_consumed);
return bits_consumed;
}
// numPrograms
if (get_bits(gb, 4)) { // numPrograms
av_log_missing_feature(latmctx->aac_ctx.avctx,
- "multiple programs are not supported\n", 1);
+ "Multiple programs", 1);
return AVERROR_PATCHWELCOME;
}
// 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);
+ "Multiple layers", 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)
+ if ((ret = latm_decode_audio_specific_config(latmctx, gb, 0)) < 0)
return ret;
} else {
int ascLen = latm_get_value(gb);
- if ((ret = latm_decode_audio_specific_config(latmctx, gb)) < 0)
+ if ((ret = latm_decode_audio_specific_config(latmctx, gb, ascLen)) < 0)
return ret;
ascLen -= ret;
skip_bits_long(gb, ascLen);
}
-static int latm_decode_frame(AVCodecContext *avctx, void *out, int *out_size,
- AVPacket *avpkt)
+static int latm_decode_frame(AVCodecContext *avctx, void *out,
+ int *got_frame_ptr, 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 (!latmctx->initialized) {
if (!avctx->extradata) {
- *out_size = 0;
+ *got_frame_ptr = 0;
return avpkt->size;
} else {
- if ((err = aac_decode_init(avctx)) < 0)
+ push_output_configuration(&latmctx->aac_ctx);
+ if ((err = decode_audio_specific_config(
+ &latmctx->aac_ctx, avctx, &latmctx->aac_ctx.oc[1].m4ac,
+ avctx->extradata, avctx->extradata_size*8, 1)) < 0) {
+ pop_output_configuration(&latmctx->aac_ctx);
return err;
+ }
latmctx->initialized = 1;
}
}
return AVERROR_INVALIDDATA;
}
- if ((err = aac_decode_frame_int(avctx, out, out_size, &gb)) < 0)
+ if ((err = aac_decode_frame_int(avctx, out, got_frame_ptr, &gb)) < 0)
return err;
return muxlength;
}
-av_cold static int latm_decode_init(AVCodecContext *avctx)
+static av_cold int latm_decode_init(AVCodecContext *avctx)
{
struct LATMContext *latmctx = avctx->priv_data;
- int ret;
+ int ret = aac_decode_init(avctx);
- ret = aac_decode_init(avctx);
-
- if (avctx->extradata_size > 0) {
+ if (avctx->extradata_size > 0)
latmctx->initialized = !ret;
- } else {
- latmctx->initialized = 0;
- }
return ret;
}
-AVCodec aac_decoder = {
- "aac",
- AVMEDIA_TYPE_AUDIO,
- CODEC_ID_AAC,
- sizeof(AACContext),
- aac_decode_init,
- NULL,
- aac_decode_close,
- aac_decode_frame,
- .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
- .sample_fmts = (const enum AVSampleFormat[]) {
- AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE
+AVCodec ff_aac_decoder = {
+ .name = "aac",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = AV_CODEC_ID_AAC,
+ .priv_data_size = sizeof(AACContext),
+ .init = aac_decode_init,
+ .close = aac_decode_close,
+ .decode = aac_decode_frame,
+ .long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
+ .sample_fmts = (const enum AVSampleFormat[]) {
+ AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
},
+ .capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
.channel_layouts = aac_channel_layout,
};
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 AVSampleFormat[]) {
- AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE
+AVCodec ff_aac_latm_decoder = {
+ .name = "aac_latm",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = AV_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 Coding LATM syntax)"),
+ .sample_fmts = (const enum AVSampleFormat[]) {
+ AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
},
+ .capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
.channel_layouts = aac_channel_layout,
};
projects / ffmpeg / blobdiff