From f497a9e84edb24c99a9043e1b21c7e48f3908e87 Mon Sep 17 00:00:00 2001 From: Jovan Zelincevic Date: Tue, 30 Jun 2015 11:53:03 +0200 Subject: libavcodec: Implementation of AAC_fixed_decoder (LC-module) [1/4] Move existing code to the new template files Signed-off-by: Nedeljko Babic Signed-off-by: Michael Niedermayer --- libavcodec/aacdec.c | 3132 +-------------------------------------------------- 1 file changed, 57 insertions(+), 3075 deletions(-) (limited to 'libavcodec/aacdec.c') diff --git a/libavcodec/aacdec.c b/libavcodec/aacdec.c index 622cc5c..1d1abc9 100644 --- a/libavcodec/aacdec.c +++ b/libavcodec/aacdec.c @@ -32,55 +32,6 @@ * @author Maxim Gavrilov ( maxim.gavrilov gmail com ) */ -/* - * supported tools - * - * Support? Name - * N (code in SoC repo) gain control - * Y block switching - * Y window shapes - standard - * N window shapes - Low Delay - * Y filterbank - standard - * N (code in SoC repo) filterbank - Scalable Sample Rate - * Y Temporal Noise Shaping - * Y Long Term Prediction - * Y intensity stereo - * Y channel coupling - * Y frequency domain prediction - * Y Perceptual Noise Substitution - * Y Mid/Side stereo - * N Scalable Inverse AAC Quantization - * N Frequency Selective Switch - * N upsampling filter - * Y quantization & coding - AAC - * N quantization & coding - TwinVQ - * N quantization & coding - BSAC - * N AAC Error Resilience tools - * N Error Resilience payload syntax - * N Error Protection tool - * N CELP - * N Silence Compression - * N HVXC - * N HVXC 4kbits/s VR - * N Structured Audio tools - * N Structured Audio Sample Bank Format - * N MIDI - * N Harmonic and Individual Lines plus Noise - * N Text-To-Speech Interface - * Y Spectral Band Replication - * Y (not in this code) Layer-1 - * Y (not in this code) Layer-2 - * Y (not in this code) Layer-3 - * N SinuSoidal Coding (Transient, Sinusoid, Noise) - * Y Parametric Stereo - * N Direct Stream Transfer - * Y Enhanced AAC Low Delay (ER AAC ELD) - * - * Note: - HE AAC v1 comprises LC AAC with Spectral Band Replication. - * - HE AAC v2 comprises LC AAC with Spectral Band Replication and - Parametric Stereo. - */ - #include "libavutil/float_dsp.h" #include "libavutil/opt.h" #include "avcodec.h" @@ -108,1450 +59,19 @@ #include #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 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) -{ - 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. - * - * @param che_pos current channel position configuration - * @param type channel element type - * @param id channel element id - * @param channels count of the number of channels in the configuration - * - * @return Returns error status. 0 - OK, !0 - error - */ -static av_cold int che_configure(AACContext *ac, - enum ChannelPosition che_pos, - int type, int id, int *channels) -{ - if (*channels >= MAX_CHANNELS) - return AVERROR_INVALIDDATA; - 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) { - 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)) { - ac->output_element[(*channels)++] = &ac->che[type][id]->ch[1]; - } - } - } else { - if (ac->che[type][id]) - ff_aac_sbr_ctx_close(&ac->che[type][id]->sbr); - av_freep(&ac->che[type][id]); - } - return 0; -} - -static int frame_configure_elements(AVCodecContext *avctx) -{ - AACContext *ac = avctx->priv_data; - int type, id, ch, ret; - - /* set channel pointers to internal buffers by default */ - for (type = 0; type < 4; type++) { - for (id = 0; id < MAX_ELEM_ID; id++) { - ChannelElement *che = ac->che[type][id]; - if (che) { - che->ch[0].ret = che->ch[0].ret_buf; - che->ch[1].ret = che->ch[1].ret_buf; - } - } - } - - /* get output buffer */ - av_frame_unref(ac->frame); - if (!avctx->channels) - return 1; - - ac->frame->nb_samples = 2048; - 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++) { - if (ac->output_element[ch]) - ac->output_element[ch]->ret = (float *)ac->frame->extended_data[ch]; - } - - 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; - - if (num_side_channels == 0 && num_back_channels >= 4) { - num_side_channels = 2; - num_back_channels -= 2; - } - - 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].status == OC_NONE) { - 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; - output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags, - ac->oc[1].status, 0); - } -} - -/** - * Configure output channel order based on the current program - * configuration element. - * - * @return Returns error status. 0 - OK, !0 - error - */ -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) -{ - AVCodecContext *avctx = ac->avctx; - int i, channels = 0, ret; - uint64_t layout = 0; - uint8_t id_map[TYPE_END][MAX_ELEM_ID] = {{ 0 }}; - uint8_t type_counts[TYPE_END] = { 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; - } - for (i = 0; i < tags; i++) { - int type = layout_map[i][0]; - int id = layout_map[i][1]; - id_map[type][id] = type_counts[type]++; - } - // 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 iid = id_map[type][id]; - 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, iid, &channels); - if (ret < 0) - return ret; - ac->tag_che_map[type][id] = ac->che[type][iid]; - } - 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; - } - } - - 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; - - if (get_new_frame) { - if ((ret = frame_configure_elements(ac->avctx)) < 0) - return ret; - } - - 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. - * - * @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 && channel_config < 11) || - channel_config > 12) { - av_log(avctx, AV_LOG_ERROR, - "invalid default channel configuration (%d)\n", - channel_config); - return AVERROR_INVALIDDATA; - } - *tags = tags_per_config[channel_config]; - memcpy(layout_map, aac_channel_layout_map[channel_config - 1], - *tags * sizeof(*layout_map)); - - /* - * AAC specification has 7.1(wide) as a default layout for 8-channel streams. - * However, at least Nero AAC encoder encodes 7.1 streams using the default - * channel config 7, mapping the side channels of the original audio stream - * to the second AAC_CHANNEL_FRONT pair in the AAC stream. Similarly, e.g. FAAD - * decodes the second AAC_CHANNEL_FRONT pair as side channels, therefore decoding - * the incorrect streams as if they were correct (and as the encoder intended). - * - * As actual intended 7.1(wide) streams are very rare, default to assuming a - * 7.1 layout was intended. - */ - if (channel_config == 7 && avctx->strict_std_compliance < FF_COMPLIANCE_STRICT) { - av_log(avctx, AV_LOG_INFO, "Assuming an incorrectly encoded 7.1 channel layout" - " instead of a spec-compliant 7.1(wide) layout, use -strict %d to decode" - " according to the specification instead.\n", FF_COMPLIANCE_STRICT); - layout_map[2][2] = AAC_CHANNEL_SIDE; - } - - return 0; -} - -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); - - 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; - if (output_configure(ac, layout_map, layout_map_tags, - OC_TRIAL_FRAME, 1) < 0) - return NULL; - - 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); - - 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; - if (output_configure(ac, layout_map, layout_map_tags, - OC_TRIAL_FRAME, 1) < 0) - return NULL; - - 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 12: - 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 11: - if (ac->tags_mapped == 2 && - ac->oc[1].m4ac.chan_config == 11 && - type == TYPE_SCE) { - ac->tags_mapped++; - return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1]; - } - 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)) { - if (!ac->warned_remapping_once && (type != TYPE_LFE || elem_id != 0)) { - av_log(ac->avctx, AV_LOG_WARNING, - "This stream seems to incorrectly report its last channel as %s[%d], mapping to LFE[0]\n", - type == TYPE_SCE ? "SCE" : "LFE", elem_id); - ac->warned_remapping_once++; - } - 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: - /* Some streams incorrectly code 4.0 audio as - * SCE[0] CPE[0] LFE[0] - * instead of - * SCE[0] CPE[0] SCE[1]. - * If we seem to have encountered such a stream, transfer - * the SCE[1] element to the LFE[0]'s mapping */ - if (ac->tags_mapped == tags_per_config[ac->oc[1].m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) { - if (!ac->warned_remapping_once && (type != TYPE_SCE || elem_id != 1)) { - av_log(ac->avctx, AV_LOG_WARNING, - "This stream seems to incorrectly report its last channel as %s[%d], mapping to SCE[1]\n", - type == TYPE_SCE ? "SCE" : "LFE", elem_id); - ac->warned_remapping_once++; - } - ac->tags_mapped++; - return ac->tag_che_map[type][elem_id] = ac->che[TYPE_SCE][1]; - } - 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 type speaker type/position for these channels - */ -static void decode_channel_map(uint8_t layout_map[][3], - enum ChannelPosition type, - GetBitContext *gb, int n) -{ - while (n--) { - 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; - default: - // AAC_CHANNEL_OFF has no channel map - av_assert0(0); - } - 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. - * - * @return Returns error status. 0 - OK, !0 - error - */ -static int decode_pce(AVCodecContext *avctx, MPEG4AudioConfig *m4ac, - uint8_t (*layout_map)[3], - GetBitContext *gb) -{ - int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc; - int sampling_index; - int comment_len; - int tags; - - skip_bits(gb, 2); // object_type - - sampling_index = get_bits(gb, 4); - if (m4ac->sampling_index != sampling_index) - av_log(avctx, AV_LOG_WARNING, - "Sample rate index in program config element does not " - "match the sample rate index configured by the container.\n"); - - num_front = get_bits(gb, 4); - num_side = get_bits(gb, 4); - num_back = get_bits(gb, 4); - num_lfe = get_bits(gb, 2); - num_assoc_data = get_bits(gb, 3); - num_cc = get_bits(gb, 4); - - if (get_bits1(gb)) - skip_bits(gb, 4); // mono_mixdown_tag - if (get_bits1(gb)) - skip_bits(gb, 4); // stereo_mixdown_tag - - 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); - 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(layout_map + tags, AAC_CHANNEL_CC, gb, num_cc); - tags += num_cc; - - align_get_bits(gb); - - /* 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, "decode_pce: " overread_err); - return AVERROR_INVALIDDATA; - } - skip_bits_long(gb, comment_len); - return tags; -} - -/** - * Decode GA "General Audio" specific configuration; reference: table 4.1. - * - * @param ac pointer to AACContext, may be null - * @param avctx pointer to AVCCodecContext, used for logging - * - * @return Returns error status. 0 - OK, !0 - error - */ -static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx, - GetBitContext *gb, - MPEG4AudioConfig *m4ac, - int channel_config) -{ - int extension_flag, ret, ep_config, res_flags; - uint8_t layout_map[MAX_ELEM_ID*4][3]; - int tags = 0; - - if (get_bits1(gb)) { // frameLengthFlag - avpriv_request_sample(avctx, "960/120 MDCT window"); - return AVERROR_PATCHWELCOME; - } - m4ac->frame_length_short = 0; - - if (get_bits1(gb)) // dependsOnCoreCoder - skip_bits(gb, 14); // coreCoderDelay - extension_flag = get_bits1(gb); - - if (m4ac->object_type == AOT_AAC_SCALABLE || - m4ac->object_type == AOT_ER_AAC_SCALABLE) - skip_bits(gb, 3); // layerNr - - if (channel_config == 0) { - skip_bits(gb, 4); // element_instance_tag - tags = decode_pce(avctx, m4ac, layout_map, gb); - if (tags < 0) - return tags; - } else { - if ((ret = set_default_channel_config(avctx, layout_map, - &tags, channel_config))) - return ret; - } - - 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, 0))) - return ret; - - if (extension_flag) { - switch (m4ac->object_type) { - case AOT_ER_BSAC: - skip_bits(gb, 5); // numOfSubFrame - skip_bits(gb, 11); // layer_length - break; - case AOT_ER_AAC_LC: - case AOT_ER_AAC_LTP: - case AOT_ER_AAC_SCALABLE: - case AOT_ER_AAC_LD: - res_flags = get_bits(gb, 3); - if (res_flags) { - avpriv_report_missing_feature(avctx, - "AAC data resilience (flags %x)", - res_flags); - return AVERROR_PATCHWELCOME; - } - break; - } - skip_bits1(gb); // extensionFlag3 (TBD in version 3) - } - switch (m4ac->object_type) { - case AOT_ER_AAC_LC: - case AOT_ER_AAC_LTP: - case AOT_ER_AAC_SCALABLE: - case AOT_ER_AAC_LD: - ep_config = get_bits(gb, 2); - if (ep_config) { - avpriv_report_missing_feature(avctx, - "epConfig %d", ep_config); - return AVERROR_PATCHWELCOME; - } - } - return 0; -} - -static int decode_eld_specific_config(AACContext *ac, AVCodecContext *avctx, - GetBitContext *gb, - MPEG4AudioConfig *m4ac, - int channel_config) -{ - int ret, ep_config, res_flags; - uint8_t layout_map[MAX_ELEM_ID*4][3]; - int tags = 0; - const int ELDEXT_TERM = 0; - - m4ac->ps = 0; - m4ac->sbr = 0; - - m4ac->frame_length_short = get_bits1(gb); - res_flags = get_bits(gb, 3); - if (res_flags) { - avpriv_report_missing_feature(avctx, - "AAC data resilience (flags %x)", - res_flags); - return AVERROR_PATCHWELCOME; - } - - if (get_bits1(gb)) { // ldSbrPresentFlag - avpriv_report_missing_feature(avctx, - "Low Delay SBR"); - return AVERROR_PATCHWELCOME; - } - - while (get_bits(gb, 4) != ELDEXT_TERM) { - int len = get_bits(gb, 4); - if (len == 15) - len += get_bits(gb, 8); - if (len == 15 + 255) - len += get_bits(gb, 16); - if (get_bits_left(gb) < len * 8 + 4) { - av_log(avctx, AV_LOG_ERROR, overread_err); - return AVERROR_INVALIDDATA; - } - skip_bits_long(gb, 8 * len); - } - - if ((ret = set_default_channel_config(avctx, layout_map, - &tags, channel_config))) - return ret; - - if (ac && (ret = output_configure(ac, layout_map, tags, OC_GLOBAL_HDR, 0))) - return ret; - - ep_config = get_bits(gb, 2); - if (ep_config) { - avpriv_report_missing_feature(avctx, - "epConfig %d", ep_config); - return AVERROR_PATCHWELCOME; - } - return 0; -} - -/** - * Decode audio specific configuration; reference: table 1.13. - * - * @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 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 bit_size, - int sync_extension) -{ - GetBitContext gb; - int i, ret; - - ff_dlog(avctx, "audio specific config size %d\n", bit_size >> 3); - for (i = 0; i < bit_size >> 3; i++) - ff_dlog(avctx, "%02x ", data[i]); - ff_dlog(avctx, "\n"); - - 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 AVERROR_INVALIDDATA; - if (m4ac->sampling_index > 12) { - av_log(avctx, AV_LOG_ERROR, - "invalid sampling rate index %d\n", - m4ac->sampling_index); - return AVERROR_INVALIDDATA; - } - if (m4ac->object_type == AOT_ER_AAC_LD && - (m4ac->sampling_index < 3 || m4ac->sampling_index > 7)) { - av_log(avctx, AV_LOG_ERROR, - "invalid low delay sampling rate index %d\n", - m4ac->sampling_index); - return AVERROR_INVALIDDATA; - } - - skip_bits_long(&gb, i); - - switch (m4ac->object_type) { - case AOT_AAC_MAIN: - case AOT_AAC_LC: - case AOT_AAC_LTP: - case AOT_ER_AAC_LC: - case AOT_ER_AAC_LD: - if ((ret = decode_ga_specific_config(ac, avctx, &gb, - m4ac, m4ac->chan_config)) < 0) - return ret; - break; - case AOT_ER_AAC_ELD: - if ((ret = decode_eld_specific_config(ac, avctx, &gb, - m4ac, m4ac->chan_config)) < 0) - return ret; - break; - default: - avpriv_report_missing_feature(avctx, - "Audio object type %s%d", - m4ac->sbr == 1 ? "SBR+" : "", - m4ac->object_type); - return AVERROR(ENOSYS); - } - - ff_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); -} - -/** - * linear congruential pseudorandom number generator - * - * @param previous_val pointer to the current state of the generator - * - * @return Returns a 32-bit pseudorandom integer - */ -static av_always_inline int lcg_random(unsigned previous_val) -{ - union { unsigned u; int s; } v = { previous_val * 1664525u + 1013904223 }; - return v.s; -} - -static av_always_inline void reset_predict_state(PredictorState *ps) -{ - ps->r0 = 0.0f; - ps->r1 = 0.0f; - ps->cor0 = 0.0f; - ps->cor1 = 0.0f; - ps->var0 = 1.0f; - ps->var1 = 1.0f; -} - -static void reset_all_predictors(PredictorState *ps) -{ - int i; - for (i = 0; i < MAX_PREDICTORS; i++) - 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; - for (i = group_num - 1; i < MAX_PREDICTORS; i += 30) - reset_predict_state(&ps[i]); -} - -#define AAC_INIT_VLC_STATIC(num, size) \ - INIT_VLC_STATIC(&vlc_spectral[num], 8, ff_aac_spectral_sizes[num], \ - ff_aac_spectral_bits[num], sizeof(ff_aac_spectral_bits[num][0]), \ - sizeof(ff_aac_spectral_bits[num][0]), \ - 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; - int ret; - - 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) { - if ((ret = decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac, - avctx->extradata, - avctx->extradata_size * 8, - 1)) < 0) - return ret; - } 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, 0); - else if (avctx->err_recognition & AV_EF_EXPLODE) - return AVERROR_INVALIDDATA; - } - } - - 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); - AAC_INIT_VLC_STATIC( 3, 300); - AAC_INIT_VLC_STATIC( 4, 328); - AAC_INIT_VLC_STATIC( 5, 294); - AAC_INIT_VLC_STATIC( 6, 306); - AAC_INIT_VLC_STATIC( 7, 268); - AAC_INIT_VLC_STATIC( 8, 510); - AAC_INIT_VLC_STATIC( 9, 366); - AAC_INIT_VLC_STATIC(10, 462); - - ff_aac_sbr_init(); - - ac->fdsp = avpriv_float_dsp_alloc(avctx->flags & CODEC_FLAG_BITEXACT); - if (!ac->fdsp) { - return AVERROR(ENOMEM); - } - - ac->random_state = 0x1f2e3d4c; - - ff_aac_tableinit(); - - INIT_VLC_STATIC(&vlc_scalefactors, 7, - FF_ARRAY_ELEMS(ff_aac_scalefactor_code), - ff_aac_scalefactor_bits, - sizeof(ff_aac_scalefactor_bits[0]), - sizeof(ff_aac_scalefactor_bits[0]), - 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 / (32768.0 * 1024.0)); - ff_mdct_init(&ac->mdct_ld, 10, 1, 1.0 / (32768.0 * 512.0)); - ff_mdct_init(&ac->mdct_small, 8, 1, 1.0 / (32768.0 * 128.0)); - ff_mdct_init(&ac->mdct_ltp, 11, 0, -2.0 * 32768.0); - ret = ff_imdct15_init(&ac->mdct480, 5); - if (ret < 0) - return ret; - - // 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); - ff_init_ff_sine_windows(10); - ff_init_ff_sine_windows( 9); - ff_init_ff_sine_windows( 7); - - cbrt_tableinit(); - - return 0; -} - -/** - * Skip data_stream_element; reference: table 4.10. - */ -static int skip_data_stream_element(AACContext *ac, GetBitContext *gb) -{ - int byte_align = get_bits1(gb); - int count = get_bits(gb, 8); - if (count == 255) - count += get_bits(gb, 8); - if (byte_align) - align_get_bits(gb); - - if (get_bits_left(gb) < 8 * count) { - av_log(ac->avctx, AV_LOG_ERROR, "skip_data_stream_element: "overread_err); - return AVERROR_INVALIDDATA; - } - skip_bits_long(gb, 8 * count); - return 0; -} - -static int decode_prediction(AACContext *ac, IndividualChannelStream *ics, - GetBitContext *gb) -{ - int sfb; - if (get_bits1(gb)) { - ics->predictor_reset_group = get_bits(gb, 5); - if (ics->predictor_reset_group == 0 || - ics->predictor_reset_group > 30) { - av_log(ac->avctx, AV_LOG_ERROR, - "Invalid Predictor Reset Group.\n"); - return AVERROR_INVALIDDATA; - } - } - 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. - */ -static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics, - GetBitContext *gb) -{ - const MPEG4AudioConfig *const m4ac = &ac->oc[1].m4ac; - const int aot = m4ac->object_type; - const int sampling_index = m4ac->sampling_index; - if (aot != AOT_ER_AAC_ELD) { - if (get_bits1(gb)) { - av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n"); - if (ac->avctx->err_recognition & AV_EF_BITSTREAM) - return AVERROR_INVALIDDATA; - } - ics->window_sequence[1] = ics->window_sequence[0]; - ics->window_sequence[0] = get_bits(gb, 2); - if (aot == AOT_ER_AAC_LD && - ics->window_sequence[0] != ONLY_LONG_SEQUENCE) { - av_log(ac->avctx, AV_LOG_ERROR, - "AAC LD is only defined for ONLY_LONG_SEQUENCE but " - "window sequence %d found.\n", ics->window_sequence[0]); - ics->window_sequence[0] = ONLY_LONG_SEQUENCE; - return AVERROR_INVALIDDATA; - } - ics->use_kb_window[1] = ics->use_kb_window[0]; - ics->use_kb_window[0] = get_bits1(gb); - } - ics->num_window_groups = 1; - ics->group_len[0] = 1; - if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { - int i; - ics->max_sfb = get_bits(gb, 4); - for (i = 0; i < 7; i++) { - if (get_bits1(gb)) { - ics->group_len[ics->num_window_groups - 1]++; - } else { - ics->num_window_groups++; - ics->group_len[ics->num_window_groups - 1] = 1; - } - } - ics->num_windows = 8; - ics->swb_offset = ff_swb_offset_128[sampling_index]; - ics->num_swb = ff_aac_num_swb_128[sampling_index]; - ics->tns_max_bands = ff_tns_max_bands_128[sampling_index]; - ics->predictor_present = 0; - } else { - ics->max_sfb = get_bits(gb, 6); - ics->num_windows = 1; - if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD) { - if (m4ac->frame_length_short) { - ics->swb_offset = ff_swb_offset_480[sampling_index]; - ics->num_swb = ff_aac_num_swb_480[sampling_index]; - ics->tns_max_bands = ff_tns_max_bands_480[sampling_index]; - } else { - ics->swb_offset = ff_swb_offset_512[sampling_index]; - ics->num_swb = ff_aac_num_swb_512[sampling_index]; - ics->tns_max_bands = ff_tns_max_bands_512[sampling_index]; - } - if (!ics->num_swb || !ics->swb_offset) - return AVERROR_BUG; - } else { - ics->swb_offset = ff_swb_offset_1024[sampling_index]; - ics->num_swb = ff_aac_num_swb_1024[sampling_index]; - ics->tns_max_bands = ff_tns_max_bands_1024[sampling_index]; - } - if (aot != AOT_ER_AAC_ELD) { - ics->predictor_present = get_bits1(gb); - ics->predictor_reset_group = 0; - } - if (ics->predictor_present) { - if (aot == AOT_AAC_MAIN) { - if (decode_prediction(ac, ics, gb)) { - goto fail; - } - } else if (aot == AOT_AAC_LC || - aot == AOT_ER_AAC_LC) { - av_log(ac->avctx, AV_LOG_ERROR, - "Prediction is not allowed in AAC-LC.\n"); - goto fail; - } else { - if (aot == AOT_ER_AAC_LD) { - av_log(ac->avctx, AV_LOG_ERROR, - "LTP in ER AAC LD not yet implemented.\n"); - return AVERROR_PATCHWELCOME; - } - if ((ics->ltp.present = get_bits(gb, 1))) - decode_ltp(&ics->ltp, gb, ics->max_sfb); - } - } - } - - if (ics->max_sfb > ics->num_swb) { - av_log(ac->avctx, AV_LOG_ERROR, - "Number of scalefactor bands in group (%d) " - "exceeds limit (%d).\n", - ics->max_sfb, ics->num_swb); - goto fail; - } - - return 0; -fail: - ics->max_sfb = 0; - return AVERROR_INVALIDDATA; -} - -/** - * Decode band types (section_data payload); reference: table 4.46. - * - * @param band_type array of the used band type - * @param band_type_run_end array of the last scalefactor band of a band type run - * - * @return Returns error status. 0 - OK, !0 - error - */ -static int decode_band_types(AACContext *ac, enum BandType band_type[120], - int band_type_run_end[120], GetBitContext *gb, - IndividualChannelStream *ics) -{ - int g, idx = 0; - const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5; - for (g = 0; g < ics->num_window_groups; g++) { - int k = 0; - while (k < ics->max_sfb) { - uint8_t sect_end = k; - int sect_len_incr; - int sect_band_type = get_bits(gb, 4); - if (sect_band_type == 12) { - av_log(ac->avctx, AV_LOG_ERROR, "invalid band type\n"); - return AVERROR_INVALIDDATA; - } - do { - 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, "decode_band_types: "overread_err); - return AVERROR_INVALIDDATA; - } - 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 AVERROR_INVALIDDATA; - } - } 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; - } - } - } - return 0; -} - -/** - * Decode scalefactors; reference: table 4.47. - * - * @param global_gain first scalefactor value as scalefactors are differentially coded - * @param band_type array of the used band type - * @param band_type_run_end array of the last scalefactor band of a band type run - * @param sf array of scalefactors or intensity stereo positions - * - * @return Returns error status. 0 - OK, !0 - error - */ -static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb, - unsigned int global_gain, - IndividualChannelStream *ics, - enum BandType band_type[120], - int band_type_run_end[120]) -{ - int g, i, idx = 0; - int offset[3] = { global_gain, global_gain - NOISE_OFFSET, 0 }; - int clipped_offset; - int noise_flag = 1; - for (g = 0; g < ics->num_window_groups; g++) { - for (i = 0; i < ics->max_sfb;) { - int run_end = band_type_run_end[idx]; - if (band_type[idx] == ZERO_BT) { - for (; i < run_end; i++, idx++) - sf[idx] = 0.0; - } 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) - SCALE_DIFF_ZERO; - clipped_offset = av_clip(offset[2], -155, 100); - if (offset[2] != clipped_offset) { - avpriv_request_sample(ac->avctx, - "If you heard an audible artifact, there may be a bug in the decoder. " - "Clipped intensity stereo position (%d -> %d)", - offset[2], clipped_offset); - } - sf[idx] = ff_aac_pow2sf_tab[-clipped_offset + POW_SF2_ZERO]; - } - } else if (band_type[idx] == NOISE_BT) { - for (; i < run_end; i++, idx++) { - if (noise_flag-- > 0) - offset[1] += get_bits(gb, NOISE_PRE_BITS) - NOISE_PRE; - else - offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - SCALE_DIFF_ZERO; - clipped_offset = av_clip(offset[1], -100, 155); - if (offset[1] != clipped_offset) { - avpriv_request_sample(ac->avctx, - "If you heard an audible artifact, there may be a bug in the decoder. " - "Clipped noise gain (%d -> %d)", - offset[1], clipped_offset); - } - 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) - SCALE_DIFF_ZERO; - if (offset[0] > 255U) { - av_log(ac->avctx, AV_LOG_ERROR, - "Scalefactor (%d) out of range.\n", offset[0]); - return AVERROR_INVALIDDATA; - } - sf[idx] = -ff_aac_pow2sf_tab[offset[0] - 100 + POW_SF2_ZERO]; - } - } - } - } - return 0; -} - -/** - * Decode pulse data; reference: table 4.7. - */ -static int decode_pulses(Pulse *pulse, GetBitContext *gb, - const uint16_t *swb_offset, int num_swb) -{ - int i, pulse_swb; - pulse->num_pulse = get_bits(gb, 2) + 1; - pulse_swb = get_bits(gb, 6); - if (pulse_swb >= num_swb) - return -1; - pulse->pos[0] = swb_offset[pulse_swb]; - pulse->pos[0] += get_bits(gb, 5); - if (pulse->pos[0] >= swb_offset[num_swb]) - return -1; - pulse->amp[0] = get_bits(gb, 4); - for (i = 1; i < pulse->num_pulse; i++) { - pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i - 1]; - if (pulse->pos[i] >= swb_offset[num_swb]) - return -1; - pulse->amp[i] = get_bits(gb, 4); - } - return 0; -} - -/** - * Decode Temporal Noise Shaping data; reference: table 4.48. - * - * @return Returns error status. 0 - OK, !0 - error - */ -static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns, - GetBitContext *gb, const IndividualChannelStream *ics) -{ - 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->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); - - for (filt = 0; filt < tns->n_filt[w]; filt++) { - int tmp2_idx; - tns->length[w][filt] = get_bits(gb, 6 - 2 * is8); - - if ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) { - av_log(ac->avctx, AV_LOG_ERROR, - "TNS filter order %d is greater than maximum %d.\n", - tns->order[w][filt], tns_max_order); - tns->order[w][filt] = 0; - return AVERROR_INVALIDDATA; - } - if (tns->order[w][filt]) { - tns->direction[w][filt] = get_bits1(gb); - coef_compress = get_bits1(gb); - coef_len = coef_res + 3 - coef_compress; - tmp2_idx = 2 * coef_compress + coef_res; - - for (i = 0; i < tns->order[w][filt]; i++) - tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)]; - } - } - } - } - return 0; -} +# include "arm/aac.h" +#elif ARCH_MIPS +# include "mips/aacdec_mips.h" +#endif -/** - * Decode Mid/Side data; reference: table 4.54. - * - * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s; - * [1] mask is decoded from bitstream; [2] mask is all 1s; - * [3] reserved for scalable AAC - */ -static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb, - int ms_present) +static av_always_inline void reset_predict_state(PredictorState *ps) { - int idx; - int max_idx = cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb; - if (ms_present == 1) { - for (idx = 0; idx < max_idx; idx++) - cpe->ms_mask[idx] = get_bits1(gb); - } else if (ms_present == 2) { - memset(cpe->ms_mask, 1, max_idx * sizeof(cpe->ms_mask[0])); - } + ps->r0 = 0.0f; + ps->r1 = 0.0f; + ps->cor0 = 0.0f; + ps->cor1 = 0.0f; + ps->var0 = 1.0f; + ps->var1 = 1.0f; } #ifndef VMUL2 @@ -1611,1062 +131,70 @@ static inline float *VMUL4S(float *dst, const float *v, unsigned idx, *dst++ = v[idx>>2 & 3] * t.f; sign <<= nz & 1; nz >>= 1; - t.i = s.i ^ (sign & 1U<<31); - *dst++ = v[idx>>4 & 3] * t.f; - - sign <<= nz & 1; - t.i = s.i ^ (sign & 1U<<31); - *dst++ = v[idx>>6 & 3] * t.f; - - return dst; -} -#endif - -/** - * Decode spectral data; reference: table 4.50. - * Dequantize and scale spectral data; reference: 4.6.3.3. - * - * @param coef array of dequantized, scaled spectral data - * @param sf array of scalefactors or intensity stereo positions - * @param pulse_present set if pulses are present - * @param pulse pointer to pulse data struct - * @param band_type array of the used band type - * - * @return Returns error status. 0 - OK, !0 - error - */ -static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024], - GetBitContext *gb, const float sf[120], - int pulse_present, const Pulse *pulse, - const IndividualChannelStream *ics, - enum BandType band_type[120]) -{ - int i, k, g, idx = 0; - const int c = 1024 / ics->num_windows; - const uint16_t *offsets = ics->swb_offset; - float *coef_base = coef; - - for (g = 0; g < ics->num_windows; g++) - memset(coef + g * 128 + offsets[ics->max_sfb], 0, - sizeof(float) * (c - offsets[ics->max_sfb])); - - for (g = 0; g < ics->num_window_groups; g++) { - unsigned g_len = ics->group_len[g]; - - for (i = 0; i < ics->max_sfb; i++, idx++) { - const unsigned cbt_m1 = band_type[idx] - 1; - float *cfo = coef + offsets[i]; - int off_len = offsets[i + 1] - offsets[i]; - int group; - - if (cbt_m1 >= INTENSITY_BT2 - 1) { - for (group = 0; group < g_len; group++, cfo+=128) { - memset(cfo, 0, off_len * sizeof(float)); - } - } else if (cbt_m1 == NOISE_BT - 1) { - for (group = 0; group < g_len; group++, cfo+=128) { - float scale; - float band_energy; - - for (k = 0; k < off_len; k++) { - ac->random_state = lcg_random(ac->random_state); - cfo[k] = ac->random_state; - } - - band_energy = ac->fdsp->scalarproduct_float(cfo, cfo, off_len); - scale = sf[idx] / sqrtf(band_energy); - ac->fdsp->vector_fmul_scalar(cfo, cfo, scale, off_len); - } - } else { - const float *vq = ff_aac_codebook_vector_vals[cbt_m1]; - const uint16_t *cb_vector_idx = ff_aac_codebook_vector_idx[cbt_m1]; - VLC_TYPE (*vlc_tab)[2] = vlc_spectral[cbt_m1].table; - OPEN_READER(re, gb); - - switch (cbt_m1 >> 1) { - case 0: - for (group = 0; group < g_len; group++, cfo+=128) { - float *cf = cfo; - int len = off_len; - - do { - int code; - unsigned cb_idx; - - UPDATE_CACHE(re, gb); - GET_VLC(code, re, gb, vlc_tab, 8, 2); - cb_idx = cb_vector_idx[code]; - cf = VMUL4(cf, vq, cb_idx, sf + idx); - } while (len -= 4); - } - break; - - case 1: - for (group = 0; group < g_len; group++, cfo+=128) { - float *cf = cfo; - int len = off_len; - - do { - int code; - unsigned nnz; - unsigned cb_idx; - uint32_t bits; - - UPDATE_CACHE(re, gb); - GET_VLC(code, re, gb, vlc_tab, 8, 2); - cb_idx = cb_vector_idx[code]; - nnz = cb_idx >> 8 & 15; - 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); - } - break; - - case 2: - for (group = 0; group < g_len; group++, cfo+=128) { - float *cf = cfo; - int len = off_len; - - do { - int code; - unsigned cb_idx; - - UPDATE_CACHE(re, gb); - GET_VLC(code, re, gb, vlc_tab, 8, 2); - cb_idx = cb_vector_idx[code]; - cf = VMUL2(cf, vq, cb_idx, sf + idx); - } while (len -= 2); - } - break; - - case 3: - case 4: - for (group = 0; group < g_len; group++, cfo+=128) { - float *cf = cfo; - int len = off_len; - - do { - int code; - unsigned nnz; - unsigned cb_idx; - unsigned sign; - - UPDATE_CACHE(re, gb); - GET_VLC(code, re, gb, vlc_tab, 8, 2); - cb_idx = cb_vector_idx[code]; - nnz = cb_idx >> 8 & 15; - 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); - } - break; - - default: - for (group = 0; group < g_len; group++, cfo+=128) { - float *cf = cfo; - uint32_t *icf = (uint32_t *) cf; - int len = off_len; - - do { - int code; - unsigned nzt, nnz; - unsigned cb_idx; - uint32_t bits; - int j; - - UPDATE_CACHE(re, gb); - GET_VLC(code, re, gb, vlc_tab, 8, 2); - - if (!code) { - *icf++ = 0; - *icf++ = 0; - continue; - } - - cb_idx = cb_vector_idx[code]; - nnz = cb_idx >> 12; - nzt = cb_idx >> 8; - bits = SHOW_UBITS(re, gb, nnz) << (32-nnz); - LAST_SKIP_BITS(re, gb, nnz); - - for (j = 0; j < 2; j++) { - if (nzt & 1< 8) { - av_log(ac->avctx, AV_LOG_ERROR, "error in spectral data, ESC overflow\n"); - return AVERROR_INVALIDDATA; - } - - SKIP_BITS(re, gb, b + 1); - b += 4; - n = (1 << b) + SHOW_UBITS(re, gb, b); - LAST_SKIP_BITS(re, gb, b); - *icf++ = cbrt_tab[n] | (bits & 1U<<31); - bits <<= 1; - } else { - unsigned v = ((const uint32_t*)vq)[cb_idx & 15]; - *icf++ = (bits & 1U<<31) | v; - bits <<= !!v; - } - cb_idx >>= 4; - } - } while (len -= 2); - - ac->fdsp->vector_fmul_scalar(cfo, cfo, sf[idx], off_len); - } - } - - CLOSE_READER(re, gb); - } - } - coef += g_len << 7; - } - - if (pulse_present) { - idx = 0; - for (i = 0; i < pulse->num_pulse; i++) { - float co = coef_base[ pulse->pos[i] ]; - while (offsets[idx + 1] <= pulse->pos[i]) - idx++; - if (band_type[idx] != NOISE_BT && sf[idx]) { - float ico = -pulse->amp[i]; - if (co) { - co /= sf[idx]; - ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico); - } - coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx]; - } - } - } - return 0; -} - -static av_always_inline float flt16_round(float pf) -{ - 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 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 av_intfloat32 pun; - pun.f = pf; - pun.i &= 0xFFFF0000U; - return pun.f; -} - -static av_always_inline void predict(PredictorState *ps, float *coef, - int output_enable) -{ - const float a = 0.953125; // 61.0 / 64 - const float alpha = 0.90625; // 29.0 / 32 - float e0, e1; - float pv; - float k1, k2; - float r0 = ps->r0, r1 = ps->r1; - float cor0 = ps->cor0, cor1 = ps->cor1; - float var0 = ps->var0, var1 = ps->var1; - - k1 = var0 > 1 ? cor0 * flt16_even(a / var0) : 0; - k2 = var1 > 1 ? cor1 * flt16_even(a / var1) : 0; - - pv = flt16_round(k1 * r0 + k2 * r1); - if (output_enable) - *coef += pv; - - e0 = *coef; - e1 = e0 - k1 * r0; - - ps->cor1 = flt16_trunc(alpha * cor1 + r1 * e1); - ps->var1 = flt16_trunc(alpha * var1 + 0.5f * (r1 * r1 + e1 * e1)); - ps->cor0 = flt16_trunc(alpha * cor0 + r0 * e0); - ps->var0 = flt16_trunc(alpha * var0 + 0.5f * (r0 * r0 + e0 * e0)); - - ps->r1 = flt16_trunc(a * (r0 - k1 * e0)); - ps->r0 = flt16_trunc(a * e0); -} - -/** - * Apply AAC-Main style frequency domain prediction. - */ -static void apply_prediction(AACContext *ac, SingleChannelElement *sce) -{ - int sfb, k; - - if (!sce->ics.predictor_initialized) { - reset_all_predictors(sce->predictor_state); - sce->ics.predictor_initialized = 1; - } - - if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) { - 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], - sce->ics.predictor_present && - sce->ics.prediction_used[sfb]); - } - } - if (sce->ics.predictor_reset_group) - reset_predictor_group(sce->predictor_state, - sce->ics.predictor_reset_group); - } else - reset_all_predictors(sce->predictor_state); -} - -/** - * Decode an individual_channel_stream payload; reference: table 4.44. - * - * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information. - * @param scale_flag scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.) - * - * @return Returns error status. 0 - OK, !0 - error - */ -static int decode_ics(AACContext *ac, SingleChannelElement *sce, - GetBitContext *gb, int common_window, int scale_flag) -{ - Pulse pulse; - TemporalNoiseShaping *tns = &sce->tns; - IndividualChannelStream *ics = &sce->ics; - float *out = sce->coeffs; - int global_gain, eld_syntax, er_syntax, pulse_present = 0; - int ret; - - eld_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD; - er_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_LC || - ac->oc[1].m4ac.object_type == AOT_ER_AAC_LTP || - ac->oc[1].m4ac.object_type == AOT_ER_AAC_LD || - ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD; - - /* This assignment is to silence a GCC warning about the variable being used - * uninitialized when in fact it always is. - */ - pulse.num_pulse = 0; - - global_gain = get_bits(gb, 8); - - if (!common_window && !scale_flag) { - if (decode_ics_info(ac, ics, gb) < 0) - return AVERROR_INVALIDDATA; - } - - if ((ret = decode_band_types(ac, sce->band_type, - sce->band_type_run_end, gb, ics)) < 0) - return ret; - if ((ret = decode_scalefactors(ac, sce->sf, gb, global_gain, ics, - sce->band_type, sce->band_type_run_end)) < 0) - return ret; - - pulse_present = 0; - if (!scale_flag) { - if (!eld_syntax && (pulse_present = get_bits1(gb))) { - if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { - av_log(ac->avctx, AV_LOG_ERROR, - "Pulse tool not allowed in eight short sequence.\n"); - return AVERROR_INVALIDDATA; - } - if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) { - av_log(ac->avctx, AV_LOG_ERROR, - "Pulse data corrupt or invalid.\n"); - return AVERROR_INVALIDDATA; - } - } - tns->present = get_bits1(gb); - if (tns->present && !er_syntax) - if (decode_tns(ac, tns, gb, ics) < 0) - return AVERROR_INVALIDDATA; - if (!eld_syntax && get_bits1(gb)) { - avpriv_request_sample(ac->avctx, "SSR"); - return AVERROR_PATCHWELCOME; - } - // I see no textual basis in the spec for this occurring after SSR gain - // control, but this is what both reference and real implmentations do - if (tns->present && er_syntax) - if (decode_tns(ac, tns, gb, ics) < 0) - return AVERROR_INVALIDDATA; - } - - if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present, - &pulse, ics, sce->band_type) < 0) - return AVERROR_INVALIDDATA; - - if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN && !common_window) - apply_prediction(ac, sce); - - return 0; -} - -/** - * Mid/Side stereo decoding; reference: 4.6.8.1.3. - */ -static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe) -{ - const IndividualChannelStream *ics = &cpe->ch[0].ics; - float *ch0 = cpe->ch[0].coeffs; - float *ch1 = cpe->ch[1].coeffs; - int g, i, group, idx = 0; - const uint16_t *offsets = ics->swb_offset; - for (g = 0; g < ics->num_window_groups; g++) { - for (i = 0; i < ics->max_sfb; i++, idx++) { - if (cpe->ms_mask[idx] && - cpe->ch[0].band_type[idx] < NOISE_BT && - cpe->ch[1].band_type[idx] < NOISE_BT) { - for (group = 0; group < ics->group_len[g]; group++) { - ac->fdsp->butterflies_float(ch0 + group * 128 + offsets[i], - ch1 + group * 128 + offsets[i], - offsets[i+1] - offsets[i]); - } - } - } - ch0 += ics->group_len[g] * 128; - ch1 += ics->group_len[g] * 128; - } -} - -/** - * intensity stereo decoding; reference: 4.6.8.2.3 - * - * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s; - * [1] mask is decoded from bitstream; [2] mask is all 1s; - * [3] reserved for scalable AAC - */ -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, idx = 0; - int c; - float scale; - for (g = 0; g < ics->num_window_groups; g++) { - for (i = 0; i < ics->max_sfb;) { - if (sce1->band_type[idx] == INTENSITY_BT || - sce1->band_type[idx] == INTENSITY_BT2) { - const int bt_run_end = sce1->band_type_run_end[idx]; - for (; i < bt_run_end; i++, idx++) { - c = -1 + 2 * (sce1->band_type[idx] - 14); - if (ms_present) - c *= 1 - 2 * cpe->ms_mask[idx]; - scale = c * sce1->sf[idx]; - for (group = 0; group < ics->group_len[g]; group++) - ac->fdsp->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]; - idx += bt_run_end - i; - i = bt_run_end; - } - } - coef0 += ics->group_len[g] * 128; - coef1 += ics->group_len[g] * 128; - } -} - -/** - * Decode a channel_pair_element; reference: table 4.4. - * - * @return Returns error status. 0 - OK, !0 - error - */ -static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe) -{ - int i, ret, common_window, ms_present = 0; - int eld_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD; - - common_window = eld_syntax || get_bits1(gb); - if (common_window) { - 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"); - return AVERROR_INVALIDDATA; - } else if (ms_present) - decode_mid_side_stereo(cpe, gb, ms_present); - } - if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0))) - return ret; - if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0))) - return ret; - - if (common_window) { - if (ms_present) - apply_mid_side_stereo(ac, cpe); - 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(ac, cpe, ms_present); - return 0; -} - -static const float cce_scale[] = { - 1.09050773266525765921, //2^(1/8) - 1.18920711500272106672, //2^(1/4) - M_SQRT2, - 2, -}; - -/** - * Decode coupling_channel_element; reference: table 4.8. - * - * @return Returns error status. 0 - OK, !0 - error - */ -static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che) -{ - int num_gain = 0; - int c, g, sfb, ret; - int sign; - float scale; - SingleChannelElement *sce = &che->ch[0]; - ChannelCoupling *coup = &che->coup; - - coup->coupling_point = 2 * get_bits1(gb); - coup->num_coupled = get_bits(gb, 3); - for (c = 0; c <= coup->num_coupled; c++) { - num_gain++; - coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE; - coup->id_select[c] = get_bits(gb, 4); - if (coup->type[c] == TYPE_CPE) { - coup->ch_select[c] = get_bits(gb, 2); - if (coup->ch_select[c] == 3) - num_gain++; - } else - coup->ch_select[c] = 2; - } - coup->coupling_point += get_bits1(gb) || (coup->coupling_point >> 1); - - sign = get_bits(gb, 1); - scale = cce_scale[get_bits(gb, 2)]; - - if ((ret = decode_ics(ac, sce, gb, 0, 0))) - return ret; - - for (c = 0; c < num_gain; c++) { - int idx = 0; - int cge = 1; - int gain = 0; - float gain_cache = 1.0; - if (c) { - cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb); - gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0; - gain_cache = powf(scale, -gain); - } - if (coup->coupling_point == AFTER_IMDCT) { - coup->gain[c][0] = gain_cache; - } else { - for (g = 0; g < sce->ics.num_window_groups; g++) { - for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++) { - if (sce->band_type[idx] != ZERO_BT) { - if (!cge) { - int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; - if (t) { - int s = 1; - t = gain += t; - if (sign) { - s -= 2 * (t & 0x1); - t >>= 1; - } - gain_cache = powf(scale, -t) * s; - } - } - coup->gain[c][idx] = gain_cache; - } - } - } - } - } - return 0; -} - -/** - * Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53. - * - * @return Returns number of bytes consumed. - */ -static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc, - GetBitContext *gb) -{ - int i; - int num_excl_chan = 0; - - do { - for (i = 0; i < 7; i++) - che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb); - } while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb)); - - return num_excl_chan / 7; -} - -/** - * Decode dynamic range information; reference: table 4.52. - * - * @return Returns number of bytes consumed. - */ -static int decode_dynamic_range(DynamicRangeControl *che_drc, - GetBitContext *gb) -{ - int n = 1; - int drc_num_bands = 1; - int i; - - /* pce_tag_present? */ - if (get_bits1(gb)) { - che_drc->pce_instance_tag = get_bits(gb, 4); - skip_bits(gb, 4); // tag_reserved_bits - n++; - } - - /* excluded_chns_present? */ - if (get_bits1(gb)) { - n += decode_drc_channel_exclusions(che_drc, gb); - } - - /* drc_bands_present? */ - if (get_bits1(gb)) { - che_drc->band_incr = get_bits(gb, 4); - che_drc->interpolation_scheme = get_bits(gb, 4); - n++; - drc_num_bands += che_drc->band_incr; - for (i = 0; i < drc_num_bands; i++) { - che_drc->band_top[i] = get_bits(gb, 8); - n++; - } - } - - /* prog_ref_level_present? */ - if (get_bits1(gb)) { - che_drc->prog_ref_level = get_bits(gb, 7); - skip_bits1(gb); // prog_ref_level_reserved_bits - n++; - } - - for (i = 0; i < drc_num_bands; i++) { - che_drc->dyn_rng_sgn[i] = get_bits1(gb); - che_drc->dyn_rng_ctl[i] = get_bits(gb, 7); - n++; - } - - 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=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. - * - * @param cnt length of TYPE_FIL syntactic element in bytes - * - * @return Returns number of bytes consumed - */ -static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt, - ChannelElement *che, enum RawDataBlockType elem_type) -{ - int crc_flag = 0; - int res = cnt; - int type = get_bits(gb, 4); - - if (ac->avctx->debug & FF_DEBUG_STARTCODE) - av_log(ac->avctx, AV_LOG_DEBUG, "extension type: %d len:%d\n", type, cnt); - - switch (type) { // extension type - case EXT_SBR_DATA_CRC: - crc_flag++; - case EXT_SBR_DATA: - if (!che) { - av_log(ac->avctx, AV_LOG_ERROR, "SBR was found before the first channel element.\n"); - return res; - } 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->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->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; - ac->avctx->profile = FF_PROFILE_AAC_HE_V2; - output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags, - ac->oc[1].status, 1); - } else { - ac->oc[1].m4ac.sbr = 1; - ac->avctx->profile = FF_PROFILE_AAC_HE; - } - 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); - break; - case EXT_FILL: - decode_fill(ac, gb, 8 * cnt - 4); - break; - case EXT_FILL_DATA: - case EXT_DATA_ELEMENT: - default: - skip_bits_long(gb, 8 * cnt - 4); - break; - }; - return res; -} + t.i = s.i ^ (sign & 1U<<31); + *dst++ = v[idx>>4 & 3] * t.f; -/** - * Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3. - * - * @param decode 1 if tool is used normally, 0 if tool is used in LTP. - * @param coef spectral coefficients - */ -static void apply_tns(float coef[1024], TemporalNoiseShaping *tns, - IndividualChannelStream *ics, int decode) -{ - const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb); - int w, filt, m, i; - int bottom, top, order, start, end, size, inc; - float lpc[TNS_MAX_ORDER]; - float tmp[TNS_MAX_ORDER+1]; - - for (w = 0; w < ics->num_windows; w++) { - bottom = ics->num_swb; - for (filt = 0; filt < tns->n_filt[w]; filt++) { - top = bottom; - bottom = FFMAX(0, top - tns->length[w][filt]); - order = tns->order[w][filt]; - if (order == 0) - continue; - - // tns_decode_coef - compute_lpc_coefs(tns->coef[w][filt], order, lpc, 0, 0, 0); - - start = ics->swb_offset[FFMIN(bottom, mmm)]; - end = ics->swb_offset[FFMIN( top, mmm)]; - if ((size = end - start) <= 0) - continue; - if (tns->direction[w][filt]) { - inc = -1; - start = end - 1; - } else { - inc = 1; - } - start += w * 128; + sign <<= nz & 1; + t.i = s.i ^ (sign & 1U<<31); + *dst++ = v[idx>>6 & 3] * t.f; - 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]; - } - } - } - } + return dst; } +#endif -/** - * 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) +static av_always_inline float flt16_round(float pf) { - 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->fdsp->vector_fmul_reverse(in + 1024, in + 1024, lwindow, 1024); - } else { - ac->fdsp->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); + union av_intfloat32 tmp; + tmp.f = pf; + tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U; + return tmp.f; } -/** - * Apply the long term prediction - */ -static void apply_ltp(AACContext *ac, SingleChannelElement *sce) +static av_always_inline float flt16_even(float pf) { - 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)); - - ac->windowing_and_mdct_ltp(ac, predFreq, predTime, &sce->ics); - - if (sce->tns.present) - 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]) - for (i = offsets[sfb]; i < offsets[sfb + 1]; i++) - sce->coeffs[i] += predFreq[i]; - } + union av_intfloat32 tmp; + tmp.f = pf; + tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U; + return tmp.f; } -/** - * Update the LTP buffer for next frame - */ -static void update_ltp(AACContext *ac, SingleChannelElement *sce) +static av_always_inline float flt16_trunc(float pf) { - 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->fdsp->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->fdsp->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->fdsp->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)); + union av_intfloat32 pun; + pun.f = pf; + pun.i &= 0xFFFF0000U; + return pun.f; } -/** - * Conduct IMDCT and windowing. - */ -static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce) +static av_always_inline void predict(PredictorState *ps, float *coef, + int output_enable) { - IndividualChannelStream *ics = &sce->ics; - float *in = sce->coeffs; - float *out = sce->ret; - float *saved = sce->saved; - 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; - float *buf = ac->buf_mdct; - float *temp = ac->temp; - int i; - - // imdct - if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { - for (i = 0; i < 1024; i += 128) - ac->mdct_small.imdct_half(&ac->mdct_small, buf + i, in + i); - } else - ac->mdct.imdct_half(&ac->mdct, buf, in); - - /* window overlapping - * NOTE: To simplify the overlapping code, all 'meaningless' short to long - * and long to short transitions are considered to be short to short - * transitions. This leaves just two cases (long to long and short to short) - * with a little special sauce for EIGHT_SHORT_SEQUENCE. - */ - 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->fdsp->vector_fmul_window( out, saved, buf, lwindow_prev, 512); - } else { - memcpy( out, saved, 448 * sizeof(float)); - - if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { - ac->fdsp->vector_fmul_window(out + 448 + 0*128, saved + 448, buf + 0*128, swindow_prev, 64); - ac->fdsp->vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow, 64); - ac->fdsp->vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow, 64); - ac->fdsp->vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow, 64); - ac->fdsp->vector_fmul_window(temp, buf + 3*128 + 64, buf + 4*128, swindow, 64); - memcpy( out + 448 + 4*128, temp, 64 * sizeof(float)); - } else { - ac->fdsp->vector_fmul_window(out + 448, saved + 448, buf, swindow_prev, 64); - memcpy( out + 576, buf + 64, 448 * sizeof(float)); - } - } + const float a = 0.953125; // 61.0 / 64 + const float alpha = 0.90625; // 29.0 / 32 + float e0, e1; + float pv; + float k1, k2; + float r0 = ps->r0, r1 = ps->r1; + float cor0 = ps->cor0, cor1 = ps->cor1; + float var0 = ps->var0, var1 = ps->var1; - // buffer update - if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { - memcpy( saved, temp + 64, 64 * sizeof(float)); - ac->fdsp->vector_fmul_window(saved + 64, buf + 4*128 + 64, buf + 5*128, swindow, 64); - ac->fdsp->vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 64); - ac->fdsp->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)); - memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(float)); - } else { // LONG_STOP or ONLY_LONG - memcpy( saved, buf + 512, 512 * sizeof(float)); - } -} + k1 = var0 > 1 ? cor0 * flt16_even(a / var0) : 0; + k2 = var1 > 1 ? cor1 * flt16_even(a / var1) : 0; -static void imdct_and_windowing_ld(AACContext *ac, SingleChannelElement *sce) -{ - IndividualChannelStream *ics = &sce->ics; - float *in = sce->coeffs; - float *out = sce->ret; - float *saved = sce->saved; - float *buf = ac->buf_mdct; - - // imdct - ac->mdct.imdct_half(&ac->mdct_ld, buf, in); - - // window overlapping - if (ics->use_kb_window[1]) { - // AAC LD uses a low overlap sine window instead of a KBD window - memcpy(out, saved, 192 * sizeof(float)); - ac->fdsp->vector_fmul_window(out + 192, saved + 192, buf, ff_sine_128, 64); - memcpy( out + 320, buf + 64, 192 * sizeof(float)); - } else { - ac->fdsp->vector_fmul_window(out, saved, buf, ff_sine_512, 256); - } + pv = flt16_round(k1 * r0 + k2 * r1); + if (output_enable) + *coef += pv; - // buffer update - memcpy(saved, buf + 256, 256 * sizeof(float)); -} + e0 = *coef; + e1 = e0 - k1 * r0; -static void imdct_and_windowing_eld(AACContext *ac, SingleChannelElement *sce) -{ - float *in = sce->coeffs; - float *out = sce->ret; - float *saved = sce->saved; - float *buf = ac->buf_mdct; - int i; - const int n = ac->oc[1].m4ac.frame_length_short ? 480 : 512; - const int n2 = n >> 1; - const int n4 = n >> 2; - const float *const window = n == 480 ? ff_aac_eld_window_480 : - ff_aac_eld_window_512; - - // Inverse transform, mapped to the conventional IMDCT by - // Chivukula, R.K.; Reznik, Y.A.; Devarajan, V., - // "Efficient algorithms for MPEG-4 AAC-ELD, AAC-LD and AAC-LC filterbanks," - // International Conference on Audio, Language and Image Processing, ICALIP 2008. - // URL: http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=&arnumber=4590245&isnumber=4589950 - for (i = 0; i < n2; i+=2) { - float temp; - temp = in[i ]; in[i ] = -in[n - 1 - i]; in[n - 1 - i] = temp; - temp = -in[i + 1]; in[i + 1] = in[n - 2 - i]; in[n - 2 - i] = temp; - } - if (n == 480) - ac->mdct480->imdct_half(ac->mdct480, buf, in, 1, -1.f/(16*1024*960)); - else - ac->mdct.imdct_half(&ac->mdct_ld, buf, in); - for (i = 0; i < n; i+=2) { - buf[i] = -buf[i]; - } - // Like with the regular IMDCT at this point we still have the middle half - // of a transform but with even symmetry on the left and odd symmetry on - // the right - - // window overlapping - // The spec says to use samples [0..511] but the reference decoder uses - // samples [128..639]. - for (i = n4; i < n2; i ++) { - out[i - n4] = buf[n2 - 1 - i] * window[i - n4] + - saved[ i + n2] * window[i + n - n4] + - -saved[ n + n2 - 1 - i] * window[i + 2*n - n4] + - -saved[2*n + n2 + i] * window[i + 3*n - n4]; - } - for (i = 0; i < n2; i ++) { - out[n4 + i] = buf[i] * window[i + n2 - n4] + - -saved[ n - 1 - i] * window[i + n2 + n - n4] + - -saved[ n + i] * window[i + n2 + 2*n - n4] + - saved[2*n + n - 1 - i] * window[i + n2 + 3*n - n4]; - } - for (i = 0; i < n4; i ++) { - out[n2 + n4 + i] = buf[ i + n2] * window[i + n - n4] + - -saved[ n2 - 1 - i] * window[i + 2*n - n4] + - -saved[ n + n2 + i] * window[i + 3*n - n4]; - } + ps->cor1 = flt16_trunc(alpha * cor1 + r1 * e1); + ps->var1 = flt16_trunc(alpha * var1 + 0.5f * (r1 * r1 + e1 * e1)); + ps->cor0 = flt16_trunc(alpha * cor0 + r0 * e0); + ps->var0 = flt16_trunc(alpha * var0 + 0.5f * (r0 * r0 + e0 * e0)); - // buffer update - memmove(saved + n, saved, 2 * n * sizeof(float)); - memcpy( saved, buf, n * sizeof(float)); + ps->r1 = flt16_trunc(a * (r0 - k1 * e0)); + ps->r0 = flt16_trunc(a * e0); } /** @@ -2724,506 +252,7 @@ static void apply_independent_coupling(AACContext *ac, dest[i] += gain * src[i]; } -/** - * channel coupling transformation interface - * - * @param apply_coupling_method pointer to (in)dependent coupling function - */ -static void apply_channel_coupling(AACContext *ac, ChannelElement *cc, - enum RawDataBlockType type, int elem_id, - enum CouplingPoint coupling_point, - void (*apply_coupling_method)(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index)) -{ - int i, c; - - for (i = 0; i < MAX_ELEM_ID; i++) { - ChannelElement *cce = ac->che[TYPE_CCE][i]; - int index = 0; - - if (cce && cce->coup.coupling_point == coupling_point) { - ChannelCoupling *coup = &cce->coup; - - for (c = 0; c <= coup->num_coupled; c++) { - if (coup->type[c] == type && coup->id_select[c] == elem_id) { - if (coup->ch_select[c] != 1) { - apply_coupling_method(ac, &cc->ch[0], cce, index); - if (coup->ch_select[c] != 0) - index++; - } - if (coup->ch_select[c] != 2) - apply_coupling_method(ac, &cc->ch[1], cce, index++); - } else - index += 1 + (coup->ch_select[c] == 3); - } - } - } -} - -/** - * Convert spectral data to float samples, applying all supported tools as appropriate. - */ -static void spectral_to_sample(AACContext *ac) -{ - int i, type; - void (*imdct_and_window)(AACContext *ac, SingleChannelElement *sce); - switch (ac->oc[1].m4ac.object_type) { - case AOT_ER_AAC_LD: - imdct_and_window = imdct_and_windowing_ld; - break; - case AOT_ER_AAC_ELD: - imdct_and_window = imdct_and_windowing_eld; - break; - default: - imdct_and_window = ac->imdct_and_windowing; - } - for (type = 3; type >= 0; type--) { - for (i = 0; i < MAX_ELEM_ID; i++) { - ChannelElement *che = ac->che[type][i]; - if (che && che->present) { - 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) - ac->apply_ltp(ac, &che->ch[0]); - if (che->ch[1].ics.ltp.present && type == TYPE_CPE) - ac->apply_ltp(ac, &che->ch[1]); - } - } - if (che->ch[0].tns.present) - ac->apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1); - if (che->ch[1].tns.present) - 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_window(ac, &che->ch[0]); - if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) - ac->update_ltp(ac, &che->ch[0]); - if (type == TYPE_CPE) { - imdct_and_window(ac, &che->ch[1]); - if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) - 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); - } - } - if (type <= TYPE_CCE) - apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, apply_independent_coupling); - che->present = 0; - } else if (che) { - av_log(ac->avctx, AV_LOG_VERBOSE, "ChannelElement %d.%d missing \n", type, i); - } - } - } -} - -static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb) -{ - int size; - AACADTSHeaderInfo hdr_info; - uint8_t layout_map[MAX_ELEM_ID*4][3]; - int layout_map_tags, ret; - - size = avpriv_aac_parse_header(gb, &hdr_info); - if (size > 0) { - 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"); - ac->warned_num_aac_frames = 1; - } - push_output_configuration(ac); - if (hdr_info.chan_config) { - ac->oc[1].m4ac.chan_config = hdr_info.chan_config; - if ((ret = set_default_channel_config(ac->avctx, - layout_map, - &layout_map_tags, - hdr_info.chan_config)) < 0) - return ret; - if ((ret = output_configure(ac, layout_map, layout_map_tags, - FFMAX(ac->oc[1].status, - OC_TRIAL_FRAME), 0)) < 0) - return ret; - } 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; - ac->oc[1].m4ac.object_type = hdr_info.object_type; - ac->oc[1].m4ac.frame_length_short = 0; - 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_er_frame(AVCodecContext *avctx, void *data, - int *got_frame_ptr, GetBitContext *gb) -{ - AACContext *ac = avctx->priv_data; - const MPEG4AudioConfig *const m4ac = &ac->oc[1].m4ac; - ChannelElement *che; - int err, i; - int samples = m4ac->frame_length_short ? 960 : 1024; - int chan_config = m4ac->chan_config; - int aot = m4ac->object_type; - - if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD) - samples >>= 1; - - ac->frame = data; - - if ((err = frame_configure_elements(avctx)) < 0) - return err; - - // The FF_PROFILE_AAC_* defines are all object_type - 1 - // This may lead to an undefined profile being signaled - ac->avctx->profile = aot - 1; - - ac->tags_mapped = 0; - - if (chan_config < 0 || (chan_config >= 8 && chan_config < 11) || chan_config >= 13) { - avpriv_request_sample(avctx, "Unknown ER channel configuration %d", - chan_config); - return AVERROR_INVALIDDATA; - } - for (i = 0; i < tags_per_config[chan_config]; i++) { - const int elem_type = aac_channel_layout_map[chan_config-1][i][0]; - const int elem_id = aac_channel_layout_map[chan_config-1][i][1]; - 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 AVERROR_INVALIDDATA; - } - che->present = 1; - if (aot != AOT_ER_AAC_ELD) - skip_bits(gb, 4); - switch (elem_type) { - case TYPE_SCE: - err = decode_ics(ac, &che->ch[0], gb, 0, 0); - break; - case TYPE_CPE: - err = decode_cpe(ac, gb, che); - break; - case TYPE_LFE: - err = decode_ics(ac, &che->ch[0], gb, 0, 0); - break; - } - if (err < 0) - return err; - } - - spectral_to_sample(ac); - - ac->frame->nb_samples = samples; - ac->frame->sample_rate = avctx->sample_rate; - *got_frame_ptr = 1; - - skip_bits_long(gb, get_bits_left(gb)); - return 0; -} - -static int aac_decode_frame_int(AVCodecContext *avctx, void *data, - 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; - - if (show_bits(gb, 12) == 0xfff) { - if ((err = parse_adts_frame_header(ac, gb)) < 0) { - av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n"); - goto fail; - } - 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 = AVERROR_INVALIDDATA; - goto fail; - } - } - - if ((err = frame_configure_elements(avctx)) < 0) - goto fail; - - // The FF_PROFILE_AAC_* defines are all object_type - 1 - // This may lead to an undefined profile being signaled - ac->avctx->profile = ac->oc[1].m4ac.object_type - 1; - - ac->tags_mapped = 0; - // parse - while ((elem_type = get_bits(gb, 3)) != TYPE_END) { - elem_id = get_bits(gb, 4); - - if (avctx->debug & FF_DEBUG_STARTCODE) - av_log(avctx, AV_LOG_DEBUG, "Elem type:%x id:%x\n", elem_type, elem_id); - - if (!avctx->channels && elem_type != TYPE_PCE) { - err = AVERROR_INVALIDDATA; - goto fail; - } - - if (elem_type < TYPE_DSE) { - 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); - err = AVERROR_INVALIDDATA; - goto fail; - } - samples = 1024; - che->present = 1; - } - - switch (elem_type) { - - case TYPE_SCE: - err = decode_ics(ac, &che->ch[0], gb, 0, 0); - audio_found = 1; - sce_count++; - break; - - case TYPE_CPE: - err = decode_cpe(ac, gb, che); - audio_found = 1; - break; - - case TYPE_CCE: - err = decode_cce(ac, gb, che); - break; - - case TYPE_LFE: - err = decode_ics(ac, &che->ch[0], gb, 0, 0); - audio_found = 1; - break; - - case TYPE_DSE: - err = skip_data_stream_element(ac, gb); - break; - - case TYPE_PCE: { - 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 (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, layout_map, tags, OC_TRIAL_PCE, 1); - if (!err) - ac->oc[1].m4ac.chan_config = 0; - pce_found = 1; - } - break; - } - - case TYPE_FIL: - 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, "TYPE_FIL: "overread_err); - err = AVERROR_INVALIDDATA; - goto fail; - } - while (elem_id > 0) - elem_id -= decode_extension_payload(ac, gb, elem_id, che_prev, elem_type_prev); - err = 0; /* FIXME */ - break; - - default: - err = AVERROR_BUG; /* should not happen, but keeps compiler happy */ - break; - } - - che_prev = che; - elem_type_prev = elem_type; - - if (err) - goto fail; - - if (get_bits_left(gb) < 3) { - av_log(avctx, AV_LOG_ERROR, overread_err); - err = AVERROR_INVALIDDATA; - goto fail; - } - } - - if (!avctx->channels) { - *got_frame_ptr = 0; - return 0; - } - - spectral_to_sample(ac); - - 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->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)); - } - - if (!ac->frame->data[0] && samples) { - av_log(avctx, AV_LOG_ERROR, "no frame data found\n"); - err = AVERROR_INVALIDDATA; - goto fail; - } - - if (samples) { - ac->frame->nb_samples = samples; - ac->frame->sample_rate = avctx->sample_rate; - } else - av_frame_unref(ac->frame); - *got_frame_ptr = !!samples; - - /* 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 (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]; - } - - return 0; -fail: - pop_output_configuration(ac); - return err; -} - -static int aac_decode_frame(AVCodecContext *avctx, void *data, - 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); - 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 && 0) { - 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; - } - } - - 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; - - if ((err = init_get_bits(&gb, buf, buf_size * 8)) < 0) - return err; - - switch (ac->oc[1].m4ac.object_type) { - case AOT_ER_AAC_LC: - case AOT_ER_AAC_LTP: - case AOT_ER_AAC_LD: - case AOT_ER_AAC_ELD: - err = aac_decode_er_frame(avctx, data, got_frame_ptr, &gb); - break; - default: - err = aac_decode_frame_int(avctx, data, got_frame_ptr, &gb, avpkt); - } - if (err < 0) - return err; - - buf_consumed = (get_bits_count(&gb) + 7) >> 3; - for (buf_offset = buf_consumed; buf_offset < buf_size; buf_offset++) - if (buf[buf_offset]) - break; - - return buf_size > buf_offset ? buf_consumed : buf_size; -} - -static av_cold int aac_decode_close(AVCodecContext *avctx) -{ - AACContext *ac = avctx->priv_data; - int i, type; - - for (i = 0; i < MAX_ELEM_ID; i++) { - for (type = 0; type < 4; type++) { - if (ac->che[type][i]) - ff_aac_sbr_ctx_close(&ac->che[type][i]->sbr); - av_freep(&ac->che[type][i]); - } - } - - ff_mdct_end(&ac->mdct); - ff_mdct_end(&ac->mdct_small); - ff_mdct_end(&ac->mdct_ld); - ff_mdct_end(&ac->mdct_ltp); - ff_imdct15_uninit(&ac->mdct480); - av_freep(&ac->fdsp); - return 0; -} - +#include "aacdec_template.c" #define LOAS_SYNC_WORD 0x2b7 ///< 11 bits LOAS sync word @@ -3505,53 +534,6 @@ 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, -}; - -static const AVProfile profiles[] = { - { FF_PROFILE_AAC_MAIN, "Main" }, - { FF_PROFILE_AAC_LOW, "LC" }, - { FF_PROFILE_AAC_SSR, "SSR" }, - { FF_PROFILE_AAC_LTP, "LTP" }, - { FF_PROFILE_AAC_HE, "HE-AAC" }, - { FF_PROFILE_AAC_HE_V2, "HE-AACv2" }, - { FF_PROFILE_AAC_LD, "LD" }, - { FF_PROFILE_AAC_ELD, "ELD" }, - { FF_PROFILE_UNKNOWN }, -}; - AVCodec ff_aac_decoder = { .name = "aac", .long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"), -- cgit v1.1