diff --git a/libavcodec/Makefile b/libavcodec/Makefile index 53c628f09c..cff6347bdb 100644 --- a/libavcodec/Makefile +++ b/libavcodec/Makefile @@ -180,7 +180,7 @@ OBJS-$(CONFIG_WMV2DSP) += wmv2dsp.o OBJS-$(CONFIG_ZERO12V_DECODER) += 012v.o OBJS-$(CONFIG_A64MULTI_ENCODER) += a64multienc.o elbg.o OBJS-$(CONFIG_A64MULTI5_ENCODER) += a64multienc.o elbg.o -OBJS-$(CONFIG_AAC_DECODER) += aacdec.o aactab.o \ +OBJS-$(CONFIG_AAC_DECODER) += aactab.o \ aacsbr.o aacps_common.o aacps_float.o \ kbdwin.o \ sbrdsp.o aacpsdsp_float.o cbrt_data.o diff --git a/libavcodec/aac/aacdec.c b/libavcodec/aac/aacdec.c index dfbc309583..01a10468fa 100644 --- a/libavcodec/aac/aacdec.c +++ b/libavcodec/aac/aacdec.c @@ -32,10 +32,20 @@ #include #include +#include "aacdec.h" +#include "aacdec_tab.h" + #include "libavcodec/aac.h" #include "libavcodec/aacsbr.h" -#include "aacdec.h" +#include "libavcodec/aactab.h" +#include "libavcodec/adts_header.h" + #include "libavcodec/avcodec.h" +#include "libavcodec/internal.h" +#include "libavcodec/codec_internal.h" +#include "libavcodec/decode.h" +#include "libavcodec/profiles.h" + #include "libavutil/attributes.h" #include "libavutil/error.h" #include "libavutil/log.h" @@ -44,14 +54,1067 @@ #include "libavutil/opt.h" #include "libavutil/tx.h" #include "libavutil/version.h" +#include "libavutil/thread.h" -extern const AACDecDSP aac_dsp; -extern const AACDecDSP aac_dsp_fixed; +/* + * 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 (not in fixed point code) 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. + */ -extern const AACDecProc aac_proc; -extern const AACDecProc aac_proc_fixed; +static int output_configure(AACDecContext *ac, + uint8_t layout_map[MAX_ELEM_ID*4][3], int tags, + enum OCStatus oc_type, int get_new_frame); -av_cold int ff_aac_decode_close(AVCodecContext *avctx) +#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(AACDecContext *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]) { + int ret; + if (ac->is_fixed) + ret = ff_aac_sbr_ctx_alloc_init_fixed(ac, &ac->che[type][id], type); + else + ret = ff_aac_sbr_ctx_alloc_init(ac, &ac->che[type][id], type); + if (ret < 0) + return ret; + } + 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]) { + if (ac->is_fixed) + ff_aac_sbr_ctx_close_fixed(ac->che[type][id]); + else + ff_aac_sbr_ctx_close(ac->che[type][id]); + } + av_freep(&ac->che[type][id]); + } + return 0; +} + +static int frame_configure_elements(AVCodecContext *avctx) +{ + AACDecContext *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].AAC_RENAME(output) = che->ch[0].AAC_RENAME(ret_buf); + che->ch[1].AAC_RENAME(output) = che->ch[1].AAC_RENAME(ret_buf); + } + } + } + + /* get output buffer */ + av_frame_unref(ac->frame); + if (!avctx->ch_layout.nb_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->ch_layout.nb_channels; ch++) { + if (ac->output_element[ch]) + ac->output_element[ch]->AAC_RENAME(output) = (INTFLOAT *)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, uint64_t *layout) +{ + 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 + }; + if (e2c_vec[offset].av_position != UINT64_MAX) + *layout |= e2c_vec[offset].av_position; + + 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 + }; + if (left != UINT64_MAX) + *layout |= left; + + if (right != UINT64_MAX) + *layout |= right; + + 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 ^= (pos != AAC_CHANNEL_LFE); + } + } + if (sce_parity && + (pos == AAC_CHANNEL_FRONT && first_cpe)) + return -1; + + return num_pos_channels; +} + +static int assign_channels(struct elem_to_channel e2c_vec[MAX_ELEM_ID], uint8_t (*layout_map)[3], + uint64_t *layout, int tags, int layer, int pos, int *current) +{ + int i = *current, j = 0; + int nb_channels = count_paired_channels(layout_map, tags, pos, i); + + if (nb_channels < 0 || nb_channels > 5) + return 0; + + if (pos == AAC_CHANNEL_LFE) { + while (nb_channels) { + if (ff_aac_channel_map[layer][pos - 1][j] == AV_CHAN_NONE) + return -1; + e2c_vec[i] = (struct elem_to_channel) { + .av_position = 1ULL << ff_aac_channel_map[layer][pos - 1][j], + .syn_ele = layout_map[i][0], + .elem_id = layout_map[i][1], + .aac_position = pos + }; + *layout |= e2c_vec[i].av_position; + i++; + j++; + nb_channels--; + } + *current = i; + + return 0; + } + + while (nb_channels & 1) { + if (ff_aac_channel_map[layer][pos - 1][0] == AV_CHAN_NONE) + return -1; + if (ff_aac_channel_map[layer][pos - 1][0] == AV_CHAN_UNUSED) + break; + e2c_vec[i] = (struct elem_to_channel) { + .av_position = 1ULL << ff_aac_channel_map[layer][pos - 1][0], + .syn_ele = layout_map[i][0], + .elem_id = layout_map[i][1], + .aac_position = pos + }; + *layout |= e2c_vec[i].av_position; + i++; + nb_channels--; + } + + j = (pos != AAC_CHANNEL_SIDE) && nb_channels <= 3 ? 3 : 1; + while (nb_channels >= 2) { + if (ff_aac_channel_map[layer][pos - 1][j] == AV_CHAN_NONE || + ff_aac_channel_map[layer][pos - 1][j+1] == AV_CHAN_NONE) + return -1; + i += assign_pair(e2c_vec, layout_map, i, + 1ULL << ff_aac_channel_map[layer][pos - 1][j], + 1ULL << ff_aac_channel_map[layer][pos - 1][j+1], + pos, layout); + j += 2; + nb_channels -= 2; + } + while (nb_channels & 1) { + if (ff_aac_channel_map[layer][pos - 1][5] == AV_CHAN_NONE) + return -1; + e2c_vec[i] = (struct elem_to_channel) { + .av_position = 1ULL << ff_aac_channel_map[layer][pos - 1][5], + .syn_ele = layout_map[i][0], + .elem_id = layout_map[i][1], + .aac_position = pos + }; + *layout |= e2c_vec[i].av_position; + i++; + nb_channels--; + } + if (nb_channels) + return -1; + + *current = i; + + return 0; +} + +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 } }; + uint64_t layout = 0; + + if (FF_ARRAY_ELEMS(e2c_vec) < tags) + return 0; + + for (n = 0, i = 0; n < 3 && i < tags; n++) { + int ret = assign_channels(e2c_vec, layout_map, &layout, tags, n, AAC_CHANNEL_FRONT, &i); + if (ret < 0) + return 0; + ret = assign_channels(e2c_vec, layout_map, &layout, tags, n, AAC_CHANNEL_SIDE, &i); + if (ret < 0) + return 0; + ret = assign_channels(e2c_vec, layout_map, &layout, tags, n, AAC_CHANNEL_BACK, &i); + if (ret < 0) + return 0; + ret = assign_channels(e2c_vec, layout_map, &layout, tags, n, AAC_CHANNEL_LFE, &i); + if (ret < 0) + return 0; + } + + total_non_cc_elements = n = i; + + if (layout == AV_CH_LAYOUT_22POINT2) { + // For 22.2 reorder the result as needed + FFSWAP(struct elem_to_channel, e2c_vec[2], e2c_vec[0]); // FL & FR first (final), FC third + FFSWAP(struct elem_to_channel, e2c_vec[2], e2c_vec[1]); // FC second (final), FLc & FRc third + FFSWAP(struct elem_to_channel, e2c_vec[6], e2c_vec[2]); // LFE1 third (final), FLc & FRc seventh + FFSWAP(struct elem_to_channel, e2c_vec[4], e2c_vec[3]); // BL & BR fourth (final), SiL & SiR fifth + FFSWAP(struct elem_to_channel, e2c_vec[6], e2c_vec[4]); // FLc & FRc fifth (final), SiL & SiR seventh + FFSWAP(struct elem_to_channel, e2c_vec[7], e2c_vec[6]); // LFE2 seventh (final), SiL & SiR eight (final) + FFSWAP(struct elem_to_channel, e2c_vec[9], e2c_vec[8]); // TpFL & TpFR ninth (final), TFC tenth (final) + FFSWAP(struct elem_to_channel, e2c_vec[11], e2c_vec[10]); // TC eleventh (final), TpSiL & TpSiR twelth + FFSWAP(struct elem_to_channel, e2c_vec[12], e2c_vec[11]); // TpBL & TpBR twelth (final), TpSiL & TpSiR thirteenth (final) + } else { + // For everything else, utilize the AV channel position define as a + // stable sort. + 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); + + } + + 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; + } + + return layout; +} + +/** + * Save current output configuration if and only if it has been locked. + */ +static int push_output_configuration(AACDecContext *ac) +{ + int pushed = 0; + + if (ac->oc[1].status == OC_LOCKED || ac->oc[0].status == OC_NONE) { + ac->oc[0] = ac->oc[1]; + pushed = 1; + } + ac->oc[1].status = OC_NONE; + return pushed; +} + +/** + * Restore the previous output configuration if and only if the current + * configuration is unlocked. + */ +static void pop_output_configuration(AACDecContext *ac) +{ + if (ac->oc[1].status != OC_LOCKED && ac->oc[0].status != OC_NONE) { + ac->oc[1] = ac->oc[0]; + ac->avctx->ch_layout = ac->oc[1].ch_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(AACDecContext *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]++; + if (id_map[type][id] >= MAX_ELEM_ID) { + avpriv_request_sample(ac->avctx, "Too large remapped id"); + return AVERROR_PATCHWELCOME; + } + } + // Try to sniff a reasonable channel order, otherwise output the + // channels in the order the PCE declared them. + if (ac->output_channel_order == CHANNEL_ORDER_DEFAULT) + 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; + } + } + + av_channel_layout_uninit(&ac->oc[1].ch_layout); + if (layout) + av_channel_layout_from_mask(&ac->oc[1].ch_layout, layout); + else { + ac->oc[1].ch_layout.order = AV_CHANNEL_ORDER_UNSPEC; + ac->oc[1].ch_layout.nb_channels = channels; + } + + av_channel_layout_copy(&avctx->ch_layout, &ac->oc[1].ch_layout); + 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) +{ + AACDecContext *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(AACDecContext *ac, AVCodecContext *avctx, + uint8_t (*layout_map)[3], + int *tags, + int channel_config) +{ + if (channel_config < 1 || (channel_config > 7 && channel_config < 11) || + channel_config > 14) { + av_log(avctx, AV_LOG_ERROR, + "invalid default channel configuration (%d)\n", + channel_config); + return AVERROR_INVALIDDATA; + } + *tags = ff_tags_per_config[channel_config]; + memcpy(layout_map, ff_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) { + layout_map[2][2] = AAC_CHANNEL_BACK; + + if (!ac || !ac->warned_71_wide++) { + 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); + } + } + + return 0; +} + +static ChannelElement *get_che(AACDecContext *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, 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"); + + 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, 1) < 0) + return NULL; + + 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 14: + if (ac->tags_mapped > 2 && ((type == TYPE_CPE && elem_id < 3) || + (type == TYPE_LFE && elem_id < 1))) { + ac->tags_mapped++; + return ac->tag_che_map[type][elem_id] = ac->che[type][elem_id]; + } + case 13: + if (ac->tags_mapped > 3 && ((type == TYPE_CPE && elem_id < 8) || + (type == TYPE_SCE && elem_id < 6) || + (type == TYPE_LFE && elem_id < 2))) { + ac->tags_mapped++; + return ac->tag_che_map[type][elem_id] = ac->che[type][elem_id]; + } + 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 == 3 && 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 == ff_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 == ff_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->tags_mapped == 1 && ac->oc[1].m4ac.chan_config == 2 && + type == TYPE_SCE) { + ac->tags_mapped++; + return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1]; + } + 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++; + } +} + +static inline void relative_align_get_bits(GetBitContext *gb, + int reference_position) { + int n = (reference_position - get_bits_count(gb) & 7); + if (n) + skip_bits(gb, n); +} + +/** + * 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 byte_align_ref) +{ + 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) < 5 * (num_front + num_side + num_back + num_cc) + 4 *(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; + + relative_align_get_bits(gb, byte_align_ref); + + /* 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 AACDecContext, 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(AACDecContext *ac, AVCodecContext *avctx, + GetBitContext *gb, + int get_bit_alignment, + 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; + + m4ac->frame_length_short = get_bits1(gb); + if (m4ac->frame_length_short && m4ac->sbr == 1) { + avpriv_report_missing_feature(avctx, "SBR with 960 frame length"); + if (ac) ac->warned_960_sbr = 1; + m4ac->sbr = 0; + m4ac->ps = 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, get_bit_alignment); + if (tags < 0) + return tags; + } else { + if ((ret = set_default_channel_config(ac, 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(AACDecContext *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(ac, 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 AACDecContext, may be null + * @param avctx pointer to AVCCodecContext, used for logging + * @param m4ac pointer to MPEG4AudioConfig, used for parsing + * @param gb buffer holding an audio specific config + * @param get_bit_alignment relative alignment for byte align operations + * @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_gb(AACDecContext *ac, + AVCodecContext *avctx, + MPEG4AudioConfig *m4ac, + GetBitContext *gb, + int get_bit_alignment, + int sync_extension) +{ + int i, ret; + GetBitContext gbc = *gb; + MPEG4AudioConfig m4ac_bak = *m4ac; + + if ((i = ff_mpeg4audio_get_config_gb(m4ac, &gbc, sync_extension, avctx)) < 0) { + *m4ac = m4ac_bak; + return AVERROR_INVALIDDATA; + } + + if (m4ac->sampling_index > 12) { + av_log(avctx, AV_LOG_ERROR, + "invalid sampling rate index %d\n", + m4ac->sampling_index); + *m4ac = m4ac_bak; + 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); + *m4ac = m4ac_bak; + return AVERROR_INVALIDDATA; + } + + skip_bits_long(gb, i); + + switch (m4ac->object_type) { + case AOT_AAC_MAIN: + case AOT_AAC_LC: + case AOT_AAC_SSR: + case AOT_AAC_LTP: + case AOT_ER_AAC_LC: + case AOT_ER_AAC_LD: + if ((ret = decode_ga_specific_config(ac, avctx, gb, get_bit_alignment, + 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); +} + +static int decode_audio_specific_config(AACDecContext *ac, + AVCodecContext *avctx, + MPEG4AudioConfig *m4ac, + const uint8_t *data, int64_t bit_size, + int sync_extension) +{ + int i, ret; + GetBitContext gb; + + if (bit_size < 0 || bit_size > INT_MAX) { + av_log(avctx, AV_LOG_ERROR, "Audio specific config size is invalid\n"); + return AVERROR_INVALIDDATA; + } + + ff_dlog(avctx, "audio specific config size %d\n", (int)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; + + return decode_audio_specific_config_gb(ac, avctx, m4ac, &gb, 0, + sync_extension); +} + +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 av_cold void aac_static_table_init(void) +{ + ff_aac_sbr_init(); + ff_aac_sbr_init_fixed(); + + ff_aacdec_common_init_once(); +} +static AVOnce aac_table_init = AV_ONCE_INIT; + +static av_cold int decode_close(AVCodecContext *avctx) { AACDecContext *ac = avctx->priv_data; int is_fixed = ac->is_fixed; @@ -84,7 +1147,7 @@ av_cold int ff_aac_decode_close(AVCodecContext *avctx) return 0; } -av_cold int ff_aac_decode_init_common(AVCodecContext *avctx) +static av_cold int init_dsp(AVCodecContext *avctx) { AACDecContext *ac = avctx->priv_data; int is_fixed = ac->is_fixed, ret; @@ -127,6 +1190,1332 @@ av_cold int ff_aac_decode_init_common(AVCodecContext *avctx) return ac->dsp.init(ac); } +static av_cold int aac_decode_init_internal(AVCodecContext *avctx) +{ + AACDecContext *ac = avctx->priv_data; + int ret; + + if (avctx->sample_rate > 96000) + return AVERROR_INVALIDDATA; + + ret = ff_thread_once(&aac_table_init, &aac_static_table_init); + if (ret != 0) + return AVERROR_UNKNOWN; + + ac->avctx = avctx; + ac->oc[1].m4ac.sample_rate = avctx->sample_rate; + + if (ac->is_fixed) + avctx->sample_fmt = AV_SAMPLE_FMT_S32P; + else + 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 * 8LL, + 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->ch_layout.nb_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->ch_layout.nb_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(ac, 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; + } + } + + return init_dsp(avctx); +} + +static av_cold int aac_decode_init(AVCodecContext *avctx) +{ + AACDecContext *ac = avctx->priv_data; + ac->is_fixed = 0; + return aac_decode_init_internal(avctx); +} + +static av_cold int aac_decode_init_fixed(AVCodecContext *avctx) +{ + AACDecContext *ac = avctx->priv_data; + ac->is_fixed = 1; + return aac_decode_init_internal(avctx); +} + +/** + * Skip data_stream_element; reference: table 4.10. + */ +static int skip_data_stream_element(AACDecContext *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(AACDecContext *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(AACDecContext *ac, LongTermPrediction *ltp, + GetBitContext *gb, uint8_t max_sfb) +{ + int sfb; + + ltp->lag = get_bits(gb, 11); + if (ac->is_fixed) + ltp->coef_fixed = Q30(ff_ltp_coef[get_bits(gb, 3)]); + else + ltp->coef = ff_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(AACDecContext *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; + int ret_fail = AVERROR_INVALIDDATA; + + 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; + if (m4ac->frame_length_short) { + ics->swb_offset = ff_swb_offset_120[sampling_index]; + ics->num_swb = ff_aac_num_swb_120[sampling_index]; + } else { + 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) { + ret_fail = AVERROR_BUG; + goto fail; + } + } else { + if (m4ac->frame_length_short) { + ics->num_swb = ff_aac_num_swb_960[sampling_index]; + ics->swb_offset = ff_swb_offset_960[sampling_index]; + } else { + ics->num_swb = ff_aac_num_swb_1024[sampling_index]; + ics->swb_offset = ff_swb_offset_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"); + ret_fail = AVERROR_PATCHWELCOME; + goto fail; + } + if ((ics->ltp.present = get_bits(gb, 1))) + decode_ltp(ac, &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 ret_fail; +} + +/** + * 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(AACDecContext *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(AACDecContext *ac, int sfo[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]; + switch (band_type[idx]) { + case ZERO_BT: + for (; i < run_end; i++, idx++) + sfo[idx] = 0; + break; + case INTENSITY_BT: /* fallthrough */ + case INTENSITY_BT2: + for (; i < run_end; i++, idx++) { + offset[2] += get_vlc2(gb, ff_vlc_scalefactors, 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); + } + sfo[idx] = clipped_offset; + } + break; + case 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, ff_vlc_scalefactors, 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); + } + sfo[idx] = clipped_offset; + } + break; + default: + for (; i < run_end; i++, idx++) { + offset[0] += get_vlc2(gb, ff_vlc_scalefactors, 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; + } + sfo[idx] = offset[0]; + } + break; + } + } + } + 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(AACDecContext *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++) { + if (ac->is_fixed) + tns->coef_fixed[w][filt][i] = Q31(ff_tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)]); + else + tns->coef[w][filt][i] = ff_tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)]; + } + } + } + } + } + return 0; +} + +/** + * 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) +{ + 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])); + } +} + +static void decode_gain_control(SingleChannelElement * sce, GetBitContext * gb) +{ + // wd_num, wd_test, aloc_size + static const uint8_t gain_mode[4][3] = { + {1, 0, 5}, // ONLY_LONG_SEQUENCE = 0, + {2, 1, 2}, // LONG_START_SEQUENCE, + {8, 0, 2}, // EIGHT_SHORT_SEQUENCE, + {2, 1, 5}, // LONG_STOP_SEQUENCE + }; + + const int mode = sce->ics.window_sequence[0]; + uint8_t bd, wd, ad; + + // FIXME: Store the gain control data on |sce| and do something with it. + uint8_t max_band = get_bits(gb, 2); + for (bd = 0; bd < max_band; bd++) { + for (wd = 0; wd < gain_mode[mode][0]; wd++) { + uint8_t adjust_num = get_bits(gb, 3); + for (ad = 0; ad < adjust_num; ad++) { + skip_bits(gb, 4 + ((wd == 0 && gain_mode[mode][1]) + ? 4 + : gain_mode[mode][2])); + } + } + } +} + +/** + * 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 + */ +int ff_aac_decode_ics(AACDecContext *ac, SingleChannelElement *sce, + GetBitContext *gb, int common_window, int scale_flag) +{ + Pulse pulse; + TemporalNoiseShaping *tns = &sce->tns; + IndividualChannelStream *ics = &sce->ics; + 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) { + ret = decode_ics_info(ac, ics, gb); + if (ret < 0) + goto fail; + } + + if ((ret = decode_band_types(ac, sce->band_type, + sce->band_type_run_end, gb, ics)) < 0) + goto fail; + if ((ret = decode_scalefactors(ac, sce->sfo, gb, global_gain, ics, + sce->band_type, sce->band_type_run_end)) < 0) + goto fail; + + ac->dsp.dequant_scalefactors(sce); + + 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"); + ret = AVERROR_INVALIDDATA; + goto fail; + } + if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) { + av_log(ac->avctx, AV_LOG_ERROR, + "Pulse data corrupt or invalid.\n"); + ret = AVERROR_INVALIDDATA; + goto fail; + } + } + tns->present = get_bits1(gb); + if (tns->present && !er_syntax) { + ret = decode_tns(ac, tns, gb, ics); + if (ret < 0) + goto fail; + } + if (!eld_syntax && get_bits1(gb)) { + decode_gain_control(sce, gb); + if (!ac->warned_gain_control) { + avpriv_report_missing_feature(ac->avctx, "Gain control"); + ac->warned_gain_control = 1; + } + } + // 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) { + ret = decode_tns(ac, tns, gb, ics); + if (ret < 0) + goto fail; + } + } + + ret = ac->proc.decode_spectrum_and_dequant(ac, gb, + pulse_present ? &pulse : NULL, + sce); + if (ret < 0) + goto fail; + + if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN && !common_window) + ac->dsp.apply_prediction(ac, sce); + + return 0; +fail: + tns->present = 0; + return ret; +} + +/** + * Decode a channel_pair_element; reference: table 4.4. + * + * @return Returns error status. 0 - OK, !0 - error + */ +static int decode_cpe(AACDecContext *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(ac, &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 = ff_aac_decode_ics(ac, &cpe->ch[0], gb, common_window, 0))) + return ret; + if ((ret = ff_aac_decode_ics(ac, &cpe->ch[1], gb, common_window, 0))) + return ret; + + if (common_window) { + if (ms_present) + ac->dsp.apply_mid_side_stereo(ac, cpe); + if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN) { + ac->dsp.apply_prediction(ac, &cpe->ch[0]); + ac->dsp.apply_prediction(ac, &cpe->ch[1]); + } + } + + ac->dsp.apply_intensity_stereo(ac, cpe, ms_present); + 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(AACDecContext *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(AACDecContext *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.frame_length_short) { + if (!ac->warned_960_sbr) + avpriv_report_missing_feature(ac->avctx, + "SBR with 960 frame length"); + ac->warned_960_sbr = 1; + skip_bits_long(gb, 8 * cnt - 4); + 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->ch_layout.nb_channels == 1) { + ac->oc[1].m4ac.sbr = 1; + ac->oc[1].m4ac.ps = 1; + ac->avctx->profile = AV_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 = AV_PROFILE_AAC_HE; + } + + if (ac->is_fixed) + res = ff_aac_sbr_decode_extension_fixed(ac, che, gb, crc_flag, cnt, elem_type); + else + res = ff_aac_sbr_decode_extension(ac, che, gb, crc_flag, cnt, elem_type); + + + if (ac->oc[1].m4ac.ps == 1 && !ac->warned_he_aac_mono) { + av_log(ac->avctx, AV_LOG_VERBOSE, "Treating HE-AAC mono as stereo.\n"); + ac->warned_he_aac_mono = 1; + } + 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; +} + +/** + * channel coupling transformation interface + * + * @param apply_coupling_method pointer to (in)dependent coupling function + */ +static void apply_channel_coupling(AACDecContext *ac, ChannelElement *cc, + enum RawDataBlockType type, int elem_id, + enum CouplingPoint coupling_point, + void (*apply_coupling_method)(AACDecContext *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 samples, applying all supported tools as appropriate. + */ +static void spectral_to_sample(AACDecContext *ac, int samples) +{ + int i, type; + void (*imdct_and_window)(AACDecContext *ac, SingleChannelElement *sce); + switch (ac->oc[1].m4ac.object_type) { + case AOT_ER_AAC_LD: + imdct_and_window = ac->dsp.imdct_and_windowing_ld; + break; + case AOT_ER_AAC_ELD: + imdct_and_window = ac->dsp.imdct_and_windowing_eld; + break; + default: + if (ac->oc[1].m4ac.frame_length_short) + imdct_and_window = ac->dsp.imdct_and_windowing_960; + else + imdct_and_window = ac->dsp.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, ac->dsp.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->dsp.apply_ltp(ac, &che->ch[0]); + if (che->ch[1].ics.ltp.present && type == TYPE_CPE) + ac->dsp.apply_ltp(ac, &che->ch[1]); + } + } + if (che->ch[0].tns.present) + ac->dsp.apply_tns(che->ch[0].AAC_RENAME(coeffs), + &che->ch[0].tns, &che->ch[0].ics, 1); + if (che->ch[1].tns.present) + ac->dsp.apply_tns(che->ch[1].AAC_RENAME(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, ac->dsp.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->dsp.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->dsp.update_ltp(ac, &che->ch[1]); + } + if (ac->oc[1].m4ac.sbr > 0) { + if (ac->is_fixed) + ff_aac_sbr_apply_fixed(ac, che, type, + che->ch[0].AAC_RENAME(output), + che->ch[1].AAC_RENAME(output)); + else + ff_aac_sbr_apply(ac, che, type, + che->ch[0].AAC_RENAME(output), + che->ch[1].AAC_RENAME(output)); + } + } + if (type <= TYPE_CCE) + apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, ac->dsp.apply_independent_coupling); + ac->dsp.clip_output(ac, che, type, samples); + 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(AACDecContext *ac, GetBitContext *gb) +{ + int size; + AACADTSHeaderInfo hdr_info; + uint8_t layout_map[MAX_ELEM_ID*4][3]; + int layout_map_tags, ret; + + size = ff_adts_header_parse(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, 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, AVFrame *frame, + int *got_frame_ptr, GetBitContext *gb) +{ + AACDecContext *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 = frame; + + if ((err = frame_configure_elements(avctx)) < 0) + return err; + + // The AV_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 < ff_tags_per_config[chan_config]; i++) { + const int elem_type = ff_aac_channel_layout_map[chan_config-1][i][0]; + const int elem_id = ff_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 = ff_aac_decode_ics(ac, &che->ch[0], gb, 0, 0); + break; + case TYPE_CPE: + err = decode_cpe(ac, gb, che); + break; + case TYPE_LFE: + err = ff_aac_decode_ics(ac, &che->ch[0], gb, 0, 0); + break; + } + if (err < 0) + return err; + } + + spectral_to_sample(ac, samples); + + if (!ac->frame->data[0] && samples) { + av_log(avctx, AV_LOG_ERROR, "no frame data found\n"); + return AVERROR_INVALIDDATA; + } + + 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, AVFrame *frame, + int *got_frame_ptr, GetBitContext *gb, + const AVPacket *avpkt) +{ + AACDecContext *ac = avctx->priv_data; + ChannelElement *che = NULL, *che_prev = NULL; + enum RawDataBlockType elem_type, che_prev_type = TYPE_END; + int err, elem_id; + int samples = 0, multiplier, audio_found = 0, pce_found = 0; + int is_dmono, sce_count = 0; + int payload_alignment; + uint8_t che_presence[4][MAX_ELEM_ID] = {{0}}; + + ac->frame = frame; + + 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 AV_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; + + payload_alignment = get_bits_count(gb); + 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->ch_layout.nb_channels && elem_type != TYPE_PCE) { + err = AVERROR_INVALIDDATA; + goto fail; + } + + if (elem_type < TYPE_DSE) { + if (che_presence[elem_type][elem_id]) { + int error = che_presence[elem_type][elem_id] > 1; + av_log(ac->avctx, error ? AV_LOG_ERROR : AV_LOG_DEBUG, "channel element %d.%d duplicate\n", + elem_type, elem_id); + if (error) { + err = AVERROR_INVALIDDATA; + goto fail; + } + } + che_presence[elem_type][elem_id]++; + + 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 = ac->oc[1].m4ac.frame_length_short ? 960 : 1024; + che->present = 1; + } + + switch (elem_type) { + + case TYPE_SCE: + err = ff_aac_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 = ac->proc.decode_cce(ac, gb, che); + break; + + case TYPE_LFE: + err = ff_aac_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] = {{0}}; + int tags; + + int pushed = push_output_configuration(ac); + if (pce_found && !pushed) { + err = AVERROR_INVALIDDATA; + goto fail; + } + + tags = decode_pce(avctx, &ac->oc[1].m4ac, layout_map, gb, + payload_alignment); + 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"); + pop_output_configuration(ac); + } else { + err = output_configure(ac, layout_map, tags, OC_TRIAL_PCE, 1); + if (!err) + ac->oc[1].m4ac.chan_config = 0; + pce_found = 1; + } + break; + } + + 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; + } + err = 0; + while (elem_id > 0) { + int ret = decode_extension_payload(ac, gb, elem_id, che_prev, che_prev_type); + if (ret < 0) { + err = ret; + break; + } + elem_id -= ret; + } + break; + + default: + err = AVERROR_BUG; /* should not happen, but keeps compiler happy */ + break; + } + + if (elem_type < TYPE_DSE) { + che_prev = che; + che_prev_type = 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->ch_layout.nb_channels) { + *got_frame_ptr = 0; + return 0; + } + + multiplier = (ac->oc[1].m4ac.sbr == 1) ? ac->oc[1].m4ac.ext_sample_rate > ac->oc[1].m4ac.sample_rate : 0; + samples <<= multiplier; + + spectral_to_sample(ac, 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; + } + + 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 && + !av_channel_layout_compare(&ac->oc[1].ch_layout, + &(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO); + if (is_dmono) { + if (ac->dmono_mode == 1) + frame->data[1] = frame->data[0]; + else if (ac->dmono_mode == 2) + frame->data[0] = frame->data[1]; + } + + return 0; +fail: + pop_output_configuration(ac); + return err; +} + +static int aac_decode_frame(AVCodecContext *avctx, AVFrame *frame, + int *got_frame_ptr, AVPacket *avpkt) +{ + AACDecContext *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; + size_t new_extradata_size; + const uint8_t *new_extradata = av_packet_get_side_data(avpkt, + AV_PKT_DATA_NEW_EXTRADATA, + &new_extradata_size); + size_t 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) { + /* discard previous configuration */ + ac->oc[1].status = OC_NONE; + err = decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac, + new_extradata, + new_extradata_size * 8LL, 1); + if (err < 0) { + return err; + } + } + + 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_bits8(&gb, buf, buf_size)) < 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, frame, got_frame_ptr, &gb); + break; + default: + err = aac_decode_frame_int(avctx, frame, 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; +} + +#include "aacdec_latm.h" + #define AACDEC_FLAGS AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM #define OFF(field) offsetof(AACDecContext, field) static const AVOption options[] = { @@ -153,9 +2542,49 @@ static const AVOption options[] = { {NULL}, }; -const AVClass ff_aac_decoder_class = { +static const AVClass decoder_class = { .class_name = "AAC decoder", .item_name = av_default_item_name, .option = options, .version = LIBAVUTIL_VERSION_INT, }; + +const FFCodec ff_aac_decoder = { + .p.name = "aac", + CODEC_LONG_NAME("AAC (Advanced Audio Coding)"), + .p.type = AVMEDIA_TYPE_AUDIO, + .p.id = AV_CODEC_ID_AAC, + .p.priv_class = &decoder_class, + .priv_data_size = sizeof(AACDecContext), + .init = aac_decode_init, + .close = decode_close, + FF_CODEC_DECODE_CB(aac_decode_frame), + .p.sample_fmts = (const enum AVSampleFormat[]) { + AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE + }, + .p.capabilities = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1, + .caps_internal = FF_CODEC_CAP_INIT_CLEANUP, + .p.ch_layouts = ff_aac_ch_layout, + .flush = flush, + .p.profiles = NULL_IF_CONFIG_SMALL(ff_aac_profiles), +}; + +const FFCodec ff_aac_fixed_decoder = { + .p.name = "aac_fixed", + CODEC_LONG_NAME("AAC (Advanced Audio Coding)"), + .p.type = AVMEDIA_TYPE_AUDIO, + .p.id = AV_CODEC_ID_AAC, + .p.priv_class = &decoder_class, + .priv_data_size = sizeof(AACDecContext), + .init = aac_decode_init_fixed, + .close = decode_close, + FF_CODEC_DECODE_CB(aac_decode_frame), + .p.sample_fmts = (const enum AVSampleFormat[]) { + AV_SAMPLE_FMT_S32P, AV_SAMPLE_FMT_NONE + }, + .p.capabilities = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1, + .caps_internal = FF_CODEC_CAP_INIT_CLEANUP, + .p.ch_layouts = ff_aac_ch_layout, + .p.profiles = NULL_IF_CONFIG_SMALL(ff_aac_profiles), + .flush = flush, +}; diff --git a/libavcodec/aac/aacdec.h b/libavcodec/aac/aacdec.h index c8107f6bce..d994fe7981 100644 --- a/libavcodec/aac/aacdec.h +++ b/libavcodec/aac/aacdec.h @@ -339,10 +339,11 @@ struct AACDecContext { #define fdsp RENAME_FIXED(fdsp) #endif -extern const struct AVClass ff_aac_decoder_class; +extern const AACDecDSP aac_dsp; +extern const AACDecDSP aac_dsp_fixed; -int ff_aac_decode_init_common(struct AVCodecContext *avctx); -int ff_aac_decode_close(struct AVCodecContext *avctx); +extern const AACDecProc aac_proc; +extern const AACDecProc aac_proc_fixed; void ff_aacdec_init_mips(AACDecContext *c); diff --git a/libavcodec/aac/aacdec_latm.h b/libavcodec/aac/aacdec_latm.h index 0226aebba4..22153dec83 100644 --- a/libavcodec/aac/aacdec_latm.h +++ b/libavcodec/aac/aacdec_latm.h @@ -335,7 +335,7 @@ const FFCodec ff_aac_latm_decoder = { .p.id = AV_CODEC_ID_AAC_LATM, .priv_data_size = sizeof(struct LATMContext), .init = latm_decode_init, - .close = ff_aac_decode_close, + .close = decode_close, FF_CODEC_DECODE_CB(latm_decode_frame), .p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE diff --git a/libavcodec/aacdec.c b/libavcodec/aacdec.c deleted file mode 100644 index 666e0c89db..0000000000 --- a/libavcodec/aacdec.c +++ /dev/null @@ -1,2495 +0,0 @@ -/* - * AAC decoder - * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org ) - * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com ) - * Copyright (c) 2008-2013 Alex Converse - * - * AAC LATM decoder - * Copyright (c) 2008-2010 Paul Kendall - * Copyright (c) 2010 Janne Grunau - * - * This file is part of FFmpeg. - * - * FFmpeg is free software; you can redistribute it and/or - * modify it under the terms of the GNU Lesser General Public - * License as published by the Free Software Foundation; either - * version 2.1 of the License, or (at your option) any later version. - * - * FFmpeg is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Lesser General Public License for more details. - * - * You should have received a copy of the GNU Lesser General Public - * License along with FFmpeg; if not, write to the Free Software - * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA - */ - -/** - * @file - * AAC decoder - * @author Oded Shimon ( ods15 ods15 dyndns org ) - * @author Maxim Gavrilov ( maxim.gavrilov gmail com ) - */ - -#define USE_FIXED 1 // aacsbr.h breaks without this - -#include "libavutil/float_dsp.h" -#include "avcodec.h" -#include "codec_internal.h" -#include "get_bits.h" -#include "kbdwin.h" -#include "sinewin.h" - -#include "aac.h" -#include "aac/aacdec.h" -#include "aactab.h" -#include "aac/aacdec_tab.h" -#include "adts_header.h" -#include "cbrt_data.h" -#include "aacsbr.h" -#include "mpeg4audio.h" -#include "profiles.h" -#include "libavutil/intfloat.h" - -#include -#include -#include -#include - -#if ARCH_ARM -# include "arm/aac.h" -#elif ARCH_MIPS -# include "mips/aacdec_mips.h" -#endif - -/* - * 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 (not in fixed point code) 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/channel_layout.h" -#include "libavutil/thread.h" -#include "decode.h" -#include "internal.h" -#include "lpc_functions.h" - -static int output_configure(AACDecContext *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(AACDecContext *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]) { - int ret; - if (ac->is_fixed) - ret = ff_aac_sbr_ctx_alloc_init_fixed(ac, &ac->che[type][id], type); - else - ret = ff_aac_sbr_ctx_alloc_init(ac, &ac->che[type][id], type); - if (ret < 0) - return ret; - } - 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]) { - if (ac->is_fixed) - ff_aac_sbr_ctx_close_fixed(ac->che[type][id]); - else - ff_aac_sbr_ctx_close(ac->che[type][id]); - } - av_freep(&ac->che[type][id]); - } - return 0; -} - -static int frame_configure_elements(AVCodecContext *avctx) -{ - AACDecContext *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].AAC_RENAME(output) = che->ch[0].AAC_RENAME(ret_buf); - che->ch[1].AAC_RENAME(output) = che->ch[1].AAC_RENAME(ret_buf); - } - } - } - - /* get output buffer */ - av_frame_unref(ac->frame); - if (!avctx->ch_layout.nb_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->ch_layout.nb_channels; ch++) { - if (ac->output_element[ch]) - ac->output_element[ch]->AAC_RENAME(output) = (INTFLOAT *)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, uint64_t *layout) -{ - 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 - }; - if (e2c_vec[offset].av_position != UINT64_MAX) - *layout |= e2c_vec[offset].av_position; - - 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 - }; - if (left != UINT64_MAX) - *layout |= left; - - if (right != UINT64_MAX) - *layout |= right; - - 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 ^= (pos != AAC_CHANNEL_LFE); - } - } - if (sce_parity && - (pos == AAC_CHANNEL_FRONT && first_cpe)) - return -1; - - return num_pos_channels; -} - -static int assign_channels(struct elem_to_channel e2c_vec[MAX_ELEM_ID], uint8_t (*layout_map)[3], - uint64_t *layout, int tags, int layer, int pos, int *current) -{ - int i = *current, j = 0; - int nb_channels = count_paired_channels(layout_map, tags, pos, i); - - if (nb_channels < 0 || nb_channels > 5) - return 0; - - if (pos == AAC_CHANNEL_LFE) { - while (nb_channels) { - if (ff_aac_channel_map[layer][pos - 1][j] == AV_CHAN_NONE) - return -1; - e2c_vec[i] = (struct elem_to_channel) { - .av_position = 1ULL << ff_aac_channel_map[layer][pos - 1][j], - .syn_ele = layout_map[i][0], - .elem_id = layout_map[i][1], - .aac_position = pos - }; - *layout |= e2c_vec[i].av_position; - i++; - j++; - nb_channels--; - } - *current = i; - - return 0; - } - - while (nb_channels & 1) { - if (ff_aac_channel_map[layer][pos - 1][0] == AV_CHAN_NONE) - return -1; - if (ff_aac_channel_map[layer][pos - 1][0] == AV_CHAN_UNUSED) - break; - e2c_vec[i] = (struct elem_to_channel) { - .av_position = 1ULL << ff_aac_channel_map[layer][pos - 1][0], - .syn_ele = layout_map[i][0], - .elem_id = layout_map[i][1], - .aac_position = pos - }; - *layout |= e2c_vec[i].av_position; - i++; - nb_channels--; - } - - j = (pos != AAC_CHANNEL_SIDE) && nb_channels <= 3 ? 3 : 1; - while (nb_channels >= 2) { - if (ff_aac_channel_map[layer][pos - 1][j] == AV_CHAN_NONE || - ff_aac_channel_map[layer][pos - 1][j+1] == AV_CHAN_NONE) - return -1; - i += assign_pair(e2c_vec, layout_map, i, - 1ULL << ff_aac_channel_map[layer][pos - 1][j], - 1ULL << ff_aac_channel_map[layer][pos - 1][j+1], - pos, layout); - j += 2; - nb_channels -= 2; - } - while (nb_channels & 1) { - if (ff_aac_channel_map[layer][pos - 1][5] == AV_CHAN_NONE) - return -1; - e2c_vec[i] = (struct elem_to_channel) { - .av_position = 1ULL << ff_aac_channel_map[layer][pos - 1][5], - .syn_ele = layout_map[i][0], - .elem_id = layout_map[i][1], - .aac_position = pos - }; - *layout |= e2c_vec[i].av_position; - i++; - nb_channels--; - } - if (nb_channels) - return -1; - - *current = i; - - return 0; -} - -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 } }; - uint64_t layout = 0; - - if (FF_ARRAY_ELEMS(e2c_vec) < tags) - return 0; - - for (n = 0, i = 0; n < 3 && i < tags; n++) { - int ret = assign_channels(e2c_vec, layout_map, &layout, tags, n, AAC_CHANNEL_FRONT, &i); - if (ret < 0) - return 0; - ret = assign_channels(e2c_vec, layout_map, &layout, tags, n, AAC_CHANNEL_SIDE, &i); - if (ret < 0) - return 0; - ret = assign_channels(e2c_vec, layout_map, &layout, tags, n, AAC_CHANNEL_BACK, &i); - if (ret < 0) - return 0; - ret = assign_channels(e2c_vec, layout_map, &layout, tags, n, AAC_CHANNEL_LFE, &i); - if (ret < 0) - return 0; - } - - total_non_cc_elements = n = i; - - if (layout == AV_CH_LAYOUT_22POINT2) { - // For 22.2 reorder the result as needed - FFSWAP(struct elem_to_channel, e2c_vec[2], e2c_vec[0]); // FL & FR first (final), FC third - FFSWAP(struct elem_to_channel, e2c_vec[2], e2c_vec[1]); // FC second (final), FLc & FRc third - FFSWAP(struct elem_to_channel, e2c_vec[6], e2c_vec[2]); // LFE1 third (final), FLc & FRc seventh - FFSWAP(struct elem_to_channel, e2c_vec[4], e2c_vec[3]); // BL & BR fourth (final), SiL & SiR fifth - FFSWAP(struct elem_to_channel, e2c_vec[6], e2c_vec[4]); // FLc & FRc fifth (final), SiL & SiR seventh - FFSWAP(struct elem_to_channel, e2c_vec[7], e2c_vec[6]); // LFE2 seventh (final), SiL & SiR eight (final) - FFSWAP(struct elem_to_channel, e2c_vec[9], e2c_vec[8]); // TpFL & TpFR ninth (final), TFC tenth (final) - FFSWAP(struct elem_to_channel, e2c_vec[11], e2c_vec[10]); // TC eleventh (final), TpSiL & TpSiR twelth - FFSWAP(struct elem_to_channel, e2c_vec[12], e2c_vec[11]); // TpBL & TpBR twelth (final), TpSiL & TpSiR thirteenth (final) - } else { - // For everything else, utilize the AV channel position define as a - // stable sort. - 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); - - } - - 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; - } - - return layout; -} - -/** - * Save current output configuration if and only if it has been locked. - */ -static int push_output_configuration(AACDecContext *ac) -{ - int pushed = 0; - - if (ac->oc[1].status == OC_LOCKED || ac->oc[0].status == OC_NONE) { - ac->oc[0] = ac->oc[1]; - pushed = 1; - } - ac->oc[1].status = OC_NONE; - return pushed; -} - -/** - * Restore the previous output configuration if and only if the current - * configuration is unlocked. - */ -static void pop_output_configuration(AACDecContext *ac) -{ - if (ac->oc[1].status != OC_LOCKED && ac->oc[0].status != OC_NONE) { - ac->oc[1] = ac->oc[0]; - ac->avctx->ch_layout = ac->oc[1].ch_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(AACDecContext *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]++; - if (id_map[type][id] >= MAX_ELEM_ID) { - avpriv_request_sample(ac->avctx, "Too large remapped id"); - return AVERROR_PATCHWELCOME; - } - } - // Try to sniff a reasonable channel order, otherwise output the - // channels in the order the PCE declared them. - if (ac->output_channel_order == CHANNEL_ORDER_DEFAULT) - 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; - } - } - - av_channel_layout_uninit(&ac->oc[1].ch_layout); - if (layout) - av_channel_layout_from_mask(&ac->oc[1].ch_layout, layout); - else { - ac->oc[1].ch_layout.order = AV_CHANNEL_ORDER_UNSPEC; - ac->oc[1].ch_layout.nb_channels = channels; - } - - av_channel_layout_copy(&avctx->ch_layout, &ac->oc[1].ch_layout); - 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) -{ - AACDecContext *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(AACDecContext *ac, AVCodecContext *avctx, - uint8_t (*layout_map)[3], - int *tags, - int channel_config) -{ - if (channel_config < 1 || (channel_config > 7 && channel_config < 11) || - channel_config > 14) { - av_log(avctx, AV_LOG_ERROR, - "invalid default channel configuration (%d)\n", - channel_config); - return AVERROR_INVALIDDATA; - } - *tags = ff_tags_per_config[channel_config]; - memcpy(layout_map, ff_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) { - layout_map[2][2] = AAC_CHANNEL_BACK; - - if (!ac || !ac->warned_71_wide++) { - 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); - } - } - - return 0; -} - -static ChannelElement *get_che(AACDecContext *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, 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"); - - 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, 1) < 0) - return NULL; - - 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 14: - if (ac->tags_mapped > 2 && ((type == TYPE_CPE && elem_id < 3) || - (type == TYPE_LFE && elem_id < 1))) { - ac->tags_mapped++; - return ac->tag_che_map[type][elem_id] = ac->che[type][elem_id]; - } - case 13: - if (ac->tags_mapped > 3 && ((type == TYPE_CPE && elem_id < 8) || - (type == TYPE_SCE && elem_id < 6) || - (type == TYPE_LFE && elem_id < 2))) { - ac->tags_mapped++; - return ac->tag_che_map[type][elem_id] = ac->che[type][elem_id]; - } - 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 == 3 && 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 == ff_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 == ff_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->tags_mapped == 1 && ac->oc[1].m4ac.chan_config == 2 && - type == TYPE_SCE) { - ac->tags_mapped++; - return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1]; - } - 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++; - } -} - -static inline void relative_align_get_bits(GetBitContext *gb, - int reference_position) { - int n = (reference_position - get_bits_count(gb) & 7); - if (n) - skip_bits(gb, n); -} - -/** - * 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 byte_align_ref) -{ - 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) < 5 * (num_front + num_side + num_back + num_cc) + 4 *(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; - - relative_align_get_bits(gb, byte_align_ref); - - /* 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 AACDecContext, 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(AACDecContext *ac, AVCodecContext *avctx, - GetBitContext *gb, - int get_bit_alignment, - 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; - - m4ac->frame_length_short = get_bits1(gb); - if (m4ac->frame_length_short && m4ac->sbr == 1) { - avpriv_report_missing_feature(avctx, "SBR with 960 frame length"); - if (ac) ac->warned_960_sbr = 1; - m4ac->sbr = 0; - m4ac->ps = 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, get_bit_alignment); - if (tags < 0) - return tags; - } else { - if ((ret = set_default_channel_config(ac, 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(AACDecContext *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(ac, 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 AACDecContext, may be null - * @param avctx pointer to AVCCodecContext, used for logging - * @param m4ac pointer to MPEG4AudioConfig, used for parsing - * @param gb buffer holding an audio specific config - * @param get_bit_alignment relative alignment for byte align operations - * @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_gb(AACDecContext *ac, - AVCodecContext *avctx, - MPEG4AudioConfig *m4ac, - GetBitContext *gb, - int get_bit_alignment, - int sync_extension) -{ - int i, ret; - GetBitContext gbc = *gb; - MPEG4AudioConfig m4ac_bak = *m4ac; - - if ((i = ff_mpeg4audio_get_config_gb(m4ac, &gbc, sync_extension, avctx)) < 0) { - *m4ac = m4ac_bak; - return AVERROR_INVALIDDATA; - } - - if (m4ac->sampling_index > 12) { - av_log(avctx, AV_LOG_ERROR, - "invalid sampling rate index %d\n", - m4ac->sampling_index); - *m4ac = m4ac_bak; - 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); - *m4ac = m4ac_bak; - return AVERROR_INVALIDDATA; - } - - skip_bits_long(gb, i); - - switch (m4ac->object_type) { - case AOT_AAC_MAIN: - case AOT_AAC_LC: - case AOT_AAC_SSR: - case AOT_AAC_LTP: - case AOT_ER_AAC_LC: - case AOT_ER_AAC_LD: - if ((ret = decode_ga_specific_config(ac, avctx, gb, get_bit_alignment, - 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); -} - -static int decode_audio_specific_config(AACDecContext *ac, - AVCodecContext *avctx, - MPEG4AudioConfig *m4ac, - const uint8_t *data, int64_t bit_size, - int sync_extension) -{ - int i, ret; - GetBitContext gb; - - if (bit_size < 0 || bit_size > INT_MAX) { - av_log(avctx, AV_LOG_ERROR, "Audio specific config size is invalid\n"); - return AVERROR_INVALIDDATA; - } - - ff_dlog(avctx, "audio specific config size %d\n", (int)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; - - return decode_audio_specific_config_gb(ac, avctx, m4ac, &gb, 0, - sync_extension); -} - -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 av_cold void aac_static_table_init(void) -{ - ff_aac_sbr_init(); - ff_aac_sbr_init_fixed(); - - ff_aacdec_common_init_once(); -} - -static AVOnce aac_table_init = AV_ONCE_INIT; - -static av_cold int aac_decode_init_internal(AVCodecContext *avctx) -{ - AACDecContext *ac = avctx->priv_data; - int ret; - - if (avctx->sample_rate > 96000) - return AVERROR_INVALIDDATA; - - ret = ff_thread_once(&aac_table_init, &aac_static_table_init); - if (ret != 0) - return AVERROR_UNKNOWN; - - ac->avctx = avctx; - ac->oc[1].m4ac.sample_rate = avctx->sample_rate; - - if (ac->is_fixed) - avctx->sample_fmt = AV_SAMPLE_FMT_S32P; - else - 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 * 8LL, - 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->ch_layout.nb_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->ch_layout.nb_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(ac, 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; - } - } - - return ff_aac_decode_init_common(avctx); -} - -static av_cold int aac_decode_init(AVCodecContext *avctx) -{ - AACDecContext *ac = avctx->priv_data; - ac->is_fixed = 0; - return aac_decode_init_internal(avctx); -} - -static av_cold int aac_decode_init_fixed(AVCodecContext *avctx) -{ - AACDecContext *ac = avctx->priv_data; - ac->is_fixed = 1; - return aac_decode_init_internal(avctx); -} - -/** - * Skip data_stream_element; reference: table 4.10. - */ -static int skip_data_stream_element(AACDecContext *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(AACDecContext *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(AACDecContext *ac, LongTermPrediction *ltp, - GetBitContext *gb, uint8_t max_sfb) -{ - int sfb; - - ltp->lag = get_bits(gb, 11); - if (ac->is_fixed) - ltp->coef_fixed = Q30(ff_ltp_coef[get_bits(gb, 3)]); - else - ltp->coef = ff_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(AACDecContext *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; - int ret_fail = AVERROR_INVALIDDATA; - - 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; - if (m4ac->frame_length_short) { - ics->swb_offset = ff_swb_offset_120[sampling_index]; - ics->num_swb = ff_aac_num_swb_120[sampling_index]; - } else { - 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) { - ret_fail = AVERROR_BUG; - goto fail; - } - } else { - if (m4ac->frame_length_short) { - ics->num_swb = ff_aac_num_swb_960[sampling_index]; - ics->swb_offset = ff_swb_offset_960[sampling_index]; - } else { - ics->num_swb = ff_aac_num_swb_1024[sampling_index]; - ics->swb_offset = ff_swb_offset_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"); - ret_fail = AVERROR_PATCHWELCOME; - goto fail; - } - if ((ics->ltp.present = get_bits(gb, 1))) - decode_ltp(ac, &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 ret_fail; -} - -/** - * 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(AACDecContext *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(AACDecContext *ac, int sfo[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]; - switch (band_type[idx]) { - case ZERO_BT: - for (; i < run_end; i++, idx++) - sfo[idx] = 0; - break; - case INTENSITY_BT: /* fallthrough */ - case INTENSITY_BT2: - for (; i < run_end; i++, idx++) { - offset[2] += get_vlc2(gb, ff_vlc_scalefactors, 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); - } - sfo[idx] = clipped_offset; - } - break; - case 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, ff_vlc_scalefactors, 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); - } - sfo[idx] = clipped_offset; - } - break; - default: - for (; i < run_end; i++, idx++) { - offset[0] += get_vlc2(gb, ff_vlc_scalefactors, 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; - } - sfo[idx] = offset[0]; - } - break; - } - } - } - 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(AACDecContext *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++) { - if (ac->is_fixed) - tns->coef_fixed[w][filt][i] = Q31(ff_tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)]); - else - tns->coef[w][filt][i] = ff_tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)]; - } - } - } - } - } - return 0; -} - -/** - * 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) -{ - 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])); - } -} - -static void decode_gain_control(SingleChannelElement * sce, GetBitContext * gb) -{ - // wd_num, wd_test, aloc_size - static const uint8_t gain_mode[4][3] = { - {1, 0, 5}, // ONLY_LONG_SEQUENCE = 0, - {2, 1, 2}, // LONG_START_SEQUENCE, - {8, 0, 2}, // EIGHT_SHORT_SEQUENCE, - {2, 1, 5}, // LONG_STOP_SEQUENCE - }; - - const int mode = sce->ics.window_sequence[0]; - uint8_t bd, wd, ad; - - // FIXME: Store the gain control data on |sce| and do something with it. - uint8_t max_band = get_bits(gb, 2); - for (bd = 0; bd < max_band; bd++) { - for (wd = 0; wd < gain_mode[mode][0]; wd++) { - uint8_t adjust_num = get_bits(gb, 3); - for (ad = 0; ad < adjust_num; ad++) { - skip_bits(gb, 4 + ((wd == 0 && gain_mode[mode][1]) - ? 4 - : gain_mode[mode][2])); - } - } - } -} - -/** - * 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 - */ -int ff_aac_decode_ics(AACDecContext *ac, SingleChannelElement *sce, - GetBitContext *gb, int common_window, int scale_flag) -{ - Pulse pulse; - TemporalNoiseShaping *tns = &sce->tns; - IndividualChannelStream *ics = &sce->ics; - 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) { - ret = decode_ics_info(ac, ics, gb); - if (ret < 0) - goto fail; - } - - if ((ret = decode_band_types(ac, sce->band_type, - sce->band_type_run_end, gb, ics)) < 0) - goto fail; - if ((ret = decode_scalefactors(ac, sce->sfo, gb, global_gain, ics, - sce->band_type, sce->band_type_run_end)) < 0) - goto fail; - - ac->dsp.dequant_scalefactors(sce); - - 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"); - ret = AVERROR_INVALIDDATA; - goto fail; - } - if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) { - av_log(ac->avctx, AV_LOG_ERROR, - "Pulse data corrupt or invalid.\n"); - ret = AVERROR_INVALIDDATA; - goto fail; - } - } - tns->present = get_bits1(gb); - if (tns->present && !er_syntax) { - ret = decode_tns(ac, tns, gb, ics); - if (ret < 0) - goto fail; - } - if (!eld_syntax && get_bits1(gb)) { - decode_gain_control(sce, gb); - if (!ac->warned_gain_control) { - avpriv_report_missing_feature(ac->avctx, "Gain control"); - ac->warned_gain_control = 1; - } - } - // 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) { - ret = decode_tns(ac, tns, gb, ics); - if (ret < 0) - goto fail; - } - } - - ret = ac->proc.decode_spectrum_and_dequant(ac, gb, - pulse_present ? &pulse : NULL, - sce); - if (ret < 0) - goto fail; - - if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN && !common_window) - ac->dsp.apply_prediction(ac, sce); - - return 0; -fail: - tns->present = 0; - return ret; -} - -/** - * Decode a channel_pair_element; reference: table 4.4. - * - * @return Returns error status. 0 - OK, !0 - error - */ -static int decode_cpe(AACDecContext *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(ac, &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 = ff_aac_decode_ics(ac, &cpe->ch[0], gb, common_window, 0))) - return ret; - if ((ret = ff_aac_decode_ics(ac, &cpe->ch[1], gb, common_window, 0))) - return ret; - - if (common_window) { - if (ms_present) - ac->dsp.apply_mid_side_stereo(ac, cpe); - if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN) { - ac->dsp.apply_prediction(ac, &cpe->ch[0]); - ac->dsp.apply_prediction(ac, &cpe->ch[1]); - } - } - - ac->dsp.apply_intensity_stereo(ac, cpe, ms_present); - 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(AACDecContext *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(AACDecContext *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.frame_length_short) { - if (!ac->warned_960_sbr) - avpriv_report_missing_feature(ac->avctx, - "SBR with 960 frame length"); - ac->warned_960_sbr = 1; - skip_bits_long(gb, 8 * cnt - 4); - 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->ch_layout.nb_channels == 1) { - ac->oc[1].m4ac.sbr = 1; - ac->oc[1].m4ac.ps = 1; - ac->avctx->profile = AV_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 = AV_PROFILE_AAC_HE; - } - - if (ac->is_fixed) - res = ff_aac_sbr_decode_extension_fixed(ac, che, gb, crc_flag, cnt, elem_type); - else - res = ff_aac_sbr_decode_extension(ac, che, gb, crc_flag, cnt, elem_type); - - - if (ac->oc[1].m4ac.ps == 1 && !ac->warned_he_aac_mono) { - av_log(ac->avctx, AV_LOG_VERBOSE, "Treating HE-AAC mono as stereo.\n"); - ac->warned_he_aac_mono = 1; - } - 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; -} - -/** - * channel coupling transformation interface - * - * @param apply_coupling_method pointer to (in)dependent coupling function - */ -static void apply_channel_coupling(AACDecContext *ac, ChannelElement *cc, - enum RawDataBlockType type, int elem_id, - enum CouplingPoint coupling_point, - void (*apply_coupling_method)(AACDecContext *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 samples, applying all supported tools as appropriate. - */ -static void spectral_to_sample(AACDecContext *ac, int samples) -{ - int i, type; - void (*imdct_and_window)(AACDecContext *ac, SingleChannelElement *sce); - switch (ac->oc[1].m4ac.object_type) { - case AOT_ER_AAC_LD: - imdct_and_window = ac->dsp.imdct_and_windowing_ld; - break; - case AOT_ER_AAC_ELD: - imdct_and_window = ac->dsp.imdct_and_windowing_eld; - break; - default: - if (ac->oc[1].m4ac.frame_length_short) - imdct_and_window = ac->dsp.imdct_and_windowing_960; - else - imdct_and_window = ac->dsp.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, ac->dsp.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->dsp.apply_ltp(ac, &che->ch[0]); - if (che->ch[1].ics.ltp.present && type == TYPE_CPE) - ac->dsp.apply_ltp(ac, &che->ch[1]); - } - } - if (che->ch[0].tns.present) - ac->dsp.apply_tns(che->ch[0].AAC_RENAME(coeffs), - &che->ch[0].tns, &che->ch[0].ics, 1); - if (che->ch[1].tns.present) - ac->dsp.apply_tns(che->ch[1].AAC_RENAME(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, ac->dsp.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->dsp.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->dsp.update_ltp(ac, &che->ch[1]); - } - if (ac->oc[1].m4ac.sbr > 0) { - if (ac->is_fixed) - ff_aac_sbr_apply_fixed(ac, che, type, - che->ch[0].AAC_RENAME(output), - che->ch[1].AAC_RENAME(output)); - else - ff_aac_sbr_apply(ac, che, type, - che->ch[0].AAC_RENAME(output), - che->ch[1].AAC_RENAME(output)); - } - } - if (type <= TYPE_CCE) - apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, ac->dsp.apply_independent_coupling); - ac->dsp.clip_output(ac, che, type, samples); - 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(AACDecContext *ac, GetBitContext *gb) -{ - int size; - AACADTSHeaderInfo hdr_info; - uint8_t layout_map[MAX_ELEM_ID*4][3]; - int layout_map_tags, ret; - - size = ff_adts_header_parse(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, 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, AVFrame *frame, - int *got_frame_ptr, GetBitContext *gb) -{ - AACDecContext *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 = frame; - - if ((err = frame_configure_elements(avctx)) < 0) - return err; - - // The AV_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 < ff_tags_per_config[chan_config]; i++) { - const int elem_type = ff_aac_channel_layout_map[chan_config-1][i][0]; - const int elem_id = ff_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 = ff_aac_decode_ics(ac, &che->ch[0], gb, 0, 0); - break; - case TYPE_CPE: - err = decode_cpe(ac, gb, che); - break; - case TYPE_LFE: - err = ff_aac_decode_ics(ac, &che->ch[0], gb, 0, 0); - break; - } - if (err < 0) - return err; - } - - spectral_to_sample(ac, samples); - - if (!ac->frame->data[0] && samples) { - av_log(avctx, AV_LOG_ERROR, "no frame data found\n"); - return AVERROR_INVALIDDATA; - } - - 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, AVFrame *frame, - int *got_frame_ptr, GetBitContext *gb, - const AVPacket *avpkt) -{ - AACDecContext *ac = avctx->priv_data; - ChannelElement *che = NULL, *che_prev = NULL; - enum RawDataBlockType elem_type, che_prev_type = TYPE_END; - int err, elem_id; - int samples = 0, multiplier, audio_found = 0, pce_found = 0; - int is_dmono, sce_count = 0; - int payload_alignment; - uint8_t che_presence[4][MAX_ELEM_ID] = {{0}}; - - ac->frame = frame; - - 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 AV_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; - - payload_alignment = get_bits_count(gb); - 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->ch_layout.nb_channels && elem_type != TYPE_PCE) { - err = AVERROR_INVALIDDATA; - goto fail; - } - - if (elem_type < TYPE_DSE) { - if (che_presence[elem_type][elem_id]) { - int error = che_presence[elem_type][elem_id] > 1; - av_log(ac->avctx, error ? AV_LOG_ERROR : AV_LOG_DEBUG, "channel element %d.%d duplicate\n", - elem_type, elem_id); - if (error) { - err = AVERROR_INVALIDDATA; - goto fail; - } - } - che_presence[elem_type][elem_id]++; - - 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 = ac->oc[1].m4ac.frame_length_short ? 960 : 1024; - che->present = 1; - } - - switch (elem_type) { - - case TYPE_SCE: - err = ff_aac_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 = ac->proc.decode_cce(ac, gb, che); - break; - - case TYPE_LFE: - err = ff_aac_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] = {{0}}; - int tags; - - int pushed = push_output_configuration(ac); - if (pce_found && !pushed) { - err = AVERROR_INVALIDDATA; - goto fail; - } - - tags = decode_pce(avctx, &ac->oc[1].m4ac, layout_map, gb, - payload_alignment); - 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"); - pop_output_configuration(ac); - } else { - err = output_configure(ac, layout_map, tags, OC_TRIAL_PCE, 1); - if (!err) - ac->oc[1].m4ac.chan_config = 0; - pce_found = 1; - } - break; - } - - 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; - } - err = 0; - while (elem_id > 0) { - int ret = decode_extension_payload(ac, gb, elem_id, che_prev, che_prev_type); - if (ret < 0) { - err = ret; - break; - } - elem_id -= ret; - } - break; - - default: - err = AVERROR_BUG; /* should not happen, but keeps compiler happy */ - break; - } - - if (elem_type < TYPE_DSE) { - che_prev = che; - che_prev_type = 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->ch_layout.nb_channels) { - *got_frame_ptr = 0; - return 0; - } - - multiplier = (ac->oc[1].m4ac.sbr == 1) ? ac->oc[1].m4ac.ext_sample_rate > ac->oc[1].m4ac.sample_rate : 0; - samples <<= multiplier; - - spectral_to_sample(ac, 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; - } - - 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 && - !av_channel_layout_compare(&ac->oc[1].ch_layout, - &(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO); - if (is_dmono) { - if (ac->dmono_mode == 1) - frame->data[1] = frame->data[0]; - else if (ac->dmono_mode == 2) - frame->data[0] = frame->data[1]; - } - - return 0; -fail: - pop_output_configuration(ac); - return err; -} - -static int aac_decode_frame(AVCodecContext *avctx, AVFrame *frame, - int *got_frame_ptr, AVPacket *avpkt) -{ - AACDecContext *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; - size_t new_extradata_size; - const uint8_t *new_extradata = av_packet_get_side_data(avpkt, - AV_PKT_DATA_NEW_EXTRADATA, - &new_extradata_size); - size_t 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) { - /* discard previous configuration */ - ac->oc[1].status = OC_NONE; - err = decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac, - new_extradata, - new_extradata_size * 8LL, 1); - if (err < 0) { - return err; - } - } - - 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_bits8(&gb, buf, buf_size)) < 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, frame, got_frame_ptr, &gb); - break; - default: - err = aac_decode_frame_int(avctx, frame, 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; -} - -#include "libavcodec/aac/aacdec_latm.h" - -const FFCodec ff_aac_decoder = { - .p.name = "aac", - CODEC_LONG_NAME("AAC (Advanced Audio Coding)"), - .p.type = AVMEDIA_TYPE_AUDIO, - .p.id = AV_CODEC_ID_AAC, - .p.priv_class = &ff_aac_decoder_class, - .priv_data_size = sizeof(AACDecContext), - .init = aac_decode_init, - .close = ff_aac_decode_close, - FF_CODEC_DECODE_CB(aac_decode_frame), - .p.sample_fmts = (const enum AVSampleFormat[]) { - AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE - }, - .p.capabilities = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1, - .caps_internal = FF_CODEC_CAP_INIT_CLEANUP, - .p.ch_layouts = ff_aac_ch_layout, - .flush = flush, - .p.profiles = NULL_IF_CONFIG_SMALL(ff_aac_profiles), -}; - -const FFCodec ff_aac_fixed_decoder = { - .p.name = "aac_fixed", - CODEC_LONG_NAME("AAC (Advanced Audio Coding)"), - .p.type = AVMEDIA_TYPE_AUDIO, - .p.id = AV_CODEC_ID_AAC, - .p.priv_class = &ff_aac_decoder_class, - .priv_data_size = sizeof(AACDecContext), - .init = aac_decode_init_fixed, - .close = ff_aac_decode_close, - FF_CODEC_DECODE_CB(aac_decode_frame), - .p.sample_fmts = (const enum AVSampleFormat[]) { - AV_SAMPLE_FMT_S32P, AV_SAMPLE_FMT_NONE - }, - .p.capabilities = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1, - .caps_internal = FF_CODEC_CAP_INIT_CLEANUP, - .p.ch_layouts = ff_aac_ch_layout, - .p.profiles = NULL_IF_CONFIG_SMALL(ff_aac_profiles), - .flush = flush, -};