More OKed hunks of the AAC decoder from SoC
Originally committed as revision 14694 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Robert Swain
parent
3f66d16821
commit
9cc04edff9
408
libavcodec/aac.c
408
libavcodec/aac.c
@ -99,6 +99,40 @@ static VLC vlc_scalefactors;
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static VLC vlc_spectral[11];
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/**
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* Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit.
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*
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* @param cpe_map Stereo (Channel Pair Element) map, NULL if stereo bit is not present.
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* @param sce_map mono (Single Channel Element) map
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* @param type speaker type/position for these channels
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*/
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static void decode_channel_map(enum ChannelPosition *cpe_map,
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enum ChannelPosition *sce_map, enum ChannelPosition type, GetBitContext * gb, int n) {
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while(n--) {
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enum ChannelPosition *map = cpe_map && get_bits1(gb) ? cpe_map : sce_map; // stereo or mono map
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map[get_bits(gb, 4)] = type;
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}
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}
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/**
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* Decode program configuration element; reference: table 4.2.
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*
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* @param new_che_pos New channel position configuration - we only do something if it differs from the current one.
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*
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* @return Returns error status. 0 - OK, !0 - error
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*/
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static int decode_pce(AACContext * ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
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GetBitContext * gb) {
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int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc;
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skip_bits(gb, 2); // object_type
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ac->m4ac.sampling_index = get_bits(gb, 4);
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if(ac->m4ac.sampling_index > 11) {
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av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
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return -1;
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}
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ac->m4ac.sample_rate = ff_mpeg4audio_sample_rates[ac->m4ac.sampling_index];
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num_front = get_bits(gb, 4);
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num_side = get_bits(gb, 4);
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num_back = get_bits(gb, 4);
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@ -130,6 +164,131 @@ static VLC vlc_spectral[11];
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return 0;
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}
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/**
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* Set up channel positions based on a default channel configuration
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* as specified in table 1.17.
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*
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* @param new_che_pos New channel position configuration - we only do something if it differs from the current one.
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*
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* @return Returns error status. 0 - OK, !0 - error
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*/
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static int set_default_channel_config(AACContext *ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
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int channel_config)
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{
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if(channel_config < 1 || channel_config > 7) {
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av_log(ac->avccontext, AV_LOG_ERROR, "invalid default channel configuration (%d)\n",
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channel_config);
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return -1;
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}
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/* default channel configurations:
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*
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* 1ch : front center (mono)
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* 2ch : L + R (stereo)
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* 3ch : front center + L + R
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* 4ch : front center + L + R + back center
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* 5ch : front center + L + R + back stereo
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* 6ch : front center + L + R + back stereo + LFE
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* 7ch : front center + L + R + outer front left + outer front right + back stereo + LFE
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*/
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if(channel_config != 2)
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new_che_pos[TYPE_SCE][0] = AAC_CHANNEL_FRONT; // front center (or mono)
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if(channel_config > 1)
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new_che_pos[TYPE_CPE][0] = AAC_CHANNEL_FRONT; // L + R (or stereo)
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if(channel_config == 4)
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new_che_pos[TYPE_SCE][1] = AAC_CHANNEL_BACK; // back center
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if(channel_config > 4)
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new_che_pos[TYPE_CPE][(channel_config == 7) + 1]
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= AAC_CHANNEL_BACK; // back stereo
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if(channel_config > 5)
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new_che_pos[TYPE_LFE][0] = AAC_CHANNEL_LFE; // LFE
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if(channel_config == 7)
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new_che_pos[TYPE_CPE][1] = AAC_CHANNEL_FRONT; // outer front left + outer front right
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return 0;
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}
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return -1;
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}
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if (get_bits1(gb)) // dependsOnCoreCoder
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skip_bits(gb, 14); // coreCoderDelay
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extension_flag = get_bits1(gb);
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if(ac->m4ac.object_type == AOT_AAC_SCALABLE ||
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ac->m4ac.object_type == AOT_ER_AAC_SCALABLE)
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skip_bits(gb, 3); // layerNr
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memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
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if (channel_config == 0) {
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skip_bits(gb, 4); // element_instance_tag
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if((ret = decode_pce(ac, new_che_pos, gb)))
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return ret;
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} else {
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if((ret = set_default_channel_config(ac, new_che_pos, channel_config)))
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return ret;
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}
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if((ret = output_configure(ac, ac->che_pos, new_che_pos)))
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return ret;
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if (extension_flag) {
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switch (ac->m4ac.object_type) {
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case AOT_ER_BSAC:
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skip_bits(gb, 5); // numOfSubFrame
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skip_bits(gb, 11); // layer_length
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break;
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case AOT_ER_AAC_LC:
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case AOT_ER_AAC_LTP:
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case AOT_ER_AAC_SCALABLE:
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case AOT_ER_AAC_LD:
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skip_bits(gb, 3); /* aacSectionDataResilienceFlag
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* aacScalefactorDataResilienceFlag
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* aacSpectralDataResilienceFlag
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*/
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break;
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}
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skip_bits1(gb); // extensionFlag3 (TBD in version 3)
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}
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return 0;
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}
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/**
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* Decode audio specific configuration; reference: table 1.13.
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*
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* @param data pointer to AVCodecContext extradata
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* @param data_size size of AVCCodecContext extradata
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*
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* @return Returns error status. 0 - OK, !0 - error
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*/
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static int decode_audio_specific_config(AACContext * ac, void *data, int data_size) {
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GetBitContext gb;
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int i;
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init_get_bits(&gb, data, data_size * 8);
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if((i = ff_mpeg4audio_get_config(&ac->m4ac, data, data_size)) < 0)
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return -1;
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if(ac->m4ac.sampling_index > 11) {
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av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
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return -1;
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}
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skip_bits_long(&gb, i);
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switch (ac->m4ac.object_type) {
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case AOT_AAC_LC:
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if (decode_ga_specific_config(ac, &gb, ac->m4ac.chan_config))
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return -1;
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break;
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default:
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av_log(ac->avccontext, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n",
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ac->m4ac.sbr == 1? "SBR+" : "", ac->m4ac.object_type);
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return -1;
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}
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return 0;
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}
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static av_cold int aac_decode_init(AVCodecContext * avccontext) {
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AACContext * ac = avccontext->priv_data;
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int i;
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@ -140,6 +299,7 @@ static av_cold int aac_decode_init(AVCodecContext * avccontext) {
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decode_audio_specific_config(ac, avccontext->extradata, avccontext->extradata_size))
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return -1;
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avccontext->sample_fmt = SAMPLE_FMT_S16;
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avccontext->sample_rate = ac->m4ac.sample_rate;
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avccontext->frame_size = 1024;
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@ -157,6 +317,8 @@ static av_cold int aac_decode_init(AVCodecContext * avccontext) {
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dsputil_init(&ac->dsp, avccontext);
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ac->random_state = 0x1f2e3d4c;
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// -1024 - Compensate wrong IMDCT method.
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// 32768 - Required to scale values to the correct range for the bias method
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// for float to int16 conversion.
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@ -188,6 +350,10 @@ static av_cold int aac_decode_init(AVCodecContext * avccontext) {
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return 0;
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}
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/**
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* Skip data_stream_element; reference: table 4.10.
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*/
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static void skip_data_stream_element(GetBitContext * gb) {
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int byte_align = get_bits1(gb);
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int count = get_bits(gb, 8);
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if (count == 255)
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@ -197,6 +363,27 @@ static av_cold int aac_decode_init(AVCodecContext * avccontext) {
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skip_bits_long(gb, 8 * count);
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}
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/**
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* Decode Individual Channel Stream info; reference: table 4.6.
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*
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* @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information.
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*/
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static int decode_ics_info(AACContext * ac, IndividualChannelStream * ics, GetBitContext * gb, int common_window) {
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if (get_bits1(gb)) {
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av_log(ac->avccontext, AV_LOG_ERROR, "Reserved bit set.\n");
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memset(ics, 0, sizeof(IndividualChannelStream));
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return -1;
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}
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ics->window_sequence[1] = ics->window_sequence[0];
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ics->window_sequence[0] = get_bits(gb, 2);
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ics->use_kb_window[1] = ics->use_kb_window[0];
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ics->use_kb_window[0] = get_bits1(gb);
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ics->num_window_groups = 1;
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ics->group_len[0] = 1;
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return 0;
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}
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/**
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* inverse quantization
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*
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@ -210,6 +397,15 @@ static inline float ivquant(int a) {
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return cbrtf(fabsf(a)) * a;
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}
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/**
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* Decode band types (section_data payload); reference: table 4.46.
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*
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* @param band_type array of the used band type
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* @param band_type_run_end array of the last scalefactor band of a band type run
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*
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* @return Returns error status. 0 - OK, !0 - error
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*/
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static int decode_band_types(AACContext * ac, enum BandType band_type[120],
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int band_type_run_end[120], GetBitContext * gb, IndividualChannelStream * ics) {
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int g, idx = 0;
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const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5;
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@ -232,7 +428,13 @@ static inline float ivquant(int a) {
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sect_len, ics->max_sfb);
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return -1;
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}
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}
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}
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return 0;
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}
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/**
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* Decode scalefactors; reference: table 4.47.
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*
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* @param mix_gain channel gain (Not used by AAC bitstream.)
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* @param global_gain first scalefactor value as scalefactors are differentially coded
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@ -313,6 +515,16 @@ static void decode_pulses(Pulse * pulse, GetBitContext * gb) {
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}
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}
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/**
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* Decode Mid/Side data; reference: table 4.54.
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*
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* @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s;
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* [1] mask is decoded from bitstream; [2] mask is all 1s;
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* [3] reserved for scalable AAC
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*/
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static void decode_mid_side_stereo(ChannelElement * cpe, GetBitContext * gb,
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int ms_present) {
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/**
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* Add pulses with particular amplitudes to the quantized spectral data; reference: 4.6.3.3.
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*
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@ -330,10 +542,109 @@ static void add_pulses(int icoef[1024], const Pulse * pulse, const IndividualCha
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}
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/**
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* Parse Spectral Band Replication extension data; reference: table 4.55.
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* Decode an individual_channel_stream payload; reference: table 4.44.
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*
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* @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information.
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* @param scale_flag scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.)
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*
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* @return Returns error status. 0 - OK, !0 - error
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*/
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static int decode_ics(AACContext * ac, SingleChannelElement * sce, GetBitContext * gb, int common_window, int scale_flag) {
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int icoeffs[1024];
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Pulse pulse;
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TemporalNoiseShaping * tns = &sce->tns;
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IndividualChannelStream * ics = &sce->ics;
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float * out = sce->coeffs;
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int global_gain, pulse_present = 0;
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/* These two assignments are to silence some GCC warnings about the
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* variables being used uninitialised when in fact they always are.
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*/
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pulse.num_pulse = 0;
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pulse.start = 0;
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global_gain = get_bits(gb, 8);
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if (!common_window && !scale_flag) {
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if (decode_ics_info(ac, ics, gb, 0) < 0)
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return -1;
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}
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if (decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics) < 0)
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return -1;
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if (decode_scalefactors(ac, sce->sf, gb, global_gain, ics, sce->band_type, sce->band_type_run_end) < 0)
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return -1;
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pulse_present = 0;
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if (!scale_flag) {
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if ((pulse_present = get_bits1(gb))) {
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if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
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av_log(ac->avccontext, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n");
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return -1;
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}
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decode_pulses(&pulse, gb);
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}
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if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics))
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return -1;
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if (get_bits1(gb)) {
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av_log_missing_feature(ac->avccontext, "SSR", 1);
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return -1;
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}
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}
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if (decode_spectrum(ac, icoeffs, gb, ics, sce->band_type) < 0)
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return -1;
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if (pulse_present)
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add_pulses(icoeffs, &pulse, ics);
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dequant(ac, out, icoeffs, sce->sf, ics, sce->band_type);
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return 0;
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}
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/**
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* Decode a channel_pair_element; reference: table 4.4.
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*
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* @param elem_id Identifies the instance of a syntax element.
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*
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* @return Returns error status. 0 - OK, !0 - error
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*/
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static int decode_cpe(AACContext * ac, GetBitContext * gb, int elem_id) {
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int i, ret, common_window, ms_present = 0;
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ChannelElement * cpe;
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cpe = ac->che[TYPE_CPE][elem_id];
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common_window = get_bits1(gb);
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if (common_window) {
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if (decode_ics_info(ac, &cpe->ch[0].ics, gb, 1))
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return -1;
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i = cpe->ch[1].ics.use_kb_window[0];
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cpe->ch[1].ics = cpe->ch[0].ics;
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cpe->ch[1].ics.use_kb_window[1] = i;
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ms_present = get_bits(gb, 2);
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if(ms_present == 3) {
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av_log(ac->avccontext, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
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return -1;
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} else if(ms_present)
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decode_mid_side_stereo(cpe, gb, ms_present);
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}
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if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0)))
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return ret;
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if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0)))
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return ret;
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if (common_window && ms_present)
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apply_mid_side_stereo(cpe);
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if (cpe->ch[1].ics.intensity_present)
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apply_intensity_stereo(cpe, ms_present);
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return 0;
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}
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/**
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* Decode Spectral Band Replication extension data; reference: table 4.55.
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*
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* @param crc flag indicating the presence of CRC checksum
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* @param cnt length of TYPE_FIL syntactic element in bytes
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*
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* @return Returns number of bytes consumed from the TYPE_FIL element.
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*/
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static int decode_sbr_extension(AACContext * ac, GetBitContext * gb, int crc, int cnt) {
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@ -343,6 +654,66 @@ static int decode_sbr_extension(AACContext * ac, GetBitContext * gb, int crc, in
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return cnt;
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}
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/**
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* Decode dynamic range information; reference: table 4.52.
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*
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* @param cnt length of TYPE_FIL syntactic element in bytes
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*
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* @return Returns number of bytes consumed.
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*/
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static int decode_dynamic_range(DynamicRangeControl *che_drc, GetBitContext * gb, int cnt) {
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int n = 1;
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int drc_num_bands = 1;
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int i;
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/* pce_tag_present? */
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if(get_bits1(gb)) {
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che_drc->pce_instance_tag = get_bits(gb, 4);
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skip_bits(gb, 4); // tag_reserved_bits
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n++;
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}
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/* excluded_chns_present? */
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if(get_bits1(gb)) {
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n += decode_drc_channel_exclusions(che_drc, gb);
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}
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/* drc_bands_present? */
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if (get_bits1(gb)) {
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che_drc->band_incr = get_bits(gb, 4);
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che_drc->interpolation_scheme = get_bits(gb, 4);
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n++;
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drc_num_bands += che_drc->band_incr;
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for (i = 0; i < drc_num_bands; i++) {
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che_drc->band_top[i] = get_bits(gb, 8);
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n++;
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}
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}
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/* 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;
|
||||
}
|
||||
|
||||
/**
|
||||
* Decode extension data (incomplete); reference: table 4.51.
|
||||
*
|
||||
* @param cnt length of TYPE_FIL syntactic element in bytes
|
||||
*
|
||||
* @return Returns number of bytes consumed
|
||||
*/
|
||||
static int decode_extension_payload(AACContext * ac, GetBitContext * gb, int cnt) {
|
||||
int crc_flag = 0;
|
||||
int res = cnt;
|
||||
switch (get_bits(gb, 4)) { // extension type
|
||||
@ -364,6 +735,21 @@ static int decode_sbr_extension(AACContext * ac, GetBitContext * gb, int crc, in
|
||||
return res;
|
||||
}
|
||||
|
||||
/**
|
||||
* Conduct IMDCT and windowing.
|
||||
*/
|
||||
static void imdct_and_windowing(AACContext * ac, SingleChannelElement * sce) {
|
||||
IndividualChannelStream * ics = &sce->ics;
|
||||
float * in = sce->coeffs;
|
||||
float * out = sce->ret;
|
||||
float * saved = sce->saved;
|
||||
const float * lwindow = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_aac_sine_long_1024;
|
||||
const float * swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_aac_sine_short_128;
|
||||
const float * lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_aac_sine_long_1024;
|
||||
const float * swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_aac_sine_short_128;
|
||||
float * buf = ac->buf_mdct;
|
||||
int i;
|
||||
|
||||
/**
|
||||
* Apply dependent channel coupling (applied before IMDCT).
|
||||
*
|
||||
@ -409,6 +795,26 @@ static void apply_independent_coupling(AACContext * ac, SingleChannelElement * s
|
||||
sce->ret[i] += gain * (cc->ch[0].ret[i] - ac->add_bias);
|
||||
}
|
||||
|
||||
if (!ac->is_saved) {
|
||||
ac->is_saved = 1;
|
||||
*data_size = 0;
|
||||
return 0;
|
||||
}
|
||||
|
||||
data_size_tmp = 1024 * avccontext->channels * sizeof(int16_t);
|
||||
if(*data_size < data_size_tmp) {
|
||||
av_log(avccontext, AV_LOG_ERROR,
|
||||
"Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n",
|
||||
*data_size, data_size_tmp);
|
||||
return -1;
|
||||
}
|
||||
*data_size = data_size_tmp;
|
||||
|
||||
ac->dsp.float_to_int16_interleave(data, (const float **)ac->output_data, 1024, avccontext->channels);
|
||||
|
||||
return buf_size;
|
||||
}
|
||||
|
||||
static av_cold int aac_decode_close(AVCodecContext * avccontext) {
|
||||
AACContext * ac = avccontext->priv_data;
|
||||
int i, j;
|
||||
|
||||
@ -43,6 +43,7 @@
|
||||
size);
|
||||
|
||||
#define MAX_CHANNELS 64
|
||||
#define MAX_ELEM_ID 16
|
||||
|
||||
#define IVQUANT_SIZE 1024
|
||||
|
||||
@ -76,6 +77,17 @@ enum AudioObjectType {
|
||||
AOT_SSC, ///< N SinuSoidal Coding
|
||||
};
|
||||
|
||||
enum RawDataBlockType {
|
||||
TYPE_SCE,
|
||||
TYPE_CPE,
|
||||
TYPE_CCE,
|
||||
TYPE_LFE,
|
||||
TYPE_DSE,
|
||||
TYPE_PCE,
|
||||
TYPE_FIL,
|
||||
TYPE_END,
|
||||
};
|
||||
|
||||
enum ExtensionPayloadID {
|
||||
EXT_FILL,
|
||||
EXT_FILL_DATA,
|
||||
@ -111,6 +123,35 @@ enum ChannelPosition {
|
||||
AAC_CHANNEL_CC = 5,
|
||||
};
|
||||
|
||||
/**
|
||||
* The point during decoding at which channel coupling is applied.
|
||||
*/
|
||||
enum CouplingPoint {
|
||||
BEFORE_TNS,
|
||||
BETWEEN_TNS_AND_IMDCT,
|
||||
AFTER_IMDCT = 3,
|
||||
};
|
||||
|
||||
/**
|
||||
* Individual Channel Stream
|
||||
*/
|
||||
|
||||
/**
|
||||
* Dynamic Range Control - decoded from the bitstream but not processed further.
|
||||
*/
|
||||
typedef struct {
|
||||
int pce_instance_tag; ///< Indicates with which program the DRC info is associated.
|
||||
int dyn_rng_sgn[17]; ///< DRC sign information; 0 - positive, 1 - negative
|
||||
int dyn_rng_ctl[17]; ///< DRC magnitude information
|
||||
int exclude_mask[MAX_CHANNELS]; ///< Channels to be excluded from DRC processing.
|
||||
int band_incr; ///< Number of DRC bands greater than 1 having DRC info.
|
||||
int interpolation_scheme; ///< Indicates the interpolation scheme used in the SBR QMF domain.
|
||||
int band_top[17]; ///< Indicates the top of the i-th DRC band in units of 4 spectral lines.
|
||||
int prog_ref_level; /**< A reference level for the long-term program audio level for all
|
||||
* channels combined.
|
||||
*/
|
||||
} DynamicRangeControl;
|
||||
|
||||
typedef struct {
|
||||
int num_pulse;
|
||||
int start;
|
||||
@ -134,9 +175,15 @@ typedef struct {
|
||||
int is_saved; ///< Set if elements have stored overlap from previous frame.
|
||||
DynamicRangeControl che_drc;
|
||||
|
||||
/**
|
||||
* @defgroup elements
|
||||
* @{
|
||||
*/
|
||||
enum ChannelPosition che_pos[4][MAX_ELEM_ID]; /**< channel element channel mapping with the
|
||||
* first index as the first 4 raw data block types
|
||||
*/
|
||||
ChannelElement * che[4][MAX_ELEM_ID];
|
||||
/** @} */
|
||||
|
||||
/**
|
||||
* @defgroup tables Computed / set up during initialization.
|
||||
@ -145,6 +192,7 @@ typedef struct {
|
||||
MDCTContext mdct;
|
||||
MDCTContext mdct_small;
|
||||
DSPContext dsp;
|
||||
int random_state;
|
||||
/** @} */
|
||||
|
||||
/**
|
||||
|
||||
@ -32,6 +32,14 @@
|
||||
|
||||
#include <stdint.h>
|
||||
|
||||
const uint8_t ff_aac_num_swb_1024[] = {
|
||||
41, 41, 47, 49, 49, 51, 47, 47, 43, 43, 43, 40
|
||||
};
|
||||
|
||||
const uint8_t ff_aac_num_swb_128[] = {
|
||||
12, 12, 12, 14, 14, 14, 15, 15, 15, 15, 15, 15
|
||||
};
|
||||
|
||||
const uint32_t ff_aac_scalefactor_code[121] = {
|
||||
0x3ffe8, 0x3ffe6, 0x3ffe7, 0x3ffe5, 0x7fff5, 0x7fff1, 0x7ffed, 0x7fff6,
|
||||
0x7ffee, 0x7ffef, 0x7fff0, 0x7fffc, 0x7fffd, 0x7ffff, 0x7fffe, 0x7fff7,
|
||||
@ -796,6 +804,13 @@ const float ff_aac_ivquant_tab[IVQUANT_SIZE] = {
|
||||
4064.0312908, 4074.6805676, 4085.3368071, 4096.0000000,
|
||||
};
|
||||
|
||||
/**
|
||||
* Table of pow(2, (i - 200)/4.) used for different purposes depending on the
|
||||
* range of indices to the table:
|
||||
* [ 0, 255] scale factor decoding when using C dsp.float_to_int16
|
||||
* [60, 315] scale factor decoding when using SIMD dsp.float_to_int16
|
||||
* [45, 300] intensity stereo position decoding mapped in reverse order i.e. 0->300, 1->299, ..., 254->46, 255->45
|
||||
*/
|
||||
const float ff_aac_pow2sf_tab[316] = {
|
||||
8.88178420e-16, 1.05622810e-15, 1.25607397e-15, 1.49373210e-15,
|
||||
1.77635684e-15, 2.11245619e-15, 2.51214793e-15, 2.98746420e-15,
|
||||
|
||||
@ -35,6 +35,18 @@
|
||||
|
||||
#include <stdint.h>
|
||||
|
||||
/* NOTE:
|
||||
* Tables in this file are used by the AAC decoder and will be used by the AAC
|
||||
* encoder.
|
||||
*/
|
||||
|
||||
/* @name number of scalefactor window bands for long and short transform windows respectively
|
||||
* @{
|
||||
*/
|
||||
extern const uint8_t ff_aac_num_swb_1024[];
|
||||
extern const uint8_t ff_aac_num_swb_128 [];
|
||||
// @}
|
||||
|
||||
extern const uint32_t ff_aac_scalefactor_code[121];
|
||||
extern const uint8_t ff_aac_scalefactor_bits[121];
|
||||
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user