static VLC vlc_spectral[11];
+/**
+ * Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit.
+ *
+ * @param cpe_map Stereo (Channel Pair Element) map, NULL if stereo bit is not present.
+ * @param sce_map mono (Single Channel Element) map
+ * @param type speaker type/position for these channels
+ */
+static void decode_channel_map(enum ChannelPosition *cpe_map,
+ enum ChannelPosition *sce_map, enum ChannelPosition type, GetBitContext * gb, int n) {
+ while(n--) {
+ enum ChannelPosition *map = cpe_map && get_bits1(gb) ? cpe_map : sce_map; // stereo or mono map
+ map[get_bits(gb, 4)] = type;
+ }
+}
+
+/**
+ * Decode program configuration element; reference: table 4.2.
+ *
+ * @param new_che_pos New channel position configuration - we only do something if it differs from the current one.
+ *
+ * @return Returns error status. 0 - OK, !0 - error
+ */
+static int decode_pce(AACContext * ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
+ GetBitContext * gb) {
+ int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc;
+
+ skip_bits(gb, 2); // object_type
+
+ ac->m4ac.sampling_index = get_bits(gb, 4);
+ if(ac->m4ac.sampling_index > 11) {
+ av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
+ return -1;
+ }
+ ac->m4ac.sample_rate = ff_mpeg4audio_sample_rates[ac->m4ac.sampling_index];
num_front = get_bits(gb, 4);
num_side = get_bits(gb, 4);
num_back = get_bits(gb, 4);
return 0;
}
+/**
+ * Set up channel positions based on a default channel configuration
+ * as specified in table 1.17.
+ *
+ * @param new_che_pos New channel position configuration - we only do something if it differs from the current one.
+ *
+ * @return Returns error status. 0 - OK, !0 - error
+ */
+static int set_default_channel_config(AACContext *ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
+ int channel_config)
+{
+ if(channel_config < 1 || channel_config > 7) {
+ av_log(ac->avccontext, AV_LOG_ERROR, "invalid default channel configuration (%d)\n",
+ channel_config);
+ return -1;
+ }
+
+ /* default channel configurations:
+ *
+ * 1ch : front center (mono)
+ * 2ch : L + R (stereo)
+ * 3ch : front center + L + R
+ * 4ch : front center + L + R + back center
+ * 5ch : front center + L + R + back stereo
+ * 6ch : front center + L + R + back stereo + LFE
+ * 7ch : front center + L + R + outer front left + outer front right + back stereo + LFE
+ */
+
+ if(channel_config != 2)
+ new_che_pos[TYPE_SCE][0] = AAC_CHANNEL_FRONT; // front center (or mono)
+ if(channel_config > 1)
+ new_che_pos[TYPE_CPE][0] = AAC_CHANNEL_FRONT; // L + R (or stereo)
+ if(channel_config == 4)
+ new_che_pos[TYPE_SCE][1] = AAC_CHANNEL_BACK; // back center
+ if(channel_config > 4)
+ new_che_pos[TYPE_CPE][(channel_config == 7) + 1]
+ = AAC_CHANNEL_BACK; // back stereo
+ if(channel_config > 5)
+ new_che_pos[TYPE_LFE][0] = AAC_CHANNEL_LFE; // LFE
+ if(channel_config == 7)
+ new_che_pos[TYPE_CPE][1] = AAC_CHANNEL_FRONT; // outer front left + outer front right
+
+ return 0;
+}
+
+ return -1;
+ }
+
+ if (get_bits1(gb)) // dependsOnCoreCoder
+ skip_bits(gb, 14); // coreCoderDelay
+ extension_flag = get_bits1(gb);
+
+ if(ac->m4ac.object_type == AOT_AAC_SCALABLE ||
+ ac->m4ac.object_type == AOT_ER_AAC_SCALABLE)
+ skip_bits(gb, 3); // layerNr
+
+ memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
+ if (channel_config == 0) {
+ skip_bits(gb, 4); // element_instance_tag
+ if((ret = decode_pce(ac, new_che_pos, gb)))
+ return ret;
+ } else {
+ if((ret = set_default_channel_config(ac, new_che_pos, channel_config)))
+ return ret;
+ }
+ if((ret = output_configure(ac, ac->che_pos, new_che_pos)))
+ return ret;
+
+ if (extension_flag) {
+ switch (ac->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:
+ skip_bits(gb, 3); /* aacSectionDataResilienceFlag
+ * aacScalefactorDataResilienceFlag
+ * aacSpectralDataResilienceFlag
+ */
+ break;
+ }
+ skip_bits1(gb); // extensionFlag3 (TBD in version 3)
+ }
+ return 0;
+}
+
+/**
+ * Decode audio specific configuration; reference: table 1.13.
+ *
+ * @param data pointer to AVCodecContext extradata
+ * @param data_size size of AVCCodecContext extradata
+ *
+ * @return Returns error status. 0 - OK, !0 - error
+ */
+static int decode_audio_specific_config(AACContext * ac, void *data, int data_size) {
+ GetBitContext gb;
+ int i;
+
+ init_get_bits(&gb, data, data_size * 8);
+
+ if((i = ff_mpeg4audio_get_config(&ac->m4ac, data, data_size)) < 0)
+ return -1;
+ if(ac->m4ac.sampling_index > 11) {
+ av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
+ return -1;
+ }
+
+ skip_bits_long(&gb, i);
+
+ switch (ac->m4ac.object_type) {
+ case AOT_AAC_LC:
+ if (decode_ga_specific_config(ac, &gb, ac->m4ac.chan_config))
+ return -1;
+ break;
+ default:
+ av_log(ac->avccontext, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n",
+ ac->m4ac.sbr == 1? "SBR+" : "", ac->m4ac.object_type);
+ return -1;
+ }
+ return 0;
+}
+
static av_cold int aac_decode_init(AVCodecContext * avccontext) {
AACContext * ac = avccontext->priv_data;
int i;
decode_audio_specific_config(ac, avccontext->extradata, avccontext->extradata_size))
return -1;
+ avccontext->sample_fmt = SAMPLE_FMT_S16;
avccontext->sample_rate = ac->m4ac.sample_rate;
avccontext->frame_size = 1024;
dsputil_init(&ac->dsp, avccontext);
+ ac->random_state = 0x1f2e3d4c;
+
// -1024 - Compensate wrong IMDCT method.
// 32768 - Required to scale values to the correct range for the bias method
// for float to int16 conversion.
return 0;
}
+/**
+ * Skip data_stream_element; reference: table 4.10.
+ */
+static void skip_data_stream_element(GetBitContext * gb) {
int byte_align = get_bits1(gb);
int count = get_bits(gb, 8);
if (count == 255)
skip_bits_long(gb, 8 * count);
}
+/**
+ * Decode Individual Channel Stream info; reference: table 4.6.
+ *
+ * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information.
+ */
+static int decode_ics_info(AACContext * ac, IndividualChannelStream * ics, GetBitContext * gb, int common_window) {
+ if (get_bits1(gb)) {
+ av_log(ac->avccontext, AV_LOG_ERROR, "Reserved bit set.\n");
+ memset(ics, 0, sizeof(IndividualChannelStream));
+ return -1;
+ }
+ ics->window_sequence[1] = ics->window_sequence[0];
+ ics->window_sequence[0] = get_bits(gb, 2);
+ 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;
+
+ return 0;
+}
+
/**
* inverse quantization
*
return cbrtf(fabsf(a)) * a;
}
+/**
+ * Decode band types (section_data payload); reference: table 4.46.
+ *
+ * @param band_type array of the used band type
+ * @param band_type_run_end array of the last scalefactor band of a band type run
+ *
+ * @return Returns error status. 0 - OK, !0 - error
+ */
+static int decode_band_types(AACContext * ac, enum BandType band_type[120],
int band_type_run_end[120], GetBitContext * gb, IndividualChannelStream * ics) {
int g, idx = 0;
const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5;
sect_len, ics->max_sfb);
return -1;
}
+ }
+ }
+ return 0;
+}
+/**
+ * Decode scalefactors; reference: table 4.47.
*
* @param mix_gain channel gain (Not used by AAC bitstream.)
* @param global_gain first scalefactor value as scalefactors are differentially coded
}
}
+/**
+ * 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) {
+
/**
* Add pulses with particular amplitudes to the quantized spectral data; reference: 4.6.3.3.
*
}
/**
- * Parse Spectral Band Replication extension data; reference: table 4.55.
+ * Decode an individual_channel_stream payload; reference: table 4.44.
+ *
+ * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information.
+ * @param scale_flag scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.)
+ *
+ * @return Returns error status. 0 - OK, !0 - error
+ */
+static int decode_ics(AACContext * ac, SingleChannelElement * sce, GetBitContext * gb, int common_window, int scale_flag) {
+ int icoeffs[1024];
+ Pulse pulse;
+ TemporalNoiseShaping * tns = &sce->tns;
+ IndividualChannelStream * ics = &sce->ics;
+ float * out = sce->coeffs;
+ int global_gain, pulse_present = 0;
+
+ /* These two assignments are to silence some GCC warnings about the
+ * variables being used uninitialised when in fact they always are.
+ */
+ pulse.num_pulse = 0;
+ pulse.start = 0;
+
+ global_gain = get_bits(gb, 8);
+
+ if (!common_window && !scale_flag) {
+ if (decode_ics_info(ac, ics, gb, 0) < 0)
+ return -1;
+ }
+
+ if (decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics) < 0)
+ return -1;
+ if (decode_scalefactors(ac, sce->sf, gb, global_gain, ics, sce->band_type, sce->band_type_run_end) < 0)
+ return -1;
+
+ pulse_present = 0;
+ if (!scale_flag) {
+ if ((pulse_present = get_bits1(gb))) {
+ if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
+ av_log(ac->avccontext, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n");
+ return -1;
+ }
+ decode_pulses(&pulse, gb);
+ }
+ if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics))
+ return -1;
+ if (get_bits1(gb)) {
+ av_log_missing_feature(ac->avccontext, "SSR", 1);
+ return -1;
+ }
+ }
+
+ if (decode_spectrum(ac, icoeffs, gb, ics, sce->band_type) < 0)
+ return -1;
+ if (pulse_present)
+ add_pulses(icoeffs, &pulse, ics);
+ dequant(ac, out, icoeffs, sce->sf, ics, sce->band_type);
+ return 0;
+}
+
+/**
+ * Decode a channel_pair_element; reference: table 4.4.
+ *
+ * @param elem_id Identifies the instance of a syntax element.
+ *
+ * @return Returns error status. 0 - OK, !0 - error
+ */
+static int decode_cpe(AACContext * ac, GetBitContext * gb, int elem_id) {
+ int i, ret, common_window, ms_present = 0;
+ ChannelElement * cpe;
+
+ cpe = ac->che[TYPE_CPE][elem_id];
+ common_window = get_bits1(gb);
+ if (common_window) {
+ if (decode_ics_info(ac, &cpe->ch[0].ics, gb, 1))
+ return -1;
+ 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;
+ ms_present = get_bits(gb, 2);
+ if(ms_present == 3) {
+ av_log(ac->avccontext, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
+ return -1;
+ } else if(ms_present)
+ decode_mid_side_stereo(cpe, gb, ms_present);
+ }
+ if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0)))
+ return ret;
+ if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0)))
+ return ret;
+
+ if (common_window && ms_present)
+ apply_mid_side_stereo(cpe);
+
+ if (cpe->ch[1].ics.intensity_present)
+ apply_intensity_stereo(cpe, ms_present);
+ return 0;
+}
+
+/**
+ * Decode Spectral Band Replication extension data; reference: table 4.55.
*
* @param crc flag indicating the presence of CRC checksum
* @param cnt length of TYPE_FIL syntactic element in bytes
+ *
* @return Returns number of bytes consumed from the TYPE_FIL element.
*/
static int decode_sbr_extension(AACContext * ac, GetBitContext * gb, int crc, int cnt) {
return cnt;
}
+/**
+ * Decode dynamic range information; reference: table 4.52.
+ *
+ * @param cnt length of TYPE_FIL syntactic element in bytes
+ *
+ * @return Returns number of bytes consumed.
+ */
+static int decode_dynamic_range(DynamicRangeControl *che_drc, GetBitContext * gb, int cnt) {
+ 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;
+}
+
+/**
+ * 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
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).
*
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;
projects / platform / upstream / libav.git / commitdiff