Commit 9cc04edf authored by Robert Swain's avatar Robert Swain
Browse files

More OKed hunks of the AAC decoder from SoC

Originally committed as revision 14694 to svn://svn.ffmpeg.org/ffmpeg/trunk
parent 3f66d168
......@@ -99,6 +99,40 @@ static VLC vlc_scalefactors;
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);
......@@ -130,6 +164,131 @@ static VLC vlc_spectral[11];
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;
......@@ -140,6 +299,7 @@ static av_cold int aac_decode_init(AVCodecContext * avccontext) {
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;
......@@ -157,6 +317,8 @@ static av_cold int aac_decode_init(AVCodecContext * avccontext) {
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.
......@@ -188,6 +350,10 @@ static av_cold int aac_decode_init(AVCodecContext * avccontext) {
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)
......@@ -197,6 +363,27 @@ static av_cold int aac_decode_init(AVCodecContext * avccontext) {
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
*
......@@ -210,6 +397,15 @@ static inline float ivquant(int a) {
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;
......@@ -232,7 +428,13 @@ static inline float ivquant(int a) {
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
......@@ -313,6 +515,16 @@ static void decode_pulses(Pulse * pulse, GetBitContext * gb) {
}
}
/**
* 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.
*
......@@ -330,10 +542,109 @@ static void add_pulses(int icoef[1024], const Pulse * pulse, const IndividualCha
}
/**
* 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) {
......@@ -343,6 +654,66 @@ static int decode_sbr_extension(AACContext * ac, GetBitContext * gb, int crc, in
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
......@@ -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 [];
// @}