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ffmpeg.wasm-core
Commit 76efedea authored by Luca Barbato's avatar Luca Barbato
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qdm2: formatting cosmetics 

Apply the usual style plus drop few unnecessary return at the end
of void functions.
parent f054e309
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Inline Side-by-side
Showing with 204 additions and 187 deletions
libavcodec/qdm2.c
View file @ 76efedea
......@@ -216,6 +216,10 @@ static const uint16_t qdm2_vlc_offs[] = {
0,260,566,598,894,1166,1230,1294,1678,1950,2214,2278,2310,2570,2834,3124,3448,3838,
};
static const int switchtable[23] = {
0, 5, 1, 5, 5, 5, 5, 5, 2, 5, 5, 5, 5, 5, 5, 5, 3, 5, 5, 5, 5, 5, 4
};
static av_cold void qdm2_init_vlc(void)
{
static VLC_TYPE qdm2_table[3838][2];
......@@ -359,7 +363,7 @@ static av_cold void qdm2_init_vlc(void)
INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
}
static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth)
static int qdm2_get_vlc(GetBitContext *gb, VLC *vlc, int flag, int depth)
{
int value;
......@@ -367,29 +371,27 @@ static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth)
/* stage-2, 3 bits exponent escape sequence */
if (value-- == 0)
value = get_bits (gb, get_bits (gb, 3) + 1);
value = get_bits(gb, get_bits(gb, 3) + 1);
/* stage-3, optional */
if (flag) {
int tmp = vlc_stage3_values[value];
if ((value & ~3) > 0)
tmp += get_bits (gb, (value >> 2));
tmp += get_bits(gb, (value >> 2));
value = tmp;
}
return value;
}
static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth)
static int qdm2_get_se_vlc(VLC *vlc, GetBitContext *gb, int depth)
{
int value = qdm2_get_vlc (gb, vlc, 0, depth);
int value = qdm2_get_vlc(gb, vlc, 0, depth);
return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
}
/**
* QDM2 checksum
*
......@@ -399,49 +401,50 @@ static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth)
*
* @return 0 if checksum is OK
*/
static uint16_t qdm2_packet_checksum (const uint8_t *data, int length, int value) {
static uint16_t qdm2_packet_checksum(const uint8_t *data, int length, int value)
{
int i;
for (i=0; i < length; i++)
for (i = 0; i < length; i++)
value -= data[i];
return (uint16_t)(value & 0xffff);
}
/**
* Fill a QDM2SubPacket structure with packet type, size, and data pointer.
*
* @param gb bitreader context
* @param sub_packet packet under analysis
*/
static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet)
static void qdm2_decode_sub_packet_header(GetBitContext *gb,
QDM2SubPacket *sub_packet)
{
sub_packet->type = get_bits (gb, 8);
sub_packet->type = get_bits(gb, 8);
if (sub_packet->type == 0) {
sub_packet->size = 0;
sub_packet->data = NULL;
} else {
sub_packet->size = get_bits (gb, 8);
sub_packet->size = get_bits(gb, 8);
if (sub_packet->type & 0x80) {
sub_packet->size <<= 8;
sub_packet->size |= get_bits (gb, 8);
sub_packet->type &= 0x7f;
}
if (sub_packet->type & 0x80) {
sub_packet->size <<= 8;
sub_packet->size |= get_bits(gb, 8);
sub_packet->type &= 0x7f;
}
if (sub_packet->type == 0x7f)
sub_packet->type |= (get_bits (gb, 8) << 8);
if (sub_packet->type == 0x7f)
sub_packet->type |= (get_bits(gb, 8) << 8);
sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data
// FIXME: this depends on bitreader-internal data
sub_packet->data = &gb->buffer[get_bits_count(gb) / 8];
}
av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n",
sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
av_log(NULL, AV_LOG_DEBUG, "Subpacket: type=%d size=%d start_offs=%x\n",
sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
}
/**
* Return node pointer to first packet of requested type in list.
*
......@@ -449,7 +452,8 @@ static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub
* @param type type of searched subpacket
* @return node pointer for subpacket if found, else NULL
*/
static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type)
static QDM2SubPNode *qdm2_search_subpacket_type_in_list(QDM2SubPNode *list,
int type)
{
while (list != NULL && list->packet != NULL) {
if (list->packet->type == type)
......@@ -459,14 +463,13 @@ static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int
return NULL;
}
/**
* Replace 8 elements with their average value.
* Called by qdm2_decode_superblock before starting subblock decoding.
*
* @param q context
*/
static void average_quantized_coeffs (QDM2Context *q)
static void average_quantized_coeffs(QDM2Context *q)
{
int i, j, n, ch, sum;
......@@ -483,12 +486,11 @@ static void average_quantized_coeffs (QDM2Context *q)
if (sum > 0)
sum--;
for (j=0; j < 8; j++)
for (j = 0; j < 8; j++)
q->quantized_coeffs[ch][i][j] = sum;
}
}
/**
* Build subband samples with noise weighted by q->tone_level.
* Called by synthfilt_build_sb_samples.
......@@ -496,7 +498,7 @@ static void average_quantized_coeffs (QDM2Context *q)
* @param q context
* @param sb subband index
*/
static void build_sb_samples_from_noise (QDM2Context *q, int sb)
static void build_sb_samples_from_noise(QDM2Context *q, int sb)
{
int ch, j;
......@@ -505,14 +507,16 @@ static void build_sb_samples_from_noise (QDM2Context *q, int sb)
if (!q->nb_channels)
return;
for (ch = 0; ch < q->nb_channels; ch++)
for (ch = 0; ch < q->nb_channels; ch++) {
for (j = 0; j < 64; j++) {
q->sb_samples[ch][j * 2][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j];
q->sb_samples[ch][j * 2 + 1][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j];
q->sb_samples[ch][j * 2][sb] =
SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j];
q->sb_samples[ch][j * 2 + 1][sb] =
SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j];
}
}
}
/**
* Called while processing data from subpackets 11 and 12.
* Used after making changes to coding_method array.
......@@ -521,44 +525,62 @@ static void build_sb_samples_from_noise (QDM2Context *q, int sb)
* @param channels number of channels
* @param coding_method q->coding_method[0][0][0]
*/
static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method)
static void fix_coding_method_array(int sb, int channels,
sb_int8_array coding_method)
{
int j,k;
int j, k;
int ch;
int run, case_val;
static const int switchtable[23] = {0,5,1,5,5,5,5,5,2,5,5,5,5,5,5,5,3,5,5,5,5,5,4};
for (ch = 0; ch < channels; ch++) {
for (j = 0; j < 64; ) {
if((coding_method[ch][sb][j] - 8) > 22) {
run = 1;
if ((coding_method[ch][sb][j] - 8) > 22) {
run = 1;
case_val = 8;
} else {
switch (switchtable[coding_method[ch][sb][j]-8]) {
case 0: run = 10; case_val = 10; break;
case 1: run = 1; case_val = 16; break;
case 2: run = 5; case_val = 24; break;
case 3: run = 3; case_val = 30; break;
case 4: run = 1; case_val = 30; break;
case 5: run = 1; case_val = 8; break;
default: run = 1; case_val = 8; break;
switch (switchtable[coding_method[ch][sb][j] - 8]) {
case 0: run = 10;
case_val = 10;
break;
case 1: run = 1;
case_val = 16;
break;
case 2: run = 5;
case_val = 24;
break;
case 3: run = 3;
case_val = 30;
break;
case 4: run = 1;
case_val = 30;
break;
case 5: run = 1;
case_val = 8;
break;
default: run = 1;
case_val = 8;
break;
}
}
for (k = 0; k < run; k++)
if (j + k < 128)
if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j])
for (k = 0; k < run; k++) {
if (j + k < 128) {
if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j]) {
if (k > 0) {
SAMPLES_NEEDED
SAMPLES_NEEDED
//not debugged, almost never used
memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t));
memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t));
memset(&coding_method[ch][sb][j + k], case_val,
k *sizeof(int8_t));
memset(&coding_method[ch][sb][j + k], case_val,
3 * sizeof(int8_t));
}
}
}
}
j += run;
}
}
}
/**
* Related to synthesis filter
* Called by process_subpacket_10
......@@ -566,7 +588,7 @@ static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_
* @param q context
* @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10
*/
static void fill_tone_level_array (QDM2Context *q, int flag)
static void fill_tone_level_array(QDM2Context *q, int flag)
{
int i, sb, ch, sb_used;
int tmp, tab;
......@@ -638,16 +660,14 @@ static void fill_tone_level_array (QDM2Context *q, int flag)
}
}
}
return;
}
/**
* Related to synthesis filter
* Called by process_subpacket_11
* c is built with data from subpacket 11
* Most of this function is used only if superblock_type_2_3 == 0, never seen it in samples
* Most of this function is used only if superblock_type_2_3 == 0,
* never seen it in samples.
*
* @param tone_level_idx
* @param tone_level_idx_temp
......@@ -657,9 +677,12 @@ static void fill_tone_level_array (QDM2Context *q, int flag)
* @param superblocktype_2_3 flag based on superblock packet type
* @param cm_table_select q->cm_table_select
*/
static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp,
sb_int8_array coding_method, int nb_channels,
int c, int superblocktype_2_3, int cm_table_select)
static void fill_coding_method_array(sb_int8_array tone_level_idx,
sb_int8_array tone_level_idx_temp,
sb_int8_array coding_method,
int nb_channels,
int c, int superblocktype_2_3,
int cm_table_select)
{
int ch, sb, j;
int tmp, acc, esp_40, comp;
......@@ -765,15 +788,14 @@ static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_arra
for (j = 0; j < 64; j++)
coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
}
return;
}
/**
*
* Called by process_subpacket_11 to process more data from subpacket 11 with sb 0-8
* Called by process_subpacket_12 to process data from subpacket 12 with sb 8-sb_used
* Called by process_subpacket_11 to process more data from subpacket 11
* with sb 0-8.
* Called by process_subpacket_12 to process data from subpacket 12 with
* sb 8-sb_used.
*
* @param q context
* @param gb bitreader context
......@@ -781,7 +803,8 @@ static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_arra
* @param sb_min lower subband processed (sb_min included)
* @param sb_max higher subband processed (sb_max excluded)
*/
static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max)
static void synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb,
int length, int sb_min, int sb_max)
{
int sb, j, k, n, ch, run, channels;
int joined_stereo, zero_encoding, chs;
......@@ -961,16 +984,18 @@ static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int l
} // subband loop
}
/**
* Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch][0]).
* This is similar to process_subpacket_9, but for a single channel and for element [0]
* Init the first element of a channel in quantized_coeffs with data
* from packet 10 (quantized_coeffs[ch][0]).
* This is similar to process_subpacket_9, but for a single channel
* and for element [0]
* same VLC tables as process_subpacket_9 are used.
*
* @param quantized_coeffs pointer to quantized_coeffs[ch][0]
* @param gb bitreader context
*/
static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb)
static void init_quantized_coeffs_elem0(int8_t *quantized_coeffs,
GetBitContext *gb)
{
int i, k, run, level, diff;
......@@ -997,11 +1022,11 @@ static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext
}
}
/**
* Related to synthesis filter, process data from packet 10
* Init part of quantized_coeffs via function init_quantized_coeffs_elem0
* Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with data from packet 10
* Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with
* data from packet 10
*
* @param q context
* @param gb bitreader context
......@@ -1069,29 +1094,29 @@ static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb)
* @param q context
* @param node pointer to node with packet
*/
static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node)
static void process_subpacket_9(QDM2Context *q, QDM2SubPNode *node)
{
GetBitContext gb;
int i, j, k, n, ch, run, level, diff;
init_get_bits(&gb, node->packet->data, node->packet->size*8);
init_get_bits(&gb, node->packet->data, node->packet->size * 8);
n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function
n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
for (i = 1; i < n; i++)
for (ch=0; ch < q->nb_channels; ch++) {
for (ch = 0; ch < q->nb_channels; ch++) {
level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
q->quantized_coeffs[ch][i][0] = level;
for (j = 0; j < (8 - 1); ) {
run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
for (k = 1; k <= run; k++)
q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run));
q->quantized_coeffs[ch][i][j + k] = (level + ((k * diff) / run));
level += diff;
j += run;
j += run;
}
}
......@@ -1100,14 +1125,13 @@ static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node)
q->quantized_coeffs[ch][0][i] = 0;
}
/**
* Process subpacket 10 if not null, else
*
* @param q context
* @param node pointer to node with packet
*/
static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node)
static void process_subpacket_10(QDM2Context *q, QDM2SubPNode *node)
{
GetBitContext gb;
......@@ -1120,14 +1144,13 @@ static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node)
}
}
/**
* Process subpacket 11
*
* @param q context
* @param node pointer to node with packet
*/
static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node)
static void process_subpacket_11(QDM2Context *q, QDM2SubPNode *node)
{
GetBitContext gb;
int length = 0;
......@@ -1138,24 +1161,25 @@ static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node)
}
if (length >= 32) {
int c = get_bits (&gb, 13);
int c = get_bits(&gb, 13);
if (c > 3)
fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method,
q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select);
fill_coding_method_array(q->tone_level_idx,
q->tone_level_idx_temp, q->coding_method,
q->nb_channels, 8 * c,
q->superblocktype_2_3, q->cm_table_select);
}
synthfilt_build_sb_samples(q, &gb, length, 0, 8);
}
/**
* Process subpacket 12
*
* @param q context
* @param node pointer to node with packet
*/
static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node)
static void process_subpacket_12(QDM2Context *q, QDM2SubPNode *node)
{
GetBitContext gb;
int length = 0;
......@@ -1174,7 +1198,7 @@ static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node)
* @param q context
* @param list list with synthesis filter packets (list D)
*/
static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list)
static void process_synthesis_subpackets(QDM2Context *q, QDM2SubPNode *list)
{
QDM2SubPNode *nodes[4];
......@@ -1201,13 +1225,12 @@ static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list)
process_subpacket_12(q, NULL);
}
/*
* Decode superblock, fill packet lists.
*
* @param q context
*/
static void qdm2_decode_super_block (QDM2Context *q)
static void qdm2_decode_super_block(QDM2Context *q)
{
GetBitContext gb;
QDM2SubPacket header, *packet;
......@@ -1219,33 +1242,33 @@ static void qdm2_decode_super_block (QDM2Context *q)
memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
q->sub_packets_B = 0;
sub_packets_D = 0;
sub_packets_D = 0;
average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
init_get_bits(&gb, q->compressed_data, q->compressed_size*8);
init_get_bits(&gb, q->compressed_data, q->compressed_size * 8);
qdm2_decode_sub_packet_header(&gb, &header);
if (header.type < 2 || header.type >= 8) {
q->has_errors = 1;
av_log(NULL,AV_LOG_ERROR,"bad superblock type\n");
av_log(NULL, AV_LOG_ERROR, "bad superblock type\n");
return;
}
q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
init_get_bits(&gb, header.data, header.size*8);
init_get_bits(&gb, header.data, header.size * 8);
if (header.type == 2 || header.type == 4 || header.type == 5) {
int csum = 257 * get_bits(&gb, 8);
csum += 2 * get_bits(&gb, 8);
int csum = 257 * get_bits(&gb, 8);
csum += 2 * get_bits(&gb, 8);
csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
if (csum != 0) {
q->has_errors = 1;
av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n");
av_log(NULL, AV_LOG_ERROR, "bad packet checksum\n");
return;
}
}
......@@ -1271,8 +1294,8 @@ static void qdm2_decode_super_block (QDM2Context *q)
q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
/* seek to next block */
init_get_bits(&gb, header.data, header.size*8);
skip_bits(&gb, next_index*8);
init_get_bits(&gb, header.data, header.size * 8);
skip_bits(&gb, next_index * 8);
if (next_index >= header.size)
break;
......@@ -1281,7 +1304,7 @@ static void qdm2_decode_super_block (QDM2Context *q)
/* decode subpacket */
packet = &q->sub_packets[i];
qdm2_decode_sub_packet_header(&gb, packet);
next_index = packet->size + get_bits_count(&gb) / 8;
next_index = packet->size + get_bits_count(&gb) / 8;
sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
if (packet->type == 0)
......@@ -1314,13 +1337,13 @@ static void qdm2_decode_super_block (QDM2Context *q)
} else if (packet->type == 15) {
SAMPLES_NEEDED_2("packet type 15")
return;
} else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) {
} else if (packet->type >= 16 && packet->type < 48 &&
!fft_subpackets[packet->type - 16]) {
/* packets for FFT */
QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
}
} // Packet bytes loop
/* **************************************************************** */
if (q->sub_packet_list_D[0].packet != NULL) {
process_synthesis_subpackets(q, q->sub_packet_list_D);
q->do_synth_filter = 1;
......@@ -1329,49 +1352,48 @@ static void qdm2_decode_super_block (QDM2Context *q)
process_subpacket_11(q, NULL);
process_subpacket_12(q, NULL);
}
/* **************************************************************** */
}
static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet,
int offset, int duration, int channel,
int exp, int phase)
static void qdm2_fft_init_coefficient(QDM2Context *q, int sub_packet,
int offset, int duration, int channel,
int exp, int phase)
{
if (q->fft_coefs_min_index[duration] < 0)
q->fft_coefs_min_index[duration] = q->fft_coefs_index;
q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
q->fft_coefs[q->fft_coefs_index].sub_packet =
((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
q->fft_coefs[q->fft_coefs_index].channel = channel;
q->fft_coefs[q->fft_coefs_index].offset = offset;
q->fft_coefs[q->fft_coefs_index].exp = exp;
q->fft_coefs[q->fft_coefs_index].phase = phase;
q->fft_coefs[q->fft_coefs_index].offset = offset;
q->fft_coefs[q->fft_coefs_index].exp = exp;
q->fft_coefs[q->fft_coefs_index].phase = phase;
q->fft_coefs_index++;
}
static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b)
static void qdm2_fft_decode_tones(QDM2Context *q, int duration,
GetBitContext *gb, int b)
{
int channel, stereo, phase, exp;
int local_int_4, local_int_8, stereo_phase, local_int_10;
int local_int_4, local_int_8, stereo_phase, local_int_10;
int local_int_14, stereo_exp, local_int_20, local_int_28;
int n, offset;
local_int_4 = 0;
local_int_4 = 0;
local_int_28 = 0;
local_int_20 = 2;
local_int_8 = (4 - duration);
local_int_8 = (4 - duration);
local_int_10 = 1 << (q->group_order - duration - 1);
offset = 1;
offset = 1;
while (1) {
if (q->superblocktype_2_3) {
while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
offset = 1;
if (n == 0) {
local_int_4 += local_int_10;
local_int_4 += local_int_10;
local_int_28 += (1 << local_int_8);
} else {
local_int_4 += 8*local_int_10;
local_int_4 += 8 * local_int_10;
local_int_28 += (8 << local_int_8);
}
}
......@@ -1379,7 +1401,7 @@ static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *
} else {
offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
while (offset >= (local_int_10 - 1)) {
offset += (1 - (local_int_10 - 1));
offset += (1 - (local_int_10 - 1));
local_int_4 += local_int_10;
local_int_28 += (1 << local_int_8);
}
......@@ -1394,22 +1416,22 @@ static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *
if (q->nb_channels > 1) {
channel = get_bits1(gb);
stereo = get_bits1(gb);
stereo = get_bits1(gb);
} else {
channel = 0;
stereo = 0;
stereo = 0;
}
exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
exp = (exp < 0) ? 0 : exp;
exp = (exp < 0) ? 0 : exp;
phase = get_bits(gb, 3);
stereo_exp = 0;
phase = get_bits(gb, 3);
stereo_exp = 0;
stereo_phase = 0;
if (stereo) {
stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
if (stereo_phase < 0)
stereo_phase += 8;
......@@ -1418,17 +1440,18 @@ static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *
if (q->frequency_range > (local_int_14 + 1)) {
int sub_packet = (local_int_20 + local_int_28);
qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase);
qdm2_fft_init_coefficient(q, sub_packet, offset, duration,
channel, exp, phase);
if (stereo)
qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase);
qdm2_fft_init_coefficient(q, sub_packet, offset, duration,
1 - channel,
stereo_exp, stereo_phase);
}
offset++;
}
}
static void qdm2_decode_fft_packets (QDM2Context *q)
static void qdm2_decode_fft_packets(QDM2Context *q)
{
int i, j, min, max, value, type, unknown_flag;
GetBitContext gb;
......@@ -1438,18 +1461,18 @@ static void qdm2_decode_fft_packets (QDM2Context *q)
/* reset minimum indexes for FFT coefficients */
q->fft_coefs_index = 0;
for (i=0; i < 5; i++)
for (i = 0; i < 5; i++)
q->fft_coefs_min_index[i] = -1;
/* process subpackets ordered by type, largest type first */
for (i = 0, max = 256; i < q->sub_packets_B; i++) {
QDM2SubPacket *packet= NULL;
QDM2SubPacket *packet = NULL;
/* find subpacket with largest type less than max */
for (j = 0, min = 0; j < q->sub_packets_B; j++) {
value = q->sub_packet_list_B[j].packet->type;
if (value > min && value < max) {
min = value;
min = value;
packet = q->sub_packet_list_B[j].packet;
}
}
......@@ -1460,11 +1483,13 @@ static void qdm2_decode_fft_packets (QDM2Context *q)
if (!packet)
return;
if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16]))
if (i == 0 &&
(packet->type < 16 || packet->type >= 48 ||
fft_subpackets[packet->type - 16]))
return;
/* decode FFT tones */
init_get_bits (&gb, packet->data, packet->size*8);
init_get_bits(&gb, packet->data, packet->size * 8);
if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
unknown_flag = 1;
......@@ -1479,13 +1504,13 @@ static void qdm2_decode_fft_packets (QDM2Context *q)
if (duration >= 0 && duration < 4)
qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
} else if (type == 31) {
for (j=0; j < 4; j++)
for (j = 0; j < 4; j++)
qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
} else if (type == 46) {
for (j=0; j < 6; j++)
for (j = 0; j < 6; j++)
q->fft_level_exp[j] = get_bits(&gb, 6);
for (j=0; j < 4; j++)
qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
for (j = 0; j < 4; j++)
qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
}
} // Loop on B packets
......@@ -1500,20 +1525,19 @@ static void qdm2_decode_fft_packets (QDM2Context *q)
q->fft_coefs_max_index[j] = q->fft_coefs_index;
}
static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone)
static void qdm2_fft_generate_tone(QDM2Context *q, FFTTone *tone)
{
float level, f[6];
int i;
QDM2Complex c;
const double iscale = 2.0*M_PI / 512.0;
float level, f[6];
int i;
QDM2Complex c;
const double iscale = 2.0 * M_PI / 512.0;
tone->phase += tone->phase_shift;
/* calculate current level (maximum amplitude) of tone */
level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
c.im = level * sin(tone->phase*iscale);
c.re = level * cos(tone->phase*iscale);
c.im = level * sin(tone->phase * iscale);
c.re = level * cos(tone->phase * iscale);
/* generate FFT coefficients for tone */
if (tone->duration >= 3 || tone->cutoff >= 3) {
......@@ -1523,30 +1547,31 @@ static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone)
tone->complex[1].re -= c.re;
} else {
f[1] = -tone->table[4];
f[0] = tone->table[3] - tone->table[0];
f[2] = 1.0 - tone->table[2] - tone->table[3];
f[3] = tone->table[1] + tone->table[4] - 1.0;
f[4] = tone->table[0] - tone->table[1];
f[5] = tone->table[2];
f[0] = tone->table[3] - tone->table[0];
f[2] = 1.0 - tone->table[2] - tone->table[3];
f[3] = tone->table[1] + tone->table[4] - 1.0;
f[4] = tone->table[0] - tone->table[1];
f[5] = tone->table[2];
for (i = 0; i < 2; i++) {
tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i];
tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]);
tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re +=
c.re * f[i];
tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im +=
c.im * ((tone->cutoff <= i) ? -f[i] : f[i]);
}
for (i = 0; i < 4; i++) {
tone->complex[i].re += c.re * f[i+2];
tone->complex[i].im += c.im * f[i+2];
tone->complex[i].re += c.re * f[i + 2];
tone->complex[i].im += c.im * f[i + 2];
}
}
/* copy the tone if it has not yet died out */
if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
}
}
static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet)
static void qdm2_fft_tone_synthesizer(QDM2Context *q, int sub_packet)
{
int i, j, ch;
const double iscale = 0.25 * M_PI;
......@@ -1617,29 +1642,27 @@ static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet)
}
}
static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet)
static void qdm2_calculate_fft(QDM2Context *q, int channel, int sub_packet)
{
const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
float *out = q->output_buffer + channel;
float *out = q->output_buffer + channel;
int i;
q->fft.complex[channel][0].re *= 2.0f;
q->fft.complex[channel][0].im = 0.0f;
q->fft.complex[channel][0].im = 0.0f;
q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
/* add samples to output buffer */
for (i = 0; i < FFALIGN(q->fft_size, 8); i++) {
out[0] += q->fft.complex[channel][i].re * gain;
out[q->channels] += q->fft.complex[channel][i].im * gain;
out += 2 * q->channels;
out += 2 * q->channels;
}
}
/**
* @param q context
* @param index subpacket number
*/
static void qdm2_synthesis_filter (QDM2Context *q, int index)
static void qdm2_synthesis_filter(QDM2Context *q, int index)
{
int i, k, ch, sb_used, sub_sampling, dither_state = 0;
......@@ -1648,7 +1671,7 @@ static void qdm2_synthesis_filter (QDM2Context *q, int index)
for (ch = 0; ch < q->channels; ch++)
for (i = 0; i < 8; i++)
for (k=sb_used; k < SBLIMIT; k++)
for (k = sb_used; k < SBLIMIT; k++)
q->sb_samples[ch][(8 * index) + i][k] = 0;
for (ch = 0; ch < q->nb_channels; ch++) {
......@@ -1656,10 +1679,10 @@ static void qdm2_synthesis_filter (QDM2Context *q, int index)
for (i = 0; i < 8; i++) {
ff_mpa_synth_filter_float(&q->mpadsp,
q->synth_buf[ch], &(q->synth_buf_offset[ch]),
ff_mpa_synth_window_float, &dither_state,
samples_ptr, q->nb_channels,
q->sb_samples[ch][(8 * index) + i]);
q->synth_buf[ch], &(q->synth_buf_offset[ch]),
ff_mpa_synth_window_float, &dither_state,
samples_ptr, q->nb_channels,
q->sb_samples[ch][(8 * index) + i]);
samples_ptr += 32 * q->nb_channels;
}
}
......@@ -1672,7 +1695,6 @@ static void qdm2_synthesis_filter (QDM2Context *q, int index)
q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch];
}
/**
* Init static data (does not depend on specific file)
*
......@@ -1686,7 +1708,6 @@ static av_cold void qdm2_init_static_data(AVCodec *codec) {
init_noise_samples();
}
/**
* Init parameters from codec extradata
*/
......@@ -1736,7 +1757,7 @@ static av_cold int qdm2_decode_init(AVCodecContext *avctx)
return -1;
}
extradata = avctx->extradata;
extradata = avctx->extradata;
extradata_size = avctx->extradata_size;
while (extradata_size > 7) {
......@@ -1869,7 +1890,6 @@ static av_cold int qdm2_decode_init(AVCodecContext *avctx)
return 0;
}
static av_cold int qdm2_decode_close(AVCodecContext *avctx)
{
QDM2Context *s = avctx->priv_data;
......@@ -1879,8 +1899,7 @@ static av_cold int qdm2_decode_close(AVCodecContext *avctx)
return 0;
}
static int qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out)
static int qdm2_decode(QDM2Context *q, const uint8_t *in, int16_t *out)
{
int ch, i;
const int frame_size = (q->frame_size * q->channels);
......@@ -1939,7 +1958,6 @@ static int qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out)
return 0;
}
static int qdm2_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
......@@ -1974,8 +1992,7 @@ static int qdm2_decode_frame(AVCodecContext *avctx, void *data,
return s->checksum_size;
}
AVCodec ff_qdm2_decoder =
{
AVCodec ff_qdm2_decoder = {
.name = "qdm2",
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_QDM2,
......
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