[FFmpeg-cvslog] qdm2: formatting cosmetics
Luca Barbato
git at videolan.org
Tue Aug 27 16:51:16 CEST 2013
ffmpeg | branch: release/1.1 | Luca Barbato <lu_zero at gentoo.org> | Thu Jun 27 02:49:15 2013 +0200| [998a0389d307109af43e452441050281a09d540c] | committer: Luca Barbato
qdm2: formatting cosmetics
Apply the usual style plus drop few unnecessary return at the end
of void functions.
(cherry picked from commit 76efedeadb1f6bf79020c44a71dd0cee13d932ad)
Signed-off-by: Luca Barbato <lu_zero at gentoo.org>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=998a0389d307109af43e452441050281a09d540c
---
libavcodec/qdm2.c | 391 ++++++++++++++++++++++++++++-------------------------
1 file changed, 204 insertions(+), 187 deletions(-)
diff --git a/libavcodec/qdm2.c b/libavcodec/qdm2.c
index 767368c..297ca65 100644
--- a/libavcodec/qdm2.c
+++ b/libavcodec/qdm2.c
@@ -219,6 +219,10 @@ 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];
@@ -362,7 +366,7 @@ 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;
@@ -370,29 +374,27 @@ 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
*
@@ -402,49 +404,50 @@ 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.
*
@@ -452,7 +455,8 @@ 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)
@@ -462,14 +466,13 @@ 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;
@@ -486,12 +489,11 @@ 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.
@@ -499,7 +501,7 @@ 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;
@@ -508,14 +510,16 @@ 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.
@@ -524,44 +528,62 @@ 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
@@ -569,7 +591,7 @@ 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;
@@ -641,16 +663,14 @@ 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
@@ -660,9 +680,12 @@ 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;
@@ -768,15 +791,14 @@ 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
@@ -784,7 +806,8 @@ 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;
@@ -964,16 +987,18 @@ 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;
@@ -1000,11 +1025,11 @@ 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
@@ -1072,29 +1097,29 @@ 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;
}
}
@@ -1103,14 +1128,13 @@ 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;
@@ -1123,14 +1147,13 @@ 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;
@@ -1141,24 +1164,25 @@ 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;
@@ -1177,7 +1201,7 @@ 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];
@@ -1204,13 +1228,12 @@ 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;
@@ -1222,33 +1245,33 @@ 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;
}
}
@@ -1274,8 +1297,8 @@ 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;
@@ -1284,7 +1307,7 @@ 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)
@@ -1317,13 +1340,13 @@ 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;
@@ -1332,49 +1355,48 @@ 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);
}
}
@@ -1382,7 +1404,7 @@ 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);
}
@@ -1397,22 +1419,22 @@ 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;
@@ -1421,17 +1443,18 @@ 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;
@@ -1441,18 +1464,18 @@ 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;
}
}
@@ -1463,11 +1486,13 @@ 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;
@@ -1482,13 +1507,13 @@ 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
@@ -1503,20 +1528,19 @@ 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) {
@@ -1526,30 +1550,31 @@ 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;
@@ -1620,29 +1645,27 @@ 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;
@@ -1651,7 +1674,7 @@ 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++) {
@@ -1659,10 +1682,10 @@ 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;
}
}
@@ -1675,7 +1698,6 @@ 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)
*
@@ -1689,7 +1711,6 @@ static av_cold void qdm2_init_static_data(AVCodec *codec) {
init_noise_samples();
}
-
/**
* Init parameters from codec extradata
*/
@@ -1739,7 +1760,7 @@ 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) {
@@ -1875,7 +1896,6 @@ 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;
@@ -1885,8 +1905,7 @@ 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);
@@ -1945,7 +1964,6 @@ 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)
{
@@ -1980,8 +1998,7 @@ 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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