[FFmpeg-cvslog] dca: K&R formatting cosmetics
Shitiz Garg
git at videolan.org
Wed Jan 4 01:39:32 CET 2012
ffmpeg | branch: master | Shitiz Garg <mail at dragooon.net> | Tue Jan 3 16:37:28 2012 +0530| [f37b4efe03b88d9b1a0865a10a4e6708556ddb07] | committer: Diego Biurrun
dca: K&R formatting cosmetics
Signed-off-by: Diego Biurrun <diego at biurrun.de>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=f37b4efe03b88d9b1a0865a10a4e6708556ddb07
---
libavcodec/dca.c | 546 +++++++++++++++++++++++++++++-------------------------
1 files changed, 289 insertions(+), 257 deletions(-)
diff --git a/libavcodec/dca.c b/libavcodec/dca.c
index e3f87b9..3735b5a 100644
--- a/libavcodec/dca.c
+++ b/libavcodec/dca.c
@@ -48,13 +48,13 @@
//#define TRACE
-#define DCA_PRIM_CHANNELS_MAX (7)
-#define DCA_SUBBANDS (32)
-#define DCA_ABITS_MAX (32) /* Should be 28 */
-#define DCA_SUBSUBFRAMES_MAX (4)
-#define DCA_SUBFRAMES_MAX (16)
-#define DCA_BLOCKS_MAX (16)
-#define DCA_LFE_MAX (3)
+#define DCA_PRIM_CHANNELS_MAX (7)
+#define DCA_SUBBANDS (32)
+#define DCA_ABITS_MAX (32) /* Should be 28 */
+#define DCA_SUBSUBFRAMES_MAX (4)
+#define DCA_SUBFRAMES_MAX (16)
+#define DCA_BLOCKS_MAX (16)
+#define DCA_LFE_MAX (3)
enum DCAMode {
DCA_MONO = 0,
@@ -127,28 +127,45 @@ static const int dca_ext_audio_descr_mask[] = {
* OV -> center back
* All 2 channel configurations -> AV_CH_LAYOUT_STEREO
*/
-
static const uint64_t dca_core_channel_layout[] = {
- AV_CH_FRONT_CENTER, ///< 1, A
- AV_CH_LAYOUT_STEREO, ///< 2, A + B (dual mono)
- AV_CH_LAYOUT_STEREO, ///< 2, L + R (stereo)
- AV_CH_LAYOUT_STEREO, ///< 2, (L+R) + (L-R) (sum-difference)
- AV_CH_LAYOUT_STEREO, ///< 2, LT +RT (left and right total)
- AV_CH_LAYOUT_STEREO|AV_CH_FRONT_CENTER, ///< 3, C+L+R
- AV_CH_LAYOUT_STEREO|AV_CH_BACK_CENTER, ///< 3, L+R+S
- AV_CH_LAYOUT_STEREO|AV_CH_FRONT_CENTER|AV_CH_BACK_CENTER, ///< 4, C + L + R+ S
- AV_CH_LAYOUT_STEREO|AV_CH_SIDE_LEFT|AV_CH_SIDE_RIGHT, ///< 4, L + R +SL+ SR
- AV_CH_LAYOUT_STEREO|AV_CH_FRONT_CENTER|AV_CH_SIDE_LEFT|AV_CH_SIDE_RIGHT, ///< 5, C + L + R+ SL+SR
- AV_CH_LAYOUT_STEREO|AV_CH_SIDE_LEFT|AV_CH_SIDE_RIGHT|AV_CH_FRONT_LEFT_OF_CENTER|AV_CH_FRONT_RIGHT_OF_CENTER, ///< 6, CL + CR + L + R + SL + SR
- AV_CH_LAYOUT_STEREO|AV_CH_BACK_LEFT|AV_CH_BACK_RIGHT|AV_CH_FRONT_CENTER|AV_CH_BACK_CENTER, ///< 6, C + L + R+ LR + RR + OV
- AV_CH_FRONT_CENTER|AV_CH_FRONT_RIGHT_OF_CENTER|AV_CH_FRONT_LEFT_OF_CENTER|AV_CH_BACK_CENTER|AV_CH_BACK_LEFT|AV_CH_BACK_RIGHT, ///< 6, CF+ CR+LF+ RF+LR + RR
- AV_CH_FRONT_LEFT_OF_CENTER|AV_CH_FRONT_CENTER|AV_CH_FRONT_RIGHT_OF_CENTER|AV_CH_LAYOUT_STEREO|AV_CH_SIDE_LEFT|AV_CH_SIDE_RIGHT, ///< 7, CL + C + CR + L + R + SL + SR
- AV_CH_FRONT_LEFT_OF_CENTER|AV_CH_FRONT_RIGHT_OF_CENTER|AV_CH_LAYOUT_STEREO|AV_CH_SIDE_LEFT|AV_CH_SIDE_RIGHT|AV_CH_BACK_LEFT|AV_CH_BACK_RIGHT, ///< 8, CL + CR + L + R + SL1 + SL2+ SR1 + SR2
- AV_CH_FRONT_LEFT_OF_CENTER|AV_CH_FRONT_CENTER|AV_CH_FRONT_RIGHT_OF_CENTER|AV_CH_LAYOUT_STEREO|AV_CH_SIDE_LEFT|AV_CH_BACK_CENTER|AV_CH_SIDE_RIGHT, ///< 8, CL + C+ CR + L + R + SL + S+ SR
+ AV_CH_FRONT_CENTER, ///< 1, A
+ AV_CH_LAYOUT_STEREO, ///< 2, A + B (dual mono)
+ AV_CH_LAYOUT_STEREO, ///< 2, L + R (stereo)
+ AV_CH_LAYOUT_STEREO, ///< 2, (L + R) + (L - R) (sum-difference)
+ AV_CH_LAYOUT_STEREO, ///< 2, LT + RT (left and right total)
+ AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER, ///< 3, C + L + R
+ AV_CH_LAYOUT_STEREO | AV_CH_BACK_CENTER, ///< 3, L + R + S
+ AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER | AV_CH_BACK_CENTER, ///< 4, C + L + R + S
+ AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT, ///< 4, L + R + SL + SR
+
+ AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER | AV_CH_SIDE_LEFT |
+ AV_CH_SIDE_RIGHT, ///< 5, C + L + R + SL + SR
+
+ AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT |
+ AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER, ///< 6, CL + CR + L + R + SL + SR
+
+ AV_CH_LAYOUT_STEREO | AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT |
+ AV_CH_FRONT_CENTER | AV_CH_BACK_CENTER, ///< 6, C + L + R + LR + RR + OV
+
+ AV_CH_FRONT_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER |
+ AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_BACK_CENTER |
+ AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT, ///< 6, CF + CR + LF + RF + LR + RR
+
+ AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_CENTER |
+ AV_CH_FRONT_RIGHT_OF_CENTER | AV_CH_LAYOUT_STEREO |
+ AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT, ///< 7, CL + C + CR + L + R + SL + SR
+
+ AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER |
+ AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT |
+ AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT, ///< 8, CL + CR + L + R + SL1 + SL2 + SR1 + SR2
+
+ AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_CENTER |
+ AV_CH_FRONT_RIGHT_OF_CENTER | AV_CH_LAYOUT_STEREO |
+ AV_CH_SIDE_LEFT | AV_CH_BACK_CENTER | AV_CH_SIDE_RIGHT, ///< 8, CL + C + CR + L + R + SL + S + SR
};
static const int8_t dca_lfe_index[] = {
- 1,2,2,2,2,3,2,3,2,3,2,3,1,3,2,3
+ 1, 2, 2, 2, 2, 3, 2, 3, 2, 3, 2, 3, 1, 3, 2, 3
};
static const int8_t dca_channel_reorder_lfe[][9] = {
@@ -227,19 +244,19 @@ static const int8_t dca_channel_reorder_nolfe_xch[][9] = {
{ 3, 2, 4, 0, 1, 5, 8, 7, 6},
};
-#define DCA_DOLBY 101 /* FIXME */
+#define DCA_DOLBY 101 /* FIXME */
-#define DCA_CHANNEL_BITS 6
-#define DCA_CHANNEL_MASK 0x3F
+#define DCA_CHANNEL_BITS 6
+#define DCA_CHANNEL_MASK 0x3F
-#define DCA_LFE 0x80
+#define DCA_LFE 0x80
-#define HEADER_SIZE 14
+#define HEADER_SIZE 14
-#define DCA_MAX_FRAME_SIZE 16384
-#define DCA_MAX_EXSS_HEADER_SIZE 4096
+#define DCA_MAX_FRAME_SIZE 16384
+#define DCA_MAX_EXSS_HEADER_SIZE 4096
-#define DCA_BUFFER_PADDING_SIZE 1024
+#define DCA_BUFFER_PADDING_SIZE 1024
/** Bit allocation */
typedef struct {
@@ -254,9 +271,11 @@ static BitAlloc dca_tmode; ///< transition mode VLCs
static BitAlloc dca_scalefactor; ///< scalefactor VLCs
static BitAlloc dca_smpl_bitalloc[11]; ///< samples VLCs
-static av_always_inline int get_bitalloc(GetBitContext *gb, BitAlloc *ba, int idx)
+static av_always_inline int get_bitalloc(GetBitContext *gb, BitAlloc *ba,
+ int idx)
{
- return get_vlc2(gb, ba->vlc[idx].table, ba->vlc[idx].bits, ba->wrap) + ba->offset;
+ return get_vlc2(gb, ba->vlc[idx].table, ba->vlc[idx].bits, ba->wrap) +
+ ba->offset;
}
typedef struct {
@@ -307,8 +326,8 @@ typedef struct {
float scalefactor_adj[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< scale factor adjustment
/* Primary audio coding side information */
- int subsubframes[DCA_SUBFRAMES_MAX]; ///< number of subsubframes
- int partial_samples[DCA_SUBFRAMES_MAX]; ///< partial subsubframe samples count
+ int subsubframes[DCA_SUBFRAMES_MAX]; ///< number of subsubframes
+ int partial_samples[DCA_SUBFRAMES_MAX]; ///< partial subsubframe samples count
int prediction_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< prediction mode (ADPCM used or not)
int prediction_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< prediction VQ coefs
int bitalloc[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< bit allocation index
@@ -335,13 +354,13 @@ typedef struct {
float scale_bias; ///< output scale
DECLARE_ALIGNED(32, float, subband_samples)[DCA_BLOCKS_MAX][DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8];
- DECLARE_ALIGNED(32, float, samples)[(DCA_PRIM_CHANNELS_MAX+1)*256];
- const float *samples_chanptr[DCA_PRIM_CHANNELS_MAX+1];
+ DECLARE_ALIGNED(32, float, samples)[(DCA_PRIM_CHANNELS_MAX + 1) * 256];
+ const float *samples_chanptr[DCA_PRIM_CHANNELS_MAX + 1];
uint8_t dca_buffer[DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE + DCA_BUFFER_PADDING_SIZE];
int dca_buffer_size; ///< how much data is in the dca_buffer
- const int8_t* channel_order_tab; ///< channel reordering table, lfe and non lfe
+ const int8_t *channel_order_tab; ///< channel reordering table, lfe and non lfe
GetBitContext gb;
/* Current position in DCA frame */
int current_subframe;
@@ -416,13 +435,15 @@ static av_cold void dca_init_vlcs(void)
}
for (i = 0; i < 10; i++)
- for (j = 0; j < 7; j++){
- if (!bitalloc_codes[i][j]) break;
- dca_smpl_bitalloc[i+1].offset = bitalloc_offsets[i];
- dca_smpl_bitalloc[i+1].wrap = 1 + (j > 4);
- dca_smpl_bitalloc[i+1].vlc[j].table = &dca_table[dca_vlc_offs[c]];
- dca_smpl_bitalloc[i+1].vlc[j].table_allocated = dca_vlc_offs[c + 1] - dca_vlc_offs[c];
- init_vlc(&dca_smpl_bitalloc[i+1].vlc[j], bitalloc_maxbits[i][j],
+ for (j = 0; j < 7; j++) {
+ if (!bitalloc_codes[i][j])
+ break;
+ dca_smpl_bitalloc[i + 1].offset = bitalloc_offsets[i];
+ dca_smpl_bitalloc[i + 1].wrap = 1 + (j > 4);
+ dca_smpl_bitalloc[i + 1].vlc[j].table = &dca_table[dca_vlc_offs[c]];
+ dca_smpl_bitalloc[i + 1].vlc[j].table_allocated = dca_vlc_offs[c + 1] - dca_vlc_offs[c];
+
+ init_vlc(&dca_smpl_bitalloc[i + 1].vlc[j], bitalloc_maxbits[i][j],
bitalloc_sizes[i],
bitalloc_bits[i][j], 1, 1,
bitalloc_codes[i][j], 2, 2, INIT_VLC_USE_NEW_STATIC);
@@ -433,19 +454,19 @@ static av_cold void dca_init_vlcs(void)
static inline void get_array(GetBitContext *gb, int *dst, int len, int bits)
{
- while(len--)
+ while (len--)
*dst++ = get_bits(gb, bits);
}
-static int dca_parse_audio_coding_header(DCAContext * s, int base_channel)
+static int dca_parse_audio_coding_header(DCAContext *s, int base_channel)
{
int i, j;
static const float adj_table[4] = { 1.0, 1.1250, 1.2500, 1.4375 };
static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 };
- static const int thr[11] = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 };
+ static const int thr[11] = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 };
- s->total_channels = get_bits(&s->gb, 3) + 1 + base_channel;
- s->prim_channels = s->total_channels;
+ s->total_channels = get_bits(&s->gb, 3) + 1 + base_channel;
+ s->prim_channels = s->total_channels;
if (s->prim_channels > DCA_PRIM_CHANNELS_MAX)
s->prim_channels = DCA_PRIM_CHANNELS_MAX;
@@ -488,23 +509,28 @@ static int dca_parse_audio_coding_header(DCAContext * s, int base_channel)
get_bits(&s->gb, 16);
}
- s->current_subframe = 0;
+ s->current_subframe = 0;
s->current_subsubframe = 0;
#ifdef TRACE
av_log(s->avctx, AV_LOG_DEBUG, "subframes: %i\n", s->subframes);
av_log(s->avctx, AV_LOG_DEBUG, "prim channels: %i\n", s->prim_channels);
- for (i = base_channel; i < s->prim_channels; i++){
- av_log(s->avctx, AV_LOG_DEBUG, "subband activity: %i\n", s->subband_activity[i]);
- av_log(s->avctx, AV_LOG_DEBUG, "vq start subband: %i\n", s->vq_start_subband[i]);
- av_log(s->avctx, AV_LOG_DEBUG, "joint intensity: %i\n", s->joint_intensity[i]);
- av_log(s->avctx, AV_LOG_DEBUG, "transient mode codebook: %i\n", s->transient_huffman[i]);
- av_log(s->avctx, AV_LOG_DEBUG, "scale factor codebook: %i\n", s->scalefactor_huffman[i]);
- av_log(s->avctx, AV_LOG_DEBUG, "bit allocation quantizer: %i\n", s->bitalloc_huffman[i]);
+ for (i = base_channel; i < s->prim_channels; i++) {
+ av_log(s->avctx, AV_LOG_DEBUG, "subband activity: %i\n",
+ s->subband_activity[i]);
+ av_log(s->avctx, AV_LOG_DEBUG, "vq start subband: %i\n",
+ s->vq_start_subband[i]);
+ av_log(s->avctx, AV_LOG_DEBUG, "joint intensity: %i\n",
+ s->joint_intensity[i]);
+ av_log(s->avctx, AV_LOG_DEBUG, "transient mode codebook: %i\n",
+ s->transient_huffman[i]);
+ av_log(s->avctx, AV_LOG_DEBUG, "scale factor codebook: %i\n",
+ s->scalefactor_huffman[i]);
+ av_log(s->avctx, AV_LOG_DEBUG, "bit allocation quantizer: %i\n",
+ s->bitalloc_huffman[i]);
av_log(s->avctx, AV_LOG_DEBUG, "quant index huff:");
for (j = 0; j < 11; j++)
- av_log(s->avctx, AV_LOG_DEBUG, " %i",
- s->quant_index_huffman[i][j]);
+ av_log(s->avctx, AV_LOG_DEBUG, " %i", s->quant_index_huffman[i][j]);
av_log(s->avctx, AV_LOG_DEBUG, "\n");
av_log(s->avctx, AV_LOG_DEBUG, "scalefac adj:");
for (j = 0; j < 11; j++)
@@ -513,10 +539,10 @@ static int dca_parse_audio_coding_header(DCAContext * s, int base_channel)
}
#endif
- return 0;
+ return 0;
}
-static int dca_parse_frame_header(DCAContext * s)
+static int dca_parse_frame_header(DCAContext *s)
{
init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);
@@ -565,7 +591,8 @@ static int dca_parse_frame_header(DCAContext * s)
/* FIXME: channels mixing levels */
s->output = s->amode;
- if (s->lfe) s->output |= DCA_LFE;
+ if (s->lfe)
+ s->output |= DCA_LFE;
#ifdef TRACE
av_log(s->avctx, AV_LOG_DEBUG, "frame type: %i\n", s->frame_type);
@@ -614,15 +641,15 @@ static int dca_parse_frame_header(DCAContext * s)
static inline int get_scale(GetBitContext *gb, int level, int value)
{
- if (level < 5) {
- /* huffman encoded */
- value += get_bitalloc(gb, &dca_scalefactor, level);
- } else if (level < 8)
- value = get_bits(gb, level + 1);
- return value;
+ if (level < 5) {
+ /* huffman encoded */
+ value += get_bitalloc(gb, &dca_scalefactor, level);
+ } else if (level < 8)
+ value = get_bits(gb, level + 1);
+ return value;
}
-static int dca_subframe_header(DCAContext * s, int base_channel, int block_index)
+static int dca_subframe_header(DCAContext *s, int base_channel, int block_index)
{
/* Primary audio coding side information */
int j, k;
@@ -631,7 +658,7 @@ static int dca_subframe_header(DCAContext * s, int base_channel, int block_index
return AVERROR_INVALIDDATA;
if (!base_channel) {
- s->subsubframes[s->current_subframe] = get_bits(&s->gb, 2) + 1;
+ s->subsubframes[s->current_subframe] = get_bits(&s->gb, 2) + 1;
s->partial_samples[s->current_subframe] = get_bits(&s->gb, 3);
}
@@ -667,8 +694,8 @@ static int dca_subframe_header(DCAContext * s, int base_channel, int block_index
}
if (s->bitalloc[j][k] > 26) {
-// av_log(s->avctx,AV_LOG_DEBUG,"bitalloc index [%i][%i] too big (%i)\n",
-// j, k, s->bitalloc[j][k]);
+ // av_log(s->avctx, AV_LOG_DEBUG, "bitalloc index [%i][%i] too big (%i)\n",
+ // j, k, s->bitalloc[j][k]);
return AVERROR_INVALIDDATA;
}
}
@@ -693,7 +720,8 @@ static int dca_subframe_header(DCAContext * s, int base_channel, int block_index
const uint32_t *scale_table;
int scale_sum;
- memset(s->scale_factor[j], 0, s->subband_activity[j] * sizeof(s->scale_factor[0][0][0]) * 2);
+ memset(s->scale_factor[j], 0,
+ s->subband_activity[j] * sizeof(s->scale_factor[0][0][0]) * 2);
if (s->scalefactor_huffman[j] == 6)
scale_table = scale_factor_quant7;
@@ -811,9 +839,11 @@ static int dca_subframe_header(DCAContext * s, int base_channel, int block_index
}
#ifdef TRACE
- av_log(s->avctx, AV_LOG_DEBUG, "subsubframes: %i\n", s->subsubframes[s->current_subframe]);
+ av_log(s->avctx, AV_LOG_DEBUG, "subsubframes: %i\n",
+ s->subsubframes[s->current_subframe]);
av_log(s->avctx, AV_LOG_DEBUG, "partial samples: %i\n",
s->partial_samples[s->current_subframe]);
+
for (j = base_channel; j < s->prim_channels; j++) {
av_log(s->avctx, AV_LOG_DEBUG, "prediction mode:");
for (k = 0; k < s->subband_activity[j]; k++)
@@ -822,12 +852,12 @@ static int dca_subframe_header(DCAContext * s, int base_channel, int block_index
}
for (j = base_channel; j < s->prim_channels; j++) {
for (k = 0; k < s->subband_activity[j]; k++)
- av_log(s->avctx, AV_LOG_DEBUG,
- "prediction coefs: %f, %f, %f, %f\n",
- (float) adpcm_vb[s->prediction_vq[j][k]][0] / 8192,
- (float) adpcm_vb[s->prediction_vq[j][k]][1] / 8192,
- (float) adpcm_vb[s->prediction_vq[j][k]][2] / 8192,
- (float) adpcm_vb[s->prediction_vq[j][k]][3] / 8192);
+ av_log(s->avctx, AV_LOG_DEBUG,
+ "prediction coefs: %f, %f, %f, %f\n",
+ (float) adpcm_vb[s->prediction_vq[j][k]][0] / 8192,
+ (float) adpcm_vb[s->prediction_vq[j][k]][1] / 8192,
+ (float) adpcm_vb[s->prediction_vq[j][k]][2] / 8192,
+ (float) adpcm_vb[s->prediction_vq[j][k]][3] / 8192);
}
for (j = base_channel; j < s->prim_channels; j++) {
av_log(s->avctx, AV_LOG_DEBUG, "bitalloc index: ");
@@ -863,8 +893,10 @@ static int dca_subframe_header(DCAContext * s, int base_channel, int block_index
if (!base_channel && s->prim_channels > 2 && s->downmix) {
av_log(s->avctx, AV_LOG_DEBUG, "Downmix coeffs:\n");
for (j = 0; j < s->prim_channels; j++) {
- av_log(s->avctx, AV_LOG_DEBUG, "Channel 0,%d = %f\n", j, dca_downmix_coeffs[s->downmix_coef[j][0]]);
- av_log(s->avctx, AV_LOG_DEBUG, "Channel 1,%d = %f\n", j, dca_downmix_coeffs[s->downmix_coef[j][1]]);
+ av_log(s->avctx, AV_LOG_DEBUG, "Channel 0, %d = %f\n", j,
+ dca_downmix_coeffs[s->downmix_coef[j][0]]);
+ av_log(s->avctx, AV_LOG_DEBUG, "Channel 1, %d = %f\n", j,
+ dca_downmix_coeffs[s->downmix_coef[j][1]]);
}
av_log(s->avctx, AV_LOG_DEBUG, "\n");
}
@@ -885,7 +917,7 @@ static int dca_subframe_header(DCAContext * s, int base_channel, int block_index
return 0;
}
-static void qmf_32_subbands(DCAContext * s, int chans,
+static void qmf_32_subbands(DCAContext *s, int chans,
float samples_in[32][8], float *samples_out,
float scale)
{
@@ -895,7 +927,7 @@ static void qmf_32_subbands(DCAContext * s, int chans,
int sb_act = s->subband_activity[chans];
int subindex;
- scale *= sqrt(1/8.0);
+ scale *= sqrt(1 / 8.0);
/* Select filter */
if (!s->multirate_inter) /* Non-perfect reconstruction */
@@ -909,18 +941,18 @@ static void qmf_32_subbands(DCAContext * s, int chans,
/* Reconstructed channel sample index */
for (subindex = 0; subindex < 8; subindex++) {
/* Load in one sample from each subband and clear inactive subbands */
- for (i = 0; i < sb_act; i++){
+ for (i = 0; i < sb_act; i++) {
unsigned sign = (i - 1) & 2;
- uint32_t v = AV_RN32A(&samples_in[i][subindex]) ^ sign << 30;
+ uint32_t v = AV_RN32A(&samples_in[i][subindex]) ^ sign << 30;
AV_WN32A(&s->raXin[i], v);
}
s->synth.synth_filter_float(&s->imdct,
- s->subband_fir_hist[chans], &s->hist_index[chans],
- s->subband_fir_noidea[chans], prCoeff,
- samples_out, s->raXin, scale);
- samples_out+= 32;
-
+ s->subband_fir_hist[chans],
+ &s->hist_index[chans],
+ s->subband_fir_noidea[chans], prCoeff,
+ samples_out, s->raXin, scale);
+ samples_out += 32;
}
}
@@ -950,45 +982,44 @@ static void lfe_interpolation_fir(DCAContext *s, int decimation_select,
}
/* Interpolation */
for (deciindex = 0; deciindex < num_deci_sample; deciindex++) {
- s->dcadsp.lfe_fir(samples_out, samples_in, prCoeff, decifactor,
- scale);
+ s->dcadsp.lfe_fir(samples_out, samples_in, prCoeff, decifactor, scale);
samples_in++;
samples_out += 2 * decifactor;
}
}
/* downmixing routines */
-#define MIX_REAR1(samples, si1, rs, coef) \
- samples[i] += samples[si1] * coef[rs][0]; \
- samples[i+256] += samples[si1] * coef[rs][1];
-
-#define MIX_REAR2(samples, si1, si2, rs, coef) \
- samples[i] += samples[si1] * coef[rs][0] + samples[si2] * coef[rs+1][0]; \
- samples[i+256] += samples[si1] * coef[rs][1] + samples[si2] * coef[rs+1][1];
-
-#define MIX_FRONT3(samples, coef) \
- t = samples[i+c]; \
- u = samples[i+l]; \
- v = samples[i+r]; \
+#define MIX_REAR1(samples, si1, rs, coef) \
+ samples[i] += samples[si1] * coef[rs][0]; \
+ samples[i+256] += samples[si1] * coef[rs][1];
+
+#define MIX_REAR2(samples, si1, si2, rs, coef) \
+ samples[i] += samples[si1] * coef[rs][0] + samples[si2] * coef[rs + 1][0]; \
+ samples[i+256] += samples[si1] * coef[rs][1] + samples[si2] * coef[rs + 1][1];
+
+#define MIX_FRONT3(samples, coef) \
+ t = samples[i + c]; \
+ u = samples[i + l]; \
+ v = samples[i + r]; \
samples[i] = t * coef[0][0] + u * coef[1][0] + v * coef[2][0]; \
samples[i+256] = t * coef[0][1] + u * coef[1][1] + v * coef[2][1];
-#define DOWNMIX_TO_STEREO(op1, op2) \
- for (i = 0; i < 256; i++){ \
- op1 \
- op2 \
+#define DOWNMIX_TO_STEREO(op1, op2) \
+ for (i = 0; i < 256; i++) { \
+ op1 \
+ op2 \
}
static void dca_downmix(float *samples, int srcfmt,
int downmix_coef[DCA_PRIM_CHANNELS_MAX][2],
const int8_t *channel_mapping)
{
- int c,l,r,sl,sr,s;
+ int c, l, r, sl, sr, s;
int i;
float t, u, v;
float coef[DCA_PRIM_CHANNELS_MAX][2];
- for (i=0; i<DCA_PRIM_CHANNELS_MAX; i++) {
+ for (i = 0; i < DCA_PRIM_CHANNELS_MAX; i++) {
coef[i][0] = dca_downmix_coeffs[downmix_coef[i][0]];
coef[i][1] = dca_downmix_coeffs[downmix_coef[i][1]];
}
@@ -1007,11 +1038,11 @@ static void dca_downmix(float *samples, int srcfmt,
c = channel_mapping[0] * 256;
l = channel_mapping[1] * 256;
r = channel_mapping[2] * 256;
- DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),);
+ DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef), );
break;
case DCA_2F1R:
s = channel_mapping[2] * 256;
- DOWNMIX_TO_STEREO(MIX_REAR1(samples, i + s, 2, coef),);
+ DOWNMIX_TO_STEREO(MIX_REAR1(samples, i + s, 2, coef), );
break;
case DCA_3F1R:
c = channel_mapping[0] * 256;
@@ -1024,12 +1055,12 @@ static void dca_downmix(float *samples, int srcfmt,
case DCA_2F2R:
sl = channel_mapping[2] * 256;
sr = channel_mapping[3] * 256;
- DOWNMIX_TO_STEREO(MIX_REAR2(samples, i + sl, i + sr, 2, coef),);
+ DOWNMIX_TO_STEREO(MIX_REAR2(samples, i + sl, i + sr, 2, coef), );
break;
case DCA_3F2R:
- c = channel_mapping[0] * 256;
- l = channel_mapping[1] * 256;
- r = channel_mapping[2] * 256;
+ c = channel_mapping[0] * 256;
+ l = channel_mapping[1] * 256;
+ r = channel_mapping[2] * 256;
sl = channel_mapping[3] * 256;
sr = channel_mapping[4] * 256;
DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
@@ -1049,7 +1080,7 @@ static int decode_blockcode(int code, int levels, int *values)
for (i = 0; i < 4; i++) {
int div = FASTDIV(code, levels);
- values[i] = code - offset - div*levels;
+ values[i] = code - offset - div * levels;
code = div;
}
@@ -1063,8 +1094,8 @@ static int decode_blockcodes(int code1, int code2, int levels, int *values)
}
#endif
-static const uint8_t abits_sizes[7] = { 7, 10, 12, 13, 15, 17, 19 };
-static const uint8_t abits_levels[7] = { 3, 5, 7, 9, 13, 17, 25 };
+static const uint8_t abits_sizes[7] = { 7, 10, 12, 13, 15, 17, 19 };
+static const uint8_t abits_levels[7] = { 3, 5, 7, 9, 13, 17, 25 };
#ifndef int8x8_fmul_int32
static inline void int8x8_fmul_int32(float *dst, const int8_t *src, int scale)
@@ -1076,7 +1107,7 @@ static inline void int8x8_fmul_int32(float *dst, const int8_t *src, int scale)
}
#endif
-static int dca_subsubframe(DCAContext * s, int base_channel, int block_index)
+static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
{
int k, l;
int subsubframe = s->current_subsubframe;
@@ -1119,20 +1150,21 @@ static int dca_subsubframe(DCAContext * s, int base_channel, int block_index)
/*
* Extract bits from the bit stream
*/
- if (!abits){
+ if (!abits) {
memset(subband_samples[k][l], 0, 8 * sizeof(subband_samples[0][0][0]));
} else {
/* Deal with transients */
int sfi = s->transition_mode[k][l] && subsubframe >= s->transition_mode[k][l];
- float rscale = quant_step_size * s->scale_factor[k][l][sfi] * s->scalefactor_adj[k][sel];
+ float rscale = quant_step_size * s->scale_factor[k][l][sfi] *
+ s->scalefactor_adj[k][sel];
- if (abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table){
- if (abits <= 7){
+ if (abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table) {
+ if (abits <= 7) {
/* Block code */
int block_code1, block_code2, size, levels, err;
- size = abits_sizes[abits-1];
- levels = abits_levels[abits-1];
+ size = abits_sizes[abits - 1];
+ levels = abits_levels[abits - 1];
block_code1 = get_bits(&s->gb, size);
block_code2 = get_bits(&s->gb, size);
@@ -1143,19 +1175,20 @@ static int dca_subsubframe(DCAContext * s, int base_channel, int block_index)
"ERROR: block code look-up failed\n");
return AVERROR_INVALIDDATA;
}
- }else{
+ } else {
/* no coding */
for (m = 0; m < 8; m++)
block[m] = get_sbits(&s->gb, abits - 3);
}
- }else{
+ } else {
/* Huffman coded */
for (m = 0; m < 8; m++)
- block[m] = get_bitalloc(&s->gb, &dca_smpl_bitalloc[abits], sel);
+ block[m] = get_bitalloc(&s->gb,
+ &dca_smpl_bitalloc[abits], sel);
}
s->fmt_conv.int32_to_float_fmul_scalar(subband_samples[k][l],
- block, rscale, 8);
+ block, rscale, 8);
}
/*
@@ -1172,8 +1205,7 @@ static int dca_subsubframe(DCAContext * s, int base_channel, int block_index)
else if (s->predictor_history)
subband_samples[k][l][m] +=
(adpcm_vb[s->prediction_vq[k][l]][n - 1] *
- s->subband_samples_hist[k][l][m - n +
- 4] / 8192);
+ s->subband_samples_hist[k][l][m - n + 4] / 8192);
}
}
}
@@ -1187,7 +1219,8 @@ static int dca_subsubframe(DCAContext * s, int base_channel, int block_index)
int hfvq = s->high_freq_vq[k][l];
if (!s->debug_flag & 0x01) {
- av_log(s->avctx, AV_LOG_DEBUG, "Stream with high frequencies VQ coding\n");
+ av_log(s->avctx, AV_LOG_DEBUG,
+ "Stream with high frequencies VQ coding\n");
s->debug_flag |= 0x01;
}
@@ -1211,23 +1244,25 @@ static int dca_subsubframe(DCAContext * s, int base_channel, int block_index)
/* Backup predictor history for adpcm */
for (k = base_channel; k < s->prim_channels; k++)
for (l = 0; l < s->vq_start_subband[k]; l++)
- memcpy(s->subband_samples_hist[k][l], &subband_samples[k][l][4],
- 4 * sizeof(subband_samples[0][0][0]));
+ memcpy(s->subband_samples_hist[k][l],
+ &subband_samples[k][l][4],
+ 4 * sizeof(subband_samples[0][0][0]));
return 0;
}
-static int dca_filter_channels(DCAContext * s, int block_index)
+static int dca_filter_channels(DCAContext *s, int block_index)
{
float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];
int k;
/* 32 subbands QMF */
for (k = 0; k < s->prim_channels; k++) {
-/* static float pcm_to_double[8] =
- {32768.0, 32768.0, 524288.0, 524288.0, 0, 8388608.0, 8388608.0};*/
- qmf_32_subbands(s, k, subband_samples[k], &s->samples[256 * s->channel_order_tab[k]],
- M_SQRT1_2*s->scale_bias /*pcm_to_double[s->source_pcm_res] */ );
+/* static float pcm_to_double[8] = { 32768.0, 32768.0, 524288.0, 524288.0,
+ 0, 8388608.0, 8388608.0 };*/
+ qmf_32_subbands(s, k, subband_samples[k],
+ &s->samples[256 * s->channel_order_tab[k]],
+ M_SQRT1_2 * s->scale_bias /* pcm_to_double[s->source_pcm_res] */);
}
/* Down mixing */
@@ -1240,7 +1275,7 @@ static int dca_filter_channels(DCAContext * s, int block_index)
lfe_interpolation_fir(s, s->lfe, 2 * s->lfe,
s->lfe_data + 2 * s->lfe * (block_index + 4),
&s->samples[256 * dca_lfe_index[s->amode]],
- (1.0/256.0)*s->scale_bias);
+ (1.0 / 256.0) * s->scale_bias);
/* Outputs 20bits pcm samples */
}
@@ -1248,7 +1283,7 @@ static int dca_filter_channels(DCAContext * s, int block_index)
}
-static int dca_subframe_footer(DCAContext * s, int base_channel)
+static int dca_subframe_footer(DCAContext *s, int base_channel)
{
int aux_data_count = 0, i;
@@ -1280,7 +1315,7 @@ static int dca_subframe_footer(DCAContext * s, int base_channel)
* @param s pointer to the DCAContext
*/
-static int dca_decode_block(DCAContext * s, int base_channel, int block_index)
+static int dca_decode_block(DCAContext *s, int base_channel, int block_index)
{
int ret;
@@ -1328,8 +1363,8 @@ static int dca_decode_block(DCAContext * s, int base_channel, int block_index)
/**
* Convert bitstream to one representation based on sync marker
*/
-static int dca_convert_bitstream(const uint8_t * src, int src_size, uint8_t * dst,
- int max_size)
+static int dca_convert_bitstream(const uint8_t *src, int src_size, uint8_t *dst,
+ int max_size)
{
uint32_t mrk;
int i, tmp;
@@ -1337,7 +1372,7 @@ static int dca_convert_bitstream(const uint8_t * src, int src_size, uint8_t * ds
uint16_t *sdst = (uint16_t *) dst;
PutBitContext pb;
- if ((unsigned)src_size > (unsigned)max_size) {
+ if ((unsigned) src_size > (unsigned) max_size) {
// av_log(NULL, AV_LOG_ERROR, "Input frame size larger than DCA_MAX_FRAME_SIZE!\n");
// return -1;
src_size = max_size;
@@ -1372,18 +1407,16 @@ static int dca_convert_bitstream(const uint8_t * src, int src_size, uint8_t * ds
static int dca_exss_mask2count(int mask)
{
/* count bits that mean speaker pairs twice */
- return av_popcount(mask)
- + av_popcount(mask & (
- DCA_EXSS_CENTER_LEFT_RIGHT
- | DCA_EXSS_FRONT_LEFT_RIGHT
- | DCA_EXSS_FRONT_HIGH_LEFT_RIGHT
- | DCA_EXSS_WIDE_LEFT_RIGHT
- | DCA_EXSS_SIDE_LEFT_RIGHT
- | DCA_EXSS_SIDE_HIGH_LEFT_RIGHT
- | DCA_EXSS_SIDE_REAR_LEFT_RIGHT
- | DCA_EXSS_REAR_LEFT_RIGHT
- | DCA_EXSS_REAR_HIGH_LEFT_RIGHT
- ));
+ return av_popcount(mask) +
+ av_popcount(mask & (DCA_EXSS_CENTER_LEFT_RIGHT |
+ DCA_EXSS_FRONT_LEFT_RIGHT |
+ DCA_EXSS_FRONT_HIGH_LEFT_RIGHT |
+ DCA_EXSS_WIDE_LEFT_RIGHT |
+ DCA_EXSS_SIDE_LEFT_RIGHT |
+ DCA_EXSS_SIDE_HIGH_LEFT_RIGHT |
+ DCA_EXSS_SIDE_REAR_LEFT_RIGHT |
+ DCA_EXSS_REAR_LEFT_RIGHT |
+ DCA_EXSS_REAR_HIGH_LEFT_RIGHT));
}
/**
@@ -1409,7 +1442,7 @@ static int dca_exss_parse_asset_header(DCAContext *s)
int header_size;
int channels;
int embedded_stereo = 0;
- int embedded_6ch = 0;
+ int embedded_6ch = 0;
int drc_code_present;
int extensions_mask;
int i, j;
@@ -1544,7 +1577,8 @@ static int dca_exss_parse_asset_header(DCAContext *s)
if (!(extensions_mask & DCA_EXT_CORE))
av_log(s->avctx, AV_LOG_WARNING, "DTS core detection mismatch.\n");
if ((extensions_mask & DCA_CORE_EXTS) != s->core_ext_mask)
- av_log(s->avctx, AV_LOG_WARNING, "DTS extensions detection mismatch (%d, %d)\n",
+ av_log(s->avctx, AV_LOG_WARNING,
+ "DTS extensions detection mismatch (%d, %d)\n",
extensions_mask & DCA_CORE_EXTS, s->core_ext_mask);
return 0;
@@ -1569,7 +1603,7 @@ static void dca_exss_parse_header(DCAContext *s)
ss_index = get_bits(&s->gb, 2);
blownup = get_bits1(&s->gb);
- skip_bits(&s->gb, 8 + 4 * blownup); // header_size
+ skip_bits(&s->gb, 8 + 4 * blownup); // header_size
skip_bits(&s->gb, 16 + 4 * blownup); // hd_size
s->static_fields = get_bits1(&s->gb);
@@ -1610,18 +1644,18 @@ static void dca_exss_parse_header(DCAContext *s)
int mix_out_mask_size;
skip_bits(&s->gb, 2); // adjustment level
- mix_out_mask_size = (get_bits(&s->gb, 2) + 1) << 2;
- s->num_mix_configs = get_bits(&s->gb, 2) + 1;
+ mix_out_mask_size = (get_bits(&s->gb, 2) + 1) << 2;
+ s->num_mix_configs = get_bits(&s->gb, 2) + 1;
for (i = 0; i < s->num_mix_configs; i++) {
- int mix_out_mask = get_bits(&s->gb, mix_out_mask_size);
+ int mix_out_mask = get_bits(&s->gb, mix_out_mask_size);
s->mix_config_num_ch[i] = dca_exss_mask2count(mix_out_mask);
}
}
}
for (i = 0; i < num_assets; i++)
- skip_bits_long(&s->gb, 16 + 4 * blownup); // asset size
+ skip_bits_long(&s->gb, 16 + 4 * blownup); // asset size
for (i = 0; i < num_assets; i++) {
if (dca_exss_parse_asset_header(s))
@@ -1668,8 +1702,8 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data,
}
//set AVCodec values with parsed data
avctx->sample_rate = s->sample_rate;
- avctx->bit_rate = s->bit_rate;
- avctx->frame_size = s->sample_blocks * 32;
+ avctx->bit_rate = s->bit_rate;
+ avctx->frame_size = s->sample_blocks * 32;
s->profile = FF_PROFILE_DTS;
@@ -1701,72 +1735,71 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data,
/* extensions start at 32-bit boundaries into bitstream */
skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
- while(core_ss_end - get_bits_count(&s->gb) >= 32) {
- uint32_t bits = get_bits_long(&s->gb, 32);
+ while (core_ss_end - get_bits_count(&s->gb) >= 32) {
+ uint32_t bits = get_bits_long(&s->gb, 32);
- switch(bits) {
- case 0x5a5a5a5a: {
- int ext_amode, xch_fsize;
+ switch (bits) {
+ case 0x5a5a5a5a: {
+ int ext_amode, xch_fsize;
- s->xch_base_channel = s->prim_channels;
+ s->xch_base_channel = s->prim_channels;
- /* validate sync word using XCHFSIZE field */
- xch_fsize = show_bits(&s->gb, 10);
- if((s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize) &&
- (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize + 1))
- continue;
-
- /* skip length-to-end-of-frame field for the moment */
- skip_bits(&s->gb, 10);
-
- s->core_ext_mask |= DCA_EXT_XCH;
+ /* validate sync word using XCHFSIZE field */
+ xch_fsize = show_bits(&s->gb, 10);
+ if ((s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize) &&
+ (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize + 1))
+ continue;
- /* extension amode should == 1, number of channels in extension */
- /* AFAIK XCh is not used for more channels */
- if ((ext_amode = get_bits(&s->gb, 4)) != 1) {
- av_log(avctx, AV_LOG_ERROR, "XCh extension amode %d not"
- " supported!\n",ext_amode);
- continue;
- }
+ /* skip length-to-end-of-frame field for the moment */
+ skip_bits(&s->gb, 10);
- /* much like core primary audio coding header */
- dca_parse_audio_coding_header(s, s->xch_base_channel);
+ s->core_ext_mask |= DCA_EXT_XCH;
- for (i = 0; i < (s->sample_blocks / 8); i++) {
- if ((ret = dca_decode_block(s, s->xch_base_channel, i))) {
- av_log(avctx, AV_LOG_ERROR, "error decoding XCh extension\n");
+ /* extension amode(number of channels in extension) should be 1 */
+ /* AFAIK XCh is not used for more channels */
+ if ((ext_amode = get_bits(&s->gb, 4)) != 1) {
+ av_log(avctx, AV_LOG_ERROR, "XCh extension amode %d not"
+ " supported!\n", ext_amode);
continue;
}
+
+ /* much like core primary audio coding header */
+ dca_parse_audio_coding_header(s, s->xch_base_channel);
+
+ for (i = 0; i < (s->sample_blocks / 8); i++)
+ if ((ret = dca_decode_block(s, s->xch_base_channel, i))) {
+ av_log(avctx, AV_LOG_ERROR, "error decoding XCh extension\n");
+ continue;
+ }
+
+ s->xch_present = 1;
+ break;
}
+ case 0x47004a03:
+ /* XXCh: extended channels */
+ /* usually found either in core or HD part in DTS-HD HRA streams,
+ * but not in DTS-ES which contains XCh extensions instead */
+ s->core_ext_mask |= DCA_EXT_XXCH;
+ break;
+
+ case 0x1d95f262: {
+ int fsize96 = show_bits(&s->gb, 12) + 1;
+ if (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + fsize96)
+ continue;
- s->xch_present = 1;
- break;
- }
- case 0x47004a03:
- /* XXCh: extended channels */
- /* usually found either in core or HD part in DTS-HD HRA streams,
- * but not in DTS-ES which contains XCh extensions instead */
- s->core_ext_mask |= DCA_EXT_XXCH;
- break;
-
- case 0x1d95f262: {
- int fsize96 = show_bits(&s->gb, 12) + 1;
- if (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + fsize96)
- continue;
-
- av_log(avctx, AV_LOG_DEBUG, "X96 extension found at %d bits\n", get_bits_count(&s->gb));
- skip_bits(&s->gb, 12);
- av_log(avctx, AV_LOG_DEBUG, "FSIZE96 = %d bytes\n", fsize96);
- av_log(avctx, AV_LOG_DEBUG, "REVNO = %d\n", get_bits(&s->gb, 4));
-
- s->core_ext_mask |= DCA_EXT_X96;
- break;
- }
- }
+ av_log(avctx, AV_LOG_DEBUG, "X96 extension found at %d bits\n",
+ get_bits_count(&s->gb));
+ skip_bits(&s->gb, 12);
+ av_log(avctx, AV_LOG_DEBUG, "FSIZE96 = %d bytes\n", fsize96);
+ av_log(avctx, AV_LOG_DEBUG, "REVNO = %d\n", get_bits(&s->gb, 4));
- skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
- }
+ s->core_ext_mask |= DCA_EXT_X96;
+ break;
+ }
+ }
+ skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
+ }
} else {
/* no supported extensions, skip the rest of the core substream */
skip_bits_long(&s->gb, core_ss_end - get_bits_count(&s->gb));
@@ -1778,15 +1811,15 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data,
s->profile = FF_PROFILE_DTS_ES;
/* check for ExSS (HD part) */
- if (s->dca_buffer_size - s->frame_size > 32
- && get_bits_long(&s->gb, 32) == DCA_HD_MARKER)
+ if (s->dca_buffer_size - s->frame_size > 32 &&
+ get_bits_long(&s->gb, 32) == DCA_HD_MARKER)
dca_exss_parse_header(s);
avctx->profile = s->profile;
channels = s->prim_channels + !!s->lfe;
- if (s->amode<16) {
+ if (s->amode < 16) {
avctx->channel_layout = dca_core_channel_layout[s->amode];
if (s->xch_present && (!avctx->request_channels ||
@@ -1818,7 +1851,7 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data,
avctx->channel_layout = AV_CH_LAYOUT_STEREO;
}
} else {
- av_log(avctx, AV_LOG_ERROR, "Non standard configuration %d !\n",s->amode);
+ av_log(avctx, AV_LOG_ERROR, "Non standard configuration %d !\n", s->amode);
return AVERROR_INVALIDDATA;
}
@@ -1844,8 +1877,8 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data,
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
- samples_flt = (float *)s->frame.data[0];
- samples_s16 = (int16_t *)s->frame.data[0];
+ samples_flt = (float *) s->frame.data[0];
+ samples_s16 = (int16_t *) s->frame.data[0];
/* filter to get final output */
for (i = 0; i < (s->sample_blocks / 8); i++) {
@@ -1853,10 +1886,10 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data,
/* If this was marked as a DTS-ES stream we need to subtract back- */
/* channel from SL & SR to remove matrixed back-channel signal */
- if((s->source_pcm_res & 1) && s->xch_present) {
- float* back_chan = s->samples + s->channel_order_tab[s->xch_base_channel] * 256;
- float* lt_chan = s->samples + s->channel_order_tab[s->xch_base_channel - 2] * 256;
- float* rt_chan = s->samples + s->channel_order_tab[s->xch_base_channel - 1] * 256;
+ if ((s->source_pcm_res & 1) && s->xch_present) {
+ float *back_chan = s->samples + s->channel_order_tab[s->xch_base_channel] * 256;
+ float *lt_chan = s->samples + s->channel_order_tab[s->xch_base_channel - 2] * 256;
+ float *rt_chan = s->samples + s->channel_order_tab[s->xch_base_channel - 1] * 256;
s->dsp.vector_fmac_scalar(lt_chan, back_chan, -M_SQRT1_2, 256);
s->dsp.vector_fmac_scalar(rt_chan, back_chan, -M_SQRT1_2, 256);
}
@@ -1875,12 +1908,11 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data,
/* update lfe history */
lfe_samples = 2 * s->lfe * (s->sample_blocks / 8);
- for (i = 0; i < 2 * s->lfe * 4; i++) {
+ for (i = 0; i < 2 * s->lfe * 4; i++)
s->lfe_data[i] = s->lfe_data[i + lfe_samples];
- }
- *got_frame_ptr = 1;
- *(AVFrame *)data = s->frame;
+ *got_frame_ptr = 1;
+ *(AVFrame *) data = s->frame;
return buf_size;
}
@@ -1893,7 +1925,7 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data,
* @param avctx pointer to the AVCodecContext
*/
-static av_cold int dca_decode_init(AVCodecContext * avctx)
+static av_cold int dca_decode_init(AVCodecContext *avctx)
{
DCAContext *s = avctx->priv_data;
int i;
@@ -1907,15 +1939,15 @@ static av_cold int dca_decode_init(AVCodecContext * avctx)
ff_dcadsp_init(&s->dcadsp);
ff_fmt_convert_init(&s->fmt_conv, avctx);
- for (i = 0; i < DCA_PRIM_CHANNELS_MAX+1; i++)
+ for (i = 0; i < DCA_PRIM_CHANNELS_MAX + 1; i++)
s->samples_chanptr[i] = s->samples + i * 256;
if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
- s->scale_bias = 1.0 / 32768.0;
+ s->scale_bias = 1.0 / 32768.0;
} else {
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
- s->scale_bias = 1.0;
+ s->scale_bias = 1.0;
}
/* allow downmixing to stereo */
@@ -1930,7 +1962,7 @@ static av_cold int dca_decode_init(AVCodecContext * avctx)
return 0;
}
-static av_cold int dca_decode_end(AVCodecContext * avctx)
+static av_cold int dca_decode_end(AVCodecContext *avctx)
{
DCAContext *s = avctx->priv_data;
ff_mdct_end(&s->imdct);
@@ -1947,17 +1979,17 @@ static const AVProfile profiles[] = {
};
AVCodec ff_dca_decoder = {
- .name = "dca",
- .type = AVMEDIA_TYPE_AUDIO,
- .id = CODEC_ID_DTS,
- .priv_data_size = sizeof(DCAContext),
- .init = dca_decode_init,
- .decode = dca_decode_frame,
- .close = dca_decode_end,
- .long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
- .capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
- .sample_fmts = (const enum AVSampleFormat[]) {
- AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
- },
- .profiles = NULL_IF_CONFIG_SMALL(profiles),
+ .name = "dca",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = CODEC_ID_DTS,
+ .priv_data_size = sizeof(DCAContext),
+ .init = dca_decode_init,
+ .decode = dca_decode_frame,
+ .close = dca_decode_end,
+ .long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
+ .capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT,
+ AV_SAMPLE_FMT_S16,
+ AV_SAMPLE_FMT_NONE },
+ .profiles = NULL_IF_CONFIG_SMALL(profiles),
};
More information about the ffmpeg-cvslog
mailing list