133 for(
i = 0;
i < 8;
i++){
136 for(j = 0; j <
i; j++)
141 for(
i = 0;
i < 8;
i++)
152 for(
i = 0;
i < 8;
i++){
157 for(
i = 0;
i < 8;
i++){
162 for(
i = 0;
i < 8;
i++){
170 int16_t
tmp[146 + 60], *ptr0, *ptr1;
179 for(
i = 0;
i < 146;
i++)
181 off = (t / 25) + dec->
offset1[quart >> 1] + 18;
183 ptr0 =
tmp + 145 - off;
186 for(
i = 0;
i < 60;
i++){
187 t = (ptr0[0] *
filter[0] + ptr0[1] *
filter[1] + 0x2000) >> 14;
202 memset(
out, 0, 60 *
sizeof(*
out));
203 for(
i = 0;
i < 7;
i++) {
212 for(
i = 0, j = 3; (
i < 30) && (j > 0);
i++){
222 coef = dec->
pulsepos[quart] & 0x7FFF;
224 for(
i = 30, j = 4; (
i < 60) && (j > 0);
i++){
242 for(
i = 0;
i < 60;
i++){
252 int16_t *ptr0, *ptr1;
255 ptr1 = dec->
filters + quart * 8;
256 for(
i = 0;
i < 60;
i++){
258 for(k = 0; k < 8; k++)
259 sum += ptr0[k] * (
unsigned)ptr1[k];
260 sum =
out[
i] + ((
int)(sum + 0x800U) >> 12);
262 for(k = 7; k > 0; k--)
263 ptr0[k] = ptr0[k - 1];
267 for(
i = 0;
i < 8;
i++)
271 for(
i = 0;
i < 60;
i++){
273 for(k = 0; k < 8; k++)
274 sum += ptr0[k] * t[k];
275 for(k = 7; k > 0; k--)
276 ptr0[k] = ptr0[k - 1];
278 out[
i] += (- sum) >> 12;
281 for(
i = 0;
i < 8;
i++)
285 for(
i = 0;
i < 60;
i++){
286 int sum =
out[
i] * (1 << 12);
287 for(k = 0; k < 8; k++)
288 sum += ptr0[k] * t[k];
289 for(k = 7; k > 0; k--)
290 ptr0[k] = ptr0[k - 1];
291 ptr0[0] =
av_clip((sum + 0x800) >> 12, -0x7FFE, 0x7FFE);
293 sum = ((ptr0[1] * (dec->
filtval - (dec->
filtval >> 2))) >> 4) + sum;
294 sum = sum - (sum >> 3);
295 out[
i] =
av_clip((sum + 0x800) >> 12, -0x7FFE, 0x7FFE);
303 for(
i = 0;
i < 8;
i++)
304 c->prevfilt[
i] =
c->cvector[
i];
308 int *got_frame_ptr,
AVPacket *avpkt)
312 int buf_size = avpkt->
size;
319 iterations = buf_size / 32;
323 "Too small input buffer (%d bytes), need at least 32 bytes\n", buf_size);
333 memset(samples, 0, iterations * 240 *
sizeof(*samples));
335 for(j = 0; j < iterations; j++) {
342 for(
i = 0;
i < 4;
i++) {
359 .
name =
"truespeech",
Libavcodec external API header.
static av_cold int init(AVCodecContext *avctx)
static av_always_inline void filter(int16_t *output, ptrdiff_t out_stride, const int16_t *low, ptrdiff_t low_stride, const int16_t *high, ptrdiff_t high_stride, int len, int clip)
audio channel layout utility functions
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
bitstream reader API header.
static unsigned int get_bits_long(GetBitContext *s, int n)
Read 0-32 bits.
static unsigned int get_bits1(GetBitContext *s)
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
#define AV_CH_LAYOUT_MONO
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
#define DECLARE_ALIGNED(n, t, v)
Declare a variable that is aligned in memory.
@ AV_SAMPLE_FMT_S16
signed 16 bits
av_cold void ff_bswapdsp_init(BswapDSPContext *c)
common internal API header
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
main external API structure.
enum AVSampleFormat sample_fmt
audio sample format
int channels
number of audio channels
uint64_t channel_layout
Audio channel layout.
const char * name
Name of the codec implementation.
This structure describes decoded (raw) audio or video data.
int nb_samples
number of audio samples (per channel) described by this frame
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
This structure stores compressed data.
void(* bswap_buf)(uint32_t *dst, const uint32_t *src, int w)
TrueSpeech decoder context.
int offset1[2]
8-bit value, used in one copying offset
int offset2[4]
7-bit value, encodes offsets for copying and for two-point filter
int flag
1-bit flag, shows how to choose filters
int pulseval[4]
7x2-bit pulse values
int pulseoff[4]
4-bit offset of pulse values block
int16_t vector[8]
input vector: 5/5/4/4/4/3/3/3
int pulsepos[4]
27-bit variable, encodes 7 pulse positions
#define avpriv_request_sample(...)
static void truespeech_apply_twopoint_filter(TSContext *dec, int quart)
static av_cold int truespeech_decode_init(AVCodecContext *avctx)
AVCodec ff_truespeech_decoder
static void truespeech_save_prevvec(TSContext *c)
static int truespeech_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
static void truespeech_place_pulses(TSContext *dec, int16_t *out, int quart)
static void truespeech_correlate_filter(TSContext *dec)
static void truespeech_synth(TSContext *dec, int16_t *out, int quart)
static void truespeech_read_frame(TSContext *dec, const uint8_t *input)
static void truespeech_filters_merge(TSContext *dec)
static void truespeech_update_filters(TSContext *dec, int16_t *out, int quart)
static const int16_t ts_order2_coeffs[25 *2]
static const int16_t *const ts_codebook[8]
static const int16_t ts_decay_35_64[8]
static const int16_t ts_pulse_values[120]
static const int16_t ts_decay_3_4[8]
static const int16_t ts_pulse_scales[64]
static const int16_t ts_decay_994_1000[8]