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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)
311 const uint8_t *buf = avpkt->
data;
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);
328 frame->nb_samples = iterations * 240;
335 for(j = 0; j < iterations; j++) {
342 for(
i = 0;
i < 4;
i++) {
359 .
name =
"truespeech",
#define FF_CODEC_CAP_INIT_THREADSAFE
The codec does not modify any global variables in the init function, allowing to call the init functi...
uint64_t channel_layout
Audio channel layout.
static av_cold int truespeech_decode_init(AVCodecContext *avctx)
#define AV_CH_LAYOUT_MONO
static unsigned int get_bits_long(GetBitContext *s, int n)
Read 0-32 bits.
int flag
1-bit flag, shows how to choose filters
This structure describes decoded (raw) audio or video data.
int pulseoff[4]
4-bit offset of pulse values block
filter_frame For filters that do not use the this method is called when a frame is pushed to the filter s input It can be called at any time except in a reentrant way If the input frame is enough to produce then the filter should push the output frames on the output link immediately As an exception to the previous rule if the input frame is enough to produce several output frames then the filter needs output only at least one per link The additional frames can be left buffered in the filter
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
int offset2[4]
7-bit value, encodes offsets for copying and for two-point filter
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
int pulsepos[4]
27-bit variable, encodes 7 pulse positions
void(* bswap_buf)(uint32_t *dst, const uint32_t *src, int w)
static void truespeech_read_frame(TSContext *dec, const uint8_t *input)
static void truespeech_update_filters(TSContext *dec, int16_t *out, int quart)
av_cold void ff_bswapdsp_init(BswapDSPContext *c)
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
static const int16_t ts_decay_35_64[8]
static unsigned int get_bits1(GetBitContext *s)
const AVCodec ff_truespeech_decoder
static const int16_t ts_decay_3_4[8]
static const int16_t ts_pulse_scales[64]
static void truespeech_place_pulses(TSContext *dec, int16_t *out, int quart)
Undefined Behavior In the C some operations are like signed integer dereferencing freed accessing outside allocated Undefined Behavior must not occur in a C it is not safe even if the output of undefined operations is unused The unsafety may seem nit picking but Optimizing compilers have in fact optimized code on the assumption that no undefined Behavior occurs Optimizing code based on wrong assumptions can and has in some cases lead to effects beyond the output of computations The signed integer overflow problem in speed critical code Code which is highly optimized and works with signed integers sometimes has the problem that often the output of the computation does not c
int offset1[2]
8-bit value, used in one copying offset
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
enum AVSampleFormat sample_fmt
audio sample format
int16_t vector[8]
input vector: 5/5/4/4/4/3/3/3
and forward the test the status of outputs and forward it to the corresponding return FFERROR_NOT_READY If the filters stores internally one or a few frame for some input
int channels
number of audio channels
#define DECLARE_ALIGNED(n, t, v)
static const int16_t ts_decay_994_1000[8]
static const int16_t ts_order2_coeffs[25 *2]
#define i(width, name, range_min, range_max)
TrueSpeech decoder context.
@ AV_SAMPLE_FMT_S16
signed 16 bits
const char * name
Name of the codec implementation.
static void truespeech_filters_merge(TSContext *dec)
static const int16_t *const ts_codebook[8]
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return or at least make progress towards producing a frame
main external API structure.
static void truespeech_save_prevvec(TSContext *c)
Filter the word “frame” indicates either a video frame or a group of audio samples
#define avpriv_request_sample(...)
This structure stores compressed data.
int pulseval[4]
7x2-bit pulse values
static void truespeech_synth(TSContext *dec, int16_t *out, int quart)
static const int16_t ts_pulse_values[120]
static int truespeech_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
static void truespeech_correlate_filter(TSContext *dec)
static void truespeech_apply_twopoint_filter(TSContext *dec, int quart)