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135 for(
i = 0;
i < 8;
i++){
138 for(j = 0; j <
i; j++)
143 for(
i = 0;
i < 8;
i++)
154 for(
i = 0;
i < 8;
i++){
159 for(
i = 0;
i < 8;
i++){
164 for(
i = 0;
i < 8;
i++){
172 int16_t
tmp[146 + 60], *ptr0, *ptr1;
181 for(
i = 0;
i < 146;
i++)
183 off = (t / 25) + dec->
offset1[quart >> 1] + 18;
185 ptr0 =
tmp + 145 - off;
188 for(
i = 0;
i < 60;
i++){
189 t = (ptr0[0] *
filter[0] + ptr0[1] *
filter[1] + 0x2000) >> 14;
204 memset(
out, 0, 60 *
sizeof(*
out));
205 for(
i = 0;
i < 7;
i++) {
214 for(
i = 0, j = 3; (
i < 30) && (j > 0);
i++){
224 coef = dec->
pulsepos[quart] & 0x7FFF;
226 for(
i = 30, j = 4; (
i < 60) && (j > 0);
i++){
244 for(
i = 0;
i < 60;
i++){
254 int16_t *ptr0, *ptr1;
257 ptr1 = dec->
filters + quart * 8;
258 for(
i = 0;
i < 60;
i++){
260 for(k = 0; k < 8; k++)
261 sum += ptr0[k] * (
unsigned)ptr1[k];
262 sum =
out[
i] + ((
int)(sum + 0x800U) >> 12);
264 for(k = 7; k > 0; k--)
265 ptr0[k] = ptr0[k - 1];
269 for(
i = 0;
i < 8;
i++)
273 for(
i = 0;
i < 60;
i++){
275 for(k = 0; k < 8; k++)
276 sum += ptr0[k] * t[k];
277 for(k = 7; k > 0; k--)
278 ptr0[k] = ptr0[k - 1];
280 out[
i] += (- sum) >> 12;
283 for(
i = 0;
i < 8;
i++)
287 for(
i = 0;
i < 60;
i++){
288 int sum =
out[
i] * (1 << 12);
289 for(k = 0; k < 8; k++)
290 sum += ptr0[k] * t[k];
291 for(k = 7; k > 0; k--)
292 ptr0[k] = ptr0[k - 1];
293 ptr0[0] =
av_clip((sum + 0x800) >> 12, -0x7FFE, 0x7FFE);
295 sum = ((ptr0[1] * (dec->
filtval - (dec->
filtval >> 2))) >> 4) + sum;
296 sum = sum - (sum >> 3);
297 out[
i] =
av_clip((sum + 0x800) >> 12, -0x7FFE, 0x7FFE);
305 for(
i = 0;
i < 8;
i++)
306 c->prevfilt[
i] =
c->cvector[
i];
310 int *got_frame_ptr,
AVPacket *avpkt)
312 const uint8_t *buf = avpkt->
data;
313 int buf_size = avpkt->
size;
320 iterations = buf_size / 32;
324 "Too small input buffer (%d bytes), need at least 32 bytes\n", buf_size);
329 frame->nb_samples = iterations * 240;
336 for(j = 0; j < iterations; j++) {
343 for(
i = 0;
i < 4;
i++) {
360 .
p.
name =
"truespeech",
static av_cold int truespeech_decode_init(AVCodecContext *avctx)
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.
static int truespeech_decode_frame(AVCodecContext *avctx, AVFrame *frame, int *got_frame_ptr, AVPacket *avpkt)
#define AV_CHANNEL_LAYOUT_MONO
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
int nb_channels
Number of channels in this layout.
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.
AVCodec p
The public AVCodec.
AVChannelLayout ch_layout
Audio channel layout.
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.
#define FF_CODEC_DECODE_CB(func)
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)
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
void av_channel_layout_uninit(AVChannelLayout *channel_layout)
Free any allocated data in the channel layout and reset the channel count to 0.
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.
An AVChannelLayout holds information about the channel layout of audio data.
enum AVSampleFormat sample_fmt
audio sample format
const FFCodec ff_truespeech_decoder
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
#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)
#define FF_CODEC_CAP_INIT_THREADSAFE
The codec does not modify any global variables in the init function, allowing to call the init functi...
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 void truespeech_correlate_filter(TSContext *dec)
static void truespeech_apply_twopoint_filter(TSContext *dec, int quart)