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134 for(
i = 0;
i < 8;
i++){
137 for(j = 0; j <
i; j++)
142 for(
i = 0;
i < 8;
i++)
153 for(
i = 0;
i < 8;
i++){
158 for(
i = 0;
i < 8;
i++){
163 for(
i = 0;
i < 8;
i++){
171 int16_t
tmp[146 + 60], *ptr0, *ptr1;
180 for(
i = 0;
i < 146;
i++)
182 off = (t / 25) + dec->
offset1[quart >> 1] + 18;
184 ptr0 =
tmp + 145 - off;
187 for(
i = 0;
i < 60;
i++){
188 t = (ptr0[0] *
filter[0] + ptr0[1] *
filter[1] + 0x2000) >> 14;
203 memset(
out, 0, 60 *
sizeof(*
out));
204 for(
i = 0;
i < 7;
i++) {
213 for(
i = 0, j = 3; (
i < 30) && (j > 0);
i++){
223 coef = dec->
pulsepos[quart] & 0x7FFF;
225 for(
i = 30, j = 4; (
i < 60) && (j > 0);
i++){
243 for(
i = 0;
i < 60;
i++){
253 int16_t *ptr0, *ptr1;
256 ptr1 = dec->
filters + quart * 8;
257 for(
i = 0;
i < 60;
i++){
259 for(k = 0; k < 8; k++)
260 sum += ptr0[k] * (
unsigned)ptr1[k];
261 sum =
out[
i] + ((int)(sum + 0x800U) >> 12);
263 for(k = 7; k > 0; k--)
264 ptr0[k] = ptr0[k - 1];
268 for(
i = 0;
i < 8;
i++)
272 for(
i = 0;
i < 60;
i++){
274 for(k = 0; k < 8; k++)
275 sum += ptr0[k] * t[k];
276 for(k = 7; k > 0; k--)
277 ptr0[k] = ptr0[k - 1];
279 out[
i] += (- sum) >> 12;
282 for(
i = 0;
i < 8;
i++)
286 for(
i = 0;
i < 60;
i++){
287 int sum =
out[
i] * (1 << 12);
288 for(k = 0; k < 8; k++)
289 sum += ptr0[k] * t[k];
290 for(k = 7; k > 0; k--)
291 ptr0[k] = ptr0[k - 1];
292 ptr0[0] =
av_clip((sum + 0x800) >> 12, -0x7FFE, 0x7FFE);
294 sum = ((ptr0[1] * (dec->
filtval - (dec->
filtval >> 2))) >> 4) + sum;
295 sum = sum - (sum >> 3);
296 out[
i] =
av_clip((sum + 0x800) >> 12, -0x7FFE, 0x7FFE);
304 for(
i = 0;
i < 8;
i++)
305 c->prevfilt[
i] =
c->cvector[
i];
309 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 .
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)
int pulseoff[4]
4-bit offset of pulse values block
void(* filter)(uint8_t *src, int stride, int qscale)
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)
#define CODEC_LONG_NAME(str)
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
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.
int(* init)(AVBSFContext *ctx)
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
An AVChannelLayout holds information about the channel layout of audio data.
#define DECLARE_ALIGNED(n, t, v)
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
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)
void av_channel_layout_uninit(AVChannelLayout *channel_layout)
Free any allocated data in the channel layout and reset the channel count to 0.
Filter the word “frame” indicates either a video frame or a group of audio samples
#define avpriv_request_sample(...)
#define AV_CHANNEL_LAYOUT_MONO
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)