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39 #define MIN_LSP_SEP (0.05 / (2.0 * M_PI))
42 #define NB_SUBFRAMES 3
43 #define SUBFRAME_SIZE 54
44 #define FILTER_ORDER 10
201 "Claimed bitrate and buffer size mismatch.\n");
207 "Buffer is too small for the claimed bitrate.\n");
214 "Bitrate byte is missing, guessing the bitrate from packet size.\n");
237 float denom = 2.0 / (2.0 * 8.0 + 1.0);
258 float tt = ((
float)
i - 8.0 / 2.0) / 8.0;
260 for (n = -8; n <= 8; n++, idx++) {
261 float arg1 =
M_PI * 0.9 * (tt - n);
262 float arg2 =
M_PI * (tt - n);
293 for (j = 0; j < row_size; j++)
318 const float *prev,
int index)
320 static const float lsp_interpolation_factors[] = { 0.1667, 0.5, 0.8333 };
322 1.0 - lsp_interpolation_factors[
index],
333 static const float d_interpolation_factors[] = { 0, 0.3313, 0.6625, 1, 1 };
334 dst[0] = (1.0 - d_interpolation_factors[
index ]) * prev
335 + d_interpolation_factors[
index ] * current;
336 dst[1] = (1.0 - d_interpolation_factors[
index + 1]) * prev
337 + d_interpolation_factors[
index + 1] * current;
338 dst[2] = (1.0 - d_interpolation_factors[
index + 2]) * prev
339 + d_interpolation_factors[
index + 2] * current;
361 a[0] = k < 2 ? 0.25 : 0;
362 b[0] = k < 2 ? k < 1 ? 0.25 : -0.25 : 0;
366 b[
i + 1] =
b[
i] - 2 * lsp[
i * 2 + 1] *
b1[
i] +
b2[
i];
386 t = (
offset - delay + 0.5) * 8.0 + 0.5;
394 coef_idx = t * (2 * 8 + 1);
397 for (
i = 0;
i < 2 * 8 + 1;
i++)
407 const float delay[3],
int length)
409 float denom, locdelay, dpr, invl;
412 invl = 1.0 / ((
float) length);
416 denom = (delay[1] - delay[0]) * invl;
417 for (
i = 0;
i < dpr;
i++) {
418 locdelay = delay[0] +
i * denom;
422 denom = (delay[2] - delay[1]) * invl;
424 for (
i = dpr;
i < dpr + 10;
i++) {
425 locdelay = delay[1] + (
i - dpr) * denom;
429 for (
i = 0;
i < length;
i++)
430 excitation[
i] *= gain;
437 offset = (fixed_index[3] >> 9) & 3;
439 for (
i = 0;
i < 3;
i++) {
440 pos1 = ((fixed_index[
i] & 0x7f) / 11) * 5 + ((
i +
offset) % 5);
441 pos2 = ((fixed_index[
i] & 0x7f) % 11) * 5 + ((
i +
offset) % 5);
443 cod[pos1] = (fixed_index[
i] & 0x80) ? -1.0 : 1.0;
446 cod[pos2] = -cod[pos1];
448 cod[pos2] += cod[pos1];
451 pos1 = ((fixed_index[3] & 0x7f) / 11) * 5 + ((3 +
offset) % 5);
452 pos2 = ((fixed_index[3] & 0x7f) % 11) * 5 + ((4 +
offset) % 5);
454 cod[pos1] = (fixed_index[3] & 0x100) ? -1.0 : 1.0;
455 cod[pos2] = (fixed_index[3] & 0x80 ) ? -1.0 : 1.0;
463 sign = (fixed_index & 0x200) ? -1.0 : 1.0;
465 pos = ((fixed_index & 0x7) * 7) + 4;
467 pos = (((fixed_index >> 3) & 0x7) * 7) + 2;
469 pos = (((fixed_index >> 6) & 0x7) * 7);
479 float *excitation,
float pitch_gain,
480 int pitch_lag,
int subframe_size)
489 pitch_gain =
av_clipf(pitch_gain, 0.2, 0.9);
491 for (
i = pitch_lag;
i < subframe_size;
i++)
492 excitation[
i] += pitch_gain * excitation[
i - pitch_lag];
507 float *memory,
int buffer_length,
float *
samples)
511 for (
i = 0;
i < buffer_length;
i++) {
514 samples[
i] -= filter_coeffs[j] * memory[j];
515 memory[j] = memory[j - 1];
517 samples[
i] -= filter_coeffs[0] * memory[0];
534 const float *coef,
float *memory,
int length)
539 for (
i = 0;
i < length;
i++) {
543 sum += coef[j] * memory[j];
544 memory[j] = memory[j - 1];
546 sum += coef[0] * memory[0];
561 { 0.0 , 0.0 , 0.0 , 0.0 },
562 { 0.0 , 0.0 , 0.57, 0.57 },
563 { 0.0 , 0.0 , 0.0 , 0.0 },
564 { 0.35, 0.50, 0.50, 0.75 },
565 { 0.20, 0.50, 0.57, 0.75 },
574 float *
out,
int idx,
const struct PfCoeff *pfc,
580 float sum1 = 0.0, sum2 = 0.0, gamma, gain;
581 float tilt = pfc->
tilt;
588 for (
i = 0;
i < length - 1;
i++)
589 sum2 += in[
i] * in[
i + 1];
593 for (
i = 0;
i < length;
i++) {
594 scratch[
i] = in[
i] - tilt * e->
last;
624 gamma =
FFMIN(gamma, 1.0);
626 for (
i = 0;
i < length;
i++) {
633 memcpy(scratch,
temp, length *
sizeof(
float));
638 for (
i = 0, sum1 = 0, sum2 = 0;
i < length;
i++) {
639 sum1 += in[
i] * in[
i];
640 sum2 += scratch[
i] * scratch[
i];
642 gain = sum2 ? sqrt(sum1 / sum2) : 1.0;
644 for (
i = 0;
i < length;
i++)
681 idelay[0] = idelay[1] = idelay[2] =
MIN_DELAY;
710 pitch_lag =
lrintf((idelay[1] + idelay[0]) / 2.0);
716 for (j = 0; j < subframe_size; j++)
720 for (j = 0; j < subframe_size; j++)
728 for (j = 0; j < subframe_size; j++)
731 for (j = 0; j < subframe_size; j++)
745 int *got_frame_ptr,
AVPacket *avpkt)
747 const uint8_t *buf = avpkt->
data;
749 int buf_size = avpkt->
size;
752 int i, j,
ret, error_flag = 0;
776 uint8_t *p = (uint8_t *) &e->
frame;
834 idelay[0] = idelay[1] = idelay[2] =
MIN_DELAY;
852 pitch_lag =
lrintf((idelay[1] + idelay[0]) / 2.0);
869 acb_sum, idelay, subframe_size);
871 acb_sum, pitch_lag, subframe_size);
874 for (j = 0; j < subframe_size; j++)
878 for (j = 0; j < subframe_size; j++)
917 #define OFFSET(x) offsetof(EVRCContext, x)
918 #define AD AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_DECODING_PARAM
uint16_t lsp[4]
index into LSP codebook
static evrc_packet_rate determine_bitrate(AVCodecContext *avctx, int *buf_size, const uint8_t **buf)
Determine the bitrate from the frame size and/or the first byte of the frame.
#define AV_LOG_WARNING
Something somehow does not look correct.
static void interpolate_delay(float *dst, float current, float prev, int index)
static int decode_lspf(EVRCContext *e)
Decode the 10 vector quantized line spectral pair frequencies from the LSP transmission codes of any ...
static void fcb_excitation(EVRCContext *e, const uint16_t *codebook, float *excitation, float pitch_gain, int pitch_lag, int subframe_size)
static evrc_packet_rate buf_size2bitrate(const int buf_size)
static void residual_filter(float *output, const float *input, const float *coef, float *memory, int length)
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 output
static const float *const *const evrc_lspq_codebooks[]
This structure describes decoded (raw) audio or video data.
static int evrc_decode_frame(AVCodecContext *avctx, AVFrame *frame, int *got_frame_ptr, AVPacket *avpkt)
static const uint8_t codebooks[]
float pitch_back[ACB_SIZE]
static const float evrc_energy_quant[][3]
Rate 1/8 frame energy quantization.
float pitch[ACB_SIZE+FILTER_ORDER+SUBFRAME_SIZE]
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
uint8_t fcb_gain[3]
fixed codebook gain index
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
uint8_t tty
tty baud rate bit
static const struct PfCoeff postfilter_coeffs[5]
AVCodec p
The public AVCodec.
static double b1(void *priv, double x, double y)
AVChannelLayout ch_layout
Audio channel layout.
static const uint8_t subframe_sizes[]
static void unpack_frame(EVRCContext *e)
Frame unpacking for RATE_FULL, RATE_HALF and RATE_QUANT.
evrc_packet_rate last_valid_bitrate
static void interpolate_lsp(float *ilsp, const float *lsp, const float *prev, int index)
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
static void acb_excitation(EVRCContext *e, float *excitation, float gain, const float delay[3], int length)
uint16_t fcb_shape[3][4]
fixed codebook shape
uint8_t warned_buf_mismatch_bitrate
#define FF_CODEC_DECODE_CB(func)
static const uint8_t *const evrc_lspq_codebooks_row_sizes[]
void ff_acelp_lsf2lspd(double *lsp, const float *lsf, int lp_order)
Floating point version of ff_acelp_lsf2lsp()
uint8_t acb_gain[3]
adaptive codebook gain
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
float avg_fcb_gain
average fixed codebook gain
uint8_t lpc_flag
spectral change indicator
#define CODEC_LONG_NAME(str)
float postfilter_fir[FILTER_ORDER]
static void bl_intrp(EVRCContext *e, float *ex, float delay)
float prev_lspf[FILTER_ORDER]
float postfilter_iir[FILTER_ORDER]
#define LIBAVUTIL_VERSION_INT
uint8_t delay_diff
delay difference for entire frame
static void bandwidth_expansion(float *coeff, const float *inbuf, float gamma)
static void decode_8_pulses_35bits(const uint16_t *fixed_index, float *cod)
Describe the class of an AVClass context structure.
static __device__ float fabs(float a)
uint8_t pitch_delay
pitch delay for entire frame
const char * av_default_item_name(void *ptr)
Return the context name.
static unsigned int get_bits1(GetBitContext *s)
float interpolation_coeffs[136]
static const float estimation_delay[]
static void decode_3_pulses_10bits(uint16_t fixed_index, float *cod)
#define AV_CODEC_CAP_CHANNEL_CONF
Codec should fill in channel configuration and samplerate instead of container.
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.
An AVChannelLayout holds information about the channel layout of audio data.
EVRC-A unpacked data frame.
enum AVSampleFormat sample_fmt
audio sample format
static double b2(void *priv, double x, double y)
The reader does not expect b to be semantically here and if the code is changed by maybe adding a a division or other the signedness will almost certainly be mistaken To avoid this confusion a new type was SUINT is the C unsigned type but it holds a signed int to use the same example SUINT a
it s the only field you need to keep assuming you have a context There is some magic you don t need to care about around this just let it vf offset
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 AVOption options[]
float postfilter_residual[ACB_SIZE+SUBFRAME_SIZE]
static void synthesis_filter(const float *in, const float *filter_coeffs, float *memory, int buffer_length, float *samples)
Synthesis of the decoder output signal.
int nb_samples
number of audio samples (per channel) described by this frame
#define i(width, name, range_min, range_max)
#define xf(width, name, var, range_min, range_max, subs,...)
static void postfilter(EVRCContext *e, float *in, const float *coeff, float *out, int idx, const struct PfCoeff *pfc, int length)
void ff_weighted_vector_sumf(float *out, const float *in_a, const float *in_b, float weight_coeff_a, float weight_coeff_b, int length)
float implementation of weighted sum of two vectors.
const char * name
Name of the codec implementation.
float avg_acb_gain
average adaptive codebook gain
static void warn_insufficient_frame_quality(AVCodecContext *avctx, const char *message)
int64_t frame_num
Frame counter, set by libavcodec.
float energy_vector[NB_SUBFRAMES]
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
main external API structure.
void av_channel_layout_uninit(AVChannelLayout *channel_layout)
Free any allocated data in the channel layout and reset the channel count to 0.
static void frame_erasure(EVRCContext *e, float *samples)
Filter the word “frame” indicates either a video frame or a group of audio samples
uint8_t energy_gain
frame energy gain index
static const uint8_t evrc_lspq_nb_codebooks[]
#define AV_CHANNEL_LAYOUT_MONO
const FFCodec ff_evrc_decoder
This structure stores compressed data.
static const double coeff[2][5]
static void decode_predictor_coeffs(const float *ilspf, float *ilpc)
static const float pitch_gain_vq[]
float synthesis[FILTER_ORDER]
static const unsigned codebook[256][2]
static const AVClass evrcdec_class
static av_cold int evrc_decode_init(AVCodecContext *avctx)
Initialize the speech codec according to the specification.