37 #define MIN_LSP_SEP (0.05 / (2.0 * M_PI)) 
   40 #define NB_SUBFRAMES    3 
   41 #define SUBFRAME_SIZE  54 
   42 #define FILTER_ORDER   10 
  195         if (bitrate > **buf) {
 
  199                        "Claimed bitrate and buffer size mismatch.\n");
 
  203         } 
else if (bitrate < **buf) {
 
  205                    "Buffer is too small for the claimed bitrate.\n");
 
  212                "Bitrate byte is missing, guessing the bitrate from packet size.\n");
 
  235     float denom = 2.0 / (2.0 * 8.0 + 1.0);
 
  255     for (i = 0; i < 8; i++) {
 
  256         float tt = ((float)i - 8.0 / 2.0) / 8.0;
 
  258         for (n = -8; n <= 8; n++, idx++) {
 
  259             float arg1 = 
M_PI * 0.9 * (tt - 
n);
 
  260             float arg2 = 
M_PI * (tt - 
n);
 
  289         const float *codebook = codebooks[i];
 
  292             e->
lspf[k++] = codebook[e->
frame.
lsp[i] * row_size + j];
 
  301     for (i = 0, k = 0; i < evrc_lspq_nb_codebooks[e->
bitrate] - 1; i++) {
 
  316                             const float *prev, 
int index)
 
  318     static const float lsp_interpolation_factors[] = { 0.1667, 0.5, 0.8333 };
 
  320                             1.0 - lsp_interpolation_factors[index],
 
  331     static const float d_interpolation_factors[] = { 0, 0.3313, 0.6625, 1, 1 };
 
  332     dst[0] = (1.0 - d_interpolation_factors[
index    ]) * prev
 
  333                   + d_interpolation_factors[index    ]  * current;
 
  334     dst[1] = (1.0 - d_interpolation_factors[index + 1]) * prev
 
  335                   + d_interpolation_factors[index + 1]  * current;
 
  336     dst[2] = (1.0 - d_interpolation_factors[index + 2]) * prev
 
  337                   + d_interpolation_factors[index + 2]  * current;
 
  359         a[0] = k < 2 ? 0.25 : 0;
 
  360         b[0] = k < 2 ? k < 1 ? 0.25 : -0.25 : 0;
 
  363             a[i + 1] = a[i] - 2 * lsp[i * 2    ] * a1[i] + a2[i];
 
  364             b[i + 1] = b[i] - 2 * lsp[i * 2 + 1] * b1[i] + b2[i];
 
  384     t = (offset - delay + 0.5) * 8.0 + 0.5;
 
  392     coef_idx = t * (2 * 8 + 1);
 
  395     for (i = 0; i < 2 * 8 + 1; i++)
 
  405                            const float delay[3], 
int length)
 
  407     float denom, locdelay, dpr, invl;
 
  410     invl = 1.0 / ((float) length);
 
  414     denom = (delay[1] - delay[0]) * invl;
 
  415     for (i = 0; i < dpr; i++) {
 
  416         locdelay = delay[0] + i * denom;
 
  417         bl_intrp(e, excitation + i, locdelay);
 
  420     denom = (delay[2] - delay[1]) * invl;
 
  422     for (i = dpr; i < dpr + 10; i++) {
 
  423         locdelay = delay[1] + (i - dpr) * denom;
 
  424         bl_intrp(e, excitation + i, locdelay);
 
  427     for (i = 0; i < 
length; i++)
 
  428         excitation[i] *= gain;
 
  433     int i, pos1, pos2, 
offset;
 
  435     offset = (fixed_index[3] >> 9) & 3;
 
  437     for (i = 0; i < 3; i++) {
 
  438         pos1 = ((fixed_index[i] & 0x7f) / 11) * 5 + ((i + 
offset) % 5);
 
  439         pos2 = ((fixed_index[i] & 0x7f) % 11) * 5 + ((i + 
offset) % 5);
 
  441         cod[pos1] = (fixed_index[i] & 0x80) ? -1.0 : 1.0;
 
  444             cod[pos2]  = -cod[pos1];
 
  446             cod[pos2] +=  cod[pos1];
 
  449     pos1 = ((fixed_index[3] & 0x7f) / 11) * 5 + ((3 + 
offset) % 5);
 
  450     pos2 = ((fixed_index[3] & 0x7f) % 11) * 5 + ((4 + 
offset) % 5);
 
  452     cod[pos1] = (fixed_index[3] & 0x100) ? -1.0 : 1.0;
 
  453     cod[pos2] = (fixed_index[3] & 0x80 ) ? -1.0 : 1.0;
 
  461     sign = (fixed_index & 0x200) ? -1.0 : 1.0;
 
  463     pos = ((fixed_index        & 0x7) * 7) + 4;
 
  465     pos = (((fixed_index >> 3) & 0x7) * 7) + 2;
 
  467     pos = (((fixed_index >> 6) & 0x7) * 7);
 
  477                            float *excitation, 
float pitch_gain,
 
  478                            int pitch_lag, 
int subframe_size)
 
  487     pitch_gain = av_clipf(pitch_gain, 0.2, 0.9);
 
  489     for (i = pitch_lag; i < subframe_size; i++)
 
  490         excitation[i] += pitch_gain * excitation[i - pitch_lag];
 
  505                              float *memory, 
int buffer_length, 
float *samples)
 
  509     for (i = 0; i < buffer_length; i++) {
 
  512             samples[i] -= filter_coeffs[j] * memory[j];
 
  513             memory[j]   = memory[j - 1];
 
  515         samples[i] -= filter_coeffs[0] * memory[0];
 
  516         memory[0]   = samples[i];
 
  526         coeff[i] = inbuf[i] * fac;
 
  532                             const float *coef, 
float *memory, 
int length)
 
  537     for (i = 0; i < 
length; i++) {
 
  541             sum      += coef[j] * memory[j];
 
  542             memory[j] = memory[j - 1];
 
  544         sum += coef[0] * memory[0];
 
  545         memory[0] = input[i];
 
  559     { 0.0 , 0.0 , 0.0 , 0.0  },
 
  560     { 0.0 , 0.0 , 0.57, 0.57 },
 
  561     { 0.0 , 0.0 , 0.0 , 0.0  },
 
  562     { 0.35, 0.50, 0.50, 0.75 },
 
  563     { 0.20, 0.50, 0.57, 0.75 },
 
  572                        float *
out, 
int idx, 
const struct PfCoeff *pfc,
 
  578     float sum1 = 0.0, sum2 = 0.0, gamma, gain;
 
  579     float tilt = pfc->
tilt;
 
  586     for (i = 0; i < length - 1; i++)
 
  587         sum2 += in[i] * in[i + 1];
 
  591     for (i = 0; i < 
length; i++) {
 
  592         scratch[i] = in[i] - tilt * e->
last;
 
  622             gamma = 
FFMIN(gamma, 1.0);
 
  624             for (i = 0; i < 
length; i++) {
 
  631     memcpy(scratch, temp, length * 
sizeof(
float));
 
  636     for (i = 0, sum1 = 0, sum2 = 0; i < 
length; i++) {
 
  637         sum1 += in[i] * in[i];
 
  638         sum2 += scratch[i] * scratch[i];
 
  640     gain = sum2 ? sqrt(sum1 / sum2) : 1.0;
 
  642     for (i = 0; i < 
length; i++)
 
  660             e->
lspf[i] = e->
prev_lspf[i] * 0.875 + 0.125 * (i + 1) * 0.048;
 
  679         idelay[0] = idelay[1] = idelay[2] = 
MIN_DELAY;
 
  708         pitch_lag = 
lrintf((idelay[1] + idelay[0]) / 2.0);
 
  714             for (j = 0; j < subframe_size; j++)
 
  718             for (j = 0; j < subframe_size; j++)
 
  726             for (j = 0; j < subframe_size; j++)
 
  729             for (j = 0; j < subframe_size; j++)
 
  738         samples += subframe_size;
 
  743                              int *got_frame_ptr, 
AVPacket *avpkt)
 
  748     int buf_size       = avpkt->
size;
 
  751     int   i, j, ret, error_flag = 0;
 
  756     samples = (
float *)frame->
data[0];
 
  782     } 
else if (e->
frame.
lsp[0] == 0xf &&
 
  833         idelay[0] = idelay[1] = idelay[2] = 
MIN_DELAY;
 
  851         pitch_lag = 
lrintf((idelay[1] + idelay[0]) / 2.0);
 
  868                            acb_sum, idelay, subframe_size);
 
  870                            acb_sum, pitch_lag, subframe_size);
 
  873             for (j = 0; j < subframe_size; j++)
 
  877             for (j = 0; j < subframe_size; j++)
 
  890         samples += subframe_size;
 
  907     samples = (
float *)frame->
data[0];
 
  908     for (i = 0; i < 160; i++)
 
  916 #define OFFSET(x) offsetof(EVRCContext, x) 
  917 #define AD AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_DECODING_PARAM 
  940     .priv_class     = &evrcdec_class,
 
Data tables for the EVRC decoder. 
 
This structure describes decoded (raw) audio or video data. 
 
BYTE int const BYTE int int row_size
 
static const AVClass evrcdec_class
 
uint8_t fcb_gain[3]
fixed codebook gain index 
 
uint16_t lsp[4]
index into LSP codebook 
 
ptrdiff_t const GLvoid * data
 
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits. 
 
#define AV_LOG_WARNING
Something somehow does not look correct. 
 
#define LIBAVUTIL_VERSION_INT
 
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. 
 
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. 
 
static const float *const *const evrc_lspq_codebooks[]
 
static void fcb_excitation(EVRCContext *e, const uint16_t *codebook, float *excitation, float pitch_gain, int pitch_lag, int subframe_size)
 
float energy_vector[NB_SUBFRAMES]
 
static av_cold int evrc_decode_init(AVCodecContext *avctx)
Initialize the speech codec according to the specification. 
 
float avg_fcb_gain
average fixed codebook gain 
 
uint8_t warned_buf_mismatch_bitrate
 
uint8_t tty
tty baud rate bit 
 
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
 
static evrc_packet_rate buf_size2bitrate(const int buf_size)
 
enum AVSampleFormat sample_fmt
audio sample format 
 
static void interpolate_lsp(float *ilsp, const float *lsp, const float *prev, int index)
 
uint8_t lpc_flag
spectral change indicator 
 
static void bl_intrp(EVRCContext *e, float *ex, float delay)
 
uint16_t fcb_shape[3][4]
fixed codebook shape 
 
static const uint8_t *const evrc_lspq_codebooks_row_sizes[]
 
static const float evrc_energy_quant[][3]
Rate 1/8 frame energy quantization. 
 
bitstream reader API header. 
 
static const uint8_t subframe_sizes[]
 
float pitch[ACB_SIZE+FILTER_ORDER+SUBFRAME_SIZE]
 
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered. 
 
static const uint8_t evrc_lspq_nb_codebooks[]
 
static void acb_excitation(EVRCContext *e, float *excitation, float gain, const float delay[3], int length)
 
float interpolation_coeffs[136]
 
float postfilter_residual[ACB_SIZE+SUBFRAME_SIZE]
 
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
 
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers. 
 
static void unpack_frame(EVRCContext *e)
Frame unpacking for RATE_FULL, RATE_HALF and RATE_QUANT. 
 
const char * name
Name of the codec implementation. 
 
static int evrc_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
 
static const uint8_t offset[127][2]
 
uint8_t acb_gain[3]
adaptive codebook gain 
 
static const float pitch_gain_vq[]
 
uint64_t channel_layout
Audio channel layout. 
 
float avg_acb_gain
average adaptive codebook gain 
 
float pitch_back[ACB_SIZE]
 
float prev_lspf[FILTER_ORDER]
 
static void postfilter(EVRCContext *e, float *in, const float *coeff, float *out, int idx, const struct PfCoeff *pfc, int length)
 
float postfilter_fir[FILTER_ORDER]
 
static void bandwidth_expansion(float *coeff, const float *inbuf, float gamma)
 
uint8_t pitch_delay
pitch delay for entire frame 
 
static void decode_8_pulses_35bits(const uint16_t *fixed_index, float *cod)
 
uint8_t delay_diff
delay difference for entire frame 
 
Libavcodec external API header. 
 
static const float estimation_delay[]
 
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext. 
 
static const struct PfCoeff postfilter_coeffs[5]
 
main external API structure. 
 
static void decode_3_pulses_10bits(uint16_t fixed_index, float *cod)
 
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame. 
 
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method!=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt)>2){ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc){av_free(ac);returnNULL;}returnac;}in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> in
 
static unsigned int get_bits1(GetBitContext *s)
 
Describe the class of an AVClass context structure. 
 
float synthesis[FILTER_ORDER]
 
static const AVOption options[]
 
void ff_acelp_lsf2lspd(double *lsp, const float *lsf, int lp_order)
Floating point version of ff_acelp_lsf2lsp() 
 
static void synthesis_filter(const float *in, const float *filter_coeffs, float *memory, int buffer_length, float *samples)
Synthesis of the decoder output signal. 
 
static void decode_predictor_coeffs(const float *ilspf, float *ilpc)
 
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes. 
 
evrc_packet_rate last_valid_bitrate
 
static void warn_insufficient_frame_quality(AVCodecContext *avctx, const char *message)
 
common internal api header. 
 
int channels
number of audio channels 
 
static int decode_lspf(EVRCContext *e)
Decode the 10 vector quantized line spectral pair frequencies from the LSP transmission codes of any ...
 
static const double coeff[2][5]
 
static void frame_erasure(EVRCContext *e, float *samples)
 
uint8_t energy_gain
frame energy gain index 
 
int frame_number
Frame counter, set by libavcodec. 
 
static void residual_filter(float *output, const float *input, const float *coef, float *memory, int length)
 
float postfilter_iir[FILTER_ORDER]
 
#define AV_CH_LAYOUT_MONO
 
This structure stores compressed data. 
 
static void interpolate_delay(float *dst, float current, float prev, int index)
 
int nb_samples
number of audio samples (per channel) described by this frame 
 
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators. 
 
EVRC-A unpacked data frame.