36 for (i = 0; i < 13; i++) {
44 return (
int)(a * (
SUINT)b + (1 << 14)) >> 15;
50 const int16_t *
src = dst - lag;
52 for (i = 0; i < 40; i++)
65 int abs =
FFABS(filtered);
66 if (abs < 11059) abs <<= 1;
67 else if (abs < 20070) abs += 11059;
68 else abs = (abs >> 2) + 26112;
69 return filtered < 0 ? -abs : abs;
75 for (i = 7; i >= 0; i--) {
77 v[i + 1] = v[i] +
gsm_mult(rrp[i], in);
89 for (i = 0; i < 8; i++)
90 rrp[i] =
get_rrp((lar_prev[i] >> 2) + (lar_prev[i] >> 1) + (lar[i] >> 2));
91 for (i = 0; i < 13; i++)
94 for (i = 0; i < 8; i++)
95 rrp[i] =
get_rrp((lar_prev[i] >> 1) + (lar [i] >> 1));
96 for (i = 13; i < 27; i++)
99 for (i = 0; i < 8; i++)
100 rrp[i] =
get_rrp((lar_prev[i] >> 2) + (lar [i] >> 1) + (lar[i] >> 2));
101 for (i = 27; i < 40; i++)
104 for (i = 0; i < 8; i++)
106 for (i = 40; i < 160; i++)
115 for (i = 0; i < 160; i++) {
116 msr = av_clip_int16(data[i] +
gsm_mult(msr, 28180));
117 data[i] = av_clip_int16(msr * 2) & ~7;
127 int16_t *ref_dst = ctx->
ref_buf + 120;
138 for (i = 0; i < 4; i++) {
142 lag = av_clip(lag, 40, 120);
const char const char void * val
static void short_term_synth(GSMContext *ctx, int16_t *dst, const int16_t *src)
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
static av_noinline int get_rrp(int filtered)
static int gsm_mult(int a, int b)
static int decode_log_area(int coded, int factor, int offset)
bitstream reader API header.
static int gsm_decode_block(AVCodecContext *avctx, int16_t *samples, GetBitContext *gb, int mode)
static const uint8_t offset[127][2]
const int16_t ff_gsm_dequant_tab[64][8]
static void long_term_synth(int16_t *dst, int lag, int gain_idx)
static int filter_value(int in, int rrp[8], int v[9])
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
static int postprocess(int16_t *data, int msr)
main external API structure.
const uint16_t ff_gsm_long_term_gain_tab[4]
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) #define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac) { } void ff_audio_convert_free(AudioConvert **ac) { if(! *ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);} AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map) { AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;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);return NULL;} return ac;} 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;} else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->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);return ac;} int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in) { int use_generic=1;int len=in->nb_samples;int p;if(ac->dc) { av_log(ac->avr, AV_LOG_TRACE, "%d samples - audio_convert: %s to %s (dithered)\", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
static void apcm_dequant_add(GetBitContext *gb, int16_t *dst, const int *frame_bits)
static const int factor[16]
const int *const ff_gsm_apcm_bits[][4]
static const struct twinvq_data tab
mode
Use these values in ebur128_init (or'ed).
const uint8_t ff_gsm_requant_tab[4][8]