diff options
Diffstat (limited to 'gr-vocoder/lib/codec2')
| -rw-r--r-- | gr-vocoder/lib/codec2/c2sim.c | 2 | ||||
| -rw-r--r-- | gr-vocoder/lib/codec2/codec2.c | 2 | ||||
| -rw-r--r-- | gr-vocoder/lib/codec2/fdmdv.c | 4 | ||||
| -rw-r--r-- | gr-vocoder/lib/codec2/interp.c | 4 | ||||
| -rw-r--r-- | gr-vocoder/lib/codec2/kiss_fft.c | 2 | ||||
| -rw-r--r-- | gr-vocoder/lib/codec2/lpc.c | 2 | ||||
| -rw-r--r-- | gr-vocoder/lib/codec2/lsp.c | 2 | ||||
| -rw-r--r-- | gr-vocoder/lib/codec2/machdep.h | 2 | ||||
| -rw-r--r-- | gr-vocoder/lib/codec2/phaseexp.c | 2 | ||||
| -rw-r--r-- | gr-vocoder/lib/codec2/postfilter.c | 6 | ||||
| -rw-r--r-- | gr-vocoder/lib/codec2/quantise.c | 8 | ||||
| -rw-r--r-- | gr-vocoder/lib/codec2/sine.c | 8 |
12 files changed, 22 insertions, 22 deletions
diff --git a/gr-vocoder/lib/codec2/c2sim.c b/gr-vocoder/lib/codec2/c2sim.c index e68f2aceeb..14d7ee8eed 100644 --- a/gr-vocoder/lib/codec2/c2sim.c +++ b/gr-vocoder/lib/codec2/c2sim.c @@ -558,7 +558,7 @@ int main(int argc, char *argv[]) #endif /* experimenting with non-linear LSP spacing to see if - it's just noticable */ + it's just noticeable */ if (lspjnd) { for(i=0; i<LPC_ORD; i++) diff --git a/gr-vocoder/lib/codec2/codec2.c b/gr-vocoder/lib/codec2/codec2.c index 5bf998064b..b7e963a98a 100644 --- a/gr-vocoder/lib/codec2/codec2.c +++ b/gr-vocoder/lib/codec2/codec2.c @@ -221,7 +221,7 @@ int CODEC2_WIN32SUPPORT codec2_samples_per_frame(struct CODEC2 *c2) { if (c2->mode == CODEC2_MODE_1200) return 320; - return 0; /* shouldnt get here */ + return 0; /* shouldn't get here */ } void CODEC2_WIN32SUPPORT codec2_encode(struct CODEC2 *c2, unsigned char *bits, short speech[]) diff --git a/gr-vocoder/lib/codec2/fdmdv.c b/gr-vocoder/lib/codec2/fdmdv.c index 51d6bef544..945b15778a 100644 --- a/gr-vocoder/lib/codec2/fdmdv.c +++ b/gr-vocoder/lib/codec2/fdmdv.c @@ -1493,8 +1493,8 @@ void CODEC2_WIN32SUPPORT fdmdv_48_to_8(float out8k[], float in48k[], int n) range for plotting is 0 to -40dB. Note only the real part of the complex input signal is used at - present. A complex variable is used for input for compatability - with the other rx signal procesing. + present. A complex variable is used for input for compatibility + with the other rx signal processing. Successive calls can be used to build up a waterfall or spectrogram plot, by mapping the received levels to colours. diff --git a/gr-vocoder/lib/codec2/interp.c b/gr-vocoder/lib/codec2/interp.c index be89fc3154..8f60eaf819 100644 --- a/gr-vocoder/lib/codec2/interp.c +++ b/gr-vocoder/lib/codec2/interp.c @@ -43,7 +43,7 @@ float sample_log_amp(MODEL *model, float w); AUTHOR......: David Rowe DATE CREATED: 22/8/10 - Given two frames decribed by model parameters 20ms apart, determines + Given two frames described by model parameters 20ms apart, determines the model parameters of the 10ms frame between them. Assumes voicing is available for middle (interpolated) frame. Outputs are amplitudes and Wo for the interpolated frame. @@ -135,7 +135,7 @@ float sample_log_amp(MODEL *model, float w) AUTHOR......: David Rowe DATE CREATED: 10 Nov 2010 - Given two frames decribed by model parameters 20ms apart, determines + Given two frames described by model parameters 20ms apart, determines the model parameters of the 10ms frame between them. Assumes voicing is available for middle (interpolated) frame. Outputs are amplitudes and Wo for the interpolated frame. diff --git a/gr-vocoder/lib/codec2/kiss_fft.c b/gr-vocoder/lib/codec2/kiss_fft.c index 17b4e72112..d3b50f8702 100644 --- a/gr-vocoder/lib/codec2/kiss_fft.c +++ b/gr-vocoder/lib/codec2/kiss_fft.c @@ -15,7 +15,7 @@ THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" AND #include "_kiss_fft_guts.h" /* The guts header contains all the multiplication and addition macros that are defined for - fixed or floating point complex numbers. It also delares the kf_ internal functions. + fixed or floating point complex numbers. It also declares the kf_ internal functions. */ static void kf_bfly2( diff --git a/gr-vocoder/lib/codec2/lpc.c b/gr-vocoder/lib/codec2/lpc.c index 9a730eb4ad..c24da79595 100644 --- a/gr-vocoder/lib/codec2/lpc.c +++ b/gr-vocoder/lib/codec2/lpc.c @@ -225,7 +225,7 @@ void inverse_filter( The synthesis filter has memory as well, this is treated in the same way as the memory for the inverse filter (see inverse_filter() notes above). The difference is that the memory for the synthesis filter is stored in - the output array, wheras the memory of the inverse filter is stored in the + the output array, whereas the memory of the inverse filter is stored in the input array. Note: the calling function must update the filter memory. diff --git a/gr-vocoder/lib/codec2/lsp.c b/gr-vocoder/lib/codec2/lsp.c index 3f34444e33..c69eb4ccfa 100644 --- a/gr-vocoder/lib/codec2/lsp.c +++ b/gr-vocoder/lib/codec2/lsp.c @@ -80,7 +80,7 @@ AUTHOR......: David Rowe DATE CREATED: 24/2/93 - This function evalutes a series of chebyshev polynomials + This function evaluates a series of chebyshev polynomials FIXME: performing memory allocation at run time is very inefficient, replace with stack variables of MAX_P size. diff --git a/gr-vocoder/lib/codec2/machdep.h b/gr-vocoder/lib/codec2/machdep.h index ef2e64943e..77a18e575b 100644 --- a/gr-vocoder/lib/codec2/machdep.h +++ b/gr-vocoder/lib/codec2/machdep.h @@ -4,7 +4,7 @@ AUTHOR......: David Rowe DATE CREATED: May 2 2013 - Machine dependant functions. + Machine dependent functions. \*---------------------------------------------------------------------------*/ diff --git a/gr-vocoder/lib/codec2/phaseexp.c b/gr-vocoder/lib/codec2/phaseexp.c index 61b240df49..128f9092d6 100644 --- a/gr-vocoder/lib/codec2/phaseexp.c +++ b/gr-vocoder/lib/codec2/phaseexp.c @@ -541,7 +541,7 @@ static void top_amp(struct PEXP *pexp, MODEL *model, int start, int end, int n_h removed++; } else { - /* need to make this work thru budget of bits */ + /* need to make this work through budget of bits */ quant_phase(&model->phi[i], -PI, PI, 3); not_removed++; } diff --git a/gr-vocoder/lib/codec2/postfilter.c b/gr-vocoder/lib/codec2/postfilter.c index 7c1a6061a0..7439522408 100644 --- a/gr-vocoder/lib/codec2/postfilter.c +++ b/gr-vocoder/lib/codec2/postfilter.c @@ -82,8 +82,8 @@ 1/ If someone says "aaaaaaaahhhhhhhhh" will background estimator track up to speech level? This would be a bad thing. - 2/ If background noise suddenly dissapears from the source speech does - estimate drop quickly? What is noise suddenly re-appears? + 2/ If background noise suddenly disappears from the source speech does + estimate drop quickly? What if noise suddenly re-appears? 3/ Background noise with a non-flat sepctrum. Current algorithm just comsiders scpetrum as a whole, but this could be broken up into @@ -115,7 +115,7 @@ void postfilter( assert(e > 0.0); e = 10.0*log10f(e/model->L); - /* If beneath threhold, update bg estimate. The idea + /* If beneath threshold, update bg estimate. The idea of the threshold is to prevent updating during high level speech. */ diff --git a/gr-vocoder/lib/codec2/quantise.c b/gr-vocoder/lib/codec2/quantise.c index f7326c4fcd..919d0c0bbc 100644 --- a/gr-vocoder/lib/codec2/quantise.c +++ b/gr-vocoder/lib/codec2/quantise.c @@ -762,7 +762,7 @@ void force_min_lsp_dist(float lsp[], int lpc_order) lpc_post_filter() Applies a post filter to the LPC synthesis filter power spectrum - Pw, which supresses the inter-formant energy. + Pw, which suppresses the inter-formant energy. The algorithm is from p267 (Section 8.6) of "Digital Speech", edited by A.M. Kondoz, 1994 published by Wiley and Sons. Chapter 8 @@ -1003,7 +1003,7 @@ void aks_to_M2( just above nulls. This algorithm does the reverse to compensate - raising the amplitudes of spectral peaks, while attenuating the null. This enhances the formants, and - supresses the energy between formants. */ + suppresses the energy between formants. */ if (sim_pf) { if (Am > model->A[m]) @@ -1607,7 +1607,7 @@ void bw_expand_lsps2(float lsp[], Applies a form of Bandwidth Expansion (BW) to a vector of LSPs. Listening tests have determined that "quantising" the position of - each LSP to the non-linear steps below introduces a "just noticable + each LSP to the non-linear steps below introduces a "just noticeable difference" in the synthesised speech. This operation can be used before quantisation to limit the input @@ -1830,7 +1830,7 @@ void compute_weights2(const float *x, const float *xp, float *w, int ndim) both the pitch and energy tend to only change by small amounts during voiced speech, however it is important that these changes be coded carefully. During unvoiced speech they both change a lot but - the ear is less sensitve to errors so coarser quantisation is OK. + the ear is less sensitive to errors so coarser quantisation is OK. The ear is sensitive to log energy and loq pitch so we quantise in these domains. That way the error measure used to quantise the diff --git a/gr-vocoder/lib/codec2/sine.c b/gr-vocoder/lib/codec2/sine.c index be4df00a6d..ee77a6e0de 100644 --- a/gr-vocoder/lib/codec2/sine.c +++ b/gr-vocoder/lib/codec2/sine.c @@ -238,7 +238,7 @@ void dft_speech(kiss_fft_cfg fft_fwd_cfg, COMP Sw[], float Sn[], float w[]) void two_stage_pitch_refinement(MODEL *model, COMP Sw[]) { - float pmin,pmax,pstep; /* pitch refinment minimum, maximum and step */ + float pmin,pmax,pstep; /* pitch refinement minimum, maximum and step */ /* Coarse refinement */ @@ -461,7 +461,7 @@ float est_voicing_mbe( /* post processing, helps clean up some voicing errors ------------------*/ /* - Determine the ratio of low freqency to high frequency energy, + Determine the ratio of low frequency to high frequency energy, voiced speech tends to be dominated by low frequency energy, unvoiced by high frequency. This measure can be used to determine if we have made any gross errors. @@ -511,7 +511,7 @@ float est_voicing_mbe( AUTHOR......: David Rowe DATE CREATED: 11/5/94 - Init function that generates the trapezoidal (Parzen) sythesis window. + Init function that generates the trapezoidal (Parzen) synthesis window. \*---------------------------------------------------------------------------*/ @@ -545,7 +545,7 @@ void make_synthesis_window(float Pn[]) Synthesise a speech signal in the frequency domain from the sinusodal model parameters. Uses overlap-add with a trapezoidal - window to smoothly interpolate betwen frames. + window to smoothly interpolate between frames. \*---------------------------------------------------------------------------*/ |