From 58c42af722cebecd86e340dc3ed9ec44b1fe4a55 Mon Sep 17 00:00:00 2001 From: Diego Biurrun Date: Wed, 7 Dec 2011 13:03:53 +0100 Subject: doxygen: misc consistency, spelling and wording fixes --- libavcodec/amrwbdec.c | 68 +++++++++++++++++++++++++-------------------------- 1 file changed, 34 insertions(+), 34 deletions(-) (limited to 'libavcodec/amrwbdec.c') diff --git a/libavcodec/amrwbdec.c b/libavcodec/amrwbdec.c index d4aa557d07..6ea5d228dd 100644 --- a/libavcodec/amrwbdec.c +++ b/libavcodec/amrwbdec.c @@ -111,7 +111,7 @@ static av_cold int amrwb_decode_init(AVCodecContext *avctx) /** * Decode the frame header in the "MIME/storage" format. This format - * is simpler and does not carry the auxiliary information of the frame + * is simpler and does not carry the auxiliary frame information. * * @param[in] ctx The Context * @param[in] buf Pointer to the input buffer @@ -133,7 +133,7 @@ static int decode_mime_header(AMRWBContext *ctx, const uint8_t *buf) } /** - * Decodes quantized ISF vectors using 36-bit indexes (6K60 mode only) + * Decode quantized ISF vectors using 36-bit indexes (6K60 mode only). * * @param[in] ind Array of 5 indexes * @param[out] isf_q Buffer for isf_q[LP_ORDER] @@ -160,7 +160,7 @@ static void decode_isf_indices_36b(uint16_t *ind, float *isf_q) } /** - * Decodes quantized ISF vectors using 46-bit indexes (except 6K60 mode) + * Decode quantized ISF vectors using 46-bit indexes (except 6K60 mode). * * @param[in] ind Array of 7 indexes * @param[out] isf_q Buffer for isf_q[LP_ORDER] @@ -193,8 +193,8 @@ static void decode_isf_indices_46b(uint16_t *ind, float *isf_q) } /** - * Apply mean and past ISF values using the prediction factor - * Updates past ISF vector + * Apply mean and past ISF values using the prediction factor. + * Updates past ISF vector. * * @param[in,out] isf_q Current quantized ISF * @param[in,out] isf_past Past quantized ISF @@ -215,7 +215,7 @@ static void isf_add_mean_and_past(float *isf_q, float *isf_past) /** * Interpolate the fourth ISP vector from current and past frames - * to obtain a ISP vector for each subframe + * to obtain an ISP vector for each subframe. * * @param[in,out] isp_q ISPs for each subframe * @param[in] isp4_past Past ISP for subframe 4 @@ -232,9 +232,9 @@ static void interpolate_isp(double isp_q[4][LP_ORDER], const double *isp4_past) } /** - * Decode an adaptive codebook index into pitch lag (except 6k60, 8k85 modes) - * Calculate integer lag and fractional lag always using 1/4 resolution - * In 1st and 3rd subframes the index is relative to last subframe integer lag + * Decode an adaptive codebook index into pitch lag (except 6k60, 8k85 modes). + * Calculate integer lag and fractional lag always using 1/4 resolution. + * In 1st and 3rd subframes the index is relative to last subframe integer lag. * * @param[out] lag_int Decoded integer pitch lag * @param[out] lag_frac Decoded fractional pitch lag @@ -271,9 +271,9 @@ static void decode_pitch_lag_high(int *lag_int, int *lag_frac, int pitch_index, } /** - * Decode a adaptive codebook index into pitch lag for 8k85 and 6k60 modes - * Description is analogous to decode_pitch_lag_high, but in 6k60 relative - * index is used for all subframes except the first + * Decode an adaptive codebook index into pitch lag for 8k85 and 6k60 modes. + * The description is analogous to decode_pitch_lag_high, but in 6k60 the + * relative index is used for all subframes except the first. */ static void decode_pitch_lag_low(int *lag_int, int *lag_frac, int pitch_index, uint8_t *base_lag_int, int subframe, enum Mode mode) @@ -298,7 +298,7 @@ static void decode_pitch_lag_low(int *lag_int, int *lag_frac, int pitch_index, /** * Find the pitch vector by interpolating the past excitation at the - * pitch delay, which is obtained in this function + * pitch delay, which is obtained in this function. * * @param[in,out] ctx The context * @param[in] amr_subframe Current subframe data @@ -351,10 +351,10 @@ static void decode_pitch_vector(AMRWBContext *ctx, /** * The next six functions decode_[i]p_track decode exactly i pulses * positions and amplitudes (-1 or 1) in a subframe track using - * an encoded pulse indexing (TS 26.190 section 5.8.2) + * an encoded pulse indexing (TS 26.190 section 5.8.2). * * The results are given in out[], in which a negative number means - * amplitude -1 and vice versa (i.e., ampl(x) = x / abs(x) ) + * amplitude -1 and vice versa (i.e., ampl(x) = x / abs(x) ). * * @param[out] out Output buffer (writes i elements) * @param[in] code Pulse index (no. of bits varies, see below) @@ -470,7 +470,7 @@ static void decode_6p_track(int *out, int code, int m, int off) ///code: 6m-2 bi /** * Decode the algebraic codebook index to pulse positions and signs, - * then construct the algebraic codebook vector + * then construct the algebraic codebook vector. * * @param[out] fixed_vector Buffer for the fixed codebook excitation * @param[in] pulse_hi MSBs part of the pulse index array (higher modes only) @@ -541,7 +541,7 @@ static void decode_fixed_vector(float *fixed_vector, const uint16_t *pulse_hi, } /** - * Decode pitch gain and fixed gain correction factor + * Decode pitch gain and fixed gain correction factor. * * @param[in] vq_gain Vector-quantized index for gains * @param[in] mode Mode of the current frame @@ -559,7 +559,7 @@ static void decode_gains(const uint8_t vq_gain, const enum Mode mode, } /** - * Apply pitch sharpening filters to the fixed codebook vector + * Apply pitch sharpening filters to the fixed codebook vector. * * @param[in] ctx The context * @param[in,out] fixed_vector Fixed codebook excitation @@ -580,7 +580,7 @@ static void pitch_sharpening(AMRWBContext *ctx, float *fixed_vector) } /** - * Calculate the voicing factor (-1.0 = unvoiced to 1.0 = voiced) + * Calculate the voicing factor (-1.0 = unvoiced to 1.0 = voiced). * * @param[in] p_vector, f_vector Pitch and fixed excitation vectors * @param[in] p_gain, f_gain Pitch and fixed gains @@ -599,8 +599,8 @@ static float voice_factor(float *p_vector, float p_gain, } /** - * Reduce fixed vector sparseness by smoothing with one of three IR filters - * Also known as "adaptive phase dispersion" + * Reduce fixed vector sparseness by smoothing with one of three IR filters, + * also known as "adaptive phase dispersion". * * @param[in] ctx The context * @param[in,out] fixed_vector Unfiltered fixed vector @@ -670,7 +670,7 @@ static float *anti_sparseness(AMRWBContext *ctx, /** * Calculate a stability factor {teta} based on distance between - * current and past isf. A value of 1 shows maximum signal stability + * current and past isf. A value of 1 shows maximum signal stability. */ static float stability_factor(const float *isf, const float *isf_past) { @@ -687,7 +687,7 @@ static float stability_factor(const float *isf, const float *isf_past) /** * Apply a non-linear fixed gain smoothing in order to reduce - * fluctuation in the energy of excitation + * fluctuation in the energy of excitation. * * @param[in] fixed_gain Unsmoothed fixed gain * @param[in,out] prev_tr_gain Previous threshold gain (updated) @@ -718,7 +718,7 @@ static float noise_enhancer(float fixed_gain, float *prev_tr_gain, } /** - * Filter the fixed_vector to emphasize the higher frequencies + * Filter the fixed_vector to emphasize the higher frequencies. * * @param[in,out] fixed_vector Fixed codebook vector * @param[in] voice_fac Frame voicing factor @@ -742,7 +742,7 @@ static void pitch_enhancer(float *fixed_vector, float voice_fac) } /** - * Conduct 16th order linear predictive coding synthesis from excitation + * Conduct 16th order linear predictive coding synthesis from excitation. * * @param[in] ctx Pointer to the AMRWBContext * @param[in] lpc Pointer to the LPC coefficients @@ -802,7 +802,7 @@ static void de_emphasis(float *out, float *in, float m, float mem[1]) /** * Upsample a signal by 5/4 ratio (from 12.8kHz to 16kHz) using - * a FIR interpolation filter. Uses past data from before *in address + * a FIR interpolation filter. Uses past data from before *in address. * * @param[out] out Buffer for interpolated signal * @param[in] in Current signal data (length 0.8*o_size) @@ -832,7 +832,7 @@ static void upsample_5_4(float *out, const float *in, int o_size) /** * Calculate the high-band gain based on encoded index (23k85 mode) or - * on the low-band speech signal and the Voice Activity Detection flag + * on the low-band speech signal and the Voice Activity Detection flag. * * @param[in] ctx The context * @param[in] synth LB speech synthesis at 12.8k @@ -857,7 +857,7 @@ static float find_hb_gain(AMRWBContext *ctx, const float *synth, /** * Generate the high-band excitation with the same energy from the lower - * one and scaled by the given gain + * one and scaled by the given gain. * * @param[in] ctx The context * @param[out] hb_exc Buffer for the excitation @@ -880,7 +880,7 @@ static void scaled_hb_excitation(AMRWBContext *ctx, float *hb_exc, } /** - * Calculate the auto-correlation for the ISF difference vector + * Calculate the auto-correlation for the ISF difference vector. */ static float auto_correlation(float *diff_isf, float mean, int lag) { @@ -896,7 +896,7 @@ static float auto_correlation(float *diff_isf, float mean, int lag) /** * Extrapolate a ISF vector to the 16kHz range (20th order LP) - * used at mode 6k60 LP filter for the high frequency band + * used at mode 6k60 LP filter for the high frequency band. * * @param[out] out Buffer for extrapolated isf * @param[in] isf Input isf vector @@ -981,7 +981,7 @@ static void lpc_weighting(float *out, const float *lpc, float gamma, int size) /** * Conduct 20th order linear predictive coding synthesis for the high - * frequency band excitation at 16kHz + * frequency band excitation at 16kHz. * * @param[in] ctx The context * @param[in] subframe Current subframe index (0 to 3) @@ -1019,8 +1019,8 @@ static void hb_synthesis(AMRWBContext *ctx, int subframe, float *samples, } /** - * Apply to high-band samples a 15th order filter - * The filter characteristic depends on the given coefficients + * Apply a 15th order filter to high-band samples. + * The filter characteristic depends on the given coefficients. * * @param[out] out Buffer for filtered output * @param[in] fir_coef Filter coefficients @@ -1048,7 +1048,7 @@ static void hb_fir_filter(float *out, const float fir_coef[HB_FIR_SIZE + 1], } /** - * Update context state before the next subframe + * Update context state before the next subframe. */ static void update_sub_state(AMRWBContext *ctx) { -- cgit v1.2.3