doxygen: misc consistency, spelling and wording fixes

libavcodec/aacpsy.c

@@ -216,7 +216,7 @@ static const float psy_fir_coeffs[] = {
 };
 
 /**
- * calculates the attack threshold for ABR from the above table for the LAME psy model
+ * Calculate the ABR attack threshold from the above LAME psymodel table.
  */
 static float lame_calc_attack_threshold(int bitrate)
 {

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)
 {

libavcodec/avcodec.h

@@ -2591,7 +2591,7 @@ typedef struct AVCodecContext {
 
 #if FF_API_X264_GLOBAL_OPTS
     /**
-     * Influences how often B-frames are used.
+     * Influence how often B-frames are used.
      * - encoding: Set by user.
      * - decoding: unused
      */
@@ -2672,7 +2672,7 @@ typedef struct AVCodecContext {
     int mv0_threshold;
 
     /**
-     * Adjusts sensitivity of b_frame_strategy 1.
+     * Adjust sensitivity of b_frame_strategy 1.
      * - encoding: Set by user.
      * - decoding: unused
      */
@@ -2956,7 +2956,7 @@ typedef struct AVCodecContext {
 
 #if FF_API_FLAC_GLOBAL_OPTS
     /**
-     * Determines which LPC analysis algorithm to use.
+     * Determine which LPC analysis algorithm to use.
      * - encoding: Set by user
      * - decoding: unused
      */
@@ -4121,7 +4121,7 @@ int avcodec_decode_subtitle2(AVCodecContext *avctx, AVSubtitle *sub,
                             AVPacket *avpkt);
 
 /**
- * Frees all allocated data in the given subtitle struct.
+ * Free all allocated data in the given subtitle struct.
  *
  * @param sub AVSubtitle to free.
  */
@@ -4486,7 +4486,7 @@ int av_picture_pad(AVPicture *dst, const AVPicture *src, int height, int width,
 unsigned int av_xiphlacing(unsigned char *s, unsigned int v);
 
 /**
- * Logs a generic warning message about a missing feature. This function is
+ * Log a generic warning message about a missing feature. This function is
  * intended to be used internally by Libav (libavcodec, libavformat, etc.)
  * only, and would normally not be used by applications.
  * @param[in] avc a pointer to an arbitrary struct of which the first field is

libavcodec/cavs_parser.c

@@ -30,7 +30,7 @@
 
 
 /**
- * finds the end of the current frame in the bitstream.
+ * Find the end of the current frame in the bitstream.
  * @return the position of the first byte of the next frame, or -1
  */
 static int cavs_find_frame_end(ParseContext *pc, const uint8_t *buf,

libavcodec/celp_math.h

@@ -64,7 +64,7 @@ static inline int bidir_sal(int value, int offset)
 }
 
 /**
- * returns the dot product.
+ * Return the dot product.
  * @param a input data array
  * @param b input data array
  * @param length number of elements

libavcodec/dca_parser.c

@@ -39,7 +39,7 @@ typedef struct DCAParseContext {
  || state == DCA_MARKER_RAW_LE || state == DCA_MARKER_RAW_BE)
 
 /**
- * finds the end of the current frame in the bitstream.
+ * Find the end of the current frame in the bitstream.
  * @return the position of the first byte of the next frame, or -1
  */
 static int dca_find_frame_end(DCAParseContext * pc1, const uint8_t * buf,

libavcodec/dsputil.c

@@ -1779,7 +1779,7 @@ static void add_8x8basis_c(int16_t rem[64], int16_t basis[64], int scale){
 }
 
 /**
- * permutes an 8x8 block.
+ * Permute an 8x8 block.
  * @param block the block which will be permuted according to the given permutation vector
  * @param permutation the permutation vector
  * @param last the last non zero coefficient in scantable order, used to speed the permutation up

libavcodec/error_resilience.c

@@ -80,7 +80,7 @@ static void set_mv_strides(MpegEncContext *s, int *mv_step, int *stride){
 }
 
 /**
- * replaces the current MB with a flat dc only version.
+ * Replace the current MB with a flat dc-only version.
  */
 static void put_dc(MpegEncContext *s, uint8_t *dest_y, uint8_t *dest_cb, uint8_t *dest_cr, int mb_x, int mb_y)
 {
@@ -711,7 +711,7 @@ void ff_er_frame_start(MpegEncContext *s){
 }
 
 /**
- * adds a slice.
+ * Add a slice.
  * @param endx x component of the last macroblock, can be -1 for the last of the previous line
  * @param status the status at the end (MV_END, AC_ERROR, ...), it is assumed that no earlier end or
  *               error of the same type occurred