Make doxygen comments consistent with the rest of FFmpeg.

Originally committed as revision 14886 to svn://svn.ffmpeg.org/ffmpeg/trunk

libavcodec/acelp_filters.h

@@ -79,14 +79,14 @@
 extern const int16_t ff_acelp_interp_filter[61];
 
 /**
- * \brief Generic interpolation routine
- * \param out [out] buffer for interpolated data
- * \param in input data
- * \param filter_coeffs interpolation filter coefficients (0.15)
- * \param precision filter is able to interpolate with 1/precision precision of pitch delay
- * \param pitch_delay_frac pitch delay, fractional part [0..precision-1]
- * \param filter_length filter length
- * \param length length of speech data to process
+ * Generic interpolation routine.
+ * @param out [out] buffer for interpolated data
+ * @param in input data
+ * @param filter_coeffs interpolation filter coefficients (0.15)
+ * @param precision filter is able to interpolate with 1/precision precision of pitch delay
+ * @param pitch_delay_frac pitch delay, fractional part [0..precision-1]
+ * @param filter_length filter length
+ * @param length length of speech data to process
  *
  * filter_coeffs contains coefficients of the positive half of the symmetric
  * interpolation filter. filter_coeffs[0] should the central (unpaired) coefficient.
@@ -103,11 +103,11 @@ void ff_acelp_interpolate(
         int length);
 
 /**
- * \brief Circularly convolve fixed vector with a phase dispersion impulse
+ * Circularly convolve fixed vector with a phase dispersion impulse
  *        response filter (D.6.2 of G.729 and 6.1.5 of AMR).
- * \param fc_out vector with filter applied
- * \param fc_in source vector
- * \param filter phase filter coefficients
+ * @param fc_out vector with filter applied
+ * @param fc_in source vector
+ * @param filter phase filter coefficients
  *
  *  fc_out[n] = sum(i,0,len-1){ fc_in[i] * filter[(len + n - i)%len] }
  *
@@ -120,19 +120,19 @@ void ff_acelp_convolve_circ(
         int subframe_size);
 
 /**
- * \brief LP synthesis filter
- * \param out [out] pointer to output buffer
- * \param filter_coeffs filter coefficients (-0x8000 <= (3.12) < 0x8000)
- * \param in input signal
- * \param buffer_length amount of data to process
- * \param filter_length filter length (10 for 10th order LP filter)
- * \param stop_on_overflow   1 - return immediately if overflow occurs
+ * LP synthesis filter.
+ * @param out [out] pointer to output buffer
+ * @param filter_coeffs filter coefficients (-0x8000 <= (3.12) < 0x8000)
+ * @param in input signal
+ * @param buffer_length amount of data to process
+ * @param filter_length filter length (10 for 10th order LP filter)
+ * @param stop_on_overflow   1 - return immediately if overflow occurs
  *                           0 - ignore overflows
- * \param rounder the amount to add for rounding (usually 0x800 or 0xfff)
+ * @param rounder the amount to add for rounding (usually 0x800 or 0xfff)
  *
- * \return 1 if overflow occurred, 0 - otherwise
+ * @return 1 if overflow occurred, 0 - otherwise
  *
- * \note Output buffer must contain 10 samples of past
+ * @note Output buffer must contain 10 samples of past
  *       speech data before pointer.
  *
  * Routine applies 1/A(z) filter to given speech data.
@@ -147,12 +147,12 @@ int ff_acelp_lp_synthesis_filter(
         int rounder);
 
 /**
- * \brief Calculates coefficients of weighted A(z/weight) filter.
- * \param out [out] weighted A(z/weight) result
+ * Calculates coefficients of weighted A(z/weight) filter.
+ * @param out [out] weighted A(z/weight) result
  *                  filter (-0x8000 <= (3.12) < 0x8000)
- * \param in source filter (-0x8000 <= (3.12) < 0x8000)
- * \param weight_pow array containing weight^i (-0x8000 <= (0.15) < 0x8000)
- * \param filter_length filter length (11 for 10th order LP filter)
+ * @param in source filter (-0x8000 <= (3.12) < 0x8000)
+ * @param weight_pow array containing weight^i (-0x8000 <= (0.15) < 0x8000)
+ * @param filter_length filter length (11 for 10th order LP filter)
  *
  * out[i]=weight_pow[i]*in[i] , i=0..9
  */
@@ -163,24 +163,24 @@ void ff_acelp_weighted_filter(
         int filter_length);
 
 /**
- * \brief high-pass filtering and upscaling (4.2.5 of G.729)
- * \param out [out] output buffer for filtered speech data
- * \param hpf_f [in/out] past filtered data from previous (2 items long)
+ * high-pass filtering and upscaling (4.2.5 of G.729).
+ * @param out [out] output buffer for filtered speech data
+ * @param hpf_f [in/out] past filtered data from previous (2 items long)
  *                       frames (-0x20000000 <= (14.13) < 0x20000000)
- * \param in speech data to process
- * \param length input data size
+ * @param in speech data to process
+ * @param length input data size
  *
  * out[i] = 0.93980581 * in[i] - 1.8795834 * in[i-1] + 0.93980581 * in[i-2] +
  *          1.9330735 * out[i-1] - 0.93589199 * out[i-2]
  *
  * The filter has a cut-off frequency of 100Hz
  *
- * \note Two items before the top of the out buffer must contain two items from the
+ * @note Two items before the top of the out buffer must contain two items from the
  *       tail of the previous subframe.
  *
- * \remark It is safe to pass the same array in in and out parameters.
+ * @remark It is safe to pass the same array in in and out parameters.
  *
- * \remark AMR uses mostly the same filter (cut-off frequency 60Hz, same formula,
+ * @remark AMR uses mostly the same filter (cut-off frequency 60Hz, same formula,
  *         but constants differs in 5th sign after comma). Fortunately in
  *         fixed-point all coefficients are the same as in G.729. Thus this
  *         routine can be used for the fixed-point AMR decoder, too.