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Make doxygen comments consistent with the rest of FFmpeg.

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Originally committed as revision 14886 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Michael Niedermayer
2008-08-21 21:37:53 +00:00
parent fee37a4985
commit 2398930fe0
1 changed files with 35 additions and 35 deletions
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70
libavcodec/acelp_filters.h
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@@ -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.