fix "@param foo[in/out]" to "@param[in, out] foo"

Fix a few invalid doxygen comments:

    /**
     * @param[in,out] correctly_formatted
     * @param incorrect1[in] - [in] must come immediately after @param
     * @param incorrect2[in/out] - '/' must be ','
     */

Actual command: sed -i -e "s/\(\* .*param\)\( [^\[]*\)\(\[.*\]\)/\1\3\2/g" -e "s/in\//in,/" $( git grep -l "\* .*param .*\[\(in\|out\)" )

Signed-off-by: Andrew Sayers <ffmpeg-devel@pileofstuff.org>

2 files changed, 11 insertions, 11 deletions

diff --git a/libavcodec/g729postfilter.c b/libavcodec/g729postfilter.c
index 382db92432..b1880b2fe1 100644
--- a/libavcodec/g729postfilter.c
+++ b/libavcodec/g729postfilter.c
@@ -78,7 +78,7 @@ static const int16_t formant_pp_factor_den_pow[10] = {
 
 /**
  * \brief Residual signal calculation (4.2.1 if G.729)
- * \param out [out] output data filtered through A(z/FORMANT_PP_FACTOR_NUM)
+ * \param[out] out  output data filtered through A(z/FORMANT_PP_FACTOR_NUM)
  * \param filter_coeffs (3.12) A(z/FORMANT_PP_FACTOR_NUM) filter coefficients
  * \param in input speech data to process
  * \param subframe_size size of one subframe
@@ -105,7 +105,7 @@ static void residual_filter(int16_t* out, const int16_t* filter_coeffs, const in
  * \param dsp initialized DSP context
  * \param pitch_delay_int integer part of the pitch delay in the first subframe
  * \param residual filtering input data
- * \param residual_filt [out] speech signal with applied A(z/FORMANT_PP_FACTOR_NUM) filter
+ * \param[out] residual_filt  speech signal with applied A(z/FORMANT_PP_FACTOR_NUM) filter
  * \param subframe_size size of subframe
  *
  * \return 0 if long-term prediction gain is less than 3dB, 1 -  otherwise
@@ -472,7 +472,7 @@ static int16_t get_tilt_comp(AudioDSPContext *adsp, int16_t *lp_gn,
 
 /**
  * \brief Apply tilt compensation filter (4.2.3).
- * \param res_pst [in/out] residual signal (partially filtered)
+ * \param[in,out] res_pst  residual signal (partially filtered)
  * \param k1 (3.12) reflection coefficient
  * \param subframe_size size of subframe
  * \param ht_prev_data previous data for 4.2.3, equation 86
@@ -572,7 +572,7 @@ void ff_g729_postfilter(AudioDSPContext *adsp, int16_t* ht_prev_data, int* voici
  * \brief Adaptive gain control (4.2.4)
  * \param gain_before gain of speech before applying postfilters
  * \param gain_after  gain of speech after applying postfilters
- * \param speech [in/out] signal buffer
+ * \param[in,out] speech  signal buffer
  * \param subframe_size length of subframe
  * \param gain_prev (3.12) previous value of gain coefficient
  *
diff --git a/libavcodec/g729postfilter.h b/libavcodec/g729postfilter.h
index 69815341ed..0421ed8720 100644
--- a/libavcodec/g729postfilter.h
+++ b/libavcodec/g729postfilter.h
@@ -79,15 +79,15 @@
 /**
  * \brief Signal postfiltering (4.2)
  * \param dsp initialized DSP context
- * \param ht_prev_data [in/out] (Q12) pointer to variable receiving tilt
+ * \param[in,out] ht_prev_data  (Q12) pointer to variable receiving tilt
  *                     compensation filter data from previous subframe
- * \param voicing [in/out] (Q0) pointer to variable receiving voicing decision
+ * \param[in,out] voicing  (Q0) pointer to variable receiving voicing decision
  * \param lp_filter_coeffs (Q12) LP filter coefficients
  * \param pitch_delay_int integer part of the pitch delay
- * \param residual [in/out] (Q0) residual signal buffer (used in long-term postfilter)
- * \param res_filter_data [in/out] (Q0) speech data of previous subframe
- * \param pos_filter_data [in/out] (Q0) previous speech data for short-term postfilter
- * \param speech [in/out] (Q0) signal buffer
+ * \param[in,out] residual  (Q0) residual signal buffer (used in long-term postfilter)
+ * \param[in,out] res_filter_data  (Q0) speech data of previous subframe
+ * \param[in,out] pos_filter_data  (Q0) previous speech data for short-term postfilter
+ * \param[in,out] speech  (Q0) signal buffer
  * \param subframe_size size of subframe
  *
  * Filtering has the following  stages:
@@ -105,7 +105,7 @@ void ff_g729_postfilter(AudioDSPContext *adsp, int16_t* ht_prev_data, int* voici
  * \brief Adaptive gain control (4.2.4)
  * \param gain_before (Q0) gain of speech before applying postfilters
  * \param gain_after  (Q0) gain of speech after applying postfilters
- * \param speech [in/out] (Q0) signal buffer
+ * \param[in,out] speech  (Q0) signal buffer
  * \param subframe_size length of subframe
  * \param gain_prev (Q12) previous value of gain coefficient
  *