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authorMichael Niedermayer <michaelni@gmx.at>2011-12-13 00:29:25 +0100
committerMichael Niedermayer <michaelni@gmx.at>2011-12-13 00:39:48 +0100
commit8bc7fe4daf5c26555d77e2261c96ee14a800fad4 (patch)
tree71999a1eb8bacb609b5789b34de369ed4b049cd4 /libavcodec/amrwbdec.c
parentc83ef636a665b2e17f1b6745a98ec7a5719264df (diff)
parent58c42af722cebecd86e340dc3ed9ec44b1fe4a55 (diff)
downloadffmpeg-8bc7fe4daf5c26555d77e2261c96ee14a800fad4.tar.gz
Merge remote-tracking branch 'qatar/master'
* qatar/master: doxygen: misc consistency, spelling and wording fixes vcr1: drop unnecessary emms_c() calls without MMX code Replace all uses of av_close_input_file() with avformat_close_input(). lavf: add avformat_close_input(). lavf: deprecate av_close_input_stream(). lavf doxy: add some basic demuxing documentation. lavf doxy: add some general lavf information. lavf doxy: add misc utility functions to a group. lavf doxy: add av_guess_codec/format to the encoding group. lavf doxy: add core functions to a doxy group. Add basic libavdevice documentation. lavc: convert error_recognition to err_recognition. avconv: update -map option help text x86: Require 7 registers for the cabac asm x86: bswap: remove test for bswap instruction bswap: make generic implementation more compiler-friendly h264: remove useless cast proresdec: fix decode_slice() prototype Conflicts: configure doc/APIchanges ffprobe.c libavcodec/avcodec.h libavcodec/celp_math.h libavcodec/h264.c libavfilter/src_movie.c libavformat/anm.c libavformat/avformat.h libavformat/version.h libavutil/avstring.h libavutil/bswap.h Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavcodec/amrwbdec.c')
-rw-r--r--libavcodec/amrwbdec.c68
1 files changed, 34 insertions, 34 deletions
diff --git a/libavcodec/amrwbdec.c b/libavcodec/amrwbdec.c
index b0e54f9261..524979d755 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)
{