Author: diego Date: Sun Jan 25 16:38:52 2009 New Revision: 4000 Log: spelling/wording/grammar cosmetics Modified: amr/amrdata.h amr/amrnbdec.c amr/amrnbfloatdata.h amr/amrnbfloatdec.c Modified: amr/amrdata.h ============================================================================== --- amr/amrdata.h Sun Jan 25 16:35:48 2009 (r3999) +++ amr/amrdata.h Sun Jan 25 16:38:52 2009 (r4000) @@ -108,7 +108,8 @@ int16_t cos_table[65]; // cosine table u #define TRACKS_MODE_102 4 // number of tracks for MODE_102 -// FIXME - not quite sure how i'm going to implement this yet and some other variables may be moved around later +/* FIXME - not quite sure how i'm going to implement this yet + * and some other variables may be moved around later */ typedef struct AMRDecoderState { // Bad frame int state; @@ -244,7 +245,7 @@ static const AMROrder order_MODE_DTX[35] // encoder homing frame pattern #define EHF_MASK 0x0008 -// Homing frames for the decoder +// homing frames for the decoder static const int16_t dhf_MODE_475[PRMS_MODE_475] = { 0x00F8,0x009D,0x001C,0x0066,0x0000,0x0003,0x0028,0x000F,0x0038,0x0001, 0x000F,0x0031,0x0002,0x0008,0x000F,0x0026,0x0003 @@ -376,7 +377,7 @@ static const uint16_t pred_fac[10] = { 11651, 10588, 9767, 8593, 6484 }; -// vector quantised lsf tables +// vector quantized lsf tables static const int16_t lsf_3_MODE_515[128][4] = { { 419, 163, -30, -262}, { -455, -789,-1430, -721}, { 1006, 664, 269, 25}, { 619, 260, 183, 96}, { -968,-1358, -388, 135}, { -693, 835, 456, 154}, { 1105, 703, 569, 363}, { 1625, 1326, 985, 748}, @@ -1157,7 +1158,7 @@ static const uint16_t pitch_gain_attenua 6553 }; -// quantisation table for the adaptive codebook gain (MODE_795, MODE_122) +// quantization table for the adaptive codebook gain (MODE_795, MODE_122) #define N_PITCH_QUANTS 16 static const uint16_t pitch_gain_quant[N_PITCH_QUANTS] = { 0, Modified: amr/amrnbdec.c ============================================================================== --- amr/amrnbdec.c Sun Jan 25 16:35:48 2009 (r3999) +++ amr/amrnbdec.c Sun Jan 25 16:38:52 2009 (r4000) @@ -41,18 +41,18 @@ typedef struct AMRContext { GetBitContext gb; int16_t *sample_buffer; - int16_t *amr_prms; ///< pointer to the decoded amr parameters (lsf coefficients, codebook indices, etc) + int16_t *amr_prms; ///< pointer to the decoded AMR parameters (lsf coefficients, codebook indexes, etc) int bad_frame_indicator; ///< bad frame ? 1 : 0 int prev_frame_homing; ///< previous frame was a homing frame ? 1 : 0 enum RXFrameType prev_frame_type; ///< frame type of previous frame enum Mode prev_frame_mode; ///< mode of previous frame - int *prev_lsf_dq; ///< previous dequantised lsfs - int *prev_residual_q; ///< previous quantised residual + int *prev_lsf_dq; ///< previous dequantized lsfs + int *prev_residual_q; ///< previous quantized residual - int lsp1_q[LP_FILTER_ORDER]; ///< vector of quantised lsps - int lsp2_q[LP_FILTER_ORDER]; ///< vector of quantised lsps + int lsp1_q[LP_FILTER_ORDER]; ///< vector of quantized lsps + int lsp2_q[LP_FILTER_ORDER]; ///< vector of quantized lsps int cur_frame_homing; ///< current frame homing ? 1 : 0 enum RXFrameType cur_frame_type; ///< current frame type @@ -96,7 +96,7 @@ static int amrnb_decode_init(AVCodecCont // allocate and zero the sample buffer p->sample_buffer = av_mallocz(sizeof(int16_t)*AMR_BLOCK_SIZE); - // allocate and zero the amr parameters + // allocate and zero the AMR parameters p->amr_prms = av_mallocz(sizeof(int16_t)*PRMS_MODE_122); // allocate and zero the decoder state p->state = av_mallocz(sizeof(AMRDecoderState)); @@ -117,13 +117,13 @@ static int amrnb_decode_init(AVCodecCont /** - * Decode the bitstream into the AMR parameters and discover the frame mode + * Decode the bitstream into the AMR parameters and discover the frame mode. * * @param buf pointer to the input buffer * @param buf_size size of the input buffer * @param speech_mode pointer to the speech mode * - * @return Returns the frame mode + * @return the frame mode */ enum Mode decode_bitstream(AVCodecContext *avctx, uint8_t *buf, int buf_size, @@ -134,7 +134,7 @@ enum Mode decode_bitstream(AVCodecContex int i; const AMROrder *order; - // initialise get_bits + // initialize get_bits init_get_bits(&p->gb, buf, buf_size*8); skip_bits1(&p->gb); // set the mode @@ -207,8 +207,8 @@ enum Mode decode_bitstream(AVCodecContex /** - * DESCRIPTION FROM REF SOURCE: - * Make sure that the LSFs are properly ordered and to keep a certain minimum + * DESCRIPTION FROM REFERENCE SOURCE: + * Make sure that the LSFs are properly ordered and keep a certain minimum * distance between adjacent LSFs. * * @param lsf a vector of lsfs (range: 0<=val<=0.5) @@ -230,7 +230,7 @@ static void reorder_lsf(int *lsf, int mi /** - * Convert a vector of lsfs into the corresponding, cosine domain lsps + * Convert a vector of lsfs into the corresponding cosine domain lsps. * * @param lsf a vector of lsfs * @param lsp a vector of lsps @@ -250,7 +250,7 @@ static void lsf2lsp(int *lsf, int *lsp) /** - * Convert a set of 3 lsf quantisation indices into lsp parameters + * Convert a set of 3 lsf quantization indexes into lsp parameters. * * @param avctx pointer to the AVCodecContext for AMR */ @@ -260,12 +260,13 @@ static void decode_lsf2lsp_3(AVCodecCont AMRContext *p = avctx->priv_data; int lsf1_r[LP_FILTER_ORDER]; // vector of residual lsfs - int lsf1_q[LP_FILTER_ORDER]; // vector of quantised lsfs + int lsf1_q[LP_FILTER_ORDER]; // vector of quantized lsfs const int16_t (*lsf_3_temp1)[3], (*lsf_3_temp3)[4]; // temp ptrs for switching tables depending on mode int index_temp; // temp lsf index int i; // counter - // if the current frame is bad estimate the past quantised residual based on the past lsf shifted slightly towards the mean + /* if the current frame is bad, estimate the past quantized residual + * based on the past lsf shifted slightly towards the mean */ if(p->bad_frame_indicator) { for(i=0; i<LP_FILTER_ORDER; i++) { lsf1_q[i] = ( (p->prev_lsf_dq[i]*ALPHA)>>15 ) + ( (lsf_3_mean[i]*ONE_ALPHA)>>15 ); @@ -292,13 +293,14 @@ static void decode_lsf2lsp_3(AVCodecCont lsf_3_temp3 = lsf_3_3; } - // decode lsf residuals from lsf tables using the indices from the bitstream + // decode lsf residuals from lsf tables using the bitstream indexes lsf1_r[0] = lsf_3_temp1[ p->amr_prms[0] ][0]; lsf1_r[1] = lsf_3_temp1[ p->amr_prms[0] ][1]; lsf1_r[2] = lsf_3_temp1[ p->amr_prms[0] ][2]; index_temp = p->amr_prms[1]; - // MODE_475, MODE_515 only use every second entry - NOTE : not quite sure what they mean and the spec doesn't mention this + /* MODE_475, MODE_515 only use every second entry + * NOTE: unsure what they mean and the spec doesn't mention this */ if((p->cur_frame_mode == MODE_475) || (p->cur_frame_mode == MODE_515)) { index_temp = index_temp << 1; } @@ -325,7 +327,7 @@ static void decode_lsf2lsp_3(AVCodecCont } } - /* verification that LSFs has minimum distance of LSF_GAP Hz */ + /* verification that LSFs have minimum distance of LSF_GAP Hz */ reorder_lsf(lsf1_q, LSF_GAP); memcpy(p->prev_lsf_dq, lsf1_q, LP_FILTER_ORDER<<2); @@ -335,7 +337,7 @@ static void decode_lsf2lsp_3(AVCodecCont /** - * Convert a set of 5 lsf quantisation indices into lsp parameters + * Convert a set of 5 lsf quantization indexes into lsp parameters. * * @param avctx pointer to the AVCodecContext for AMR */ @@ -345,11 +347,12 @@ static void decode_lsf2lsp_5(AVCodecCont AMRContext *p = avctx->priv_data; int lsf1_r[LP_FILTER_ORDER], lsf2_r[LP_FILTER_ORDER]; // vectors of residual lsfs - int lsf1_q[LP_FILTER_ORDER], lsf2_q[LP_FILTER_ORDER]; // vectors of quantised lsfs + int lsf1_q[LP_FILTER_ORDER], lsf2_q[LP_FILTER_ORDER]; // vectors of quantized lsfs int temp; int i, sign; // counter and sign of 3rd lsf table - // if the current frame is bad estimate the past quantised residual based on the past lsf shifted slightly towards the mean + /* if the current frame is bad, estimate the past quantized residual + * based on the past lsf shifted slightly towards the mean */ if(p->bad_frame_indicator) { for(i=0; i<LP_FILTER_ORDER; i++) { lsf1_q[i] = ( (p->prev_lsf_dq[i]*ALPHA_122)>>15 ) + ( (lsf_5_mean[i]*ONE_ALPHA_122)>>15 ); @@ -359,7 +362,7 @@ static void decode_lsf2lsp_5(AVCodecCont p->prev_residual_q[i] = lsf2_q[i] - lsf_5_mean[i] - ( (p->prev_residual_q[i]*LSP_PRED_FAC_MODE_122)>>15 ); } }else { - /* decode prediction residuals from 5 received indices */ + /* decode prediction residuals from 5 received indexes */ lsf1_r[0] = lsf_5_1[ p->amr_prms[0] ][0]; lsf1_r[1] = lsf_5_1[ p->amr_prms[0] ][1]; lsf2_r[0] = lsf_5_1[ p->amr_prms[0] ][2]; @@ -436,7 +439,8 @@ static void lsp2poly(int *lsp, int *f) f[1] -= lsp[2*i-2]<<10; } } -/* FIXME - check rounding sensitivity of F1(z) and F2(z) as the code below is faster but rounds differently +/* FIXME - check rounding sensitivity of F1(z) and F2(z) + * as the code below is faster but rounds differently static void lsp2poly(int *lsp, int *f) { int i, j; f[0] = 1<<24; @@ -452,7 +456,7 @@ static void lsp2poly(int *lsp, int *f) { /** - * Convert a vector of lsps to lpc coefficients for a 10th order filter + * Convert a vector of lsps to lpc coefficients for a 10th order filter. * * @param lsp vector of lsps * @param lpc_coeffs pointer to a subframe of lpc coefficients @@ -494,7 +498,7 @@ static void lsp2lpc(int *lsp, int *lpc_c /** - * Interpolate lsps for subframes 1 and 3 and convert them to lpc coefficients + * Interpolate lsps for subframes 1 and 3 and convert them to lpc coefficients. * * @param avctx pointer to AVCodecContext * @param lpc_coeffs array of lpc coefficients for the four subframes @@ -531,7 +535,7 @@ static void lpc_interp_13(AVCodecContext /** - * Interpolate lsps for subframes 1, 2 and 3 and convert them to lpc coefficients + * Interpolate lsps for subframes 1, 2 and 3 and convert them to lpc coefficients. * * @param avctx pointer to AVCodecContext * @param lpc_coeffs array of lpc coefficients for the four subframes @@ -573,10 +577,10 @@ static void lpc_interp_123(AVCodecContex /** - * Decode the adaptive codebook index to the integer and fractional pitch lags for one subframe - * at 1/3 resolution + * Decode the adaptive codebook index to the integer and fractional + * pitch lags for one subframe at 1/3 resolution. * - * Description from ref source: + * Description from reference source: * The fractional lag in 1st and 3rd subframes is encoded with 8 bits * while that in 2nd and 4th subframes is relatively encoded with 4, 5 * and 6 bits depending on the mode. @@ -593,16 +597,17 @@ static void decode_pitch_lag_3(AVCodecCo AMRContext *p = avctx->priv_data; int tmp_lag; - // FIXME - find out where these constants come from and add appropriate comments - // such that people reading the code can understand why these particular values - // are used + // FIXME: find out where these constants come from and add appropriate + // comments such that people reading the code can understand why these + // particular values are used // subframe 1 or 3 if(p->cur_subframe & 1) { if(pitch_index < 197) { *pitch_lag_int = (( (pitch_index + 2)*10923 )>>15) + 19; *pitch_lag_frac = pitch_index - *pitch_lag_int*3 + 58; - // For the sake of deciphering where this comes from, the above code probably means: + // For the sake of deciphering where this comes from, + // the above code probably means: // *pitch_lag_int = (pitch_index + 2)/3 + 19; // *pitch_lag_frac = (pitch_index + 2)%3 + 56; }else { @@ -639,16 +644,16 @@ static void decode_pitch_lag_3(AVCodecCo /** - * Decode the adaptive codebook index to the integer and fractional pitch lag for one subframe - * at 1/6 resolution + * Decode the adaptive codebook index to the integer and fractional + * pitch lag for one subframe at 1/6 resolution. * - * Description from ref source: + * Description from reference source: * The fractional lag in 1st and 3rd subframes is encoded with 9 bits * while that in 2nd and 4th subframes is relatively encoded with 6 bits. * Note that in relative encoding only 61 values are used. If the * decoder receives 61, 62, or 63 as the relative pitch index, it means * that a transmission error occurred. In this case, the pitch lag from - * previous subframe (actually from previous frame) is used. + * the previous subframe (actually from the previous frame) is used. * * @param avctx pointer to AVCodecContext * @param pitch_index received adaptive codebook (pitch) index @@ -662,9 +667,9 @@ static void decode_pitch_lag_6(AVCodecCo AMRContext *p = avctx->priv_data; int temp; - // FIXME - find out where these constants come from and add appropriate comments - // such that people reading the code can understand why these particular values - // are used + // FIXME - find out where these constants come from and add appropriate + // comments such that people reading the code can understand why these + // particular values are used. // subframe 1 or 3 if(p->cur_subframe & 1) { @@ -694,11 +699,12 @@ static void decode_pitch_lag_6(AVCodecCo /** - * Find the adaptive codebook (pitch) vector from interpolating the past excitation at the - * decoded pitch lag with corresponding resolution of 1/3 or 1/6 + * Find the adaptive codebook (pitch) vector from interpolating the past + * excitation at the decoded pitch lag with corresponding resolution + * of 1/3 or 1/6. * * See the specification for details of the underlying mathematics and the used - * FIR filter + * FIR filter. * * @param avctx pointer to AVCodecContext * @param excitation pointer to the excitation buffer @@ -732,7 +738,7 @@ static void decode_pitch_vector(AVCodecC /** - * Reconstruct the algebraic codebook vector + * Reconstruct the algebraic codebook vector. * * @param fixed_code algebraic codebook vector * @param pulse_position vector of pulse positions @@ -755,7 +761,7 @@ static void reconstruct_fixed_code(int * /** - * Decode the algebraic codebook index to pulse position indices for MODE_102 + * Decode the algebraic codebook index to pulse position indexes for MODE_102. * * @param fixed_index positions of the eight pulses (encoded) * @param position_index position index of the eight pulses @@ -765,7 +771,7 @@ static void fixed2position(int16_t *fixe { int MSBs, LSBs, MSBs0_24, divMSB; - // indices from track 1 (7+3 bits) + // indexes from track 1 (7+3 bits) MSBs= FFMIN(fixed_index[0] >> 3, 124); // LSBs = fixed_index[0]%8; LSBs = fixed_index[0] & 0x7; @@ -777,7 +783,7 @@ static void fixed2position(int16_t *fixe position_index[4] = (( (MSBs - 25*divMSB)/5 )<<1) + ((LSBs & 0x2)>>1); position_index[1] = (divMSB<<1) + (LSBs>>2); - // indices from track 2 (7+3 bits) + // indexes from track 2 (7+3 bits) MSBs= FFMIN(fixed_index[1] >> 3, 124); // LSBs = fixed_index[1]%8; LSBs = fixed_index[1] & 0x7; @@ -789,7 +795,7 @@ static void fixed2position(int16_t *fixe position_index[6] = (( (MSBs - 25*divMSB)/5 )<<1) + ((LSBs & 0x2)>>1); position_index[5] = (divMSB<<1) + (LSBs>>2); - // indices from track 3 (5+2 bits) + // indexes from track 3 (5+2 bits) MSBs = fixed_index[2] >> 2; // LSBs = fixed_index[2]%4; LSBs = fixed_index[2] & 0x3; @@ -806,8 +812,8 @@ static void fixed2position(int16_t *fixe /** - * Decode the algebraic codebook index to pulse positions and signs and construct - * the algebraic codebook vector for MODE_475 and MODE_515 + * Decode the algebraic codebook index to pulse positions and signs and + * construct the algebraic codebook vector for MODE_475 and MODE_515. * * @param avctx pointer to AVCodecContext * @param sign signs of the two pulses @@ -835,8 +841,8 @@ static void decode_2_pulses_9bits(AVCode /** - * Decode the algebraic codebook index to pulse positions and signs and construct - * the algebraic codebook vector for MODE_59 + * Decode the algebraic codebook index to pulse positions and signs + * and construct the algebraic codebook vector for MODE_59. * * @param sign signs of the two pulses * @param fixed_index positions of the two pulses @@ -867,8 +873,8 @@ static void decode_2_pulses_11bits(int s /** - * Decode the algebraic codebook index to pulse positions and signs and construct - * the algebraic codebook vector for MODE_67 + * Decode the algebraic codebook index to pulse positions and signs + * and construct the algebraic codebook vector for MODE_67. * * @param sign signs of the three pulses * @param fixed_index positions of the three pulses @@ -897,8 +903,8 @@ static void decode_3_pulses_14bits(int s /** - * Decode the algebraic codebook index to pulse positions and signs and construct - * the algebraic codebook vector for MODE_74 and MODE_795 + * Decode the algebraic codebook index to pulse positions and signs and + * construct the algebraic codebook vector for MODE_74 and MODE_795. * * @param sign signs of the four pulses * @param fixed_index positions of the four pulses @@ -927,8 +933,8 @@ static void decode_4_pulses_17bits(int s /** - * Decode the algebraic codebook index to pulse positions and signs and construct - * the algebraic codebook vector for MODE_102 + * Decode the algebraic codebook index to pulse positions and signs + * and construct the algebraic codebook vector for MODE_102. * * @param fixed_index positions of the eight pulses * @param fixed_code pointer to the algebraic codebook vector @@ -961,8 +967,8 @@ static void decode_8_pulses_31bits(int16 /** - * Decode the algebraic codebook index to pulse positions and signs and construct - * the algebraic codebook vector for MODE_122 + * Decode the algebraic codebook index to pulse positions and signs + * and construct the algebraic codebook vector for MODE_122. * * @param fixed_index positions of the ten pulses * @param fixed_code pointer to the algebraic codebook vector @@ -979,7 +985,7 @@ static void decode_10_pulses_35bits(int1 // find the position of the ith pulse pos1 = dgray[fixed_index[i ] & 7]*5 + i; // find the sign of the ith pulse - sign = (fixed_index[i] & 8) ? -4096 : 4096; // +/-1 : 4096 is used here in the ref source + sign = (fixed_index[i] & 8) ? -4096 : 4096; // +/-1 : 4096 is used here in the reference source // find the position of the i+5th pulse pos2 = dgray[fixed_index[i+5] & 7]*5 + i; // assign the ith pulse (+/-1) to its appropriate position @@ -995,10 +1001,10 @@ static void decode_10_pulses_35bits(int1 // general functions FIXME - useful enough to put into libavutil? /** - * Comparison function for use with qsort + * Comparison function for use with qsort. * - * @param a First value for comparison - * @param b Second value for comparison + * @param a first value for comparison + * @param b second value for comparison * @return a-b : the result of the comparison */ @@ -1008,11 +1014,11 @@ int qsort_compare(const int *a, const in } /** - * Find the median of some values + * Find the median of some values. * * @param values pointer to the values of which to find the median * @param n number of values - * @return Returns the median value + * @return the median value */ static int median(int *values, int n) @@ -1030,10 +1036,10 @@ static int median(int *values, int n) // gain functions /** - * Calculate the pitch gain from previous values + * Calculate the pitch gain from previous values. * * @param state_ptr pointer to the current state - * @return Returns the pitch gain + * @return the pitch gain */ static int find_pitch_gain(AMRDecoderState *state_ptr) @@ -1052,11 +1058,11 @@ static int find_pitch_gain(AMRDecoderSta /** - * Decode the pitch gain using the received index + * Decode the pitch gain using the received index. * * @param mode current mode - * @param index quantisation index - * @return Returns the pitch gain + * @param index quantization index + * @return the pitch gain */ static int decode_pitch_gain(enum Mode mode, int index) @@ -1075,7 +1081,7 @@ static int decode_pitch_gain(enum Mode m /** - * Update the pitch gain and limit pitch_gain if the previous frame was bad + * Update the pitch gain and limit pitch_gain if the previous frame was bad. * * @param state_ptr pointer to the current state * @param bad_frame_indicator bad frame indicator @@ -1113,7 +1119,7 @@ static void pitch_gain_update(AMRDecoder /** - * Reset the AMR frame parameters + * Reset the AMR frame parameters. * * @param avctx pointer to the AVCodecContext for AMR */ @@ -1139,7 +1145,7 @@ static int amrnb_decode_frame(AVCodecCon const int16_t *homing_frame; // pointer to the homing frame int homing_frame_size; // homing frame size - // decode the bitstream to amr parameters + // decode the bitstream to AMR parameters p->cur_frame_mode = decode_bitstream(avctx, buf, buf_size, &speech_mode); // guess the mode from the previous frame if no data or bad data @@ -1267,14 +1273,15 @@ static int amrnb_decode_frame(AVCodecCon p->prev_frame_type = p->cur_frame_type; p->prev_frame_mode = p->cur_frame_mode; - /* To make it easy the stream can only be 16 bits mono, so let's convert it to that */ + /* To make it easy the stream can only be 16 bits mono, + * so let's convert it to that */ for (i=0 ; i<buf_size; i++) outbuffer[i] = (int16_t)p->sample_buffer[i]; /* Report how many samples we got */ *data_size = buf_size; - /* Return the amount of bytes consumed if everything was ok */ + /* Return the amount of bytes consumed if everything was OK */ return *data_size*sizeof(int16_t); } @@ -1288,7 +1295,7 @@ static int amrnb_decode_close(AVCodecCon av_free(p->amr_prms); av_free(p->state); - /* Return 0 if everything is ok, -1 if not */ + /* Return 0 if everything is OK, -1 if not */ return 0; } Modified: amr/amrnbfloatdata.h ============================================================================== --- amr/amrnbfloatdata.h Sun Jan 25 16:35:48 2009 (r3999) +++ amr/amrnbfloatdata.h Sun Jan 25 16:38:52 2009 (r4000) @@ -98,9 +98,9 @@ typedef struct AMROrder { /**************************** tables *****************************/ -// The following order* tables are used to reorder the bitstream into the bit allocation -// format as in the specification. The bits are grouped into three classes of importance. -// See the specification, for more information. +// The following order* tables are used to reorder the bitstream into the bit +// allocation format as in the specification. The bits are grouped into three +// classes of importance. See the specification for more information. static const AMROrder order_MODE_475[95] = { { 0, 7}, { 0, 6}, { 0, 5}, { 0, 4}, { 0, 3}, { 0, 2}, { 0, 1}, { 0, 0}, @@ -1814,7 +1814,7 @@ static const uint8_t gray_decode[8] = { #define MIN_ENERGY -14.0 -// scalar quantised pitch gain table for 7.95 and 12.2 kbps modes +// scalar quantized pitch gain table for 7.95 and 12.2 kbps modes static const float qua_gain_pit[16] = { 0.0 , 0.20001220703125, 0.400146484375 , 0.5 , 0.5999755859375 , 0.70001220703125, 0.75 , 0.79998779296875, @@ -1822,10 +1822,11 @@ static const float qua_gain_pit[16] = { 1.04998779296875, 1.0999755859375 , 1.1500244140625 , 1.20001220703125, }; -// scalar quantised fixed gain table for 7.95 and 12.2 kbps modes -// note: this table was generated from the integer qua_gain_code tables in the -// ref source that store Q10 qua_ener = 20*log10(g_fac). The table below stores -// g_fac as a float from printf("%10.6f,\n", powf(10.0, qua_ener/(20.0*(1<<10)))) +// scalar quantized fixed gain table for 7.95 and 12.2 kbps modes +// NOTE: This table was generated from the integer qua_gain_code tables +// in the reference source that store Q10 qua_ener = 20*log10(g_fac). +// The table below stores g_fac as a float from +// printf("%10.6f,\n", powf(10.0, qua_ener/(20.0*(1<<10)))). static const float qua_gain_code[32] = { 0.077640, 0.100586, 0.130857, 0.170410, 0.204592, 0.235358, 0.270507, 0.311044, 0.357898, 0.411116, 0.473151, 0.543935, 0.625519, 0.719258, 0.827138, 0.951199, @@ -1841,7 +1842,7 @@ static const float energy_mean[8] = { 33 static const float energy_pred_fac[4] = { 0.68, 0.58, 0.34, 0.19 }; // gain table for 4.75 kbps mode -// first index has even/odd indices for subframes 0,2/1,3 +// first index has even/odd indexes for subframes 0,2/1,3 // second index is {pitch_gain, fixed_gain_factor} static const float gains_MODE_475[512][2] = { {0.049561, 0.031250}, {0.033081, 0.034180}, {0.175354, 0.277100}, {0.138306, 0.830566}, Modified: amr/amrnbfloatdec.c ============================================================================== --- amr/amrnbfloatdec.c Sun Jan 25 16:35:48 2009 (r3999) +++ amr/amrnbfloatdec.c Sun Jan 25 16:38:52 2009 (r4000) @@ -41,7 +41,7 @@ typedef struct AMRContext { GetBitContext gb; - int16_t amr_prms[PRMS_MODE_122]; ///< decoded amr parameters (lsf coefficients, codebook indices, etc) + int16_t amr_prms[PRMS_MODE_122]; ///< decoded AMR parameters (lsf coefficients, codebook indexes, etc) int bad_frame_indicator; ///< bad frame ? 1 : 0 int cur_frame_mode; ///< current frame mode int cur_frame_type; ///< current frame type @@ -52,7 +52,7 @@ typedef struct AMRContext { float lsp_avg[LP_FILTER_ORDER]; ///< vector of averaged lsp coefficients - float lpc[4][LP_FILTER_ORDER]; ///< vectors of lpc coefficients for 4 subframes + float lpc[4][LP_FILTER_ORDER]; ///< lpc coefficient vectors for 4 subframes int search_range_min; ///< minimum pitch lag search range int search_range_max; ///< maximum pitch lag search range @@ -90,9 +90,9 @@ static void reset_state(AMRContext *p) { int i; - // initialise values for the lsp vector from the 4th subframe of the - // previous subframe values - // taken from Decoder_amr_reset using val/(float)(1<<15) + // Initialize values for the lsp vector from the 4th subframe of the + // previous subframe values. + // Taken from Decoder_amr_reset using val/(float)(1<<15). p->prev_lsp_sub4[0] = 0.91552734375; p->prev_lsp_sub4[1] = 0.79345703125; p->prev_lsp_sub4[2] = 0.640869140625; @@ -104,8 +104,8 @@ static void reset_state(AMRContext *p) p->prev_lsp_sub4[8] = -0.640869140625; p->prev_lsp_sub4[9] = -0.79345703125; - // initialise mean lsp values - // taken from Decoder_amr_reset using val/(float)(1<<15) + // Initialize mean lsp values. + // Taken from Decoder_amr_reset using val/(float)(1<<15). p->lsp_avg[0] = 0.042236328125; p->lsp_avg[1] = 0.063385009765625; p->lsp_avg[2] = 0.1043701171875; @@ -148,13 +148,13 @@ static int amrnb_decode_init(AVCodecCont /** - * Decode the bitstream into the AMR parameters and discover the frame mode + * Decode the bitstream into the AMR parameters and discover the frame mode. * * @param buf pointer to the input buffer * @param buf_size size of the input buffer * @param speech_mode pointer to the speech mode * - * @return Returns the frame mode + * @return the frame mode */ enum Mode decode_bitstream(AMRContext *p, uint8_t *buf, int buf_size, @@ -167,7 +167,7 @@ enum Mode decode_bitstream(AMRContext *p // reset amr_prms memset(p->amr_prms, 0, PRMS_MODE_122 * sizeof(int16_t)); - // initialise get_bits + // initialize get_bits init_get_bits(&p->gb, buf, buf_size*8); skip_bits(&p->gb, 1); // set the mode @@ -242,7 +242,7 @@ enum Mode decode_bitstream(AMRContext *p /*** LPC coefficient decoding functions ***/ /** - * Convert an lsf vector into an lsp vector + * Convert an lsf vector into an lsp vector. * * @param lsf input lsf vector * @param lsp output lsp vector @@ -258,7 +258,7 @@ static void lsf2lsp(float *lsf, float *l } /** - * Decode a set of 5 split-matrix quantised lsf indices into an lsp vector + * Decode a set of 5 split-matrix quantized lsf indexes into an lsp vector. * * @param p pointer to the AMRContext */ @@ -319,7 +319,7 @@ static void lsf2lsp_5(AMRContext *p) } /** - * Decode a set of 3 split-matrix quantised lsf indices into an lsp vector + * Decode a set of 3 split-matrix quantized lsf indexes into an lsp vector. * * @param p pointer to the AMRContext */ @@ -351,8 +351,8 @@ static void lsf2lsp_3(AMRContext *p) lsf_r[2] = lsf_3_1_tmp[ idx ][2]; idx = p->amr_prms[1]; - // MODE_475, MODE_515 only use every other entry as their indices are stored - // using 1 less bit (8-bits vs 9-bits) + // MODE_475, MODE_515 only use every other entry as their indexes + // are stored using 1 less bit (8 bits vs 9 bits) if((p->cur_frame_mode == MODE_475) || (p->cur_frame_mode == MODE_515)) { idx <<= 1; } @@ -377,7 +377,7 @@ static void lsf2lsp_3(AMRContext *p) } /** - * Interpolate lsp vectors for subframes 1 and 3 + * Interpolate lsp vectors for subframes 1 and 3. * * @param p pointer to the AMRContext */ @@ -393,7 +393,7 @@ static void interp_lsp_13(AMRContext *p) } /** - * Interpolate lsp vectors for subframes 1, 2 and 3 + * Interpolate lsp vectors for subframes 1, 2 and 3. * * @param p pointer to the AMRContext */ @@ -410,7 +410,7 @@ static void interp_lsp_123(AMRContext *p } /** - * Find the polynomial F1(z) or F2(z) from the lsp vectors + * Find the polynomial F1(z) or F2(z) from the lsp vectors. * * @param lsp input lsp vector * @param f pointer to the polynomial F1(z) or F2(z) @@ -433,7 +433,7 @@ static void lsp2poly(float *lsp, float * } /** - * Convert an lsp vector to lpc coefficients + * Convert an lsp vector to lpc coefficients. * * @param lsp input lsp vector * @param lpc output lpc coefficients @@ -457,8 +457,8 @@ static void lsp2lpc(float *lsp, float *l // A(z) = ( F1'(z) + F2'(z) )/2 // note f1 and f2 are actually f1' and f2' for(i=0; i<5; i++) { - lpc_coeffs[i] = 0.5*(f1[i+2] + f2[i+2]); // lpc 0..4 uses indices to f, 2..6 - lpc_coeffs[i+5] = 0.5*(f1[6-i] - f2[6-i]); // lpc 5..9 uses indices to f, 6..2 + lpc_coeffs[i] = 0.5*(f1[i+2] + f2[i+2]); // lpc 0..4 uses indexes to f, 2..6 + lpc_coeffs[i+5] = 0.5*(f1[6-i] - f2[6-i]); // lpc 5..9 uses indexes to f, 6..2 } } @@ -468,8 +468,8 @@ static void lsp2lpc(float *lsp, float *l /*** pitch vector decoding functions ***/ /** - * Decode the adaptive codebook index to the integer and fractional parts of the - * pitch lag for one subframe at 1/3 resolution + * Decode the adaptive codebook index to the integer and fractional parts + * of the pitch lag for one subframe at 1/3 resolution. * * @param p pointer to the AMRContext * @param pitch_index parsed adaptive codebook (pitch) index @@ -492,7 +492,7 @@ static void decode_pitch_lag_3(AMRContex }else { if( (p->cur_frame_mode == MODE_475) || (p->cur_frame_mode == MODE_515) || (p->cur_frame_mode == MODE_59) || (p->cur_frame_mode == MODE_67) ) { - // decoding with 4 bit resolution + // decoding with 4-bit resolution int t1_temp = FFMAX(FFMIN(p->prev_pitch_lag_int, p->search_range_max-4), p->search_range_min+5); if(pitch_index < 4) { @@ -519,8 +519,8 @@ static void decode_pitch_lag_3(AMRContex } /** - * Decode the adaptive codebook index to the integer and fractional parts of the - * pitch lag for one subframe at 1/6 resolution + * Decode the adaptive codebook index to the integer and fractional parts + * of the pitch lag for one subframe at 1/6 resolution. * * @param p pointer to the AMRContext * @param pitch_index parsed adaptive codebook (pitch) index @@ -550,7 +550,7 @@ static void decode_pitch_lag_6(AMRContex /** * Calculate the pitch vector by interpolating the past excitation at the pitch - * pitch lag using a b60 hamming windowed sinc function + * pitch lag using a b60 hamming windowed sinc function. * * @param prev_excitation pointer to the element after the previous excitations * @param lag_int integer part of pitch lag @@ -599,7 +599,7 @@ static void interp_pitch_vector(float *p /*** algebraic code book (fixed) vector decoding functions ***/ /** - * Reconstruct the algebraic codebook vector + * Reconstruct the algebraic codebook vector. * * @param pulse_position vector of pulse positions * @param sign signs of the pulses @@ -620,8 +620,8 @@ static void reconstruct_fixed_vector(int } /** - * Decode the algebraic codebook index to pulse positions and signs and construct - * the algebraic codebook vector for MODE_475 and MODE_515 + * Decode the algebraic codebook index to pulse positions and signs and + * construct the algebraic codebook vector for MODE_475 and MODE_515. * * @param fixed_index positions of the two pulses * @param sign signs of the two pulses @@ -647,8 +647,8 @@ static void decode_2_pulses_9bits(int fi } /** - * Decode the algebraic codebook index to pulse positions and signs and construct - * the algebraic codebook vector for MODE_59 + * Decode the algebraic codebook index to pulse positions and signs and + * construct the algebraic codebook vector for MODE_59. * * @param fixed_index positions of the two pulses * @param sign signs of the two pulses @@ -676,8 +676,8 @@ static void decode_2_pulses_11bits(int f } /** - * Decode the algebraic codebook index to pulse positions and signs and construct - * the algebraic codebook vector for MODE_67 + * Decode the algebraic codebook index to pulse positions and signs and + * construct the algebraic codebook vector for MODE_67. * * @param fixed_index positions of the three pulses * @param sign signs of the three pulses @@ -706,8 +706,8 @@ static void decode_3_pulses_14bits(int f } /** - * Decode the algebraic codebook index to pulse positions and signs and construct - * the algebraic codebook vector for MODE_74 and MODE_795 + * Decode the algebraic codebook index to pulse positions and signs and + * construct the algebraic codebook vector for MODE_74 and MODE_795. * * @param fixed_index positions of the four pulses * @param sign signs of the four pulses @@ -736,8 +736,8 @@ static void decode_4_pulses_17bits(int f } /** - * Decode the algebraic codebook index to pulse positions and signs and construct - * the algebraic codebook vector for MODE_102 + * Decode the algebraic codebook index to pulse positions and signs and + * construct the algebraic codebook vector for MODE_102. * * @param fixed_index positions of the eight pulses * @param fixed_vector pointer to the algebraic codebook vector @@ -791,8 +791,8 @@ static void decode_8_pulses_31bits(int16 } /** - * Decode the algebraic codebook index to pulse positions and signs and construct - * the algebraic codebook vector for MODE_122 + * Decode the algebraic codebook index to pulse positions and signs and + * construct the algebraic codebook vector for MODE_122. * * @param fixed_index positions of the ten pulses * @param fixed_vector pointer to the algebraic codebook vector @@ -827,12 +827,12 @@ static void decode_10_pulses_35bits(int1 /*** gain decoding functions ***/ /** - * Predict the fixed gain + * Predict the fixed gain. * * @param fixed_vector pointer to the algebraic codebook vector * @param prev_pred_error pointer to the quantified prediction errors from the previous four subframes * - * @return Returns the predicted fixed gain + * @return the predicted fixed gain */ static float fixed_gain_prediction(float *fixed_vector, float *prev_pred_error, @@ -863,10 +863,10 @@ static float fixed_gain_prediction(float /*** pre-processing functions ***/ /** - * Comparison function for use with qsort + * Comparison function for use with qsort. * - * @param a First value for comparison - * @param b Second value for comparison + * @param a first value for comparison + * @param b second value for comparison * @return a-b : the result of the comparison */ @@ -881,11 +881,11 @@ int qsort_compare(const float *a, const } /** - * Find the median of some float values + * Find the median of some float values. * * @param values pointer to the values of which to find the median * @param n number of values - * @return Returns the median value + * @return the median value */ static float medianf(float *values, int n) @@ -905,7 +905,7 @@ static float medianf(float *values, int /** * Circularly convolve the fixed vector with a phase dispersion impulse response - * filter + * filter. * * @param fixed_vector pointer to the fixed vector * @param ir_filter pointer to the impulse response filter @@ -945,7 +945,7 @@ static void convolve_circ(float *fixed_v /*** synthesis functions ***/ /** - * Conduct 10th order linear predictive coding synthesis + * Conduct 10th order linear predictive coding synthesis. * * @param p pointer to the AMRContext * @param excitation pointer to the excitation vector @@ -980,7 +980,7 @@ static int synthesis(AMRContext *p, floa float eta, temp1 = 0.0, temp2 = 0.0; for(i=0; i<AMR_SUBFRAME_SIZE; i++) { - // emphasise pitch vector contribution + // emphasize pitch vector contribution excitation_temp[i] = excitation[i] + pitch_factor*p->pitch_vector[i]; // find gain scale temp1 += excitation[i]*excitation[i]; @@ -999,7 +999,7 @@ static int synthesis(AMRContext *p, floa for(j=0; j<LP_FILTER_ORDER; j++) { samples[i] -= lpc[j]*samples[i-j-1]; } - // Detect overflow + // detect overflow if(fabsf(samples[i])>1.0) { overflow_temp = 1; samples[i] = av_clipf(samples[i], -1.0, 1.0); @@ -1015,7 +1015,7 @@ static int synthesis(AMRContext *p, floa /*** update functions ***/ /** - * Update buffers and history at the end of decoding a subframe + * Update buffers and history at the end of decoding a subframe. * * @param p pointer to the AMRContext */ @@ -1066,20 +1066,20 @@ static int amrnb_decode_frame(AVCodecCon int gains_index_MODE_475 = 0; // MODE_475 gains index coded every other subframe enum Mode speech_mode = MODE_475; // ??? - // decode the bitstream to amr parameters + // decode the bitstream to AMR parameters p->cur_frame_mode = decode_bitstream(p, buf, buf_size, &speech_mode); /*** LPC coefficient decoding ***/ if(p->cur_frame_mode == MODE_122) { - // decode split-matrix quantised lsf vector indices to lsp vectors + // decode split-matrix quantized lsf vector indexes to lsp vectors lsf2lsp_5(p); // interpolate LSP vectors at subframes 1 and 3 interp_lsp_13(p); // advance index into amr_prms index += 5; }else { - // decode split-matrix quantised lsf vector indices to an lsp vector + // decode split-matrix quantized lsf vector indexes to an lsp vector lsf2lsp_3(p); // interpolate LSP vectors at subframes 1, 2 and 3 interp_lsp_123(p); @@ -1305,10 +1305,10 @@ static int amrnb_decode_frame(AVCodecCon } - /* Report how many samples we got */ + /* report how many samples we got */ *data_size = AMR_BLOCK_SIZE * sizeof(int16_t); - /* Return the amount of bytes consumed if everything was ok */ + /* return the amount of bytes consumed if everything was OK */ return (mode_bits[p->cur_frame_mode] + 15)>>3; // +7 for rounding and +8 for TOC } @@ -1317,7 +1317,7 @@ static int amrnb_decode_close(AVCodecCon { AMRContext *p = avctx->priv_data; - /* Return 0 if everything is ok, -1 if not */ + /* return 0 if everything is OK, -1 if not */ return 0; }