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Diffstat (limited to 'ffmpeg1/libavcodec/amrnbdec.c')
| -rw-r--r-- | ffmpeg1/libavcodec/amrnbdec.c | 1094 |
1 files changed, 0 insertions, 1094 deletions
diff --git a/ffmpeg1/libavcodec/amrnbdec.c b/ffmpeg1/libavcodec/amrnbdec.c deleted file mode 100644 index 6376db1..0000000 --- a/ffmpeg1/libavcodec/amrnbdec.c +++ /dev/null @@ -1,1094 +0,0 @@ -/* - * AMR narrowband decoder - * Copyright (c) 2006-2007 Robert Swain - * Copyright (c) 2009 Colin McQuillan - * - * This file is part of FFmpeg. - * - * FFmpeg is free software; you can redistribute it and/or - * modify it under the terms of the GNU Lesser General Public - * License as published by the Free Software Foundation; either - * version 2.1 of the License, or (at your option) any later version. - * - * FFmpeg is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Lesser General Public License for more details. - * - * You should have received a copy of the GNU Lesser General Public - * License along with FFmpeg; if not, write to the Free Software - * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA - */ - - -/** - * @file - * AMR narrowband decoder - * - * This decoder uses floats for simplicity and so is not bit-exact. One - * difference is that differences in phase can accumulate. The test sequences - * in 3GPP TS 26.074 can still be useful. - * - * - Comparing this file's output to the output of the ref decoder gives a - * PSNR of 30 to 80. Plotting the output samples shows a difference in - * phase in some areas. - * - * - Comparing both decoders against their input, this decoder gives a similar - * PSNR. If the test sequence homing frames are removed (this decoder does - * not detect them), the PSNR is at least as good as the reference on 140 - * out of 169 tests. - */ - - -#include <string.h> -#include <math.h> - -#include "libavutil/channel_layout.h" -#include "libavutil/float_dsp.h" -#include "avcodec.h" -#include "libavutil/common.h" -#include "libavutil/avassert.h" -#include "celp_math.h" -#include "celp_filters.h" -#include "acelp_filters.h" -#include "acelp_vectors.h" -#include "acelp_pitch_delay.h" -#include "lsp.h" -#include "amr.h" -#include "internal.h" - -#include "amrnbdata.h" - -#define AMR_BLOCK_SIZE 160 ///< samples per frame -#define AMR_SAMPLE_BOUND 32768.0 ///< threshold for synthesis overflow - -/** - * Scale from constructed speech to [-1,1] - * - * AMR is designed to produce 16-bit PCM samples (3GPP TS 26.090 4.2) but - * upscales by two (section 6.2.2). - * - * Fundamentally, this scale is determined by energy_mean through - * the fixed vector contribution to the excitation vector. - */ -#define AMR_SAMPLE_SCALE (2.0 / 32768.0) - -/** Prediction factor for 12.2kbit/s mode */ -#define PRED_FAC_MODE_12k2 0.65 - -#define LSF_R_FAC (8000.0 / 32768.0) ///< LSF residual tables to Hertz -#define MIN_LSF_SPACING (50.0488 / 8000.0) ///< Ensures stability of LPC filter -#define PITCH_LAG_MIN_MODE_12k2 18 ///< Lower bound on decoded lag search in 12.2kbit/s mode - -/** Initial energy in dB. Also used for bad frames (unimplemented). */ -#define MIN_ENERGY -14.0 - -/** Maximum sharpening factor - * - * The specification says 0.8, which should be 13107, but the reference C code - * uses 13017 instead. (Amusingly the same applies to SHARP_MAX in g729dec.c.) - */ -#define SHARP_MAX 0.79449462890625 - -/** Number of impulse response coefficients used for tilt factor */ -#define AMR_TILT_RESPONSE 22 -/** Tilt factor = 1st reflection coefficient * gamma_t */ -#define AMR_TILT_GAMMA_T 0.8 -/** Adaptive gain control factor used in post-filter */ -#define AMR_AGC_ALPHA 0.9 - -typedef struct AMRContext { - AMRNBFrame frame; ///< decoded AMR parameters (lsf coefficients, codebook indexes, etc) - uint8_t bad_frame_indicator; ///< bad frame ? 1 : 0 - enum Mode cur_frame_mode; - - int16_t prev_lsf_r[LP_FILTER_ORDER]; ///< residual LSF vector from previous subframe - double lsp[4][LP_FILTER_ORDER]; ///< lsp vectors from current frame - double prev_lsp_sub4[LP_FILTER_ORDER]; ///< lsp vector for the 4th subframe of the previous frame - - float lsf_q[4][LP_FILTER_ORDER]; ///< Interpolated LSF vector for fixed gain smoothing - float lsf_avg[LP_FILTER_ORDER]; ///< vector of averaged lsf vector - - float lpc[4][LP_FILTER_ORDER]; ///< lpc coefficient vectors for 4 subframes - - uint8_t pitch_lag_int; ///< integer part of pitch lag from current subframe - - float excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1 + AMR_SUBFRAME_SIZE]; ///< current excitation and all necessary excitation history - float *excitation; ///< pointer to the current excitation vector in excitation_buf - - float pitch_vector[AMR_SUBFRAME_SIZE]; ///< adaptive code book (pitch) vector - float fixed_vector[AMR_SUBFRAME_SIZE]; ///< algebraic codebook (fixed) vector (must be kept zero between frames) - - float prediction_error[4]; ///< quantified prediction errors {20log10(^gamma_gc)} for previous four subframes - float pitch_gain[5]; ///< quantified pitch gains for the current and previous four subframes - float fixed_gain[5]; ///< quantified fixed gains for the current and previous four subframes - - float beta; ///< previous pitch_gain, bounded by [0.0,SHARP_MAX] - uint8_t diff_count; ///< the number of subframes for which diff has been above 0.65 - uint8_t hang_count; ///< the number of subframes since a hangover period started - - float prev_sparse_fixed_gain; ///< previous fixed gain; used by anti-sparseness processing to determine "onset" - uint8_t prev_ir_filter_nr; ///< previous impulse response filter "impNr": 0 - strong, 1 - medium, 2 - none - uint8_t ir_filter_onset; ///< flag for impulse response filter strength - - float postfilter_mem[10]; ///< previous intermediate values in the formant filter - float tilt_mem; ///< previous input to tilt compensation filter - float postfilter_agc; ///< previous factor used for adaptive gain control - float high_pass_mem[2]; ///< previous intermediate values in the high-pass filter - - float samples_in[LP_FILTER_ORDER + AMR_SUBFRAME_SIZE]; ///< floating point samples - - ACELPFContext acelpf_ctx; ///< context for filters for ACELP-based codecs - ACELPVContext acelpv_ctx; ///< context for vector operations for ACELP-based codecs - CELPFContext celpf_ctx; ///< context for filters for CELP-based codecs - CELPMContext celpm_ctx; ///< context for fixed point math operations - -} AMRContext; - -/** Double version of ff_weighted_vector_sumf() */ -static void weighted_vector_sumd(double *out, const double *in_a, - const double *in_b, double weight_coeff_a, - double weight_coeff_b, int length) -{ - int i; - - for (i = 0; i < length; i++) - out[i] = weight_coeff_a * in_a[i] - + weight_coeff_b * in_b[i]; -} - -static av_cold int amrnb_decode_init(AVCodecContext *avctx) -{ - AMRContext *p = avctx->priv_data; - int i; - - if (avctx->channels > 1) { - avpriv_report_missing_feature(avctx, "multi-channel AMR"); - return AVERROR_PATCHWELCOME; - } - - avctx->channels = 1; - avctx->channel_layout = AV_CH_LAYOUT_MONO; - if (!avctx->sample_rate) - avctx->sample_rate = 8000; - avctx->sample_fmt = AV_SAMPLE_FMT_FLT; - - // p->excitation always points to the same position in p->excitation_buf - p->excitation = &p->excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1]; - - for (i = 0; i < LP_FILTER_ORDER; i++) { - p->prev_lsp_sub4[i] = lsp_sub4_init[i] * 1000 / (float)(1 << 15); - p->lsf_avg[i] = p->lsf_q[3][i] = lsp_avg_init[i] / (float)(1 << 15); - } - - for (i = 0; i < 4; i++) - p->prediction_error[i] = MIN_ENERGY; - - ff_acelp_filter_init(&p->acelpf_ctx); - ff_acelp_vectors_init(&p->acelpv_ctx); - ff_celp_filter_init(&p->celpf_ctx); - ff_celp_math_init(&p->celpm_ctx); - - return 0; -} - - -/** - * Unpack an RFC4867 speech frame into the AMR frame mode and parameters. - * - * The order of speech bits is specified by 3GPP TS 26.101. - * - * @param p the context - * @param buf pointer to the input buffer - * @param buf_size size of the input buffer - * - * @return the frame mode - */ -static enum Mode unpack_bitstream(AMRContext *p, const uint8_t *buf, - int buf_size) -{ - enum Mode mode; - - // Decode the first octet. - mode = buf[0] >> 3 & 0x0F; // frame type - p->bad_frame_indicator = (buf[0] & 0x4) != 0x4; // quality bit - - if (mode >= N_MODES || buf_size < frame_sizes_nb[mode] + 1) { - return NO_DATA; - } - - if (mode < MODE_DTX) - ff_amr_bit_reorder((uint16_t *) &p->frame, sizeof(AMRNBFrame), buf + 1, - amr_unpacking_bitmaps_per_mode[mode]); - - return mode; -} - - -/// @name AMR pitch LPC coefficient decoding functions -/// @{ - -/** - * Interpolate the LSF vector (used for fixed gain smoothing). - * The interpolation is done over all four subframes even in MODE_12k2. - * - * @param[in] ctx The Context - * @param[in,out] lsf_q LSFs in [0,1] for each subframe - * @param[in] lsf_new New LSFs in [0,1] for subframe 4 - */ -static void interpolate_lsf(ACELPVContext *ctx, float lsf_q[4][LP_FILTER_ORDER], float *lsf_new) -{ - int i; - - for (i = 0; i < 4; i++) - ctx->weighted_vector_sumf(lsf_q[i], lsf_q[3], lsf_new, - 0.25 * (3 - i), 0.25 * (i + 1), - LP_FILTER_ORDER); -} - -/** - * Decode a set of 5 split-matrix quantized lsf indexes into an lsp vector. - * - * @param p the context - * @param lsp output LSP vector - * @param lsf_no_r LSF vector without the residual vector added - * @param lsf_quantizer pointers to LSF dictionary tables - * @param quantizer_offset offset in tables - * @param sign for the 3 dictionary table - * @param update store data for computing the next frame's LSFs - */ -static void lsf2lsp_for_mode12k2(AMRContext *p, double lsp[LP_FILTER_ORDER], - const float lsf_no_r[LP_FILTER_ORDER], - const int16_t *lsf_quantizer[5], - const int quantizer_offset, - const int sign, const int update) -{ - int16_t lsf_r[LP_FILTER_ORDER]; // residual LSF vector - float lsf_q[LP_FILTER_ORDER]; // quantified LSF vector - int i; - - for (i = 0; i < LP_FILTER_ORDER >> 1; i++) - memcpy(&lsf_r[i << 1], &lsf_quantizer[i][quantizer_offset], - 2 * sizeof(*lsf_r)); - - if (sign) { - lsf_r[4] *= -1; - lsf_r[5] *= -1; - } - - if (update) - memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(*lsf_r)); - - for (i = 0; i < LP_FILTER_ORDER; i++) - lsf_q[i] = lsf_r[i] * (LSF_R_FAC / 8000.0) + lsf_no_r[i] * (1.0 / 8000.0); - - ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER); - - if (update) - interpolate_lsf(&p->acelpv_ctx, p->lsf_q, lsf_q); - - ff_acelp_lsf2lspd(lsp, lsf_q, LP_FILTER_ORDER); -} - -/** - * Decode a set of 5 split-matrix quantized lsf indexes into 2 lsp vectors. - * - * @param p pointer to the AMRContext - */ -static void lsf2lsp_5(AMRContext *p) -{ - const uint16_t *lsf_param = p->frame.lsf; - float lsf_no_r[LP_FILTER_ORDER]; // LSFs without the residual vector - const int16_t *lsf_quantizer[5]; - int i; - - lsf_quantizer[0] = lsf_5_1[lsf_param[0]]; - lsf_quantizer[1] = lsf_5_2[lsf_param[1]]; - lsf_quantizer[2] = lsf_5_3[lsf_param[2] >> 1]; - lsf_quantizer[3] = lsf_5_4[lsf_param[3]]; - lsf_quantizer[4] = lsf_5_5[lsf_param[4]]; - - for (i = 0; i < LP_FILTER_ORDER; i++) - lsf_no_r[i] = p->prev_lsf_r[i] * LSF_R_FAC * PRED_FAC_MODE_12k2 + lsf_5_mean[i]; - - lsf2lsp_for_mode12k2(p, p->lsp[1], lsf_no_r, lsf_quantizer, 0, lsf_param[2] & 1, 0); - lsf2lsp_for_mode12k2(p, p->lsp[3], lsf_no_r, lsf_quantizer, 2, lsf_param[2] & 1, 1); - - // interpolate LSP vectors at subframes 1 and 3 - weighted_vector_sumd(p->lsp[0], p->prev_lsp_sub4, p->lsp[1], 0.5, 0.5, LP_FILTER_ORDER); - weighted_vector_sumd(p->lsp[2], p->lsp[1] , p->lsp[3], 0.5, 0.5, LP_FILTER_ORDER); -} - -/** - * Decode a set of 3 split-matrix quantized lsf indexes into an lsp vector. - * - * @param p pointer to the AMRContext - */ -static void lsf2lsp_3(AMRContext *p) -{ - const uint16_t *lsf_param = p->frame.lsf; - int16_t lsf_r[LP_FILTER_ORDER]; // residual LSF vector - float lsf_q[LP_FILTER_ORDER]; // quantified LSF vector - const int16_t *lsf_quantizer; - int i, j; - - lsf_quantizer = (p->cur_frame_mode == MODE_7k95 ? lsf_3_1_MODE_7k95 : lsf_3_1)[lsf_param[0]]; - memcpy(lsf_r, lsf_quantizer, 3 * sizeof(*lsf_r)); - - lsf_quantizer = lsf_3_2[lsf_param[1] << (p->cur_frame_mode <= MODE_5k15)]; - memcpy(lsf_r + 3, lsf_quantizer, 3 * sizeof(*lsf_r)); - - lsf_quantizer = (p->cur_frame_mode <= MODE_5k15 ? lsf_3_3_MODE_5k15 : lsf_3_3)[lsf_param[2]]; - memcpy(lsf_r + 6, lsf_quantizer, 4 * sizeof(*lsf_r)); - - // calculate mean-removed LSF vector and add mean - for (i = 0; i < LP_FILTER_ORDER; i++) - lsf_q[i] = (lsf_r[i] + p->prev_lsf_r[i] * pred_fac[i]) * (LSF_R_FAC / 8000.0) + lsf_3_mean[i] * (1.0 / 8000.0); - - ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER); - - // store data for computing the next frame's LSFs - interpolate_lsf(&p->acelpv_ctx, p->lsf_q, lsf_q); - memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(*lsf_r)); - - ff_acelp_lsf2lspd(p->lsp[3], lsf_q, LP_FILTER_ORDER); - - // interpolate LSP vectors at subframes 1, 2 and 3 - for (i = 1; i <= 3; i++) - for(j = 0; j < LP_FILTER_ORDER; j++) - p->lsp[i-1][j] = p->prev_lsp_sub4[j] + - (p->lsp[3][j] - p->prev_lsp_sub4[j]) * 0.25 * i; -} - -/// @} - - -/// @name AMR pitch vector decoding functions -/// @{ - -/** - * Like ff_decode_pitch_lag(), but with 1/6 resolution - */ -static void decode_pitch_lag_1_6(int *lag_int, int *lag_frac, int pitch_index, - const int prev_lag_int, const int subframe) -{ - if (subframe == 0 || subframe == 2) { - if (pitch_index < 463) { - *lag_int = (pitch_index + 107) * 10923 >> 16; - *lag_frac = pitch_index - *lag_int * 6 + 105; - } else { - *lag_int = pitch_index - 368; - *lag_frac = 0; - } - } else { - *lag_int = ((pitch_index + 5) * 10923 >> 16) - 1; - *lag_frac = pitch_index - *lag_int * 6 - 3; - *lag_int += av_clip(prev_lag_int - 5, PITCH_LAG_MIN_MODE_12k2, - PITCH_DELAY_MAX - 9); - } -} - -static void decode_pitch_vector(AMRContext *p, - const AMRNBSubframe *amr_subframe, - const int subframe) -{ - int pitch_lag_int, pitch_lag_frac; - enum Mode mode = p->cur_frame_mode; - - if (p->cur_frame_mode == MODE_12k2) { - decode_pitch_lag_1_6(&pitch_lag_int, &pitch_lag_frac, - amr_subframe->p_lag, p->pitch_lag_int, - subframe); - } else - ff_decode_pitch_lag(&pitch_lag_int, &pitch_lag_frac, - amr_subframe->p_lag, - p->pitch_lag_int, subframe, - mode != MODE_4k75 && mode != MODE_5k15, - mode <= MODE_6k7 ? 4 : (mode == MODE_7k95 ? 5 : 6)); - - p->pitch_lag_int = pitch_lag_int; // store previous lag in a uint8_t - - pitch_lag_frac <<= (p->cur_frame_mode != MODE_12k2); - - pitch_lag_int += pitch_lag_frac > 0; - - /* Calculate the pitch vector by interpolating the past excitation at the - pitch lag using a b60 hamming windowed sinc function. */ - p->acelpf_ctx.acelp_interpolatef(p->excitation, - p->excitation + 1 - pitch_lag_int, - ff_b60_sinc, 6, - pitch_lag_frac + 6 - 6*(pitch_lag_frac > 0), - 10, AMR_SUBFRAME_SIZE); - - memcpy(p->pitch_vector, p->excitation, AMR_SUBFRAME_SIZE * sizeof(float)); -} - -/// @} - - -/// @name AMR algebraic code book (fixed) vector decoding functions -/// @{ - -/** - * Decode a 10-bit algebraic codebook index from a 10.2 kbit/s frame. - */ -static void decode_10bit_pulse(int code, int pulse_position[8], - int i1, int i2, int i3) -{ - // coded using 7+3 bits with the 3 LSBs being, individually, the LSB of 1 of - // the 3 pulses and the upper 7 bits being coded in base 5 - const uint8_t *positions = base_five_table[code >> 3]; - pulse_position[i1] = (positions[2] << 1) + ( code & 1); - pulse_position[i2] = (positions[1] << 1) + ((code >> 1) & 1); - pulse_position[i3] = (positions[0] << 1) + ((code >> 2) & 1); -} - -/** - * Decode the algebraic codebook index to pulse positions and signs and - * construct the algebraic codebook vector for MODE_10k2. - * - * @param fixed_index positions of the eight pulses - * @param fixed_sparse pointer to the algebraic codebook vector - */ -static void decode_8_pulses_31bits(const int16_t *fixed_index, - AMRFixed *fixed_sparse) -{ - int pulse_position[8]; - int i, temp; - - decode_10bit_pulse(fixed_index[4], pulse_position, 0, 4, 1); - decode_10bit_pulse(fixed_index[5], pulse_position, 2, 6, 5); - - // coded using 5+2 bits with the 2 LSBs being, individually, the LSB of 1 of - // the 2 pulses and the upper 5 bits being coded in base 5 - temp = ((fixed_index[6] >> 2) * 25 + 12) >> 5; - pulse_position[3] = temp % 5; - pulse_position[7] = temp / 5; - if (pulse_position[7] & 1) - pulse_position[3] = 4 - pulse_position[3]; - pulse_position[3] = (pulse_position[3] << 1) + ( fixed_index[6] & 1); - pulse_position[7] = (pulse_position[7] << 1) + ((fixed_index[6] >> 1) & 1); - - fixed_sparse->n = 8; - for (i = 0; i < 4; i++) { - const int pos1 = (pulse_position[i] << 2) + i; - const int pos2 = (pulse_position[i + 4] << 2) + i; - const float sign = fixed_index[i] ? -1.0 : 1.0; - fixed_sparse->x[i ] = pos1; - fixed_sparse->x[i + 4] = pos2; - fixed_sparse->y[i ] = sign; - fixed_sparse->y[i + 4] = pos2 < pos1 ? -sign : sign; - } -} - -/** - * Decode the algebraic codebook index to pulse positions and signs, - * then construct the algebraic codebook vector. - * - * nb of pulses | bits encoding pulses - * For MODE_4k75 or MODE_5k15, 2 | 1-3, 4-6, 7 - * MODE_5k9, 2 | 1, 2-4, 5-6, 7-9 - * MODE_6k7, 3 | 1-3, 4, 5-7, 8, 9-11 - * MODE_7k4 or MODE_7k95, 4 | 1-3, 4-6, 7-9, 10, 11-13 - * - * @param fixed_sparse pointer to the algebraic codebook vector - * @param pulses algebraic codebook indexes - * @param mode mode of the current frame - * @param subframe current subframe number - */ -static void decode_fixed_sparse(AMRFixed *fixed_sparse, const uint16_t *pulses, - const enum Mode mode, const int subframe) -{ - av_assert1(MODE_4k75 <= (signed)mode && mode <= MODE_12k2); - - if (mode == MODE_12k2) { - ff_decode_10_pulses_35bits(pulses, fixed_sparse, gray_decode, 5, 3); - } else if (mode == MODE_10k2) { - decode_8_pulses_31bits(pulses, fixed_sparse); - } else { - int *pulse_position = fixed_sparse->x; - int i, pulse_subset; - const int fixed_index = pulses[0]; - - if (mode <= MODE_5k15) { - pulse_subset = ((fixed_index >> 3) & 8) + (subframe << 1); - pulse_position[0] = ( fixed_index & 7) * 5 + track_position[pulse_subset]; - pulse_position[1] = ((fixed_index >> 3) & 7) * 5 + track_position[pulse_subset + 1]; - fixed_sparse->n = 2; - } else if (mode == MODE_5k9) { - pulse_subset = ((fixed_index & 1) << 1) + 1; - pulse_position[0] = ((fixed_index >> 1) & 7) * 5 + pulse_subset; - pulse_subset = (fixed_index >> 4) & 3; - pulse_position[1] = ((fixed_index >> 6) & 7) * 5 + pulse_subset + (pulse_subset == 3 ? 1 : 0); - fixed_sparse->n = pulse_position[0] == pulse_position[1] ? 1 : 2; - } else if (mode == MODE_6k7) { - pulse_position[0] = (fixed_index & 7) * 5; - pulse_subset = (fixed_index >> 2) & 2; - pulse_position[1] = ((fixed_index >> 4) & 7) * 5 + pulse_subset + 1; - pulse_subset = (fixed_index >> 6) & 2; - pulse_position[2] = ((fixed_index >> 8) & 7) * 5 + pulse_subset + 2; - fixed_sparse->n = 3; - } else { // mode <= MODE_7k95 - pulse_position[0] = gray_decode[ fixed_index & 7]; - pulse_position[1] = gray_decode[(fixed_index >> 3) & 7] + 1; - pulse_position[2] = gray_decode[(fixed_index >> 6) & 7] + 2; - pulse_subset = (fixed_index >> 9) & 1; - pulse_position[3] = gray_decode[(fixed_index >> 10) & 7] + pulse_subset + 3; - fixed_sparse->n = 4; - } - for (i = 0; i < fixed_sparse->n; i++) - fixed_sparse->y[i] = (pulses[1] >> i) & 1 ? 1.0 : -1.0; - } -} - -/** - * Apply pitch lag to obtain the sharpened fixed vector (section 6.1.2) - * - * @param p the context - * @param subframe unpacked amr subframe - * @param mode mode of the current frame - * @param fixed_sparse sparse respresentation of the fixed vector - */ -static void pitch_sharpening(AMRContext *p, int subframe, enum Mode mode, - AMRFixed *fixed_sparse) -{ - // The spec suggests the current pitch gain is always used, but in other - // modes the pitch and codebook gains are joinly quantized (sec 5.8.2) - // so the codebook gain cannot depend on the quantized pitch gain. - if (mode == MODE_12k2) - p->beta = FFMIN(p->pitch_gain[4], 1.0); - - fixed_sparse->pitch_lag = p->pitch_lag_int; - fixed_sparse->pitch_fac = p->beta; - - // Save pitch sharpening factor for the next subframe - // MODE_4k75 only updates on the 2nd and 4th subframes - this follows from - // the fact that the gains for two subframes are jointly quantized. - if (mode != MODE_4k75 || subframe & 1) - p->beta = av_clipf(p->pitch_gain[4], 0.0, SHARP_MAX); -} -/// @} - - -/// @name AMR gain decoding functions -/// @{ - -/** - * fixed gain smoothing - * Note that where the spec specifies the "spectrum in the q domain" - * in section 6.1.4, in fact frequencies should be used. - * - * @param p the context - * @param lsf LSFs for the current subframe, in the range [0,1] - * @param lsf_avg averaged LSFs - * @param mode mode of the current frame - * - * @return fixed gain smoothed - */ -static float fixed_gain_smooth(AMRContext *p , const float *lsf, - const float *lsf_avg, const enum Mode mode) -{ - float diff = 0.0; - int i; - - for (i = 0; i < LP_FILTER_ORDER; i++) - diff += fabs(lsf_avg[i] - lsf[i]) / lsf_avg[i]; - - // If diff is large for ten subframes, disable smoothing for a 40-subframe - // hangover period. - p->diff_count++; - if (diff <= 0.65) - p->diff_count = 0; - - if (p->diff_count > 10) { - p->hang_count = 0; - p->diff_count--; // don't let diff_count overflow - } - - if (p->hang_count < 40) { - p->hang_count++; - } else if (mode < MODE_7k4 || mode == MODE_10k2) { - const float smoothing_factor = av_clipf(4.0 * diff - 1.6, 0.0, 1.0); - const float fixed_gain_mean = (p->fixed_gain[0] + p->fixed_gain[1] + - p->fixed_gain[2] + p->fixed_gain[3] + - p->fixed_gain[4]) * 0.2; - return smoothing_factor * p->fixed_gain[4] + - (1.0 - smoothing_factor) * fixed_gain_mean; - } - return p->fixed_gain[4]; -} - -/** - * Decode pitch gain and fixed gain factor (part of section 6.1.3). - * - * @param p the context - * @param amr_subframe unpacked amr subframe - * @param mode mode of the current frame - * @param subframe current subframe number - * @param fixed_gain_factor decoded gain correction factor - */ -static void decode_gains(AMRContext *p, const AMRNBSubframe *amr_subframe, - const enum Mode mode, const int subframe, - float *fixed_gain_factor) -{ - if (mode == MODE_12k2 || mode == MODE_7k95) { - p->pitch_gain[4] = qua_gain_pit [amr_subframe->p_gain ] - * (1.0 / 16384.0); - *fixed_gain_factor = qua_gain_code[amr_subframe->fixed_gain] - * (1.0 / 2048.0); - } else { - const uint16_t *gains; - - if (mode >= MODE_6k7) { - gains = gains_high[amr_subframe->p_gain]; - } else if (mode >= MODE_5k15) { - gains = gains_low [amr_subframe->p_gain]; - } else { - // gain index is only coded in subframes 0,2 for MODE_4k75 - gains = gains_MODE_4k75[(p->frame.subframe[subframe & 2].p_gain << 1) + (subframe & 1)]; - } - - p->pitch_gain[4] = gains[0] * (1.0 / 16384.0); - *fixed_gain_factor = gains[1] * (1.0 / 4096.0); - } -} - -/// @} - - -/// @name AMR preprocessing functions -/// @{ - -/** - * Circularly convolve a sparse fixed vector with a phase dispersion impulse - * response filter (D.6.2 of G.729 and 6.1.5 of AMR). - * - * @param out vector with filter applied - * @param in source vector - * @param filter phase filter coefficients - * - * out[n] = sum(i,0,len-1){ in[i] * filter[(len + n - i)%len] } - */ -static void apply_ir_filter(float *out, const AMRFixed *in, - const float *filter) -{ - float filter1[AMR_SUBFRAME_SIZE], ///< filters at pitch lag*1 and *2 - filter2[AMR_SUBFRAME_SIZE]; - int lag = in->pitch_lag; - float fac = in->pitch_fac; - int i; - - if (lag < AMR_SUBFRAME_SIZE) { - ff_celp_circ_addf(filter1, filter, filter, lag, fac, - AMR_SUBFRAME_SIZE); - - if (lag < AMR_SUBFRAME_SIZE >> 1) - ff_celp_circ_addf(filter2, filter, filter1, lag, fac, - AMR_SUBFRAME_SIZE); - } - - memset(out, 0, sizeof(float) * AMR_SUBFRAME_SIZE); - for (i = 0; i < in->n; i++) { - int x = in->x[i]; - float y = in->y[i]; - const float *filterp; - - if (x >= AMR_SUBFRAME_SIZE - lag) { - filterp = filter; - } else if (x >= AMR_SUBFRAME_SIZE - (lag << 1)) { - filterp = filter1; - } else - filterp = filter2; - - ff_celp_circ_addf(out, out, filterp, x, y, AMR_SUBFRAME_SIZE); - } -} - -/** - * Reduce fixed vector sparseness by smoothing with one of three IR filters. - * Also know as "adaptive phase dispersion". - * - * This implements 3GPP TS 26.090 section 6.1(5). - * - * @param p the context - * @param fixed_sparse algebraic codebook vector - * @param fixed_vector unfiltered fixed vector - * @param fixed_gain smoothed gain - * @param out space for modified vector if necessary - */ -static const float *anti_sparseness(AMRContext *p, AMRFixed *fixed_sparse, - const float *fixed_vector, - float fixed_gain, float *out) -{ - int ir_filter_nr; - - if (p->pitch_gain[4] < 0.6) { - ir_filter_nr = 0; // strong filtering - } else if (p->pitch_gain[4] < 0.9) { - ir_filter_nr = 1; // medium filtering - } else - ir_filter_nr = 2; // no filtering - - // detect 'onset' - if (fixed_gain > 2.0 * p->prev_sparse_fixed_gain) { - p->ir_filter_onset = 2; - } else if (p->ir_filter_onset) - p->ir_filter_onset--; - - if (!p->ir_filter_onset) { - int i, count = 0; - - for (i = 0; i < 5; i++) - if (p->pitch_gain[i] < 0.6) - count++; - if (count > 2) - ir_filter_nr = 0; - - if (ir_filter_nr > p->prev_ir_filter_nr + 1) - ir_filter_nr--; - } else if (ir_filter_nr < 2) - ir_filter_nr++; - - // Disable filtering for very low level of fixed_gain. - // Note this step is not specified in the technical description but is in - // the reference source in the function Ph_disp. - if (fixed_gain < 5.0) - ir_filter_nr = 2; - - if (p->cur_frame_mode != MODE_7k4 && p->cur_frame_mode < MODE_10k2 - && ir_filter_nr < 2) { - apply_ir_filter(out, fixed_sparse, - (p->cur_frame_mode == MODE_7k95 ? - ir_filters_lookup_MODE_7k95 : - ir_filters_lookup)[ir_filter_nr]); - fixed_vector = out; - } - - // update ir filter strength history - p->prev_ir_filter_nr = ir_filter_nr; - p->prev_sparse_fixed_gain = fixed_gain; - - return fixed_vector; -} - -/// @} - - -/// @name AMR synthesis functions -/// @{ - -/** - * Conduct 10th order linear predictive coding synthesis. - * - * @param p pointer to the AMRContext - * @param lpc pointer to the LPC coefficients - * @param fixed_gain fixed codebook gain for synthesis - * @param fixed_vector algebraic codebook vector - * @param samples pointer to the output speech samples - * @param overflow 16-bit overflow flag - */ -static int synthesis(AMRContext *p, float *lpc, - float fixed_gain, const float *fixed_vector, - float *samples, uint8_t overflow) -{ - int i; - float excitation[AMR_SUBFRAME_SIZE]; - - // if an overflow has been detected, the pitch vector is scaled down by a - // factor of 4 - if (overflow) - for (i = 0; i < AMR_SUBFRAME_SIZE; i++) - p->pitch_vector[i] *= 0.25; - - p->acelpv_ctx.weighted_vector_sumf(excitation, p->pitch_vector, fixed_vector, - p->pitch_gain[4], fixed_gain, AMR_SUBFRAME_SIZE); - - // emphasize pitch vector contribution - if (p->pitch_gain[4] > 0.5 && !overflow) { - float energy = p->celpm_ctx.dot_productf(excitation, excitation, - AMR_SUBFRAME_SIZE); - float pitch_factor = - p->pitch_gain[4] * - (p->cur_frame_mode == MODE_12k2 ? - 0.25 * FFMIN(p->pitch_gain[4], 1.0) : - 0.5 * FFMIN(p->pitch_gain[4], SHARP_MAX)); - - for (i = 0; i < AMR_SUBFRAME_SIZE; i++) - excitation[i] += pitch_factor * p->pitch_vector[i]; - - ff_scale_vector_to_given_sum_of_squares(excitation, excitation, energy, - AMR_SUBFRAME_SIZE); - } - - p->celpf_ctx.celp_lp_synthesis_filterf(samples, lpc, excitation, - AMR_SUBFRAME_SIZE, - LP_FILTER_ORDER); - - // detect overflow - for (i = 0; i < AMR_SUBFRAME_SIZE; i++) - if (fabsf(samples[i]) > AMR_SAMPLE_BOUND) { - return 1; - } - - return 0; -} - -/// @} - - -/// @name AMR update functions -/// @{ - -/** - * Update buffers and history at the end of decoding a subframe. - * - * @param p pointer to the AMRContext - */ -static void update_state(AMRContext *p) -{ - memcpy(p->prev_lsp_sub4, p->lsp[3], LP_FILTER_ORDER * sizeof(p->lsp[3][0])); - - memmove(&p->excitation_buf[0], &p->excitation_buf[AMR_SUBFRAME_SIZE], - (PITCH_DELAY_MAX + LP_FILTER_ORDER + 1) * sizeof(float)); - - memmove(&p->pitch_gain[0], &p->pitch_gain[1], 4 * sizeof(float)); - memmove(&p->fixed_gain[0], &p->fixed_gain[1], 4 * sizeof(float)); - - memmove(&p->samples_in[0], &p->samples_in[AMR_SUBFRAME_SIZE], - LP_FILTER_ORDER * sizeof(float)); -} - -/// @} - - -/// @name AMR Postprocessing functions -/// @{ - -/** - * Get the tilt factor of a formant filter from its transfer function - * - * @param p The Context - * @param lpc_n LP_FILTER_ORDER coefficients of the numerator - * @param lpc_d LP_FILTER_ORDER coefficients of the denominator - */ -static float tilt_factor(AMRContext *p, float *lpc_n, float *lpc_d) -{ - float rh0, rh1; // autocorrelation at lag 0 and 1 - - // LP_FILTER_ORDER prior zeros are needed for ff_celp_lp_synthesis_filterf - float impulse_buffer[LP_FILTER_ORDER + AMR_TILT_RESPONSE] = { 0 }; - float *hf = impulse_buffer + LP_FILTER_ORDER; // start of impulse response - - hf[0] = 1.0; - memcpy(hf + 1, lpc_n, sizeof(float) * LP_FILTER_ORDER); - p->celpf_ctx.celp_lp_synthesis_filterf(hf, lpc_d, hf, - AMR_TILT_RESPONSE, - LP_FILTER_ORDER); - - rh0 = p->celpm_ctx.dot_productf(hf, hf, AMR_TILT_RESPONSE); - rh1 = p->celpm_ctx.dot_productf(hf, hf + 1, AMR_TILT_RESPONSE - 1); - - // The spec only specifies this check for 12.2 and 10.2 kbit/s - // modes. But in the ref source the tilt is always non-negative. - return rh1 >= 0.0 ? rh1 / rh0 * AMR_TILT_GAMMA_T : 0.0; -} - -/** - * Perform adaptive post-filtering to enhance the quality of the speech. - * See section 6.2.1. - * - * @param p pointer to the AMRContext - * @param lpc interpolated LP coefficients for this subframe - * @param buf_out output of the filter - */ -static void postfilter(AMRContext *p, float *lpc, float *buf_out) -{ - int i; - float *samples = p->samples_in + LP_FILTER_ORDER; // Start of input - - float speech_gain = p->celpm_ctx.dot_productf(samples, samples, - AMR_SUBFRAME_SIZE); - - float pole_out[AMR_SUBFRAME_SIZE + LP_FILTER_ORDER]; // Output of pole filter - const float *gamma_n, *gamma_d; // Formant filter factor table - float lpc_n[LP_FILTER_ORDER], lpc_d[LP_FILTER_ORDER]; // Transfer function coefficients - - if (p->cur_frame_mode == MODE_12k2 || p->cur_frame_mode == MODE_10k2) { - gamma_n = ff_pow_0_7; - gamma_d = ff_pow_0_75; - } else { - gamma_n = ff_pow_0_55; - gamma_d = ff_pow_0_7; - } - - for (i = 0; i < LP_FILTER_ORDER; i++) { - lpc_n[i] = lpc[i] * gamma_n[i]; - lpc_d[i] = lpc[i] * gamma_d[i]; - } - - memcpy(pole_out, p->postfilter_mem, sizeof(float) * LP_FILTER_ORDER); - p->celpf_ctx.celp_lp_synthesis_filterf(pole_out + LP_FILTER_ORDER, lpc_d, samples, - AMR_SUBFRAME_SIZE, LP_FILTER_ORDER); - memcpy(p->postfilter_mem, pole_out + AMR_SUBFRAME_SIZE, - sizeof(float) * LP_FILTER_ORDER); - - p->celpf_ctx.celp_lp_zero_synthesis_filterf(buf_out, lpc_n, - pole_out + LP_FILTER_ORDER, - AMR_SUBFRAME_SIZE, LP_FILTER_ORDER); - - ff_tilt_compensation(&p->tilt_mem, tilt_factor(p, lpc_n, lpc_d), buf_out, - AMR_SUBFRAME_SIZE); - - ff_adaptive_gain_control(buf_out, buf_out, speech_gain, AMR_SUBFRAME_SIZE, - AMR_AGC_ALPHA, &p->postfilter_agc); -} - -/// @} - -static int amrnb_decode_frame(AVCodecContext *avctx, void *data, - int *got_frame_ptr, AVPacket *avpkt) -{ - - AMRContext *p = avctx->priv_data; // pointer to private data - AVFrame *frame = data; - const uint8_t *buf = avpkt->data; - int buf_size = avpkt->size; - float *buf_out; // pointer to the output data buffer - int i, subframe, ret; - float fixed_gain_factor; - AMRFixed fixed_sparse = {0}; // fixed vector up to anti-sparseness processing - float spare_vector[AMR_SUBFRAME_SIZE]; // extra stack space to hold result from anti-sparseness processing - float synth_fixed_gain; // the fixed gain that synthesis should use - const float *synth_fixed_vector; // pointer to the fixed vector that synthesis should use - - /* get output buffer */ - frame->nb_samples = AMR_BLOCK_SIZE; - if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) - return ret; - buf_out = (float *)frame->data[0]; - - p->cur_frame_mode = unpack_bitstream(p, buf, buf_size); - if (p->cur_frame_mode == NO_DATA) { - av_log(avctx, AV_LOG_ERROR, "Corrupt bitstream\n"); - return AVERROR_INVALIDDATA; - } - if (p->cur_frame_mode == MODE_DTX) { - avpriv_report_missing_feature(avctx, "dtx mode"); - av_log(avctx, AV_LOG_INFO, "Note: libopencore_amrnb supports dtx\n"); - return AVERROR_PATCHWELCOME; - } - - if (p->cur_frame_mode == MODE_12k2) { - lsf2lsp_5(p); - } else - lsf2lsp_3(p); - - for (i = 0; i < 4; i++) - ff_acelp_lspd2lpc(p->lsp[i], p->lpc[i], 5); - - for (subframe = 0; subframe < 4; subframe++) { - const AMRNBSubframe *amr_subframe = &p->frame.subframe[subframe]; - - decode_pitch_vector(p, amr_subframe, subframe); - - decode_fixed_sparse(&fixed_sparse, amr_subframe->pulses, - p->cur_frame_mode, subframe); - - // The fixed gain (section 6.1.3) depends on the fixed vector - // (section 6.1.2), but the fixed vector calculation uses - // pitch sharpening based on the on the pitch gain (section 6.1.3). - // So the correct order is: pitch gain, pitch sharpening, fixed gain. - decode_gains(p, amr_subframe, p->cur_frame_mode, subframe, - &fixed_gain_factor); - - pitch_sharpening(p, subframe, p->cur_frame_mode, &fixed_sparse); - - if (fixed_sparse.pitch_lag == 0) { - av_log(avctx, AV_LOG_ERROR, "The file is corrupted, pitch_lag = 0 is not allowed\n"); - return AVERROR_INVALIDDATA; - } - ff_set_fixed_vector(p->fixed_vector, &fixed_sparse, 1.0, - AMR_SUBFRAME_SIZE); - - p->fixed_gain[4] = - ff_amr_set_fixed_gain(fixed_gain_factor, - p->celpm_ctx.dot_productf(p->fixed_vector, - p->fixed_vector, - AMR_SUBFRAME_SIZE) / - AMR_SUBFRAME_SIZE, - p->prediction_error, - energy_mean[p->cur_frame_mode], energy_pred_fac); - - // The excitation feedback is calculated without any processing such - // as fixed gain smoothing. This isn't mentioned in the specification. - for (i = 0; i < AMR_SUBFRAME_SIZE; i++) - p->excitation[i] *= p->pitch_gain[4]; - ff_set_fixed_vector(p->excitation, &fixed_sparse, p->fixed_gain[4], - AMR_SUBFRAME_SIZE); - - // In the ref decoder, excitation is stored with no fractional bits. - // This step prevents buzz in silent periods. The ref encoder can - // emit long sequences with pitch factor greater than one. This - // creates unwanted feedback if the excitation vector is nonzero. - // (e.g. test sequence T19_795.COD in 3GPP TS 26.074) - for (i = 0; i < AMR_SUBFRAME_SIZE; i++) - p->excitation[i] = truncf(p->excitation[i]); - - // Smooth fixed gain. - // The specification is ambiguous, but in the reference source, the - // smoothed value is NOT fed back into later fixed gain smoothing. - synth_fixed_gain = fixed_gain_smooth(p, p->lsf_q[subframe], - p->lsf_avg, p->cur_frame_mode); - - synth_fixed_vector = anti_sparseness(p, &fixed_sparse, p->fixed_vector, - synth_fixed_gain, spare_vector); - - if (synthesis(p, p->lpc[subframe], synth_fixed_gain, - synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 0)) - // overflow detected -> rerun synthesis scaling pitch vector down - // by a factor of 4, skipping pitch vector contribution emphasis - // and adaptive gain control - synthesis(p, p->lpc[subframe], synth_fixed_gain, - synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 1); - - postfilter(p, p->lpc[subframe], buf_out + subframe * AMR_SUBFRAME_SIZE); - - // update buffers and history - ff_clear_fixed_vector(p->fixed_vector, &fixed_sparse, AMR_SUBFRAME_SIZE); - update_state(p); - } - - p->acelpf_ctx.acelp_apply_order_2_transfer_function(buf_out, - buf_out, highpass_zeros, - highpass_poles, - highpass_gain * AMR_SAMPLE_SCALE, - p->high_pass_mem, AMR_BLOCK_SIZE); - - /* Update averaged lsf vector (used for fixed gain smoothing). - * - * Note that lsf_avg should not incorporate the current frame's LSFs - * for fixed_gain_smooth. - * The specification has an incorrect formula: the reference decoder uses - * qbar(n-1) rather than qbar(n) in section 6.1(4) equation 71. */ - p->acelpv_ctx.weighted_vector_sumf(p->lsf_avg, p->lsf_avg, p->lsf_q[3], - 0.84, 0.16, LP_FILTER_ORDER); - - *got_frame_ptr = 1; - - /* return the amount of bytes consumed if everything was OK */ - return frame_sizes_nb[p->cur_frame_mode] + 1; // +7 for rounding and +8 for TOC -} - - -AVCodec ff_amrnb_decoder = { - .name = "amrnb", - .type = AVMEDIA_TYPE_AUDIO, - .id = AV_CODEC_ID_AMR_NB, - .priv_data_size = sizeof(AMRContext), - .init = amrnb_decode_init, - .decode = amrnb_decode_frame, - .capabilities = CODEC_CAP_DR1, - .long_name = NULL_IF_CONFIG_SMALL("AMR-NB (Adaptive Multi-Rate NarrowBand)"), - .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT, - AV_SAMPLE_FMT_NONE }, -}; |