| /* |
| * 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)); |
| pitch_lag_frac *= 2; |
| } |
| |
| p->pitch_lag_int = pitch_lag_int; // store previous lag in a uint8_t |
| |
| 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 representation 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 jointly 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", |
| .long_name = NULL_IF_CONFIG_SMALL("AMR-NB (Adaptive Multi-Rate NarrowBand)"), |
| .type = AVMEDIA_TYPE_AUDIO, |
| .id = AV_CODEC_ID_AMR_NB, |
| .priv_data_size = sizeof(AMRContext), |
| .init = amrnb_decode_init, |
| .decode = amrnb_decode_frame, |
| .capabilities = AV_CODEC_CAP_DR1, |
| .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT, |
| AV_SAMPLE_FMT_NONE }, |
| }; |
