[FFmpeg-soc] [PATCH] AMR-WB Decoder

[Marcelo Galvão Póvoa]

On 11 September 2010 12:37, Vitor Sessak <vitor1001 at gmail.com> wrote:
> On 09/11/2010 04:47 PM, Marcelo Galvão Póvoa wrote:
>>
>> On 9 September 2010 16:11, Vitor Sessak<vitor1001 at gmail.com>  wrote:
>>>
>>> On 09/07/2010 02:05 AM, Marcelo Galvão Póvoa wrote:
>>>>
>>>> On 6 September 2010 11:31, Vitor Sessak<vitor1001 at gmail.com>    wrote:
>>>>>
>>>>> On 09/06/2010 02:46 AM, Marcelo Galvăo Póvoa wrote:
>>>>>>
>
> [...]
>
>>>>> Can celp_filters.c:ff_celp_circ_addf() simplify this?
>>>>>
>>>>
>>>> I already gave some thought to that but I couldn't figure out how to
>>>> use ff_celp_circ_addf() there. I wrote the algorithm the simplest way
>>>> I could think of.
>>>
>>> Try
>>>
>>>        for (i = 0; i<  AMRWB_SFR_SIZE; i++)
>>>            if (fixed_vector[i])
>>>                ff_celp_circ_addf(buf, buf, coef, i, fixed_vector[i],
>>>                                  AMRWB_SFR_SIZE);
>>>
>>>
>>
>> That's right indeed, thanks!
>
>> /*
>>  * AMR wideband decoder
>>  * Copyright (c) 2010 Marcelo Galvao Povoa
>>  *
>>  * 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
>>  */
>>
>> #include "libavutil/lfg.h"
>>
>> #include "avcodec.h"
>> #include "get_bits.h"
>> #include "lsp.h"
>> #include "celp_math.h"
>> #include "celp_filters.h"
>> #include "acelp_filters.h"
>> #include "acelp_vectors.h"
>> #include "acelp_pitch_delay.h"
>>
>> #include "amrwbdata.h"
>
> No need to include amr.h?
>

Oops, fixed

>>
>> typedef struct {
>>    AMRWBFrame                             frame; ///< AMRWB parameters
>> decoded from bitstream
>>    enum Mode                        fr_cur_mode; ///< mode index of
>> current frame
>>    uint8_t                           fr_quality; ///< frame quality index
>> (FQI)
>>    float                      isf_cur[LP_ORDER]; ///< working ISF vector
>> from current frame
>>    float                   isf_q_past[LP_ORDER]; ///< quantized ISF vector
>> of the previous frame
>>    float               isf_past_final[LP_ORDER]; ///< final processed ISF
>> vector of the previous frame
>>    double                      isp[4][LP_ORDER]; ///< ISP vectors from
>> current frame
>>    double               isp_sub4_past[LP_ORDER]; ///< ISP vector for the
>> 4th subframe of the previous frame
>>
>>    float                   lp_coef[4][LP_ORDER]; ///< Linear Prediction
>> Coefficients from ISP vector
>>
>>    uint8_t                       base_pitch_lag; ///< integer part of
>> pitch lag for the next relative subframe
>>    uint8_t                        pitch_lag_int; ///< integer part of
>> pitch lag of the previous subframe
>>
>>    float excitation_buf[AMRWB_P_DELAY_MAX + LP_ORDER + 2 +
>> AMRWB_SFR_SIZE]; ///< current excitation and all necessary excitation
>> history
>>    float                            *excitation; ///< points to current
>> excitation in excitation_buf[]
>>
>>    float           pitch_vector[AMRWB_SFR_SIZE]; ///< adaptive codebook
>> (pitch) vector for current subframe
>>    float           fixed_vector[AMRWB_SFR_SIZE]; ///< algebraic codebook
>> (fixed) vector for current subframe
>>
>>    float                    prediction_error[4]; ///< quantified
>> prediction errors {20log10(^gamma_gc)} for previous four subframes
>>    float                          pitch_gain[6]; ///< quantified pitch
>> gains for the current and previous five subframes
>>    float                          fixed_gain[2]; ///< quantified fixed
>> gains for the current and previous subframes
>>
>>    float                              tilt_coef; ///< {beta_1} related to
>> the voicing of the previous subframe
>>
>>    float                 prev_sparse_fixed_gain; ///< previous fixed gain;
>> used by anti-sparseness to determine "onset"
>>    uint8_t                    prev_ir_filter_nr; ///< previous impulse
>> response filter "impNr": 0 - strong, 1 - medium, 2 - none
>>    float                           prev_tr_gain; ///< previous initial
>> gain used by noise enhancer for threshold
>>
>>    float samples_az[LP_ORDER + AMRWB_SFR_SIZE];         ///< low-band
>> samples and memory from synthesis at 12.8kHz
>>    float samples_up[UPS_MEM_SIZE + AMRWB_SFR_SIZE];     ///< low-band
>> samples and memory processed for upsampling
>>    float samples_hb[LP_ORDER_16k + AMRWB_SFR_SIZE_16k]; ///< high-band
>> samples and memory from synthesis at 16kHz
>>
>
>>    float          hpf_31_mem[4], hpf_400_mem[4]; ///< previous values in
>> the high pass filters
>
> I think now just two elements for the filter memory should be enough.
>

Right, fixed

>> diff --git a/libavcodec/amrwbdata.h b/libavcodec/amrwbdata.h
>> new file mode 100644
>> index 0000000..6344763
>> --- /dev/null
>> +++ b/libavcodec/amrwbdata.h
>
>> /** [i][j] is the number of pulses present in track j at mode i */
>> static const int pulses_nb_per_mode_tr[][4] = {
>>    {1, 1, 0, 0}, {1, 1, 1, 1}, {2, 2, 2, 2},
>>    {3, 3, 2, 2}, {3, 3, 3, 3}, {4, 4, 4, 4},
>>    {5, 5, 4, 4}, {6, 6, 6, 6}, {6, 6, 6, 6}
>> };
>
> uint8_t?
>

Fine, fixed

-- 
Marcelo
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