[FFmpeg-devel] [PATCH 1/2][RFC] avcodec/g729dec: move definitions to header file
Ganesh Ajjanagadde
gajjanagadde at gmail.com
Tue Aug 11 14:09:53 CEST 2015
On Tue, Aug 11, 2015 at 3:03 AM, Paul B Mahol <onemda at gmail.com> wrote:
>
> Dana 11. 8. 2015. 03:51 osoba "Ganesh Ajjanagadde" <gajjanagadde at gmail.com>
> napisala je:
>
>
>>
>> Moves structure definitions and related macros to g729dec.h
>> Also exports format in priv_data
>>
>> Signed-off-by: Ganesh Ajjanagadde <gajjanagadde at gmail.com>
>> ---
>> libavcodec/g729dec.c | 111 ++---------------------------------------
>> libavcodec/g729dec.h | 138
>> +++++++++++++++++++++++++++++++++++++++++++++++++++
>> 2 files changed, 141 insertions(+), 108 deletions(-)
>> create mode 100644 libavcodec/g729dec.h
>>
>> diff --git a/libavcodec/g729dec.c b/libavcodec/g729dec.c
>> index 99053ad..e5b2de0 100644
>> --- a/libavcodec/g729dec.c
>> +++ b/libavcodec/g729dec.c
>> @@ -30,6 +30,7 @@
>>
>>
>> #include "g729.h"
>> +#include "g729dec.h"
>> #include "lsp.h"
>> #include "celp_math.h"
>> #include "celp_filters.h"
>> @@ -39,57 +40,6 @@
>> #include "g729data.h"
>> #include "g729postfilter.h"
>>
>> -/**
>> - * minimum quantized LSF value (3.2.4)
>> - * 0.005 in Q13
>> - */
>> -#define LSFQ_MIN 40
>> -
>> -/**
>> - * maximum quantized LSF value (3.2.4)
>> - * 3.135 in Q13
>> - */
>> -#define LSFQ_MAX 25681
>> -
>> -/**
>> - * minimum LSF distance (3.2.4)
>> - * 0.0391 in Q13
>> - */
>> -#define LSFQ_DIFF_MIN 321
>> -
>> -/// interpolation filter length
>> -#define INTERPOL_LEN 11
>> -
>> -/**
>> - * minimum gain pitch value (3.8, Equation 47)
>> - * 0.2 in (1.14)
>> - */
>> -#define SHARP_MIN 3277
>> -
>> -/**
>> - * maximum gain pitch value (3.8, Equation 47)
>> - * (EE) This does not comply with the specification.
>> - * Specification says about 0.8, which should be
>> - * 13107 in (1.14), but reference C code uses
>> - * 13017 (equals to 0.7945) instead of it.
>> - */
>> -#define SHARP_MAX 13017
>> -
>> -/**
>> - * MR_ENERGY (mean removed energy) = mean_energy + 10 * log10(2^26 *
>> subframe_size) in (7.13)
>> - */
>> -#define MR_ENERGY 1018156
>> -
>> -#define DECISION_NOISE 0
>> -#define DECISION_INTERMEDIATE 1
>> -#define DECISION_VOICE 2
>> -
>> -typedef enum {
>> - FORMAT_G729_8K = 0,
>> - FORMAT_G729D_6K4,
>> - FORMAT_COUNT,
>> -} G729Formats;
>> -
>> typedef struct {
>> uint8_t ac_index_bits[2]; ///< adaptive codebook index for second
>> subframe (size in bits)
>> uint8_t parity_bit; ///< parity bit for pitch delay
>> @@ -99,61 +49,6 @@ typedef struct {
>> uint8_t fc_indexes_bits; ///< size (in bits) of fixed-codebook
>> index entry
>> } G729FormatDescription;
>>
>> -typedef struct {
>> - AudioDSPContext adsp;
>> -
>> - /// past excitation signal buffer
>> - int16_t exc_base[2*SUBFRAME_SIZE+PITCH_DELAY_MAX+INTERPOL_LEN];
>> -
>> - int16_t* exc; ///< start of past excitation data in
>> buffer
>> - int pitch_delay_int_prev; ///< integer part of previous subframe's
>> pitch delay (4.1.3)
>> -
>> - /// (2.13) LSP quantizer outputs
>> - int16_t past_quantizer_output_buf[MA_NP + 1][10];
>> - int16_t* past_quantizer_outputs[MA_NP + 1];
>> -
>> - int16_t lsfq[10]; ///< (2.13) quantized LSF coefficients
>> from previous frame
>> - int16_t lsp_buf[2][10]; ///< (0.15) LSP coefficients (previous
>> and current frames) (3.2.5)
>> - int16_t *lsp[2]; ///< pointers to lsp_buf
>> -
>> - int16_t quant_energy[4]; ///< (5.10) past quantized energy
>> -
>> - /// previous speech data for LP synthesis filter
>> - int16_t syn_filter_data[10];
>> -
>> -
>> - /// residual signal buffer (used in long-term postfilter)
>> - int16_t residual[SUBFRAME_SIZE + RES_PREV_DATA_SIZE];
>> -
>> - /// previous speech data for residual calculation filter
>> - int16_t res_filter_data[SUBFRAME_SIZE+10];
>> -
>> - /// previous speech data for short-term postfilter
>> - int16_t pos_filter_data[SUBFRAME_SIZE+10];
>> -
>> - /// (1.14) pitch gain of current and five previous subframes
>> - int16_t past_gain_pitch[6];
>> -
>> - /// (14.1) gain code from current and previous subframe
>> - int16_t past_gain_code[2];
>> -
>> - /// voice decision on previous subframe (0-noise, 1-intermediate,
>> 2-voice), G.729D
>> - int16_t voice_decision;
>> -
>> - int16_t onset; ///< detected onset level (0-2)
>> - int16_t was_periodic; ///< whether previous frame was declared
>> as periodic or not (4.4)
>> - int16_t ht_prev_data; ///< previous data for 4.2.3, equation 86
>> - int gain_coeff; ///< (1.14) gain coefficient (4.2.4)
>> - uint16_t rand_value; ///< random number generator value
>> (4.4.4)
>> - int ma_predictor_prev; ///< switched MA predictor of LSP
>> quantizer from last good frame
>> -
>> - /// (14.14) high-pass filter data (past input)
>> - int hpf_f[2];
>> -
>> - /// high-pass filter data (past output)
>> - int16_t hpf_z[2];
>> -} G729Context;
>> -
>> static const G729FormatDescription format_g729_8k = {
>> .ac_index_bits = {8,5},
>> .parity_bit = 1,
>> @@ -422,14 +317,14 @@ static int decode_frame(AVCodecContext *avctx, void
>> *data, int *got_frame_ptr,
>> out_frame = (int16_t*) frame->data[0];
>>
>> if (buf_size % 10 == 0) {
>> - packet_type = FORMAT_G729_8K;
>> + ctx->format = packet_type = FORMAT_G729_8K;
>> format = &format_g729_8k;
>> //Reset voice decision
>> ctx->onset = 0;
>> ctx->voice_decision = DECISION_VOICE;
>> av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729 @
>> 8kbit/s");
>> } else if (buf_size == 8) {
>> - packet_type = FORMAT_G729D_6K4;
>> + ctx->format = packet_type = FORMAT_G729D_6K4;
>> format = &format_g729d_6k4;
>> av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729D @
>> 6.4kbit/s");
>> } else {
>> diff --git a/libavcodec/g729dec.h b/libavcodec/g729dec.h
>> new file mode 100644
>> index 0000000..06cbebf
>> --- /dev/null
>> +++ b/libavcodec/g729dec.h
>> @@ -0,0 +1,138 @@
>> +/*
>> + * G.729 decoder
>> + * Copyright (C) 2015 Ganesh Ajjanagadde
>
> Unacceptable, moving things around can not give you right to do this.
Ok, I will preserve the original author's copyright.
>
>> + *
>> + * 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
>> + */
>> +
>> +#ifndef AVCODEC_G729DEC_H
>> +#define AVCODEC_G729DEC_H
>> +
>> +#include "acelp_pitch_delay.h"
>> +#include "g729data.h"
>> +#include "g729postfilter.h"
>> +
>> +typedef enum {
>> + FORMAT_G729_8K = 0,
>> + FORMAT_G729D_6K4,
>> + FORMAT_COUNT,
>> +} G729Formats;
>> +
>> +/**
>> + * minimum quantized LSF value (3.2.4)
>> + * 0.005 in Q13
>> + */
>> +#define LSFQ_MIN 40
>> +
>> +/**
>> + * maximum quantized LSF value (3.2.4)
>> + * 3.135 in Q13
>> + */
>> +#define LSFQ_MAX 25681
>> +
>> +/**
>> + * minimum LSF distance (3.2.4)
>> + * 0.0391 in Q13
>> + */
>> +#define LSFQ_DIFF_MIN 321
>> +
>> +/// interpolation filter length
>> +#define INTERPOL_LEN 11
>> +
>> +/**
>> + * minimum gain pitch value (3.8, Equation 47)
>> + * 0.2 in (1.14)
>> + */
>> +#define SHARP_MIN 3277
>> +
>> +/**
>> + * maximum gain pitch value (3.8, Equation 47)
>> + * (EE) This does not comply with the specification.
>> + * Specification says about 0.8, which should be
>> + * 13107 in (1.14), but reference C code uses
>> + * 13017 (equals to 0.7945) instead of it.
>> + */
>> +#define SHARP_MAX 13017
>> +
>> +/**
>> + * MR_ENERGY (mean removed energy) = mean_energy + 10 * log10(2^26 *
>> subframe_size) in (7.13)
>> + */
>> +#define MR_ENERGY 1018156
>> +
>> +#define DECISION_NOISE 0
>> +#define DECISION_INTERMEDIATE 1
>> +#define DECISION_VOICE 2
>> +
>> +typedef struct {
>> + AudioDSPContext adsp;
>> +
>> + /// past excitation signal buffer
>> + int16_t exc_base[2*SUBFRAME_SIZE+PITCH_DELAY_MAX+INTERPOL_LEN];
>> +
>> + int16_t* exc; ///< start of past excitation data in
>> buffer
>> + int pitch_delay_int_prev; ///< integer part of previous subframe's
>> pitch delay (4.1.3)
>> +
>> + /// (2.13) LSP quantizer outputs
>> + int16_t past_quantizer_output_buf[MA_NP + 1][10];
>> + int16_t* past_quantizer_outputs[MA_NP + 1];
>> +
>> + int16_t lsfq[10]; ///< (2.13) quantized LSF coefficients
>> from previous frame
>> + int16_t lsp_buf[2][10]; ///< (0.15) LSP coefficients (previous
>> and current frames) (3.2.5)
>> + int16_t *lsp[2]; ///< pointers to lsp_buf
>> +
>> + int16_t quant_energy[4]; ///< (5.10) past quantized energy
>> +
>> + /// previous speech data for LP synthesis filter
>> + int16_t syn_filter_data[10];
>> +
>> +
>> + /// residual signal buffer (used in long-term postfilter)
>> + int16_t residual[SUBFRAME_SIZE + RES_PREV_DATA_SIZE];
>> +
>> + /// previous speech data for residual calculation filter
>> + int16_t res_filter_data[SUBFRAME_SIZE+10];
>> +
>> + /// previous speech data for short-term postfilter
>> + int16_t pos_filter_data[SUBFRAME_SIZE+10];
>> +
>> + /// (1.14) pitch gain of current and five previous subframes
>> + int16_t past_gain_pitch[6];
>> +
>> + /// (14.1) gain code from current and previous subframe
>> + int16_t past_gain_code[2];
>> +
>> + /// voice decision on previous subframe (0-noise, 1-intermediate,
>> 2-voice), G.729D
>> + int16_t voice_decision;
>> +
>> + int16_t onset; ///< detected onset level (0-2)
>> + int16_t was_periodic; ///< whether previous frame was declared
>> as periodic or not (4.4)
>> + int16_t ht_prev_data; ///< previous data for 4.2.3, equation 86
>> + int gain_coeff; ///< (1.14) gain coefficient (4.2.4)
>> + uint16_t rand_value; ///< random number generator value
>> (4.4.4)
>> + int ma_predictor_prev; ///< switched MA predictor of LSP
>> quantizer from last good frame
>> +
>> + /// (14.14) high-pass filter data (past input)
>> + int hpf_f[2];
>> +
>> + /// high-pass filter data (past output)
>> + int16_t hpf_z[2];
>> +
>> + /// format type
>> + G729Formats format;
>> +} G729Context;
>> +
>> +#endif /* AVCODEC_G729DEC_H */
>> --
>> 2.5.0
>>
>> _______________________________________________
>> ffmpeg-devel mailing list
>> ffmpeg-devel at ffmpeg.org
>> http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
More information about the ffmpeg-devel
mailing list