Commit 2383021a authored by James Almer's avatar James Almer

avcodec/aptx: split decoder and encoder into separate files

Signed-off-by: 's avatarJames Almer <jamrial@gmail.com>
parent a8a05340
...@@ -194,10 +194,10 @@ OBJS-$(CONFIG_AMV_ENCODER) += mjpegenc.o mjpegenc_common.o \ ...@@ -194,10 +194,10 @@ OBJS-$(CONFIG_AMV_ENCODER) += mjpegenc.o mjpegenc_common.o \
OBJS-$(CONFIG_ANM_DECODER) += anm.o OBJS-$(CONFIG_ANM_DECODER) += anm.o
OBJS-$(CONFIG_ANSI_DECODER) += ansi.o cga_data.o OBJS-$(CONFIG_ANSI_DECODER) += ansi.o cga_data.o
OBJS-$(CONFIG_APE_DECODER) += apedec.o OBJS-$(CONFIG_APE_DECODER) += apedec.o
OBJS-$(CONFIG_APTX_DECODER) += aptx.o OBJS-$(CONFIG_APTX_DECODER) += aptxdec.o aptx.o
OBJS-$(CONFIG_APTX_ENCODER) += aptx.o OBJS-$(CONFIG_APTX_ENCODER) += aptxenc.o aptx.o
OBJS-$(CONFIG_APTX_HD_DECODER) += aptx.o OBJS-$(CONFIG_APTX_HD_DECODER) += aptxdec.o aptx.o
OBJS-$(CONFIG_APTX_HD_ENCODER) += aptx.o OBJS-$(CONFIG_APTX_HD_ENCODER) += aptxenc.o aptx.o
OBJS-$(CONFIG_APNG_DECODER) += png.o pngdec.o pngdsp.o OBJS-$(CONFIG_APNG_DECODER) += png.o pngdec.o pngdsp.o
OBJS-$(CONFIG_APNG_ENCODER) += png.o pngenc.o OBJS-$(CONFIG_APNG_ENCODER) += png.o pngenc.o
OBJS-$(CONFIG_ARBC_DECODER) += arbc.o OBJS-$(CONFIG_ARBC_DECODER) += arbc.o
......
This diff is collapsed.
/*
* Audio Processing Technology codec for Bluetooth (aptX)
*
* Copyright (C) 2017 Aurelien Jacobs <aurel@gnuage.org>
*
* 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_APTX_H
#define AVCODEC_APTX_H
#include "libavutil/intreadwrite.h"
#include "avcodec.h"
#include "internal.h"
#include "mathops.h"
#include "audio_frame_queue.h"
enum channels {
LEFT,
RIGHT,
NB_CHANNELS
};
enum subbands {
LF, // Low Frequency (0-5.5 kHz)
MLF, // Medium-Low Frequency (5.5-11kHz)
MHF, // Medium-High Frequency (11-16.5kHz)
HF, // High Frequency (16.5-22kHz)
NB_SUBBANDS
};
#define NB_FILTERS 2
#define FILTER_TAPS 16
typedef struct {
int pos;
int32_t buffer[2*FILTER_TAPS];
} FilterSignal;
typedef struct {
FilterSignal outer_filter_signal[NB_FILTERS];
FilterSignal inner_filter_signal[NB_FILTERS][NB_FILTERS];
} QMFAnalysis;
typedef struct {
int32_t quantized_sample;
int32_t quantized_sample_parity_change;
int32_t error;
} Quantize;
typedef struct {
int32_t quantization_factor;
int32_t factor_select;
int32_t reconstructed_difference;
} InvertQuantize;
typedef struct {
int32_t prev_sign[2];
int32_t s_weight[2];
int32_t d_weight[24];
int32_t pos;
int32_t reconstructed_differences[48];
int32_t previous_reconstructed_sample;
int32_t predicted_difference;
int32_t predicted_sample;
} Prediction;
typedef struct {
int32_t codeword_history;
int32_t dither_parity;
int32_t dither[NB_SUBBANDS];
QMFAnalysis qmf;
Quantize quantize[NB_SUBBANDS];
InvertQuantize invert_quantize[NB_SUBBANDS];
Prediction prediction[NB_SUBBANDS];
} Channel;
typedef struct {
int hd;
int block_size;
int32_t sync_idx;
Channel channels[NB_CHANNELS];
AudioFrameQueue afq;
} AptXContext;
typedef const struct {
const int32_t *quantize_intervals;
const int32_t *invert_quantize_dither_factors;
const int32_t *quantize_dither_factors;
const int16_t *quantize_factor_select_offset;
int tables_size;
int32_t factor_max;
int32_t prediction_order;
} ConstTables;
extern ConstTables ff_aptx_quant_tables[2][NB_SUBBANDS];
/* Rounded right shift with optionnal clipping */
#define RSHIFT_SIZE(size) \
av_always_inline \
static int##size##_t rshift##size(int##size##_t value, int shift) \
{ \
int##size##_t rounding = (int##size##_t)1 << (shift - 1); \
int##size##_t mask = ((int##size##_t)1 << (shift + 1)) - 1; \
return ((value + rounding) >> shift) - ((value & mask) == rounding); \
} \
av_always_inline \
static int##size##_t rshift##size##_clip24(int##size##_t value, int shift) \
{ \
return av_clip_intp2(rshift##size(value, shift), 23); \
}
RSHIFT_SIZE(32)
RSHIFT_SIZE(64)
/*
* Convolution filter coefficients for the outer QMF of the QMF tree.
* The 2 sets are a mirror of each other.
*/
static const int32_t aptx_qmf_outer_coeffs[NB_FILTERS][FILTER_TAPS] = {
{
730, -413, -9611, 43626, -121026, 269973, -585547, 2801966,
697128, -160481, 27611, 8478, -10043, 3511, 688, -897,
},
{
-897, 688, 3511, -10043, 8478, 27611, -160481, 697128,
2801966, -585547, 269973, -121026, 43626, -9611, -413, 730,
},
};
/*
* Convolution filter coefficients for the inner QMF of the QMF tree.
* The 2 sets are a mirror of each other.
*/
static const int32_t aptx_qmf_inner_coeffs[NB_FILTERS][FILTER_TAPS] = {
{
1033, -584, -13592, 61697, -171156, 381799, -828088, 3962579,
985888, -226954, 39048, 11990, -14203, 4966, 973, -1268,
},
{
-1268, 973, 4966, -14203, 11990, 39048, -226954, 985888,
3962579, -828088, 381799, -171156, 61697, -13592, -584, 1033,
},
};
/*
* Push one sample into a circular signal buffer.
*/
av_always_inline
static void aptx_qmf_filter_signal_push(FilterSignal *signal, int32_t sample)
{
signal->buffer[signal->pos ] = sample;
signal->buffer[signal->pos+FILTER_TAPS] = sample;
signal->pos = (signal->pos + 1) & (FILTER_TAPS - 1);
}
/*
* Compute the convolution of the signal with the coefficients, and reduce
* to 24 bits by applying the specified right shifting.
*/
av_always_inline
static int32_t aptx_qmf_convolution(FilterSignal *signal,
const int32_t coeffs[FILTER_TAPS],
int shift)
{
int32_t *sig = &signal->buffer[signal->pos];
int64_t e = 0;
int i;
for (i = 0; i < FILTER_TAPS; i++)
e += MUL64(sig[i], coeffs[i]);
return rshift64_clip24(e, shift);
}
static inline int32_t aptx_quantized_parity(Channel *channel)
{
int32_t parity = channel->dither_parity;
int subband;
for (subband = 0; subband < NB_SUBBANDS; subband++)
parity ^= channel->quantize[subband].quantized_sample;
return parity & 1;
}
/* For each sample, ensure that the parity of all subbands of all channels
* is 0 except once every 8 samples where the parity is forced to 1. */
static inline int aptx_check_parity(Channel channels[NB_CHANNELS], int32_t *idx)
{
int32_t parity = aptx_quantized_parity(&channels[LEFT])
^ aptx_quantized_parity(&channels[RIGHT]);
int eighth = *idx == 7;
*idx = (*idx + 1) & 7;
return parity ^ eighth;
}
void ff_aptx_invert_quantize_and_prediction(Channel *channel, int hd);
void ff_aptx_generate_dither(Channel *channel);
int ff_aptx_init(AVCodecContext *avctx);
#endif /* AVCODEC_APTX_H */
/*
* Audio Processing Technology codec for Bluetooth (aptX)
*
* Copyright (C) 2017 Aurelien Jacobs <aurel@gnuage.org>
*
* 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 "aptx.h"
/*
* Half-band QMF synthesis filter realized with a polyphase FIR filter.
* Join 2 subbands and upsample by 2.
* So for each 2 subbands sample that goes in, a pair of samples goes out.
*/
av_always_inline
static void aptx_qmf_polyphase_synthesis(FilterSignal signal[NB_FILTERS],
const int32_t coeffs[NB_FILTERS][FILTER_TAPS],
int shift,
int32_t low_subband_input,
int32_t high_subband_input,
int32_t samples[NB_FILTERS])
{
int32_t subbands[NB_FILTERS];
int i;
subbands[0] = low_subband_input + high_subband_input;
subbands[1] = low_subband_input - high_subband_input;
for (i = 0; i < NB_FILTERS; i++) {
aptx_qmf_filter_signal_push(&signal[i], subbands[1-i]);
samples[i] = aptx_qmf_convolution(&signal[i], coeffs[i], shift);
}
}
/*
* Two stage QMF synthesis tree.
* Join 4 subbands and upsample by 4.
* So for each 4 subbands sample that goes in, a group of 4 samples goes out.
*/
static void aptx_qmf_tree_synthesis(QMFAnalysis *qmf,
int32_t subband_samples[4],
int32_t samples[4])
{
int32_t intermediate_samples[4];
int i;
/* Join 4 subbands into 2 intermediate subbands upsampled to 2 samples. */
for (i = 0; i < 2; i++)
aptx_qmf_polyphase_synthesis(qmf->inner_filter_signal[i],
aptx_qmf_inner_coeffs, 22,
subband_samples[2*i+0],
subband_samples[2*i+1],
&intermediate_samples[2*i]);
/* Join 2 samples from intermediate subbands upsampled to 4 samples. */
for (i = 0; i < 2; i++)
aptx_qmf_polyphase_synthesis(qmf->outer_filter_signal,
aptx_qmf_outer_coeffs, 21,
intermediate_samples[0+i],
intermediate_samples[2+i],
&samples[2*i]);
}
static void aptx_decode_channel(Channel *channel, int32_t samples[4])
{
int32_t subband_samples[4];
int subband;
for (subband = 0; subband < NB_SUBBANDS; subband++)
subband_samples[subband] = channel->prediction[subband].previous_reconstructed_sample;
aptx_qmf_tree_synthesis(&channel->qmf, subband_samples, samples);
}
static void aptx_unpack_codeword(Channel *channel, uint16_t codeword)
{
channel->quantize[0].quantized_sample = sign_extend(codeword >> 0, 7);
channel->quantize[1].quantized_sample = sign_extend(codeword >> 7, 4);
channel->quantize[2].quantized_sample = sign_extend(codeword >> 11, 2);
channel->quantize[3].quantized_sample = sign_extend(codeword >> 13, 3);
channel->quantize[3].quantized_sample = (channel->quantize[3].quantized_sample & ~1)
| aptx_quantized_parity(channel);
}
static void aptxhd_unpack_codeword(Channel *channel, uint32_t codeword)
{
channel->quantize[0].quantized_sample = sign_extend(codeword >> 0, 9);
channel->quantize[1].quantized_sample = sign_extend(codeword >> 9, 6);
channel->quantize[2].quantized_sample = sign_extend(codeword >> 15, 4);
channel->quantize[3].quantized_sample = sign_extend(codeword >> 19, 5);
channel->quantize[3].quantized_sample = (channel->quantize[3].quantized_sample & ~1)
| aptx_quantized_parity(channel);
}
static int aptx_decode_samples(AptXContext *ctx,
const uint8_t *input,
int32_t samples[NB_CHANNELS][4])
{
int channel, ret;
for (channel = 0; channel < NB_CHANNELS; channel++) {
ff_aptx_generate_dither(&ctx->channels[channel]);
if (ctx->hd)
aptxhd_unpack_codeword(&ctx->channels[channel],
AV_RB24(input + 3*channel));
else
aptx_unpack_codeword(&ctx->channels[channel],
AV_RB16(input + 2*channel));
ff_aptx_invert_quantize_and_prediction(&ctx->channels[channel], ctx->hd);
}
ret = aptx_check_parity(ctx->channels, &ctx->sync_idx);
for (channel = 0; channel < NB_CHANNELS; channel++)
aptx_decode_channel(&ctx->channels[channel], samples[channel]);
return ret;
}
static int aptx_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
AptXContext *s = avctx->priv_data;
AVFrame *frame = data;
int pos, opos, channel, sample, ret;
if (avpkt->size < s->block_size) {
av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
return AVERROR_INVALIDDATA;
}
/* get output buffer */
frame->channels = NB_CHANNELS;
frame->format = AV_SAMPLE_FMT_S32P;
frame->nb_samples = 4 * avpkt->size / s->block_size;
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
return ret;
for (pos = 0, opos = 0; opos < frame->nb_samples; pos += s->block_size, opos += 4) {
int32_t samples[NB_CHANNELS][4];
if (aptx_decode_samples(s, &avpkt->data[pos], samples)) {
av_log(avctx, AV_LOG_ERROR, "Synchronization error\n");
return AVERROR_INVALIDDATA;
}
for (channel = 0; channel < NB_CHANNELS; channel++)
for (sample = 0; sample < 4; sample++)
AV_WN32A(&frame->data[channel][4*(opos+sample)],
samples[channel][sample] * 256);
}
*got_frame_ptr = 1;
return s->block_size * frame->nb_samples / 4;
}
#if CONFIG_APTX_DECODER
AVCodec ff_aptx_decoder = {
.name = "aptx",
.long_name = NULL_IF_CONFIG_SMALL("aptX (Audio Processing Technology for Bluetooth)"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_APTX,
.priv_data_size = sizeof(AptXContext),
.init = ff_aptx_init,
.decode = aptx_decode_frame,
.capabilities = AV_CODEC_CAP_DR1,
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
.channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_STEREO, 0},
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P,
AV_SAMPLE_FMT_NONE },
};
#endif
#if CONFIG_APTX_HD_DECODER
AVCodec ff_aptx_hd_decoder = {
.name = "aptx_hd",
.long_name = NULL_IF_CONFIG_SMALL("aptX HD (Audio Processing Technology for Bluetooth)"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_APTX_HD,
.priv_data_size = sizeof(AptXContext),
.init = ff_aptx_init,
.decode = aptx_decode_frame,
.capabilities = AV_CODEC_CAP_DR1,
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
.channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_STEREO, 0},
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P,
AV_SAMPLE_FMT_NONE },
};
#endif
This diff is collapsed.
Markdown is supported
0% or
You are about to add 0 people to the discussion. Proceed with caution.
Finish editing this message first!
Please register or to comment