mirror of https://git.ffmpeg.org/ffmpeg.git
303 lines
12 KiB
C
303 lines
12 KiB
C
/*
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* Audio Processing Technology codec for Bluetooth (aptX)
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*
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* Copyright (C) 2017 Aurelien Jacobs <aurel@gnuage.org>
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "config_components.h"
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#include "libavutil/channel_layout.h"
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#include "aptx.h"
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#include "audio_frame_queue.h"
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#include "codec_internal.h"
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#include "encode.h"
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#include "internal.h"
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typedef struct AptXEncContext {
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AptXContext common;
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AudioFrameQueue afq;
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} AptXEncContext;
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/*
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* Half-band QMF analysis filter realized with a polyphase FIR filter.
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* Split into 2 subbands and downsample by 2.
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* So for each pair of samples that goes in, one sample goes out,
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* split into 2 separate subbands.
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*/
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av_always_inline
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static void aptx_qmf_polyphase_analysis(FilterSignal signal[NB_FILTERS],
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const int32_t coeffs[NB_FILTERS][FILTER_TAPS],
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int shift,
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int32_t samples[NB_FILTERS],
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int32_t *low_subband_output,
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int32_t *high_subband_output)
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{
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int32_t subbands[NB_FILTERS];
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int i;
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for (i = 0; i < NB_FILTERS; i++) {
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aptx_qmf_filter_signal_push(&signal[i], samples[NB_FILTERS-1-i]);
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subbands[i] = aptx_qmf_convolution(&signal[i], coeffs[i], shift);
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}
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*low_subband_output = av_clip_intp2(subbands[0] + subbands[1], 23);
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*high_subband_output = av_clip_intp2(subbands[0] - subbands[1], 23);
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}
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/*
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* Two stage QMF analysis tree.
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* Split 4 input samples into 4 subbands and downsample by 4.
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* So for each group of 4 samples that goes in, one sample goes out,
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* split into 4 separate subbands.
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*/
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static void aptx_qmf_tree_analysis(QMFAnalysis *qmf,
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int32_t samples[4],
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int32_t subband_samples[4])
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{
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int32_t intermediate_samples[4];
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int i;
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/* Split 4 input samples into 2 intermediate subbands downsampled to 2 samples */
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for (i = 0; i < 2; i++)
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aptx_qmf_polyphase_analysis(qmf->outer_filter_signal,
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aptx_qmf_outer_coeffs, 23,
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&samples[2*i],
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&intermediate_samples[0+i],
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&intermediate_samples[2+i]);
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/* Split 2 intermediate subband samples into 4 final subbands downsampled to 1 sample */
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for (i = 0; i < 2; i++)
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aptx_qmf_polyphase_analysis(qmf->inner_filter_signal[i],
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aptx_qmf_inner_coeffs, 23,
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&intermediate_samples[2*i],
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&subband_samples[2*i+0],
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&subband_samples[2*i+1]);
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}
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av_always_inline
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static int32_t aptx_bin_search(int32_t value, int32_t factor,
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const int32_t *intervals, int32_t nb_intervals)
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{
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int32_t idx = 0;
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int i;
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for (i = nb_intervals >> 1; i > 0; i >>= 1)
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if (MUL64(factor, intervals[idx + i]) <= ((int64_t)value << 24))
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idx += i;
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return idx;
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}
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static void aptx_quantize_difference(Quantize *quantize,
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int32_t sample_difference,
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int32_t dither,
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int32_t quantization_factor,
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ConstTables *tables)
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{
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const int32_t *intervals = tables->quantize_intervals;
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int32_t quantized_sample, dithered_sample, parity_change;
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int32_t d, mean, interval, inv, sample_difference_abs;
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int64_t error;
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sample_difference_abs = FFABS(sample_difference);
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sample_difference_abs = FFMIN(sample_difference_abs, (1 << 23) - 1);
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quantized_sample = aptx_bin_search(sample_difference_abs >> 4,
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quantization_factor,
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intervals, tables->tables_size);
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d = rshift32_clip24(MULH(dither, dither), 7) - (1 << 23);
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d = rshift64(MUL64(d, tables->quantize_dither_factors[quantized_sample]), 23);
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intervals += quantized_sample;
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mean = (intervals[1] + intervals[0]) / 2;
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interval = (intervals[1] - intervals[0]) * (-(sample_difference < 0) | 1);
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dithered_sample = rshift64_clip24(MUL64(dither, interval) + ((int64_t)av_clip_intp2(mean + d, 23) << 32), 32);
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error = ((int64_t)sample_difference_abs << 20) - MUL64(dithered_sample, quantization_factor);
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quantize->error = FFABS(rshift64(error, 23));
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parity_change = quantized_sample;
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if (error < 0)
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quantized_sample--;
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else
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parity_change--;
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inv = -(sample_difference < 0);
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quantize->quantized_sample = quantized_sample ^ inv;
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quantize->quantized_sample_parity_change = parity_change ^ inv;
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}
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static void aptx_encode_channel(Channel *channel, int32_t samples[4], int hd)
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{
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int32_t subband_samples[4];
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int subband;
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aptx_qmf_tree_analysis(&channel->qmf, samples, subband_samples);
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ff_aptx_generate_dither(channel);
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for (subband = 0; subband < NB_SUBBANDS; subband++) {
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int32_t diff = av_clip_intp2(subband_samples[subband] - channel->prediction[subband].predicted_sample, 23);
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aptx_quantize_difference(&channel->quantize[subband], diff,
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channel->dither[subband],
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channel->invert_quantize[subband].quantization_factor,
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&ff_aptx_quant_tables[hd][subband]);
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}
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}
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static void aptx_insert_sync(Channel channels[NB_CHANNELS], int32_t *idx)
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{
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if (aptx_check_parity(channels, idx)) {
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int i;
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Channel *c;
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static const int map[] = { 1, 2, 0, 3 };
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Quantize *min = &channels[NB_CHANNELS-1].quantize[map[0]];
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for (c = &channels[NB_CHANNELS-1]; c >= channels; c--)
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for (i = 0; i < NB_SUBBANDS; i++)
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if (c->quantize[map[i]].error < min->error)
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min = &c->quantize[map[i]];
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/* Forcing the desired parity is done by offsetting by 1 the quantized
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* sample from the subband featuring the smallest quantization error. */
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min->quantized_sample = min->quantized_sample_parity_change;
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}
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}
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static uint16_t aptx_pack_codeword(Channel *channel)
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{
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int32_t parity = aptx_quantized_parity(channel);
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return (((channel->quantize[3].quantized_sample & 0x06) | parity) << 13)
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| (((channel->quantize[2].quantized_sample & 0x03) ) << 11)
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| (((channel->quantize[1].quantized_sample & 0x0F) ) << 7)
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| (((channel->quantize[0].quantized_sample & 0x7F) ) << 0);
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}
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static uint32_t aptxhd_pack_codeword(Channel *channel)
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{
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int32_t parity = aptx_quantized_parity(channel);
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return (((channel->quantize[3].quantized_sample & 0x01E) | parity) << 19)
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| (((channel->quantize[2].quantized_sample & 0x00F) ) << 15)
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| (((channel->quantize[1].quantized_sample & 0x03F) ) << 9)
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| (((channel->quantize[0].quantized_sample & 0x1FF) ) << 0);
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}
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static void aptx_encode_samples(AptXContext *ctx,
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int32_t samples[NB_CHANNELS][4],
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uint8_t *output)
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{
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int channel;
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for (channel = 0; channel < NB_CHANNELS; channel++)
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aptx_encode_channel(&ctx->channels[channel], samples[channel], ctx->hd);
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aptx_insert_sync(ctx->channels, &ctx->sync_idx);
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for (channel = 0; channel < NB_CHANNELS; channel++) {
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ff_aptx_invert_quantize_and_prediction(&ctx->channels[channel], ctx->hd);
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if (ctx->hd)
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AV_WB24(output + 3*channel,
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aptxhd_pack_codeword(&ctx->channels[channel]));
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else
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AV_WB16(output + 2*channel,
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aptx_pack_codeword(&ctx->channels[channel]));
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}
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}
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static int aptx_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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const AVFrame *frame, int *got_packet_ptr)
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{
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AptXEncContext *const s0 = avctx->priv_data;
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AptXContext *const s = &s0->common;
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int pos, ipos, channel, sample, output_size, ret;
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if ((ret = ff_af_queue_add(&s0->afq, frame)) < 0)
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return ret;
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output_size = s->block_size * frame->nb_samples/4;
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if ((ret = ff_get_encode_buffer(avctx, avpkt, output_size, 0)) < 0)
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return ret;
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for (pos = 0, ipos = 0; pos < output_size; pos += s->block_size, ipos += 4) {
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int32_t samples[NB_CHANNELS][4];
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for (channel = 0; channel < NB_CHANNELS; channel++)
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for (sample = 0; sample < 4; sample++)
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samples[channel][sample] = (int32_t)AV_RN32A(&frame->data[channel][4*(ipos+sample)]) >> 8;
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aptx_encode_samples(s, samples, avpkt->data + pos);
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}
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ff_af_queue_remove(&s0->afq, frame->nb_samples, &avpkt->pts, &avpkt->duration);
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*got_packet_ptr = 1;
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return 0;
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}
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static av_cold int aptx_close(AVCodecContext *avctx)
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{
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AptXEncContext *const s = avctx->priv_data;
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ff_af_queue_close(&s->afq);
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return 0;
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}
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static av_cold int aptx_encode_init(AVCodecContext *avctx)
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{
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AptXEncContext *const s = avctx->priv_data;
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ff_af_queue_init(avctx, &s->afq);
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if (!avctx->frame_size || avctx->frame_size % 4)
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avctx->frame_size = 1024;
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avctx->internal->pad_samples = 4;
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return ff_aptx_init(avctx);
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}
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#if CONFIG_APTX_ENCODER
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const FFCodec ff_aptx_encoder = {
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.p.name = "aptx",
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CODEC_LONG_NAME("aptX (Audio Processing Technology for Bluetooth)"),
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.p.type = AVMEDIA_TYPE_AUDIO,
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.p.id = AV_CODEC_ID_APTX,
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.p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_ENCODER_REORDERED_OPAQUE,
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.priv_data_size = sizeof(AptXEncContext),
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.init = aptx_encode_init,
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FF_CODEC_ENCODE_CB(aptx_encode_frame),
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.close = aptx_close,
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.p.ch_layouts = (const AVChannelLayout[]) { AV_CHANNEL_LAYOUT_STEREO, { 0 } },
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.p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P,
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AV_SAMPLE_FMT_NONE },
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.p.supported_samplerates = (const int[]) {8000, 16000, 24000, 32000, 44100, 48000, 0},
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};
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#endif
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#if CONFIG_APTX_HD_ENCODER
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const FFCodec ff_aptx_hd_encoder = {
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.p.name = "aptx_hd",
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CODEC_LONG_NAME("aptX HD (Audio Processing Technology for Bluetooth)"),
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.p.type = AVMEDIA_TYPE_AUDIO,
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.p.id = AV_CODEC_ID_APTX_HD,
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.p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_ENCODER_REORDERED_OPAQUE,
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.priv_data_size = sizeof(AptXEncContext),
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.init = aptx_encode_init,
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FF_CODEC_ENCODE_CB(aptx_encode_frame),
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.close = aptx_close,
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.p.ch_layouts = (const AVChannelLayout[]) { AV_CHANNEL_LAYOUT_STEREO, { 0 } },
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.p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P,
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AV_SAMPLE_FMT_NONE },
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.p.supported_samplerates = (const int[]) {8000, 16000, 24000, 32000, 44100, 48000, 0},
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};
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#endif
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