mirror of https://github.com/FFmpeg/FFmpeg.git
Signed-off-by: James Almer <jamrial@gmail.com>pull/343/head
James Almer
5 years ago
parent
a8a05340de
commit
2383021a7a
5 changed files with 712 additions and 637 deletions
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/*
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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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#ifndef AVCODEC_APTX_H |
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#define AVCODEC_APTX_H |
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#include "libavutil/intreadwrite.h" |
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#include "avcodec.h" |
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#include "internal.h" |
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#include "mathops.h" |
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#include "audio_frame_queue.h" |
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enum channels { |
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LEFT, |
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RIGHT, |
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NB_CHANNELS |
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}; |
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enum subbands { |
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LF, // Low Frequency (0-5.5 kHz)
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MLF, // Medium-Low Frequency (5.5-11kHz)
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MHF, // Medium-High Frequency (11-16.5kHz)
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HF, // High Frequency (16.5-22kHz)
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NB_SUBBANDS |
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}; |
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#define NB_FILTERS 2 |
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#define FILTER_TAPS 16 |
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typedef struct { |
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int pos; |
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int32_t buffer[2*FILTER_TAPS]; |
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} FilterSignal; |
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typedef struct { |
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FilterSignal outer_filter_signal[NB_FILTERS]; |
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FilterSignal inner_filter_signal[NB_FILTERS][NB_FILTERS]; |
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} QMFAnalysis; |
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typedef struct { |
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int32_t quantized_sample; |
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int32_t quantized_sample_parity_change; |
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int32_t error; |
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} Quantize; |
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typedef struct { |
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int32_t quantization_factor; |
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int32_t factor_select; |
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int32_t reconstructed_difference; |
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} InvertQuantize; |
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typedef struct { |
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int32_t prev_sign[2]; |
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int32_t s_weight[2]; |
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int32_t d_weight[24]; |
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int32_t pos; |
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int32_t reconstructed_differences[48]; |
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int32_t previous_reconstructed_sample; |
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int32_t predicted_difference; |
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int32_t predicted_sample; |
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} Prediction; |
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typedef struct { |
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int32_t codeword_history; |
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int32_t dither_parity; |
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int32_t dither[NB_SUBBANDS]; |
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QMFAnalysis qmf; |
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Quantize quantize[NB_SUBBANDS]; |
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InvertQuantize invert_quantize[NB_SUBBANDS]; |
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Prediction prediction[NB_SUBBANDS]; |
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} Channel; |
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typedef struct { |
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int hd; |
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int block_size; |
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int32_t sync_idx; |
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Channel channels[NB_CHANNELS]; |
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AudioFrameQueue afq; |
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} AptXContext; |
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typedef const struct { |
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const int32_t *quantize_intervals; |
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const int32_t *invert_quantize_dither_factors; |
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const int32_t *quantize_dither_factors; |
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const int16_t *quantize_factor_select_offset; |
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int tables_size; |
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int32_t factor_max; |
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int32_t prediction_order; |
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} ConstTables; |
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extern ConstTables ff_aptx_quant_tables[2][NB_SUBBANDS]; |
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/* Rounded right shift with optionnal clipping */ |
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#define RSHIFT_SIZE(size) \ |
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av_always_inline \
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static int##size##_t rshift##size(int##size##_t value, int shift) \
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{ \
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int##size##_t rounding = (int##size##_t)1 << (shift - 1); \
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int##size##_t mask = ((int##size##_t)1 << (shift + 1)) - 1; \
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return ((value + rounding) >> shift) - ((value & mask) == rounding); \
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} \
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av_always_inline \
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static int##size##_t rshift##size##_clip24(int##size##_t value, int shift) \
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{ \
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return av_clip_intp2(rshift##size(value, shift), 23); \
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} |
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RSHIFT_SIZE(32) |
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RSHIFT_SIZE(64) |
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/*
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* Convolution filter coefficients for the outer QMF of the QMF tree. |
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* The 2 sets are a mirror of each other. |
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*/ |
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static const int32_t aptx_qmf_outer_coeffs[NB_FILTERS][FILTER_TAPS] = { |
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{ |
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730, -413, -9611, 43626, -121026, 269973, -585547, 2801966, |
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697128, -160481, 27611, 8478, -10043, 3511, 688, -897, |
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}, |
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{ |
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-897, 688, 3511, -10043, 8478, 27611, -160481, 697128, |
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2801966, -585547, 269973, -121026, 43626, -9611, -413, 730, |
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}, |
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}; |
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/*
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* Convolution filter coefficients for the inner QMF of the QMF tree. |
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* The 2 sets are a mirror of each other. |
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*/ |
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static const int32_t aptx_qmf_inner_coeffs[NB_FILTERS][FILTER_TAPS] = { |
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{ |
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1033, -584, -13592, 61697, -171156, 381799, -828088, 3962579, |
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985888, -226954, 39048, 11990, -14203, 4966, 973, -1268, |
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}, |
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{ |
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-1268, 973, 4966, -14203, 11990, 39048, -226954, 985888, |
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3962579, -828088, 381799, -171156, 61697, -13592, -584, 1033, |
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}, |
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}; |
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/*
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* Push one sample into a circular signal buffer. |
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*/ |
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av_always_inline |
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static void aptx_qmf_filter_signal_push(FilterSignal *signal, int32_t sample) |
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{ |
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signal->buffer[signal->pos ] = sample; |
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signal->buffer[signal->pos+FILTER_TAPS] = sample; |
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signal->pos = (signal->pos + 1) & (FILTER_TAPS - 1); |
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} |
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/*
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* Compute the convolution of the signal with the coefficients, and reduce |
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* to 24 bits by applying the specified right shifting. |
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*/ |
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av_always_inline |
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static int32_t aptx_qmf_convolution(FilterSignal *signal, |
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const int32_t coeffs[FILTER_TAPS], |
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int shift) |
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{ |
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int32_t *sig = &signal->buffer[signal->pos]; |
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int64_t e = 0; |
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int i; |
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for (i = 0; i < FILTER_TAPS; i++) |
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e += MUL64(sig[i], coeffs[i]); |
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return rshift64_clip24(e, shift); |
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} |
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static inline int32_t aptx_quantized_parity(Channel *channel) |
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{ |
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int32_t parity = channel->dither_parity; |
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int subband; |
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for (subband = 0; subband < NB_SUBBANDS; subband++) |
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parity ^= channel->quantize[subband].quantized_sample; |
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return parity & 1; |
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} |
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/* For each sample, ensure that the parity of all subbands of all channels
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* is 0 except once every 8 samples where the parity is forced to 1. */ |
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static inline int aptx_check_parity(Channel channels[NB_CHANNELS], int32_t *idx) |
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{ |
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int32_t parity = aptx_quantized_parity(&channels[LEFT]) |
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^ aptx_quantized_parity(&channels[RIGHT]); |
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int eighth = *idx == 7; |
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*idx = (*idx + 1) & 7; |
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return parity ^ eighth; |
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} |
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void ff_aptx_invert_quantize_and_prediction(Channel *channel, int hd); |
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void ff_aptx_generate_dither(Channel *channel); |
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int ff_aptx_init(AVCodecContext *avctx); |
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#endif /* AVCODEC_APTX_H */ |
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@ -0,0 +1,204 @@ |
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/*
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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 "aptx.h" |
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/*
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* Half-band QMF synthesis filter realized with a polyphase FIR filter. |
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* Join 2 subbands and upsample by 2. |
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* So for each 2 subbands sample that goes in, a pair of samples goes out. |
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*/ |
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av_always_inline |
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static void aptx_qmf_polyphase_synthesis(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 low_subband_input, |
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int32_t high_subband_input, |
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int32_t samples[NB_FILTERS]) |
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{ |
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int32_t subbands[NB_FILTERS]; |
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int i; |
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subbands[0] = low_subband_input + high_subband_input; |
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subbands[1] = low_subband_input - high_subband_input; |
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for (i = 0; i < NB_FILTERS; i++) { |
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aptx_qmf_filter_signal_push(&signal[i], subbands[1-i]); |
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samples[i] = aptx_qmf_convolution(&signal[i], coeffs[i], shift); |
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} |
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} |
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/*
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* Two stage QMF synthesis tree. |
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* Join 4 subbands and upsample by 4. |
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* So for each 4 subbands sample that goes in, a group of 4 samples goes out. |
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*/ |
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static void aptx_qmf_tree_synthesis(QMFAnalysis *qmf, |
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int32_t subband_samples[4], |
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int32_t 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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/* Join 4 subbands into 2 intermediate subbands upsampled to 2 samples. */ |
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for (i = 0; i < 2; i++) |
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aptx_qmf_polyphase_synthesis(qmf->inner_filter_signal[i], |
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aptx_qmf_inner_coeffs, 22, |
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subband_samples[2*i+0], |
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subband_samples[2*i+1], |
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&intermediate_samples[2*i]); |
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/* Join 2 samples from intermediate subbands upsampled to 4 samples. */ |
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for (i = 0; i < 2; i++) |
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aptx_qmf_polyphase_synthesis(qmf->outer_filter_signal, |
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aptx_qmf_outer_coeffs, 21, |
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intermediate_samples[0+i], |
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intermediate_samples[2+i], |
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&samples[2*i]); |
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} |
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static void aptx_decode_channel(Channel *channel, int32_t samples[4]) |
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{ |
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int32_t subband_samples[4]; |
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int subband; |
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for (subband = 0; subband < NB_SUBBANDS; subband++) |
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subband_samples[subband] = channel->prediction[subband].previous_reconstructed_sample; |
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aptx_qmf_tree_synthesis(&channel->qmf, subband_samples, samples); |
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} |
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static void aptx_unpack_codeword(Channel *channel, uint16_t codeword) |
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{ |
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channel->quantize[0].quantized_sample = sign_extend(codeword >> 0, 7); |
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channel->quantize[1].quantized_sample = sign_extend(codeword >> 7, 4); |
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channel->quantize[2].quantized_sample = sign_extend(codeword >> 11, 2); |
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channel->quantize[3].quantized_sample = sign_extend(codeword >> 13, 3); |
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channel->quantize[3].quantized_sample = (channel->quantize[3].quantized_sample & ~1) |
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| aptx_quantized_parity(channel); |
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} |
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static void aptxhd_unpack_codeword(Channel *channel, uint32_t codeword) |
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{ |
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channel->quantize[0].quantized_sample = sign_extend(codeword >> 0, 9); |
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channel->quantize[1].quantized_sample = sign_extend(codeword >> 9, 6); |
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channel->quantize[2].quantized_sample = sign_extend(codeword >> 15, 4); |
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channel->quantize[3].quantized_sample = sign_extend(codeword >> 19, 5); |
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channel->quantize[3].quantized_sample = (channel->quantize[3].quantized_sample & ~1) |
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| aptx_quantized_parity(channel); |
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} |
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static int aptx_decode_samples(AptXContext *ctx, |
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const uint8_t *input, |
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int32_t samples[NB_CHANNELS][4]) |
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{ |
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int channel, ret; |
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for (channel = 0; channel < NB_CHANNELS; channel++) { |
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ff_aptx_generate_dither(&ctx->channels[channel]); |
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if (ctx->hd) |
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aptxhd_unpack_codeword(&ctx->channels[channel], |
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AV_RB24(input + 3*channel)); |
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else |
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aptx_unpack_codeword(&ctx->channels[channel], |
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AV_RB16(input + 2*channel)); |
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ff_aptx_invert_quantize_and_prediction(&ctx->channels[channel], ctx->hd); |
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} |
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ret = aptx_check_parity(ctx->channels, &ctx->sync_idx); |
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for (channel = 0; channel < NB_CHANNELS; channel++) |
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aptx_decode_channel(&ctx->channels[channel], samples[channel]); |
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return ret; |
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} |
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static int aptx_decode_frame(AVCodecContext *avctx, void *data, |
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int *got_frame_ptr, AVPacket *avpkt) |
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{ |
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AptXContext *s = avctx->priv_data; |
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AVFrame *frame = data; |
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int pos, opos, channel, sample, ret; |
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if (avpkt->size < s->block_size) { |
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av_log(avctx, AV_LOG_ERROR, "Packet is too small\n"); |
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return AVERROR_INVALIDDATA; |
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} |
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/* get output buffer */ |
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frame->channels = NB_CHANNELS; |
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frame->format = AV_SAMPLE_FMT_S32P; |
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frame->nb_samples = 4 * avpkt->size / s->block_size; |
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if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) |
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return ret; |
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for (pos = 0, opos = 0; opos < frame->nb_samples; pos += s->block_size, opos += 4) { |
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int32_t samples[NB_CHANNELS][4]; |
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if (aptx_decode_samples(s, &avpkt->data[pos], samples)) { |
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av_log(avctx, AV_LOG_ERROR, "Synchronization error\n"); |
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return AVERROR_INVALIDDATA; |
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} |
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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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AV_WN32A(&frame->data[channel][4*(opos+sample)], |
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samples[channel][sample] * 256); |
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} |
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*got_frame_ptr = 1; |
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return s->block_size * frame->nb_samples / 4; |
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} |
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#if CONFIG_APTX_DECODER |
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AVCodec ff_aptx_decoder = { |
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.name = "aptx", |
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.long_name = NULL_IF_CONFIG_SMALL("aptX (Audio Processing Technology for Bluetooth)"), |
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.type = AVMEDIA_TYPE_AUDIO, |
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.id = AV_CODEC_ID_APTX, |
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.priv_data_size = sizeof(AptXContext), |
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.init = ff_aptx_init, |
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.decode = aptx_decode_frame, |
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.capabilities = AV_CODEC_CAP_DR1, |
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.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE, |
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.channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_STEREO, 0}, |
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.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P, |
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AV_SAMPLE_FMT_NONE }, |
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}; |
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#endif |
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#if CONFIG_APTX_HD_DECODER |
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AVCodec ff_aptx_hd_decoder = { |
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.name = "aptx_hd", |
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.long_name = NULL_IF_CONFIG_SMALL("aptX HD (Audio Processing Technology for Bluetooth)"), |
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.type = AVMEDIA_TYPE_AUDIO, |
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.id = AV_CODEC_ID_APTX_HD, |
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.priv_data_size = sizeof(AptXContext), |
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.init = ff_aptx_init, |
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.decode = aptx_decode_frame, |
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.capabilities = AV_CODEC_CAP_DR1, |
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.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE, |
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.channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_STEREO, 0}, |
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.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P, |
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AV_SAMPLE_FMT_NONE }, |
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}; |
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#endif |
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@ -0,0 +1,278 @@ |
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/*
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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 |
||||
* 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 "aptx.h" |
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|
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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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/*
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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], |
||||
int32_t subband_samples[4]) |
||||
{ |
||||
int32_t intermediate_samples[4]; |
||||
int i; |
||||
|
||||
/* Split 4 input samples into 2 intermediate subbands downsampled to 2 samples */ |
||||
for (i = 0; i < 2; i++) |
||||
aptx_qmf_polyphase_analysis(qmf->outer_filter_signal, |
||||
aptx_qmf_outer_coeffs, 23, |
||||
&samples[2*i], |
||||
&intermediate_samples[0+i], |
||||
&intermediate_samples[2+i]); |
||||
|
||||
/* Split 2 intermediate subband samples into 4 final subbands downsampled to 1 sample */ |
||||
for (i = 0; i < 2; i++) |
||||
aptx_qmf_polyphase_analysis(qmf->inner_filter_signal[i], |
||||
aptx_qmf_inner_coeffs, 23, |
||||
&intermediate_samples[2*i], |
||||
&subband_samples[2*i+0], |
||||
&subband_samples[2*i+1]); |
||||
} |
||||
|
||||
av_always_inline |
||||
static int32_t aptx_bin_search(int32_t value, int32_t factor, |
||||
const int32_t *intervals, int32_t nb_intervals) |
||||
{ |
||||
int32_t idx = 0; |
||||
int i; |
||||
|
||||
for (i = nb_intervals >> 1; i > 0; i >>= 1) |
||||
if (MUL64(factor, intervals[idx + i]) <= ((int64_t)value << 24)) |
||||
idx += i; |
||||
|
||||
return idx; |
||||
} |
||||
|
||||
static void aptx_quantize_difference(Quantize *quantize, |
||||
int32_t sample_difference, |
||||
int32_t dither, |
||||
int32_t quantization_factor, |
||||
ConstTables *tables) |
||||
{ |
||||
const int32_t *intervals = tables->quantize_intervals; |
||||
int32_t quantized_sample, dithered_sample, parity_change; |
||||
int32_t d, mean, interval, inv, sample_difference_abs; |
||||
int64_t error; |
||||
|
||||
sample_difference_abs = FFABS(sample_difference); |
||||
sample_difference_abs = FFMIN(sample_difference_abs, (1 << 23) - 1); |
||||
|
||||
quantized_sample = aptx_bin_search(sample_difference_abs >> 4, |
||||
quantization_factor, |
||||
intervals, tables->tables_size); |
||||
|
||||
d = rshift32_clip24(MULH(dither, dither), 7) - (1 << 23); |
||||
d = rshift64(MUL64(d, tables->quantize_dither_factors[quantized_sample]), 23); |
||||
|
||||
intervals += quantized_sample; |
||||
mean = (intervals[1] + intervals[0]) / 2; |
||||
interval = (intervals[1] - intervals[0]) * (-(sample_difference < 0) | 1); |
||||
|
||||
dithered_sample = rshift64_clip24(MUL64(dither, interval) + ((int64_t)av_clip_intp2(mean + d, 23) << 32), 32); |
||||
error = ((int64_t)sample_difference_abs << 20) - MUL64(dithered_sample, quantization_factor); |
||||
quantize->error = FFABS(rshift64(error, 23)); |
||||
|
||||
parity_change = quantized_sample; |
||||
if (error < 0) |
||||
quantized_sample--; |
||||
else |
||||
parity_change--; |
||||
|
||||
inv = -(sample_difference < 0); |
||||
quantize->quantized_sample = quantized_sample ^ inv; |
||||
quantize->quantized_sample_parity_change = parity_change ^ inv; |
||||
} |
||||
|
||||
static void aptx_encode_channel(Channel *channel, int32_t samples[4], int hd) |
||||
{ |
||||
int32_t subband_samples[4]; |
||||
int subband; |
||||
aptx_qmf_tree_analysis(&channel->qmf, samples, subband_samples); |
||||
ff_aptx_generate_dither(channel); |
||||
for (subband = 0; subband < NB_SUBBANDS; subband++) { |
||||
int32_t diff = av_clip_intp2(subband_samples[subband] - channel->prediction[subband].predicted_sample, 23); |
||||
aptx_quantize_difference(&channel->quantize[subband], diff, |
||||
channel->dither[subband], |
||||
channel->invert_quantize[subband].quantization_factor, |
||||
&ff_aptx_quant_tables[hd][subband]); |
||||
} |
||||
} |
||||
|
||||
static void aptx_insert_sync(Channel channels[NB_CHANNELS], int32_t *idx) |
||||
{ |
||||
if (aptx_check_parity(channels, idx)) { |
||||
int i; |
||||
Channel *c; |
||||
static const int map[] = { 1, 2, 0, 3 }; |
||||
Quantize *min = &channels[NB_CHANNELS-1].quantize[map[0]]; |
||||
for (c = &channels[NB_CHANNELS-1]; c >= channels; c--) |
||||
for (i = 0; i < NB_SUBBANDS; i++) |
||||
if (c->quantize[map[i]].error < min->error) |
||||
min = &c->quantize[map[i]]; |
||||
|
||||
/* Forcing the desired parity is done by offsetting by 1 the quantized
|
||||
* sample from the subband featuring the smallest quantization error. */ |
||||
min->quantized_sample = min->quantized_sample_parity_change; |
||||
} |
||||
} |
||||
|
||||
static uint16_t aptx_pack_codeword(Channel *channel) |
||||
{ |
||||
int32_t parity = aptx_quantized_parity(channel); |
||||
return (((channel->quantize[3].quantized_sample & 0x06) | parity) << 13) |
||||
| (((channel->quantize[2].quantized_sample & 0x03) ) << 11) |
||||
| (((channel->quantize[1].quantized_sample & 0x0F) ) << 7) |
||||
| (((channel->quantize[0].quantized_sample & 0x7F) ) << 0); |
||||
} |
||||
|
||||
static uint32_t aptxhd_pack_codeword(Channel *channel) |
||||
{ |
||||
int32_t parity = aptx_quantized_parity(channel); |
||||
return (((channel->quantize[3].quantized_sample & 0x01E) | parity) << 19) |
||||
| (((channel->quantize[2].quantized_sample & 0x00F) ) << 15) |
||||
| (((channel->quantize[1].quantized_sample & 0x03F) ) << 9) |
||||
| (((channel->quantize[0].quantized_sample & 0x1FF) ) << 0); |
||||
} |
||||
|
||||
static void aptx_encode_samples(AptXContext *ctx, |
||||
int32_t samples[NB_CHANNELS][4], |
||||
uint8_t *output) |
||||
{ |
||||
int channel; |
||||
for (channel = 0; channel < NB_CHANNELS; channel++) |
||||
aptx_encode_channel(&ctx->channels[channel], samples[channel], ctx->hd); |
||||
|
||||
aptx_insert_sync(ctx->channels, &ctx->sync_idx); |
||||
|
||||
for (channel = 0; channel < NB_CHANNELS; channel++) { |
||||
ff_aptx_invert_quantize_and_prediction(&ctx->channels[channel], ctx->hd); |
||||
if (ctx->hd) |
||||
AV_WB24(output + 3*channel, |
||||
aptxhd_pack_codeword(&ctx->channels[channel])); |
||||
else |
||||
AV_WB16(output + 2*channel, |
||||
aptx_pack_codeword(&ctx->channels[channel])); |
||||
} |
||||
} |
||||
|
||||
static int aptx_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, |
||||
const AVFrame *frame, int *got_packet_ptr) |
||||
{ |
||||
AptXContext *s = avctx->priv_data; |
||||
int pos, ipos, channel, sample, output_size, ret; |
||||
|
||||
if ((ret = ff_af_queue_add(&s->afq, frame)) < 0) |
||||
return ret; |
||||
|
||||
output_size = s->block_size * frame->nb_samples/4; |
||||
if ((ret = ff_alloc_packet2(avctx, avpkt, output_size, 0)) < 0) |
||||
return ret; |
||||
|
||||
for (pos = 0, ipos = 0; pos < output_size; pos += s->block_size, ipos += 4) { |
||||
int32_t samples[NB_CHANNELS][4]; |
||||
|
||||
for (channel = 0; channel < NB_CHANNELS; channel++) |
||||
for (sample = 0; sample < 4; sample++) |
||||
samples[channel][sample] = (int32_t)AV_RN32A(&frame->data[channel][4*(ipos+sample)]) >> 8; |
||||
|
||||
aptx_encode_samples(s, samples, avpkt->data + pos); |
||||
} |
||||
|
||||
ff_af_queue_remove(&s->afq, frame->nb_samples, &avpkt->pts, &avpkt->duration); |
||||
*got_packet_ptr = 1; |
||||
return 0; |
||||
} |
||||
|
||||
static av_cold int aptx_close(AVCodecContext *avctx) |
||||
{ |
||||
AptXContext *s = avctx->priv_data; |
||||
ff_af_queue_close(&s->afq); |
||||
return 0; |
||||
} |
||||
|
||||
#if CONFIG_APTX_ENCODER |
||||
AVCodec ff_aptx_encoder = { |
||||
.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, |
||||
.encode2 = aptx_encode_frame, |
||||
.close = aptx_close, |
||||
.capabilities = AV_CODEC_CAP_SMALL_LAST_FRAME, |
||||
.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 }, |
||||
.supported_samplerates = (const int[]) {8000, 16000, 24000, 32000, 44100, 48000, 0}, |
||||
}; |
||||
#endif |
||||
|
||||
#if CONFIG_APTX_HD_ENCODER |
||||
AVCodec ff_aptx_hd_encoder = { |
||||
.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, |
||||
.encode2 = aptx_encode_frame, |
||||
.close = aptx_close, |
||||
.capabilities = AV_CODEC_CAP_SMALL_LAST_FRAME, |
||||
.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 }, |
||||
.supported_samplerates = (const int[]) {8000, 16000, 24000, 32000, 44100, 48000, 0}, |
||||
}; |
||||
#endif |
||||
Loading…
Reference in new issue