ffmpeg/libavcodec/aptxenc.c

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/*
* 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 "config_components.h"
#include "libavutil/channel_layout.h"
#include "aptx.h"
#include "audio_frame_queue.h"
#include "codec_internal.h"
#include "encode.h"
#include "internal.h"
typedef struct AptXEncContext {
AptXContext common;
AudioFrameQueue afq;
} AptXEncContext;
/*
* Half-band QMF analysis filter realized with a polyphase FIR filter.
* Split into 2 subbands and downsample by 2.
* So for each pair of samples that goes in, one sample goes out,
* split into 2 separate subbands.
*/
av_always_inline
static void aptx_qmf_polyphase_analysis(FilterSignal signal[NB_FILTERS],
const int32_t coeffs[NB_FILTERS][FILTER_TAPS],
int shift,
int32_t samples[NB_FILTERS],
int32_t *low_subband_output,
int32_t *high_subband_output)
{
int32_t subbands[NB_FILTERS];
int i;
for (i = 0; i < NB_FILTERS; i++) {
aptx_qmf_filter_signal_push(&signal[i], samples[NB_FILTERS-1-i]);
subbands[i] = aptx_qmf_convolution(&signal[i], coeffs[i], shift);
}
*low_subband_output = av_clip_intp2(subbands[0] + subbands[1], 23);
*high_subband_output = av_clip_intp2(subbands[0] - subbands[1], 23);
}
/*
* Two stage QMF analysis tree.
* Split 4 input samples into 4 subbands and downsample by 4.
* So for each group of 4 samples that goes in, one sample goes out,
* split into 4 separate subbands.
*/
static void aptx_qmf_tree_analysis(QMFAnalysis *qmf,
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)
{
AptXEncContext *const s0 = avctx->priv_data;
AptXContext *const s = &s0->common;
int pos, ipos, channel, sample, output_size, ret;
if ((ret = ff_af_queue_add(&s0->afq, frame)) < 0)
return ret;
output_size = s->block_size * frame->nb_samples/4;
if ((ret = ff_get_encode_buffer(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(&s0->afq, frame->nb_samples, &avpkt->pts, &avpkt->duration);
*got_packet_ptr = 1;
return 0;
}
static av_cold int aptx_close(AVCodecContext *avctx)
{
AptXEncContext *const s = avctx->priv_data;
ff_af_queue_close(&s->afq);
return 0;
}
static av_cold int aptx_encode_init(AVCodecContext *avctx)
{
AptXEncContext *const s = avctx->priv_data;
ff_af_queue_init(avctx, &s->afq);
if (!avctx->frame_size || avctx->frame_size % 4)
avctx->frame_size = 1024;
avctx->internal->pad_samples = 4;
return ff_aptx_init(avctx);
}
#if CONFIG_APTX_ENCODER
const FFCodec ff_aptx_encoder = {
.p.name = "aptx",
CODEC_LONG_NAME("aptX (Audio Processing Technology for Bluetooth)"),
.p.type = AVMEDIA_TYPE_AUDIO,
.p.id = AV_CODEC_ID_APTX,
.p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_ENCODER_REORDERED_OPAQUE,
.priv_data_size = sizeof(AptXEncContext),
.init = aptx_encode_init,
FF_CODEC_ENCODE_CB(aptx_encode_frame),
.close = aptx_close,
CODEC_OLD_CHANNEL_LAYOUTS(AV_CH_LAYOUT_STEREO)
.p.ch_layouts = (const AVChannelLayout[]) { AV_CHANNEL_LAYOUT_STEREO, { 0 } },
.p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P,
AV_SAMPLE_FMT_NONE },
.p.supported_samplerates = (const int[]) {8000, 16000, 24000, 32000, 44100, 48000, 0},
};
#endif
#if CONFIG_APTX_HD_ENCODER
const FFCodec ff_aptx_hd_encoder = {
.p.name = "aptx_hd",
CODEC_LONG_NAME("aptX HD (Audio Processing Technology for Bluetooth)"),
.p.type = AVMEDIA_TYPE_AUDIO,
.p.id = AV_CODEC_ID_APTX_HD,
.p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_ENCODER_REORDERED_OPAQUE,
.priv_data_size = sizeof(AptXEncContext),
.init = aptx_encode_init,
FF_CODEC_ENCODE_CB(aptx_encode_frame),
.close = aptx_close,
CODEC_OLD_CHANNEL_LAYOUTS(AV_CH_LAYOUT_STEREO)
.p.ch_layouts = (const AVChannelLayout[]) { AV_CHANNEL_LAYOUT_STEREO, { 0 } },
.p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P,
AV_SAMPLE_FMT_NONE },
.p.supported_samplerates = (const int[]) {8000, 16000, 24000, 32000, 44100, 48000, 0},
};
#endif