1
0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-18 03:19:31 +02:00
FFmpeg/libavcodec/aptxenc.c
Andreas Rheinhardt 4243da4ff4 avcodec/codec_internal: Use union for FFCodec decode/encode callbacks
This is possible, because every given FFCodec has to implement
exactly one of these. Doing so decreases sizeof(FFCodec) and
therefore decreases the size of the binary.
Notice that in case of position-independent code the decrease
is in .data.rel.ro, so that this translates to decreased
memory consumption.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2022-04-05 20:02:37 +02:00

290 lines
12 KiB
C

/*
* 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 "codec_internal.h"
#include "encode.h"
/*
* 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)
{
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_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(&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
const FFCodec ff_aptx_encoder = {
.p.name = "aptx",
.p.long_name = NULL_IF_CONFIG_SMALL("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_SMALL_LAST_FRAME,
.priv_data_size = sizeof(AptXContext),
.init = ff_aptx_init,
FF_CODEC_ENCODE_CB(aptx_encode_frame),
.close = aptx_close,
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
#if FF_API_OLD_CHANNEL_LAYOUT
.p.channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_STEREO, 0},
#endif
.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",
.p.long_name = NULL_IF_CONFIG_SMALL("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_SMALL_LAST_FRAME,
.priv_data_size = sizeof(AptXContext),
.init = ff_aptx_init,
FF_CODEC_ENCODE_CB(aptx_encode_frame),
.close = aptx_close,
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
#if FF_API_OLD_CHANNEL_LAYOUT
.p.channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_STEREO, 0},
#endif
.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