mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-12-28 20:53:54 +02:00
4243da4ff4
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>
215 lines
8.4 KiB
C
215 lines
8.4 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 "internal.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, AVFrame *frame,
|
|
int *got_frame_ptr, AVPacket *avpkt)
|
|
{
|
|
AptXContext *s = avctx->priv_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->ch_layout.nb_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
|
|
const FFCodec ff_aptx_decoder = {
|
|
.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,
|
|
.priv_data_size = sizeof(AptXContext),
|
|
.init = ff_aptx_init,
|
|
FF_CODEC_DECODE_CB(aptx_decode_frame),
|
|
.p.capabilities = AV_CODEC_CAP_DR1,
|
|
.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 },
|
|
};
|
|
#endif
|
|
|
|
#if CONFIG_APTX_HD_DECODER
|
|
const FFCodec ff_aptx_hd_decoder = {
|
|
.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,
|
|
.priv_data_size = sizeof(AptXContext),
|
|
.init = ff_aptx_init,
|
|
FF_CODEC_DECODE_CB(aptx_decode_frame),
|
|
.p.capabilities = AV_CODEC_CAP_DR1,
|
|
.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 },
|
|
};
|
|
#endif
|