FFmpeg
aptxdec.c
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1 /*
2  * Audio Processing Technology codec for Bluetooth (aptX)
3  *
4  * Copyright (C) 2017 Aurelien Jacobs <aurel@gnuage.org>
5  *
6  * This file is part of FFmpeg.
7  *
8  * FFmpeg is free software; you can redistribute it and/or
9  * modify it under the terms of the GNU Lesser General Public
10  * License as published by the Free Software Foundation; either
11  * version 2.1 of the License, or (at your option) any later version.
12  *
13  * FFmpeg is distributed in the hope that it will be useful,
14  * but WITHOUT ANY WARRANTY; without even the implied warranty of
15  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
16  * Lesser General Public License for more details.
17  *
18  * You should have received a copy of the GNU Lesser General Public
19  * License along with FFmpeg; if not, write to the Free Software
20  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
21  */
22 
23 #include "config_components.h"
24 
26 #include "aptx.h"
27 #include "codec_internal.h"
28 #include "decode.h"
29 
30 /*
31  * Half-band QMF synthesis filter realized with a polyphase FIR filter.
32  * Join 2 subbands and upsample by 2.
33  * So for each 2 subbands sample that goes in, a pair of samples goes out.
34  */
37  const int32_t coeffs[NB_FILTERS][FILTER_TAPS],
38  int shift,
39  int32_t low_subband_input,
40  int32_t high_subband_input,
42 {
44  int i;
45 
46  subbands[0] = low_subband_input + high_subband_input;
47  subbands[1] = low_subband_input - high_subband_input;
48 
49  for (i = 0; i < NB_FILTERS; i++) {
51  samples[i] = aptx_qmf_convolution(&signal[i], coeffs[i], shift);
52  }
53 }
54 
55 /*
56  * Two stage QMF synthesis tree.
57  * Join 4 subbands and upsample by 4.
58  * So for each 4 subbands sample that goes in, a group of 4 samples goes out.
59  */
61  int32_t subband_samples[4],
62  int32_t samples[4])
63 {
64  int32_t intermediate_samples[4];
65  int i;
66 
67  /* Join 4 subbands into 2 intermediate subbands upsampled to 2 samples. */
68  for (i = 0; i < 2; i++)
71  subband_samples[2*i+0],
72  subband_samples[2*i+1],
73  &intermediate_samples[2*i]);
74 
75  /* Join 2 samples from intermediate subbands upsampled to 4 samples. */
76  for (i = 0; i < 2; i++)
79  intermediate_samples[0+i],
80  intermediate_samples[2+i],
81  &samples[2*i]);
82 }
83 
84 
86 {
87  int32_t subband_samples[4];
88  int subband;
89  for (subband = 0; subband < NB_SUBBANDS; subband++)
90  subband_samples[subband] = channel->prediction[subband].previous_reconstructed_sample;
91  aptx_qmf_tree_synthesis(&channel->qmf, subband_samples, samples);
92 }
93 
94 static void aptx_unpack_codeword(Channel *channel, uint16_t codeword)
95 {
96  channel->quantize[0].quantized_sample = sign_extend(codeword >> 0, 7);
97  channel->quantize[1].quantized_sample = sign_extend(codeword >> 7, 4);
98  channel->quantize[2].quantized_sample = sign_extend(codeword >> 11, 2);
99  channel->quantize[3].quantized_sample = sign_extend(codeword >> 13, 3);
100  channel->quantize[3].quantized_sample = (channel->quantize[3].quantized_sample & ~1)
102 }
103 
104 static void aptxhd_unpack_codeword(Channel *channel, uint32_t codeword)
105 {
106  channel->quantize[0].quantized_sample = sign_extend(codeword >> 0, 9);
107  channel->quantize[1].quantized_sample = sign_extend(codeword >> 9, 6);
108  channel->quantize[2].quantized_sample = sign_extend(codeword >> 15, 4);
109  channel->quantize[3].quantized_sample = sign_extend(codeword >> 19, 5);
110  channel->quantize[3].quantized_sample = (channel->quantize[3].quantized_sample & ~1)
112 }
113 
115  const uint8_t *input,
117 {
118  int channel, ret;
119 
120  for (channel = 0; channel < NB_CHANNELS; channel++) {
121  ff_aptx_generate_dither(&ctx->channels[channel]);
122 
123  if (ctx->hd)
124  aptxhd_unpack_codeword(&ctx->channels[channel],
125  AV_RB24(input + 3*channel));
126  else
127  aptx_unpack_codeword(&ctx->channels[channel],
128  AV_RB16(input + 2*channel));
130  }
131 
132  ret = aptx_check_parity(ctx->channels, &ctx->sync_idx);
133 
134  for (channel = 0; channel < NB_CHANNELS; channel++)
136 
137  return ret;
138 }
139 
141  int *got_frame_ptr, AVPacket *avpkt)
142 {
143  AptXContext *s = avctx->priv_data;
144  int pos, opos, channel, sample, ret;
145 
146  if (avpkt->size < s->block_size) {
147  av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
148  return AVERROR_INVALIDDATA;
149  }
150 
151  /* get output buffer */
152  frame->ch_layout.nb_channels = NB_CHANNELS;
153  frame->format = AV_SAMPLE_FMT_S32P;
154  frame->nb_samples = 4 * (avpkt->size / s->block_size);
155  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
156  return ret;
157 
158  for (pos = 0, opos = 0; opos < frame->nb_samples; pos += s->block_size, opos += 4) {
160 
161  if (aptx_decode_samples(s, &avpkt->data[pos], samples)) {
162  av_log(avctx, AV_LOG_ERROR, "Synchronization error\n");
163  return AVERROR_INVALIDDATA;
164  }
165 
166  for (channel = 0; channel < NB_CHANNELS; channel++)
167  for (sample = 0; sample < 4; sample++)
168  AV_WN32A(&frame->data[channel][4*(opos+sample)],
169  samples[channel][sample] * 256);
170  }
171 
172  *got_frame_ptr = 1;
173  return s->block_size * frame->nb_samples / 4;
174 }
175 
176 #if CONFIG_APTX_DECODER
177 const FFCodec ff_aptx_decoder = {
178  .p.name = "aptx",
179  CODEC_LONG_NAME("aptX (Audio Processing Technology for Bluetooth)"),
180  .p.type = AVMEDIA_TYPE_AUDIO,
181  .p.id = AV_CODEC_ID_APTX,
182  .priv_data_size = sizeof(AptXContext),
183  .init = ff_aptx_init,
185  .p.capabilities = AV_CODEC_CAP_DR1,
187  .p.ch_layouts = (const AVChannelLayout[]) { AV_CHANNEL_LAYOUT_STEREO, { 0 } },
188  .p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P,
190 };
191 #endif
192 
193 #if CONFIG_APTX_HD_DECODER
194 const FFCodec ff_aptx_hd_decoder = {
195  .p.name = "aptx_hd",
196  CODEC_LONG_NAME("aptX HD (Audio Processing Technology for Bluetooth)"),
197  .p.type = AVMEDIA_TYPE_AUDIO,
198  .p.id = AV_CODEC_ID_APTX_HD,
199  .priv_data_size = sizeof(AptXContext),
200  .init = ff_aptx_init,
202  .p.capabilities = AV_CODEC_CAP_DR1,
204  .p.ch_layouts = (const AVChannelLayout[]) { AV_CHANNEL_LAYOUT_STEREO, { 0 } },
205  .p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P,
207 };
208 #endif
Channel
Definition: aptx.h:81
FILTER_TAPS
#define FILTER_TAPS
Definition: aptx.h:46
AV_CHANNEL_LAYOUT_STEREO
#define AV_CHANNEL_LAYOUT_STEREO
Definition: channel_layout.h:369
aptx_quantized_parity
static int32_t aptx_quantized_parity(Channel *channel)
Definition: aptx.h:188
QMFAnalysis
Definition: aptx.h:53
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:330
AVPacket::data
uint8_t * data
Definition: packet.h:374
ff_aptx_generate_dither
void ff_aptx_generate_dither(Channel *channel)
Definition: aptx.c:385
AV_SAMPLE_FMT_S32P
@ AV_SAMPLE_FMT_S32P
signed 32 bits, planar
Definition: samplefmt.h:65
aptx_decode_channel
static void aptx_decode_channel(Channel *channel, int32_t samples[4])
Definition: aptxdec.c:85
FFCodec
Definition: codec_internal.h:127
AV_WN32A
#define AV_WN32A(p, v)
Definition: intreadwrite.h:538
subbands
subbands
Definition: aptx.h:37
AptXContext
Definition: aptx.h:92
QMFAnalysis::inner_filter_signal
FilterSignal inner_filter_signal[NB_FILTERS][NB_FILTERS]
Definition: aptx.h:55
FFCodec::p
AVCodec p
The public AVCodec.
Definition: codec_internal.h:131
NB_FILTERS
@ NB_FILTERS
Definition: vf_waveform.c:54
AV_CH_LAYOUT_STEREO
#define AV_CH_LAYOUT_STEREO
Definition: channel_layout.h:211
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:180
FF_CODEC_DECODE_CB
#define FF_CODEC_DECODE_CB(func)
Definition: codec_internal.h:306
s
#define s(width, name)
Definition: cbs_vp9.c:256
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
NB_CHANNELS
@ NB_CHANNELS
Definition: aptx.h:34
init
int(* init)(AVBSFContext *ctx)
Definition: dts2pts_bsf.c:365
CODEC_OLD_CHANNEL_LAYOUTS
#define CODEC_OLD_CHANNEL_LAYOUTS(...)
Definition: codec_internal.h:302
ctx
AVFormatContext * ctx
Definition: movenc.c:48
aptx_qmf_tree_synthesis
static void aptx_qmf_tree_synthesis(QMFAnalysis *qmf, int32_t subband_samples[4], int32_t samples[4])
Definition: aptxdec.c:60
decode.h
CODEC_LONG_NAME
#define CODEC_LONG_NAME(str)
Definition: codec_internal.h:272
ff_aptx_decoder
const FFCodec ff_aptx_decoder
QMFAnalysis::outer_filter_signal
FilterSignal outer_filter_signal[NB_FILTERS]
Definition: aptx.h:54
aptx_qmf_convolution
static av_always_inline int32_t aptx_qmf_convolution(FilterSignal *signal, const int32_t coeffs[FILTER_TAPS], int shift)
Definition: aptx.h:174
aptx_qmf_outer_coeffs
static const int32_t aptx_qmf_outer_coeffs[NB_FILTERS][FILTER_TAPS]
Definition: aptx.h:132
FilterSignal
Definition: aptx.h:48
aptx.h
ff_get_buffer
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1473
AV_CODEC_CAP_DR1
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
Definition: codec.h:52
AVPacket::size
int size
Definition: packet.h:375
AVChannelLayout
An AVChannelLayout holds information about the channel layout of audio data.
Definition: channel_layout.h:301
codec_internal.h
shift
static int shift(int a, int b)
Definition: bonk.c:257
aptx_qmf_inner_coeffs
static const int32_t aptx_qmf_inner_coeffs[NB_FILTERS][FILTER_TAPS]
Definition: aptx.h:147
AV_SAMPLE_FMT_NONE
@ AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:56
sample
#define sample
Definition: flacdsp_template.c:44
aptx_decode_frame
static int aptx_decode_frame(AVCodecContext *avctx, AVFrame *frame, int *got_frame_ptr, AVPacket *avpkt)
Definition: aptxdec.c:140
input
and forward the test the status of outputs and forward it to the corresponding return FFERROR_NOT_READY If the filters stores internally one or a few frame for some input
Definition: filter_design.txt:172
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:269
aptx_unpack_codeword
static void aptx_unpack_codeword(Channel *channel, uint16_t codeword)
Definition: aptxdec.c:94
AVSampleFormat
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:55
av_always_inline
#define av_always_inline
Definition: attributes.h:49
aptxhd_unpack_codeword
static void aptxhd_unpack_codeword(Channel *channel, uint32_t codeword)
Definition: aptxdec.c:104
aptx_qmf_filter_signal_push
static av_always_inline void aptx_qmf_filter_signal_push(FilterSignal *signal, int32_t sample)
Definition: aptx.h:162
AVCodec::name
const char * name
Name of the codec implementation.
Definition: codec.h:191
ret
ret
Definition: filter_design.txt:187
frame
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return or at least make progress towards producing a frame
Definition: filter_design.txt:264
pos
unsigned int pos
Definition: spdifenc.c:413
ff_aptx_init
av_cold int ff_aptx_init(AVCodecContext *avctx)
Definition: aptx.c:508
AVCodecContext
main external API structure.
Definition: avcodec.h:426
channel_layout.h
ff_aptx_invert_quantize_and_prediction
void ff_aptx_invert_quantize_and_prediction(Channel *channel, int hd)
Definition: aptx.c:497
aptx_decode_samples
static int aptx_decode_samples(AptXContext *ctx, const uint8_t *input, int32_t samples[NB_CHANNELS][4])
Definition: aptxdec.c:114
sign_extend
static av_const int sign_extend(int val, unsigned bits)
Definition: mathops.h:133
ff_aptx_hd_decoder
const FFCodec ff_aptx_hd_decoder
samples
Filter the word “frame” indicates either a video frame or a group of audio samples
Definition: filter_design.txt:8
AVCodecContext::priv_data
void * priv_data
Definition: avcodec.h:453
AVPacket
This structure stores compressed data.
Definition: packet.h:351
int32_t
int32_t
Definition: audioconvert.c:56
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:27
AVERROR_INVALIDDATA
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:61
AV_CODEC_ID_APTX
@ AV_CODEC_ID_APTX
Definition: codec_id.h:523
NB_SUBBANDS
@ NB_SUBBANDS
Definition: aptx.h:42
AV_RB24
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_WL16 uint64_t_TMPL AV_WB64 unsigned int_TMPL AV_WB32 unsigned int_TMPL AV_RB24
Definition: bytestream.h:97
AV_CODEC_ID_APTX_HD
@ AV_CODEC_ID_APTX_HD
Definition: codec_id.h:524
channel
channel
Definition: ebur128.h:39
AV_RB16
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_WL16 uint64_t_TMPL AV_WB64 unsigned int_TMPL AV_WB32 unsigned int_TMPL AV_WB24 unsigned int_TMPL AV_RB16
Definition: bytestream.h:98
aptx_qmf_polyphase_synthesis
static av_always_inline 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])
Definition: aptxdec.c:36
aptx_check_parity
static int aptx_check_parity(Channel channels[NB_CHANNELS], int32_t *idx)
Definition: aptx.h:201