FFmpeg  4.3
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 "aptx.h"
24 
25 /*
26  * Half-band QMF synthesis filter realized with a polyphase FIR filter.
27  * Join 2 subbands and upsample by 2.
28  * So for each 2 subbands sample that goes in, a pair of samples goes out.
29  */
32  const int32_t coeffs[NB_FILTERS][FILTER_TAPS],
33  int shift,
34  int32_t low_subband_input,
35  int32_t high_subband_input,
37 {
39  int i;
40 
41  subbands[0] = low_subband_input + high_subband_input;
42  subbands[1] = low_subband_input - high_subband_input;
43 
44  for (i = 0; i < NB_FILTERS; i++) {
46  samples[i] = aptx_qmf_convolution(&signal[i], coeffs[i], shift);
47  }
48 }
49 
50 /*
51  * Two stage QMF synthesis tree.
52  * Join 4 subbands and upsample by 4.
53  * So for each 4 subbands sample that goes in, a group of 4 samples goes out.
54  */
56  int32_t subband_samples[4],
57  int32_t samples[4])
58 {
59  int32_t intermediate_samples[4];
60  int i;
61 
62  /* Join 4 subbands into 2 intermediate subbands upsampled to 2 samples. */
63  for (i = 0; i < 2; i++)
66  subband_samples[2*i+0],
67  subband_samples[2*i+1],
68  &intermediate_samples[2*i]);
69 
70  /* Join 2 samples from intermediate subbands upsampled to 4 samples. */
71  for (i = 0; i < 2; i++)
74  intermediate_samples[0+i],
75  intermediate_samples[2+i],
76  &samples[2*i]);
77 }
78 
79 
81 {
82  int32_t subband_samples[4];
83  int subband;
84  for (subband = 0; subband < NB_SUBBANDS; subband++)
85  subband_samples[subband] = channel->prediction[subband].previous_reconstructed_sample;
86  aptx_qmf_tree_synthesis(&channel->qmf, subband_samples, samples);
87 }
88 
89 static void aptx_unpack_codeword(Channel *channel, uint16_t codeword)
90 {
91  channel->quantize[0].quantized_sample = sign_extend(codeword >> 0, 7);
92  channel->quantize[1].quantized_sample = sign_extend(codeword >> 7, 4);
93  channel->quantize[2].quantized_sample = sign_extend(codeword >> 11, 2);
94  channel->quantize[3].quantized_sample = sign_extend(codeword >> 13, 3);
95  channel->quantize[3].quantized_sample = (channel->quantize[3].quantized_sample & ~1)
97 }
98 
99 static void aptxhd_unpack_codeword(Channel *channel, uint32_t codeword)
100 {
101  channel->quantize[0].quantized_sample = sign_extend(codeword >> 0, 9);
102  channel->quantize[1].quantized_sample = sign_extend(codeword >> 9, 6);
103  channel->quantize[2].quantized_sample = sign_extend(codeword >> 15, 4);
104  channel->quantize[3].quantized_sample = sign_extend(codeword >> 19, 5);
105  channel->quantize[3].quantized_sample = (channel->quantize[3].quantized_sample & ~1)
107 }
108 
110  const uint8_t *input,
112 {
113  int channel, ret;
114 
115  for (channel = 0; channel < NB_CHANNELS; channel++) {
116  ff_aptx_generate_dither(&ctx->channels[channel]);
117 
118  if (ctx->hd)
119  aptxhd_unpack_codeword(&ctx->channels[channel],
120  AV_RB24(input + 3*channel));
121  else
122  aptx_unpack_codeword(&ctx->channels[channel],
123  AV_RB16(input + 2*channel));
125  }
126 
127  ret = aptx_check_parity(ctx->channels, &ctx->sync_idx);
128 
129  for (channel = 0; channel < NB_CHANNELS; channel++)
131 
132  return ret;
133 }
134 
135 static int aptx_decode_frame(AVCodecContext *avctx, void *data,
136  int *got_frame_ptr, AVPacket *avpkt)
137 {
138  AptXContext *s = avctx->priv_data;
139  AVFrame *frame = data;
140  int pos, opos, channel, sample, ret;
141 
142  if (avpkt->size < s->block_size) {
143  av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
144  return AVERROR_INVALIDDATA;
145  }
146 
147  /* get output buffer */
148  frame->channels = NB_CHANNELS;
149  frame->format = AV_SAMPLE_FMT_S32P;
150  frame->nb_samples = 4 * avpkt->size / s->block_size;
151  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
152  return ret;
153 
154  for (pos = 0, opos = 0; opos < frame->nb_samples; pos += s->block_size, opos += 4) {
156 
157  if (aptx_decode_samples(s, &avpkt->data[pos], samples)) {
158  av_log(avctx, AV_LOG_ERROR, "Synchronization error\n");
159  return AVERROR_INVALIDDATA;
160  }
161 
162  for (channel = 0; channel < NB_CHANNELS; channel++)
163  for (sample = 0; sample < 4; sample++)
164  AV_WN32A(&frame->data[channel][4*(opos+sample)],
165  samples[channel][sample] * 256);
166  }
167 
168  *got_frame_ptr = 1;
169  return s->block_size * frame->nb_samples / 4;
170 }
171 
172 #if CONFIG_APTX_DECODER
174  .name = "aptx",
175  .long_name = NULL_IF_CONFIG_SMALL("aptX (Audio Processing Technology for Bluetooth)"),
176  .type = AVMEDIA_TYPE_AUDIO,
177  .id = AV_CODEC_ID_APTX,
178  .priv_data_size = sizeof(AptXContext),
179  .init = ff_aptx_init,
181  .capabilities = AV_CODEC_CAP_DR1,
182  .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
183  .channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_STEREO, 0},
184  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P,
186 };
187 #endif
188 
189 #if CONFIG_APTX_HD_DECODER
191  .name = "aptx_hd",
192  .long_name = NULL_IF_CONFIG_SMALL("aptX HD (Audio Processing Technology for Bluetooth)"),
193  .type = AVMEDIA_TYPE_AUDIO,
194  .id = AV_CODEC_ID_APTX_HD,
195  .priv_data_size = sizeof(AptXContext),
196  .init = ff_aptx_init,
198  .capabilities = AV_CODEC_CAP_DR1,
199  .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
200  .channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_STEREO, 0},
201  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P,
203 };
204 #endif
ff_aptx_decoder
AVCodec ff_aptx_decoder
AVCodec
AVCodec.
Definition: codec.h:190
Channel
Definition: aptx.h:83
FF_CODEC_CAP_INIT_THREADSAFE
#define FF_CODEC_CAP_INIT_THREADSAFE
The codec does not modify any global variables in the init function, allowing to call the init functi...
Definition: internal.h:40
init
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
FILTER_TAPS
#define FILTER_TAPS
Definition: aptx.h:48
aptx_quantized_parity
static int32_t aptx_quantized_parity(Channel *channel)
Definition: aptx.h:191
QMFAnalysis
Definition: aptx.h:55
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:300
AVPacket::data
uint8_t * data
Definition: packet.h:355
ff_aptx_generate_dither
void ff_aptx_generate_dither(Channel *channel)
Definition: aptx.c:384
AV_SAMPLE_FMT_S32P
@ AV_SAMPLE_FMT_S32P
signed 32 bits, planar
Definition: samplefmt.h:68
data
const char data[16]
Definition: mxf.c:91
aptx_decode_channel
static void aptx_decode_channel(Channel *channel, int32_t samples[4])
Definition: aptxdec.c:80
AV_RB16
#define AV_RB16
Definition: intreadwrite.h:53
AV_WN32A
#define AV_WN32A(p, v)
Definition: intreadwrite.h:538
samples
FFmpeg Automated Testing Environment ************************************Introduction Using FATE from your FFmpeg source directory Submitting the results to the FFmpeg result aggregation server Uploading new samples to the fate suite FATE makefile targets and variables Makefile targets Makefile variables Examples Introduction **************FATE is an extended regression suite on the client side and a means for results aggregation and presentation on the server side The first part of this document explains how you can use FATE from your FFmpeg source directory to test your ffmpeg binary The second part describes how you can run FATE to submit the results to FFmpeg’s FATE server In any way you can have a look at the publicly viewable FATE results by visiting this as it can be seen if some test on some platform broke with their recent contribution This usually happens on the platforms the developers could not test on The second part of this document describes how you can run FATE to submit your results to FFmpeg’s FATE server If you want to submit your results be sure to check that your combination of OS and compiler is not already listed on the above mentioned website In the third part you can find a comprehensive listing of FATE makefile targets and variables Using FATE from your FFmpeg source directory **********************************************If you want to run FATE on your machine you need to have the samples in place You can get the samples via the build target fate rsync Use this command from the top level source this will cause FATE to fail NOTE To use a custom wrapper to run the pass ‘ target exec’ to ‘configure’ or set the TARGET_EXEC Make variable Submitting the results to the FFmpeg result aggregation server ****************************************************************To submit your results to the server you should run fate through the shell script ‘tests fate sh’ from the FFmpeg sources This script needs to be invoked with a configuration file as its first argument tests fate sh path to fate_config A configuration file template with comments describing the individual configuration variables can be found at ‘doc fate_config sh template’ Create a configuration that suits your based on the configuration template The ‘slot’ configuration variable can be any string that is not yet but it is suggested that you name it adhering to the following pattern ‘ARCH OS COMPILER COMPILER VERSION’ The configuration file itself will be sourced in a shell therefore all shell features may be used This enables you to setup the environment as you need it for your build For your first test runs the ‘fate_recv’ variable should be empty or commented out This will run everything as normal except that it will omit the submission of the results to the server The following files should be present in $workdir as specified in the configuration it may help to try out the ‘ssh’ command with one or more ‘ v’ options You should get detailed output concerning your SSH configuration and the authentication process The only thing left is to automate the execution of the fate sh script and the synchronisation of the samples directory Uploading new samples to the fate suite *****************************************If you need a sample uploaded send a mail to samples request This is for developers who have an account on the fate suite server If you upload new samples
Definition: fate.txt:139
subbands
subbands
Definition: aptx.h:39
AptXContext
Definition: aptx.h:94
QMFAnalysis::inner_filter_signal
FilterSignal inner_filter_signal[NB_FILTERS][NB_FILTERS]
Definition: aptx.h:57
NB_FILTERS
@ NB_FILTERS
Definition: vf_waveform.c:49
ff_aptx_hd_decoder
AVCodec ff_aptx_hd_decoder
AV_RB24
#define AV_RB24
Definition: intreadwrite.h:64
AV_CH_LAYOUT_STEREO
#define AV_CH_LAYOUT_STEREO
Definition: channel_layout.h:86
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
decode
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
Definition: decode_audio.c:71
s
#define s(width, name)
Definition: cbs_vp9.c:257
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
NB_CHANNELS
@ NB_CHANNELS
Definition: aptx.h:36
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:55
int32_t
int32_t
Definition: audio_convert.c:194
QMFAnalysis::outer_filter_signal
FilterSignal outer_filter_signal[NB_FILTERS]
Definition: aptx.h:56
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:177
aptx_qmf_outer_coeffs
static const int32_t aptx_qmf_outer_coeffs[NB_FILTERS][FILTER_TAPS]
Definition: aptx.h:135
FilterSignal
Definition: aptx.h:50
aptx.h
ff_get_buffer
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1854
AV_CODEC_CAP_DR1
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
Definition: codec.h:50
AVPacket::size
int size
Definition: packet.h:356
NULL_IF_CONFIG_SMALL
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:186
aptx_qmf_inner_coeffs
static const int32_t aptx_qmf_inner_coeffs[NB_FILTERS][FILTER_TAPS]
Definition: aptx.h:150
AV_SAMPLE_FMT_NONE
@ AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:59
sample
#define sample
Definition: flacdsp_template.c:44
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:89
AVSampleFormat
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
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:99
uint8_t
uint8_t
Definition: audio_convert.c:194
aptx_qmf_filter_signal_push
static av_always_inline void aptx_qmf_filter_signal_push(FilterSignal *signal, int32_t sample)
Definition: aptx.h:165
AVCodec::name
const char * name
Name of the codec implementation.
Definition: codec.h:197
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:410
ff_aptx_init
av_cold int ff_aptx_init(AVCodecContext *avctx)
Definition: aptx.c:507
AVCodecContext
main external API structure.
Definition: avcodec.h:526
ff_aptx_invert_quantize_and_prediction
void ff_aptx_invert_quantize_and_prediction(Channel *channel, int hd)
Definition: aptx.c:496
aptx_decode_samples
static int aptx_decode_samples(AptXContext *ctx, const uint8_t *input, int32_t samples[NB_CHANNELS][4])
Definition: aptxdec.c:109
sign_extend
static av_const int sign_extend(int val, unsigned bits)
Definition: mathops.h:130
aptx_decode_frame
static int aptx_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: aptxdec.c:135
shift
static int shift(int a, int b)
Definition: sonic.c:82
AVCodecContext::priv_data
void * priv_data
Definition: avcodec.h:553
AVPacket
This structure stores compressed data.
Definition: packet.h:332
channel_layouts
static const uint16_t channel_layouts[7]
Definition: dca_lbr.c:113
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:28
AVERROR_INVALIDDATA
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
AV_CODEC_ID_APTX
@ AV_CODEC_ID_APTX
Definition: codec_id.h:496
NB_SUBBANDS
@ NB_SUBBANDS
Definition: aptx.h:44
AV_CODEC_ID_APTX_HD
@ AV_CODEC_ID_APTX_HD
Definition: codec_id.h:497
channel
channel
Definition: ebur128.h:39
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:31
aptx_check_parity
static int aptx_check_parity(Channel channels[NB_CHANNELS], int32_t *idx)
Definition: aptx.h:204