FFmpeg
dsddec.c
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1 /*
2  * Direct Stream Digital (DSD) decoder
3  * based on BSD licensed dsd2pcm by Sebastian Gesemann
4  * Copyright (c) 2009, 2011 Sebastian Gesemann. All rights reserved.
5  * Copyright (c) 2014 Peter Ross
6  *
7  * This file is part of FFmpeg.
8  *
9  * FFmpeg is free software; you can redistribute it and/or
10  * modify it under the terms of the GNU Lesser General Public
11  * License as published by the Free Software Foundation; either
12  * version 2.1 of the License, or (at your option) any later version.
13  *
14  * FFmpeg is distributed in the hope that it will be useful,
15  * but WITHOUT ANY WARRANTY; without even the implied warranty of
16  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
17  * Lesser General Public License for more details.
18  *
19  * You should have received a copy of the GNU Lesser General Public
20  * License along with FFmpeg; if not, write to the Free Software
21  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
22  */
23 
24 /**
25  * @file
26  * Direct Stream Digital (DSD) decoder
27  */
28 
29 #include "libavcodec/internal.h"
30 #include "avcodec.h"
31 #include "dsd.h"
32 
33 #define DSD_SILENCE 0x69
34 #define DSD_SILENCE_REVERSED 0x96
35 /* 0x69 = 01101001
36  * This pattern "on repeat" makes a low energy 352.8 kHz tone
37  * and a high energy 1.0584 MHz tone which should be filtered
38  * out completely by any playback system --> silence
39  */
40 
42 {
43  DSDContext * s;
44  int i;
45  uint8_t silence;
46 
47  if (!avctx->channels)
48  return AVERROR_INVALIDDATA;
49 
51 
52  s = av_malloc_array(sizeof(DSDContext), avctx->channels);
53  if (!s)
54  return AVERROR(ENOMEM);
55 
57  for (i = 0; i < avctx->channels; i++) {
58  s[i].pos = 0;
59  memset(s[i].buf, silence, sizeof(s[i].buf));
60  }
61 
63  avctx->priv_data = s;
64  return 0;
65 }
66 
67 typedef struct ThreadData {
69  const AVPacket *avpkt;
70 } ThreadData;
71 
72 static int dsd_channel(AVCodecContext *avctx, void *tdata, int j, int threadnr)
73 {
74  int lsbf = avctx->codec_id == AV_CODEC_ID_DSD_LSBF || avctx->codec_id == AV_CODEC_ID_DSD_LSBF_PLANAR;
75  DSDContext *s = avctx->priv_data;
76  ThreadData *td = tdata;
77  AVFrame *frame = td->frame;
78  const AVPacket *avpkt = td->avpkt;
79  int src_next, src_stride;
80  float *dst = ((float **)frame->extended_data)[j];
81 
83  src_next = frame->nb_samples;
84  src_stride = 1;
85  } else {
86  src_next = 1;
87  src_stride = avctx->channels;
88  }
89 
90  ff_dsd2pcm_translate(&s[j], frame->nb_samples, lsbf,
91  avpkt->data + j * src_next, src_stride,
92  dst, 1);
93 
94  return 0;
95 }
96 
97 static int decode_frame(AVCodecContext *avctx, void *data,
98  int *got_frame_ptr, AVPacket *avpkt)
99 {
100  ThreadData td;
101  AVFrame *frame = data;
102  int ret;
103 
104  frame->nb_samples = avpkt->size / avctx->channels;
105 
106  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
107  return ret;
108 
109  td.frame = frame;
110  td.avpkt = avpkt;
111  avctx->execute2(avctx, dsd_channel, &td, NULL, avctx->channels);
112 
113  *got_frame_ptr = 1;
114  return frame->nb_samples * avctx->channels;
115 }
116 
117 #define DSD_DECODER(id_, name_, long_name_) \
118 const AVCodec ff_ ## name_ ## _decoder = { \
119  .name = #name_, \
120  .long_name = NULL_IF_CONFIG_SMALL(long_name_), \
121  .type = AVMEDIA_TYPE_AUDIO, \
122  .id = AV_CODEC_ID_##id_, \
123  .init = decode_init, \
124  .decode = decode_frame, \
125  .capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_SLICE_THREADS, \
126  .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP, \
127  AV_SAMPLE_FMT_NONE }, \
128  .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE, \
129 };
130 
131 DSD_DECODER(DSD_LSBF, dsd_lsbf, "DSD (Direct Stream Digital), least significant bit first")
132 DSD_DECODER(DSD_MSBF, dsd_msbf, "DSD (Direct Stream Digital), most significant bit first")
133 DSD_DECODER(DSD_MSBF_PLANAR, dsd_msbf_planar, "DSD (Direct Stream Digital), most significant bit first, planar")
134 DSD_DECODER(DSD_LSBF_PLANAR, dsd_lsbf_planar, "DSD (Direct Stream Digital), least significant bit first, planar")
AV_SAMPLE_FMT_FLTP
@ AV_SAMPLE_FMT_FLTP
float, planar
Definition: samplefmt.h:69
td
#define td
Definition: regdef.h:70
AV_CODEC_ID_DSD_LSBF
@ AV_CODEC_ID_DSD_LSBF
Definition: codec_id.h:495
AVERROR
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later. That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another. Frame references ownership and permissions
dsd_channel
static int dsd_channel(AVCodecContext *avctx, void *tdata, int j, int threadnr)
Definition: dsddec.c:72
DSD_SILENCE
#define DSD_SILENCE
Definition: dsddec.c:33
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:303
internal.h
AVPacket::data
uint8_t * data
Definition: packet.h:373
data
const char data[16]
Definition: mxf.c:143
ThreadData::frame
AVFrame * frame
Definition: dsddec.c:68
dsd.h
bit
#define bit(string, value)
Definition: cbs_mpeg2.c:58
AV_CODEC_ID_DSD_MSBF_PLANAR
@ AV_CODEC_ID_DSD_MSBF_PLANAR
Definition: codec_id.h:498
ThreadData::avpkt
const AVPacket * avpkt
Definition: dsddec.c:69
first
trying all byte sequences megabyte in length and selecting the best looking sequence will yield cases to try But first
Definition: rate_distortion.txt:12
DSD_SILENCE_REVERSED
#define DSD_SILENCE_REVERSED
Definition: dsddec.c:34
av_cold
#define av_cold
Definition: attributes.h:90
s
#define s(width, name)
Definition: cbs_vp9.c:257
AVCodecContext::codec_id
enum AVCodecID codec_id
Definition: avcodec.h:389
NULL
#define NULL
Definition: coverity.c:32
DSD_DECODER
#define DSD_DECODER(id_, name_, long_name_)
Definition: dsddec.c:117
ff_init_dsd_data
av_cold void ff_init_dsd_data(void)
Definition: dsd.c:89
ff_get_buffer
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1648
AVPacket::size
int size
Definition: packet.h:374
AVCodecContext::sample_fmt
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:996
decode_frame
static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: dsddec.c:97
AVCodecContext::channels
int channels
number of audio channels
Definition: avcodec.h:989
i
int i
Definition: input.c:406
av_malloc_array
#define av_malloc_array(a, b)
Definition: tableprint_vlc.h:32
AV_CODEC_ID_DSD_LSBF_PLANAR
@ AV_CODEC_ID_DSD_LSBF_PLANAR
Definition: codec_id.h:497
ThreadData
Used for passing data between threads.
Definition: dsddec.c:67
DSDContext
Per-channel buffer.
Definition: dsd.h:41
avcodec.h
ff_dsd2pcm_translate
void ff_dsd2pcm_translate(DSDContext *s, size_t samples, int lsbf, const uint8_t *src, ptrdiff_t src_stride, float *dst, ptrdiff_t dst_stride)
Definition: dsd.c:95
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
decode_init
static av_cold int decode_init(AVCodecContext *avctx)
Definition: dsddec.c:41
AVCodecContext
main external API structure.
Definition: avcodec.h:379
AVPacket
This structure stores compressed data.
Definition: packet.h:350
AVCodecContext::priv_data
void * priv_data
Definition: avcodec.h:406
Stream
Definition: mpegts.c:110
AVERROR_INVALIDDATA
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
planar
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(const uint8_t *) pi - 0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(const int16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_S16, *(const int16_t *) pi *(1<< 16)) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(const int16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(const int32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(const int32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(const int64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64, *(const int64_t *) pi *(1.0f/(UINT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64, *(const int64_t *) pi *(1.0/(UINT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(const float *) pi *(UINT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(const double *) pi *(UINT64_C(1)<< 63))) #define FMT_PAIR_FUNC(out, in) static conv_func_type *const fmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB *AV_SAMPLE_FMT_NB]={ FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S64), };static void cpy1(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, len);} static void cpy2(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, 2 *len);} static void cpy4(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, 4 *len);} static void cpy8(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, 8 *len);} AudioConvert *swri_audio_convert_alloc(enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, const int *ch_map, int flags) { AudioConvert *ctx;conv_func_type *f=fmt_pair_to_conv_functions[av_get_packed_sample_fmt(out_fmt)+AV_SAMPLE_FMT_NB *av_get_packed_sample_fmt(in_fmt)];if(!f) return NULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) return NULL;if(channels==1){ in_fmt=av_get_planar_sample_fmt(in_fmt);out_fmt=av_get_planar_sample_fmt(out_fmt);} ctx->channels=channels;ctx->conv_f=f;ctx->ch_map=ch_map;if(in_fmt==AV_SAMPLE_FMT_U8||in_fmt==AV_SAMPLE_FMT_U8P) memset(ctx->silence, 0x80, sizeof(ctx->silence));if(out_fmt==in_fmt &&!ch_map) { switch(av_get_bytes_per_sample(in_fmt)){ case 1:ctx->simd_f=cpy1;break;case 2:ctx->simd_f=cpy2;break;case 4:ctx->simd_f=cpy4;break;case 8:ctx->simd_f=cpy8;break;} } if(HAVE_X86ASM &&1) swri_audio_convert_init_x86(ctx, out_fmt, in_fmt, channels);if(ARCH_ARM) swri_audio_convert_init_arm(ctx, out_fmt, in_fmt, channels);if(ARCH_AARCH64) swri_audio_convert_init_aarch64(ctx, out_fmt, in_fmt, channels);return ctx;} void swri_audio_convert_free(AudioConvert **ctx) { av_freep(ctx);} int swri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, int len) { int ch;int off=0;const int os=(out->planar ? 1 :out->ch_count) *out->bps;unsigned misaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask) { int planes=in->planar ? in->ch_count :1;unsigned m=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) in->ch[ch];misaligned|=m &ctx->in_simd_align_mask;} if(ctx->out_simd_align_mask) { int planes=out->planar ? out->ch_count :1;unsigned m=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) out->ch[ch];misaligned|=m &ctx->out_simd_align_mask;} if(ctx->simd_f &&!ctx->ch_map &&!misaligned){ off=len &~15;av_assert1(off >=0);av_assert1(off<=len);av_assert2(ctx->channels==SWR_CH_MAX||!in->ch[ctx->channels]);if(off >0){ if(out->planar==in->planar){ int planes=out->planar ? out->ch_count :1;for(ch=0;ch< planes;ch++){ ctx->simd_f(out->ch+ch,(const uint8_t **) in->ch+ch, off *(out-> planar
Definition: audioconvert.c:56
AVCodecContext::execute2
int(* execute2)(struct AVCodecContext *c, int(*func)(struct AVCodecContext *c2, void *arg, int jobnr, int threadnr), void *arg2, int *ret, int count)
The codec may call this to execute several independent things.
Definition: avcodec.h:1507