FFmpeg: libavcodec/dsddec.c Source File

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libavcodec

dsddec.c

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1 /*

2 * Direct Stream Digital (DSD) decoder

3 * based on BSD licensed dsd2pcm by Sebastian Gesemann

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 */

24 /**

25 * @file

26 * Direct Stream Digital (DSD) decoder

27 */

29 #include "libavcodec/internal.h"

30 #include "avcodec.h"

31 #include "dsd.h"

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 */

41 static av_cold int decode_init(AVCodecContext *avctx)

42 {

43 DSDContext * s;

44 int i;

45 uint8_t silence;

47 if (!avctx->channels)

48 return AVERROR_INVALIDDATA;

50 ff_init_dsd_data();

52 s = av_malloc_array(sizeof(DSDContext), avctx->channels);

53 if (!s)

54 return AVERROR(ENOMEM);

56 silence = avctx->codec_id == AV_CODEC_ID_DSD_LSBF || avctx->codec_id == AV_CODEC_ID_DSD_LSBF_PLANAR ? DSD_SILENCE_REVERSED : DSD_SILENCE;

57 for (i = 0; i < avctx->channels; i++) {

58 s[i].pos = 0;

59 memset(s[i].buf, silence, sizeof(s[i].buf));

60 }

62 avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;

63 avctx->priv_data = s;

64 return 0;

65 }

67 typedef struct ThreadData {

68 AVFrame *frame;

69 const AVPacket *avpkt;

70 } ThreadData;

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];

82 if (avctx->codec_id == AV_CODEC_ID_DSD_LSBF_PLANAR || avctx->codec_id == AV_CODEC_ID_DSD_MSBF_PLANAR) {

83 src_next = frame->nb_samples;

84 src_stride = 1;

85 } else {

86 src_next = 1;

87 src_stride = avctx->channels;

88 }

90 ff_dsd2pcm_translate(&s[j], frame->nb_samples, lsbf,

91 avpkt->data + j * src_next, src_stride,

92 dst, 1);

94 return 0;

95 }

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

#define td

Definition: regdef.h:70

AV_CODEC_ID_DSD_LSBF

@ AV_CODEC_ID_DSD_LSBF

Definition: codec_id.h:496

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:317

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:499

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

#define s(width, name)

Definition: cbs_vp9.c:257

AVCodecContext::codec_id

enum AVCodecID codec_id

Definition: avcodec.h:393

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:1652

AVPacket::size

int size

Definition: packet.h:374

AVCodecContext::sample_fmt

enum AVSampleFormat sample_fmt

audio sample format

Definition: avcodec.h:1000

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:993

#define i(width, name, range_min, range_max)

Definition: cbs_h2645.c:271

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:498

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

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:383

AVPacket

This structure stores compressed data.

Definition: packet.h:350

AVCodecContext::priv_data

void * priv_data

Definition: avcodec.h:410

Stream

Definition: mpegts.c:110

AVERROR_INVALIDDATA

#define AVERROR_INVALIDDATA

Invalid data found when processing input.

Definition: error.h:61

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:1511

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