FFmpeg
vmdaudio.c
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1 /*
2  * Sierra VMD audio decoder
3  * Copyright (c) 2004 The FFmpeg Project
4  *
5  * This file is part of FFmpeg.
6  *
7  * FFmpeg is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * FFmpeg is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with FFmpeg; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
22 /**
23  * @file
24  * Sierra VMD audio decoder
25  * by Vladimir "VAG" Gneushev (vagsoft at mail.ru)
26  * for more information on the Sierra VMD format, visit:
27  * http://www.pcisys.net/~melanson/codecs/
28  *
29  * The audio decoder, expects each encoded data
30  * chunk to be prepended with the appropriate 16-byte frame information
31  * record from the VMD file. It does not require the 0x330-byte VMD file
32  * header, but it does need the audio setup parameters passed in through
33  * normal libavcodec API means.
34  */
35 
36 #include <string.h>
37 
38 #include "libavutil/avassert.h"
40 #include "libavutil/common.h"
41 #include "libavutil/intreadwrite.h"
42 
43 #include "avcodec.h"
44 #include "internal.h"
45 
46 #define BLOCK_TYPE_AUDIO 1
47 #define BLOCK_TYPE_INITIAL 2
48 #define BLOCK_TYPE_SILENCE 3
49 
50 typedef struct VmdAudioContext {
51  int out_bps;
54 
55 static const uint16_t vmdaudio_table[128] = {
56  0x000, 0x008, 0x010, 0x020, 0x030, 0x040, 0x050, 0x060, 0x070, 0x080,
57  0x090, 0x0A0, 0x0B0, 0x0C0, 0x0D0, 0x0E0, 0x0F0, 0x100, 0x110, 0x120,
58  0x130, 0x140, 0x150, 0x160, 0x170, 0x180, 0x190, 0x1A0, 0x1B0, 0x1C0,
59  0x1D0, 0x1E0, 0x1F0, 0x200, 0x208, 0x210, 0x218, 0x220, 0x228, 0x230,
60  0x238, 0x240, 0x248, 0x250, 0x258, 0x260, 0x268, 0x270, 0x278, 0x280,
61  0x288, 0x290, 0x298, 0x2A0, 0x2A8, 0x2B0, 0x2B8, 0x2C0, 0x2C8, 0x2D0,
62  0x2D8, 0x2E0, 0x2E8, 0x2F0, 0x2F8, 0x300, 0x308, 0x310, 0x318, 0x320,
63  0x328, 0x330, 0x338, 0x340, 0x348, 0x350, 0x358, 0x360, 0x368, 0x370,
64  0x378, 0x380, 0x388, 0x390, 0x398, 0x3A0, 0x3A8, 0x3B0, 0x3B8, 0x3C0,
65  0x3C8, 0x3D0, 0x3D8, 0x3E0, 0x3E8, 0x3F0, 0x3F8, 0x400, 0x440, 0x480,
66  0x4C0, 0x500, 0x540, 0x580, 0x5C0, 0x600, 0x640, 0x680, 0x6C0, 0x700,
67  0x740, 0x780, 0x7C0, 0x800, 0x900, 0xA00, 0xB00, 0xC00, 0xD00, 0xE00,
68  0xF00, 0x1000, 0x1400, 0x1800, 0x1C00, 0x2000, 0x3000, 0x4000
69 };
70 
72 {
73  VmdAudioContext *s = avctx->priv_data;
74 
75  if (avctx->channels < 1 || avctx->channels > 2) {
76  av_log(avctx, AV_LOG_ERROR, "invalid number of channels\n");
77  return AVERROR(EINVAL);
78  }
79  if (avctx->block_align < 1 || avctx->block_align % avctx->channels ||
80  avctx->block_align > INT_MAX - avctx->channels
81  ) {
82  av_log(avctx, AV_LOG_ERROR, "invalid block align\n");
83  return AVERROR(EINVAL);
84  }
85 
86  avctx->channel_layout = avctx->channels == 1 ? AV_CH_LAYOUT_MONO :
88 
89  if (avctx->bits_per_coded_sample == 16)
91  else
94 
95  s->chunk_size = avctx->block_align + avctx->channels * (s->out_bps == 2);
96 
97  av_log(avctx, AV_LOG_DEBUG, "%d channels, %d bits/sample, "
98  "block align = %d, sample rate = %d\n",
99  avctx->channels, avctx->bits_per_coded_sample, avctx->block_align,
100  avctx->sample_rate);
101 
102  return 0;
103 }
104 
105 static void decode_audio_s16(int16_t *out, const uint8_t *buf, int buf_size,
106  int channels)
107 {
108  int ch;
109  const uint8_t *buf_end = buf + buf_size;
110  int predictor[2];
111  int st = channels - 1;
112 
113  /* decode initial raw sample */
114  for (ch = 0; ch < channels; ch++) {
115  predictor[ch] = (int16_t)AV_RL16(buf);
116  buf += 2;
117  *out++ = predictor[ch];
118  }
119 
120  /* decode DPCM samples */
121  ch = 0;
122  while (buf < buf_end) {
123  uint8_t b = *buf++;
124  if (b & 0x80)
125  predictor[ch] -= vmdaudio_table[b & 0x7F];
126  else
127  predictor[ch] += vmdaudio_table[b];
128  predictor[ch] = av_clip_int16(predictor[ch]);
129  *out++ = predictor[ch];
130  ch ^= st;
131  }
132 }
133 
134 static int vmdaudio_decode_frame(AVCodecContext *avctx, void *data,
135  int *got_frame_ptr, AVPacket *avpkt)
136 {
137  AVFrame *frame = data;
138  const uint8_t *buf = avpkt->data;
139  const uint8_t *buf_end;
140  int buf_size = avpkt->size;
141  VmdAudioContext *s = avctx->priv_data;
142  int block_type, silent_chunks, audio_chunks;
143  int ret;
144  uint8_t *output_samples_u8;
145  int16_t *output_samples_s16;
146 
147  if (buf_size < 16) {
148  av_log(avctx, AV_LOG_WARNING, "skipping small junk packet\n");
149  *got_frame_ptr = 0;
150  return buf_size;
151  }
152 
153  block_type = buf[6];
154  if (block_type < BLOCK_TYPE_AUDIO || block_type > BLOCK_TYPE_SILENCE) {
155  av_log(avctx, AV_LOG_ERROR, "unknown block type: %d\n", block_type);
156  return AVERROR(EINVAL);
157  }
158  buf += 16;
159  buf_size -= 16;
160 
161  /* get number of silent chunks */
162  silent_chunks = 0;
163  if (block_type == BLOCK_TYPE_INITIAL) {
164  uint32_t flags;
165  if (buf_size < 4) {
166  av_log(avctx, AV_LOG_ERROR, "packet is too small\n");
167  return AVERROR(EINVAL);
168  }
169  flags = AV_RB32(buf);
170  silent_chunks = av_popcount(flags);
171  buf += 4;
172  buf_size -= 4;
173  } else if (block_type == BLOCK_TYPE_SILENCE) {
174  silent_chunks = 1;
175  buf_size = 0; // should already be zero but set it just to be sure
176  }
177 
178  /* ensure output buffer is large enough */
179  audio_chunks = buf_size / s->chunk_size;
180 
181  /* drop incomplete chunks */
182  buf_size = audio_chunks * s->chunk_size;
183 
184  if (silent_chunks + audio_chunks >= INT_MAX / avctx->block_align)
185  return AVERROR_INVALIDDATA;
186 
187  /* get output buffer */
188  frame->nb_samples = ((silent_chunks + audio_chunks) * avctx->block_align) /
189  avctx->channels;
190  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
191  return ret;
192  output_samples_u8 = frame->data[0];
193  output_samples_s16 = (int16_t *)frame->data[0];
194 
195  /* decode silent chunks */
196  if (silent_chunks > 0) {
197  int silent_size = avctx->block_align * silent_chunks;
198  av_assert0(avctx->block_align * silent_chunks <= frame->nb_samples * avctx->channels);
199 
200  if (s->out_bps == 2) {
201  memset(output_samples_s16, 0x00, silent_size * 2);
202  output_samples_s16 += silent_size;
203  } else {
204  memset(output_samples_u8, 0x80, silent_size);
205  output_samples_u8 += silent_size;
206  }
207  }
208 
209  /* decode audio chunks */
210  if (audio_chunks > 0) {
211  buf_end = buf + buf_size;
212  av_assert0((buf_size & (avctx->channels > 1)) == 0);
213  while (buf_end - buf >= s->chunk_size) {
214  if (s->out_bps == 2) {
215  decode_audio_s16(output_samples_s16, buf, s->chunk_size,
216  avctx->channels);
217  output_samples_s16 += avctx->block_align;
218  } else {
219  memcpy(output_samples_u8, buf, s->chunk_size);
220  output_samples_u8 += avctx->block_align;
221  }
222  buf += s->chunk_size;
223  }
224  }
225 
226  *got_frame_ptr = 1;
227 
228  return avpkt->size;
229 }
230 
232  .name = "vmdaudio",
233  .long_name = NULL_IF_CONFIG_SMALL("Sierra VMD audio"),
234  .type = AVMEDIA_TYPE_AUDIO,
235  .id = AV_CODEC_ID_VMDAUDIO,
236  .priv_data_size = sizeof(VmdAudioContext),
239  .capabilities = AV_CODEC_CAP_DR1,
240 };
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
This structure describes decoded (raw) audio or video data.
Definition: frame.h:308
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:100
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:200
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
int size
Definition: packet.h:356
#define AV_CH_LAYOUT_STEREO
AVCodec.
Definition: codec.h:190
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_RL16
Definition: bytestream.h:87
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs...
Definition: avcodec.h:1223
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
Definition: decode_audio.c:71
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1194
uint8_t
#define av_cold
Definition: attributes.h:88
AV_SAMPLE_FMT_U8
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_RB32
Definition: bytestream.h:87
uint8_t * data
Definition: packet.h:355
int bits_per_coded_sample
bits per sample/pixel from the demuxer (needed for huffyuv).
Definition: avcodec.h:1750
channels
Definition: aptx.h:33
#define av_log(a,...)
static av_cold int vmdaudio_decode_init(AVCodecContext *avctx)
Definition: vmdaudio.c:71
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:194
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:153
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:215
simple assert() macros that are a bit more flexible than ISO C assert().
const char * name
Name of the codec implementation.
Definition: codec.h:197
static void decode_audio_s16(int16_t *out, const uint8_t *buf, int buf_size, int channels)
Definition: vmdaudio.c:105
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:1237
#define b
Definition: input.c:41
audio channel layout utility functions
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
#define s(width, name)
Definition: cbs_vp9.c:257
#define BLOCK_TYPE_SILENCE
Definition: vmdaudio.c:48
static const uint16_t vmdaudio_table[128]
Definition: vmdaudio.c:55
if(ret)
Libavcodec external API header.
int sample_rate
samples per second
Definition: avcodec.h:1186
main external API structure.
Definition: avcodec.h:526
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1854
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_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
Definition: audioconvert.c:56
#define flags(name, subs,...)
Definition: cbs_av1.c:560
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:322
int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)
Return number of bytes per sample.
Definition: samplefmt.c:106
#define BLOCK_TYPE_INITIAL
Definition: vmdaudio.c:47
common internal api header.
common internal and external API header
signed 16 bits
Definition: samplefmt.h:61
AVCodec ff_vmdaudio_decoder
Definition: vmdaudio.c:231
void * priv_data
Definition: avcodec.h:553
int channels
number of audio channels
Definition: avcodec.h:1187
FILE * out
Definition: movenc.c:54
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
#define AV_CH_LAYOUT_MONO
This structure stores compressed data.
Definition: packet.h:332
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:374
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
Definition: codec.h:50
static int vmdaudio_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: vmdaudio.c:134