FFmpeg  4.3
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:300
if(ret< 0)
Definition: vf_mcdeint.c:279
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
int size
Definition: packet.h:356
const char * b
Definition: vf_curves.c:116
#define AV_RL16
Definition: intreadwrite.h:42
#define AV_CH_LAYOUT_STEREO
AVCodec.
Definition: codec.h:190
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
#define AV_RB32
Definition: intreadwrite.h:130
static AVFrame * frame
const char data[16]
Definition: mxf.c:91
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:176
#define AVERROR(e)
Definition: error.h:43
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:186
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:197
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
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 &&HAVE_MMX) 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
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:1237
audio channel layout utility functions
#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
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
#define flags(name, subs,...)
Definition: cbs_av1.c:564
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:314
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
#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:366
#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