Libav
vmdaudio.c
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1 /*
2  * Sierra VMD audio decoder
3  *
4  * This file is part of Libav.
5  *
6  * Libav is free software; you can redistribute it and/or
7  * modify it under the terms of the GNU Lesser General Public
8  * License as published by the Free Software Foundation; either
9  * version 2.1 of the License, or (at your option) any later version.
10  *
11  * Libav is distributed in the hope that it will be useful,
12  * but WITHOUT ANY WARRANTY; without even the implied warranty of
13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14  * Lesser General Public License for more details.
15  *
16  * You should have received a copy of the GNU Lesser General Public
17  * License along with Libav; if not, write to the Free Software
18  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19  */
20 
35 #include <string.h>
36 
38 #include "libavutil/common.h"
39 #include "libavutil/intreadwrite.h"
40 
41 #include "avcodec.h"
42 #include "internal.h"
43 
44 #define BLOCK_TYPE_AUDIO 1
45 #define BLOCK_TYPE_INITIAL 2
46 #define BLOCK_TYPE_SILENCE 3
47 
48 typedef struct VmdAudioContext {
49  int out_bps;
52 
53 static const uint16_t vmdaudio_table[128] = {
54  0x000, 0x008, 0x010, 0x020, 0x030, 0x040, 0x050, 0x060, 0x070, 0x080,
55  0x090, 0x0A0, 0x0B0, 0x0C0, 0x0D0, 0x0E0, 0x0F0, 0x100, 0x110, 0x120,
56  0x130, 0x140, 0x150, 0x160, 0x170, 0x180, 0x190, 0x1A0, 0x1B0, 0x1C0,
57  0x1D0, 0x1E0, 0x1F0, 0x200, 0x208, 0x210, 0x218, 0x220, 0x228, 0x230,
58  0x238, 0x240, 0x248, 0x250, 0x258, 0x260, 0x268, 0x270, 0x278, 0x280,
59  0x288, 0x290, 0x298, 0x2A0, 0x2A8, 0x2B0, 0x2B8, 0x2C0, 0x2C8, 0x2D0,
60  0x2D8, 0x2E0, 0x2E8, 0x2F0, 0x2F8, 0x300, 0x308, 0x310, 0x318, 0x320,
61  0x328, 0x330, 0x338, 0x340, 0x348, 0x350, 0x358, 0x360, 0x368, 0x370,
62  0x378, 0x380, 0x388, 0x390, 0x398, 0x3A0, 0x3A8, 0x3B0, 0x3B8, 0x3C0,
63  0x3C8, 0x3D0, 0x3D8, 0x3E0, 0x3E8, 0x3F0, 0x3F8, 0x400, 0x440, 0x480,
64  0x4C0, 0x500, 0x540, 0x580, 0x5C0, 0x600, 0x640, 0x680, 0x6C0, 0x700,
65  0x740, 0x780, 0x7C0, 0x800, 0x900, 0xA00, 0xB00, 0xC00, 0xD00, 0xE00,
66  0xF00, 0x1000, 0x1400, 0x1800, 0x1C00, 0x2000, 0x3000, 0x4000
67 };
68 
70 {
71  VmdAudioContext *s = avctx->priv_data;
72 
73  if (avctx->channels < 1 || avctx->channels > 2) {
74  av_log(avctx, AV_LOG_ERROR, "invalid number of channels\n");
75  return AVERROR(EINVAL);
76  }
77  if (avctx->block_align < 1) {
78  av_log(avctx, AV_LOG_ERROR, "invalid block align\n");
79  return AVERROR(EINVAL);
80  }
81 
82  avctx->channel_layout = avctx->channels == 1 ? AV_CH_LAYOUT_MONO :
84 
85  if (avctx->bits_per_coded_sample == 16)
87  else
90 
91  s->chunk_size = avctx->block_align + avctx->channels * (s->out_bps == 2);
92 
93  av_log(avctx, AV_LOG_DEBUG, "%d channels, %d bits/sample, "
94  "block align = %d, sample rate = %d\n",
95  avctx->channels, avctx->bits_per_coded_sample, avctx->block_align,
96  avctx->sample_rate);
97 
98  return 0;
99 }
100 
101 static void decode_audio_s16(int16_t *out, const uint8_t *buf, int buf_size,
102  int channels)
103 {
104  int ch;
105  const uint8_t *buf_end = buf + buf_size;
106  int predictor[2];
107  int st = channels - 1;
108 
109  /* decode initial raw sample */
110  for (ch = 0; ch < channels; ch++) {
111  predictor[ch] = (int16_t)AV_RL16(buf);
112  buf += 2;
113  *out++ = predictor[ch];
114  }
115 
116  /* decode DPCM samples */
117  ch = 0;
118  while (buf < buf_end) {
119  uint8_t b = *buf++;
120  if (b & 0x80)
121  predictor[ch] -= vmdaudio_table[b & 0x7F];
122  else
123  predictor[ch] += vmdaudio_table[b];
124  predictor[ch] = av_clip_int16(predictor[ch]);
125  *out++ = predictor[ch];
126  ch ^= st;
127  }
128 }
129 
130 static int vmdaudio_decode_frame(AVCodecContext *avctx, void *data,
131  int *got_frame_ptr, AVPacket *avpkt)
132 {
133  AVFrame *frame = data;
134  const uint8_t *buf = avpkt->data;
135  const uint8_t *buf_end;
136  int buf_size = avpkt->size;
137  VmdAudioContext *s = avctx->priv_data;
138  int block_type, silent_chunks, audio_chunks;
139  int ret;
140  uint8_t *output_samples_u8;
141  int16_t *output_samples_s16;
142 
143  if (buf_size < 16) {
144  av_log(avctx, AV_LOG_WARNING, "skipping small junk packet\n");
145  *got_frame_ptr = 0;
146  return buf_size;
147  }
148 
149  block_type = buf[6];
150  if (block_type < BLOCK_TYPE_AUDIO || block_type > BLOCK_TYPE_SILENCE) {
151  av_log(avctx, AV_LOG_ERROR, "unknown block type: %d\n", block_type);
152  return AVERROR(EINVAL);
153  }
154  buf += 16;
155  buf_size -= 16;
156 
157  /* get number of silent chunks */
158  silent_chunks = 0;
159  if (block_type == BLOCK_TYPE_INITIAL) {
160  uint32_t flags;
161  if (buf_size < 4) {
162  av_log(avctx, AV_LOG_ERROR, "packet is too small\n");
163  return AVERROR(EINVAL);
164  }
165  flags = AV_RB32(buf);
166  silent_chunks = av_popcount(flags);
167  buf += 4;
168  buf_size -= 4;
169  } else if (block_type == BLOCK_TYPE_SILENCE) {
170  silent_chunks = 1;
171  buf_size = 0; // should already be zero but set it just to be sure
172  }
173 
174  /* ensure output buffer is large enough */
175  audio_chunks = buf_size / s->chunk_size;
176 
177  /* drop incomplete chunks */
178  buf_size = audio_chunks * s->chunk_size;
179 
180  /* get output buffer */
181  frame->nb_samples = ((silent_chunks + audio_chunks) * avctx->block_align) /
182  avctx->channels;
183  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
184  av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
185  return ret;
186  }
187  output_samples_u8 = frame->data[0];
188  output_samples_s16 = (int16_t *)frame->data[0];
189 
190  /* decode silent chunks */
191  if (silent_chunks > 0) {
192  int silent_size = FFMIN(avctx->block_align * silent_chunks,
193  frame->nb_samples * avctx->channels);
194  if (s->out_bps == 2) {
195  memset(output_samples_s16, 0x00, silent_size * 2);
196  output_samples_s16 += silent_size;
197  } else {
198  memset(output_samples_u8, 0x80, silent_size);
199  output_samples_u8 += silent_size;
200  }
201  }
202 
203  /* decode audio chunks */
204  if (audio_chunks > 0) {
205  buf_end = buf + (buf_size & ~(avctx->channels > 1));
206  while (buf + s->chunk_size <= buf_end) {
207  if (s->out_bps == 2) {
208  decode_audio_s16(output_samples_s16, buf, s->chunk_size,
209  avctx->channels);
210  output_samples_s16 += avctx->block_align;
211  } else {
212  memcpy(output_samples_u8, buf, s->chunk_size);
213  output_samples_u8 += avctx->block_align;
214  }
215  buf += s->chunk_size;
216  }
217  }
218 
219  *got_frame_ptr = 1;
220 
221  return avpkt->size;
222 }
223 
225  .name = "vmdaudio",
226  .long_name = NULL_IF_CONFIG_SMALL("Sierra VMD audio"),
227  .type = AVMEDIA_TYPE_AUDIO,
228  .id = AV_CODEC_ID_VMDAUDIO,
229  .priv_data_size = sizeof(VmdAudioContext),
232  .capabilities = CODEC_CAP_DR1,
233 };
This structure describes decoded (raw) audio or video data.
Definition: frame.h:135
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:129
int size
Definition: avcodec.h:974
#define AV_RL16
Definition: intreadwrite.h:42
#define AV_CH_LAYOUT_STEREO
AVCodec.
Definition: avcodec.h:2796
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs...
Definition: avcodec.h:1828
static int decode(MimicContext *ctx, int quality, int num_coeffs, int is_iframe)
Definition: mimic.c:275
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1799
uint8_t
#define av_cold
Definition: attributes.h:66
AV_SAMPLE_FMT_U8
#define AV_RB32
Definition: intreadwrite.h:130
#define b
Definition: input.c:52
#define CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
Definition: avcodec.h:684
const char data[16]
Definition: mxf.c:70
uint8_t * data
Definition: avcodec.h:973
static int flags
Definition: log.c:44
int bits_per_coded_sample
bits per sample/pixel from the demuxer (needed for huffyuv).
Definition: avcodec.h:2507
static av_cold int vmdaudio_decode_init(AVCodecContext *avctx)
Definition: vmdaudio.c:69
static void predictor(uint8_t *src, int size)
Definition: exr.c:151
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:123
#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:150
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:144
void av_log(void *avcl, int level, const char *fmt,...)
Definition: log.c:169
const char * name
Name of the codec implementation.
Definition: avcodec.h:2803
static void decode_audio_s16(int16_t *out, const uint8_t *buf, int buf_size, int channels)
Definition: vmdaudio.c:101
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:1852
audio channel layout utility functions
#define FFMIN(a, b)
Definition: common.h:57
#define BLOCK_TYPE_SILENCE
Definition: vmdaudio.c:46
static const uint16_t vmdaudio_table[128]
Definition: vmdaudio.c:53
if(ac->has_optimized_func)
Libavcodec external API header.
int sample_rate
samples per second
Definition: avcodec.h:1791
main external API structure.
Definition: avcodec.h:1050
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: utils.c:612
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:141
int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)
Return number of bytes per sample.
Definition: samplefmt.c:95
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method!=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt)>2){ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc){av_free(ac);returnNULL;}returnac;}in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_dlog(ac->avr,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> out
#define BLOCK_TYPE_INITIAL
Definition: vmdaudio.c:45
common internal api header.
common internal and external API header
signed 16 bits
Definition: samplefmt.h:64
AVCodec ff_vmdaudio_decoder
Definition: vmdaudio.c:224
static av_cold int init(AVCodecParserContext *s)
Definition: h264_parser.c:499
void * priv_data
Definition: avcodec.h:1092
int channels
number of audio channels
Definition: avcodec.h:1792
#define AV_CH_LAYOUT_MONO
This structure stores compressed data.
Definition: avcodec.h:950
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:179
static int vmdaudio_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: vmdaudio.c:130