FFmpeg
samplefmt.h
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1 /*
2  * This file is part of FFmpeg.
3  *
4  * FFmpeg is free software; you can redistribute it and/or
5  * modify it under the terms of the GNU Lesser General Public
6  * License as published by the Free Software Foundation; either
7  * version 2.1 of the License, or (at your option) any later version.
8  *
9  * FFmpeg is distributed in the hope that it will be useful,
10  * but WITHOUT ANY WARRANTY; without even the implied warranty of
11  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12  * Lesser General Public License for more details.
13  *
14  * You should have received a copy of the GNU Lesser General Public
15  * License along with FFmpeg; if not, write to the Free Software
16  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
17  */
18 
19 #ifndef AVUTIL_SAMPLEFMT_H
20 #define AVUTIL_SAMPLEFMT_H
21 
22 #include <stdint.h>
23 
24 #include "avutil.h"
25 #include "attributes.h"
26 
27 /**
28  * @addtogroup lavu_audio
29  * @{
30  *
31  * @defgroup lavu_sampfmts Audio sample formats
32  *
33  * Audio sample format enumeration and related convenience functions.
34  * @{
35  */
36 
37 /**
38  * Audio sample formats
39  *
40  * - The data described by the sample format is always in native-endian order.
41  * Sample values can be expressed by native C types, hence the lack of a signed
42  * 24-bit sample format even though it is a common raw audio data format.
43  *
44  * - The floating-point formats are based on full volume being in the range
45  * [-1.0, 1.0]. Any values outside this range are beyond full volume level.
46  *
47  * - The data layout as used in av_samples_fill_arrays() and elsewhere in FFmpeg
48  * (such as AVFrame in libavcodec) is as follows:
49  *
50  * @par
51  * For planar sample formats, each audio channel is in a separate data plane,
52  * and linesize is the buffer size, in bytes, for a single plane. All data
53  * planes must be the same size. For packed sample formats, only the first data
54  * plane is used, and samples for each channel are interleaved. In this case,
55  * linesize is the buffer size, in bytes, for the 1 plane.
56  *
57  */
60  AV_SAMPLE_FMT_U8, ///< unsigned 8 bits
61  AV_SAMPLE_FMT_S16, ///< signed 16 bits
62  AV_SAMPLE_FMT_S32, ///< signed 32 bits
63  AV_SAMPLE_FMT_FLT, ///< float
64  AV_SAMPLE_FMT_DBL, ///< double
65 
66  AV_SAMPLE_FMT_U8P, ///< unsigned 8 bits, planar
67  AV_SAMPLE_FMT_S16P, ///< signed 16 bits, planar
68  AV_SAMPLE_FMT_S32P, ///< signed 32 bits, planar
69  AV_SAMPLE_FMT_FLTP, ///< float, planar
70  AV_SAMPLE_FMT_DBLP, ///< double, planar
71  AV_SAMPLE_FMT_S64, ///< signed 64 bits
72  AV_SAMPLE_FMT_S64P, ///< signed 64 bits, planar
73 
74  AV_SAMPLE_FMT_NB ///< Number of sample formats. DO NOT USE if linking dynamically
75 };
76 
77 /**
78  * Return the name of sample_fmt, or NULL if sample_fmt is not
79  * recognized.
80  */
81 const char *av_get_sample_fmt_name(enum AVSampleFormat sample_fmt);
82 
83 /**
84  * Return a sample format corresponding to name, or AV_SAMPLE_FMT_NONE
85  * on error.
86  */
87 enum AVSampleFormat av_get_sample_fmt(const char *name);
88 
89 /**
90  * Return the planar<->packed alternative form of the given sample format, or
91  * AV_SAMPLE_FMT_NONE on error. If the passed sample_fmt is already in the
92  * requested planar/packed format, the format returned is the same as the
93  * input.
94  */
96 
97 /**
98  * Get the packed alternative form of the given sample format.
99  *
100  * If the passed sample_fmt is already in packed format, the format returned is
101  * the same as the input.
102  *
103  * @return the packed alternative form of the given sample format or
104  AV_SAMPLE_FMT_NONE on error.
105  */
107 
108 /**
109  * Get the planar alternative form of the given sample format.
110  *
111  * If the passed sample_fmt is already in planar format, the format returned is
112  * the same as the input.
113  *
114  * @return the planar alternative form of the given sample format or
115  AV_SAMPLE_FMT_NONE on error.
116  */
118 
119 /**
120  * Generate a string corresponding to the sample format with
121  * sample_fmt, or a header if sample_fmt is negative.
122  *
123  * @param buf the buffer where to write the string
124  * @param buf_size the size of buf
125  * @param sample_fmt the number of the sample format to print the
126  * corresponding info string, or a negative value to print the
127  * corresponding header.
128  * @return the pointer to the filled buffer or NULL if sample_fmt is
129  * unknown or in case of other errors
130  */
131 char *av_get_sample_fmt_string(char *buf, int buf_size, enum AVSampleFormat sample_fmt);
132 
133 /**
134  * Return number of bytes per sample.
135  *
136  * @param sample_fmt the sample format
137  * @return number of bytes per sample or zero if unknown for the given
138  * sample format
139  */
140 int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt);
141 
142 /**
143  * Check if the sample format is planar.
144  *
145  * @param sample_fmt the sample format to inspect
146  * @return 1 if the sample format is planar, 0 if it is interleaved
147  */
148 int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt);
149 
150 /**
151  * Get the required buffer size for the given audio parameters.
152  *
153  * @param[out] linesize calculated linesize, may be NULL
154  * @param nb_channels the number of channels
155  * @param nb_samples the number of samples in a single channel
156  * @param sample_fmt the sample format
157  * @param align buffer size alignment (0 = default, 1 = no alignment)
158  * @return required buffer size, or negative error code on failure
159  */
160 int av_samples_get_buffer_size(int *linesize, int nb_channels, int nb_samples,
161  enum AVSampleFormat sample_fmt, int align);
162 
163 /**
164  * @}
165  *
166  * @defgroup lavu_sampmanip Samples manipulation
167  *
168  * Functions that manipulate audio samples
169  * @{
170  */
171 
172 /**
173  * Fill plane data pointers and linesize for samples with sample
174  * format sample_fmt.
175  *
176  * The audio_data array is filled with the pointers to the samples data planes:
177  * for planar, set the start point of each channel's data within the buffer,
178  * for packed, set the start point of the entire buffer only.
179  *
180  * The value pointed to by linesize is set to the aligned size of each
181  * channel's data buffer for planar layout, or to the aligned size of the
182  * buffer for all channels for packed layout.
183  *
184  * The buffer in buf must be big enough to contain all the samples
185  * (use av_samples_get_buffer_size() to compute its minimum size),
186  * otherwise the audio_data pointers will point to invalid data.
187  *
188  * @see enum AVSampleFormat
189  * The documentation for AVSampleFormat describes the data layout.
190  *
191  * @param[out] audio_data array to be filled with the pointer for each channel
192  * @param[out] linesize calculated linesize, may be NULL
193  * @param buf the pointer to a buffer containing the samples
194  * @param nb_channels the number of channels
195  * @param nb_samples the number of samples in a single channel
196  * @param sample_fmt the sample format
197  * @param align buffer size alignment (0 = default, 1 = no alignment)
198  * @return >=0 on success or a negative error code on failure
199  * @todo return minimum size in bytes required for the buffer in case
200  * of success at the next bump
201  */
202 int av_samples_fill_arrays(uint8_t **audio_data, int *linesize,
203  const uint8_t *buf,
204  int nb_channels, int nb_samples,
205  enum AVSampleFormat sample_fmt, int align);
206 
207 /**
208  * Allocate a samples buffer for nb_samples samples, and fill data pointers and
209  * linesize accordingly.
210  * The allocated samples buffer can be freed by using av_freep(&audio_data[0])
211  * Allocated data will be initialized to silence.
212  *
213  * @see enum AVSampleFormat
214  * The documentation for AVSampleFormat describes the data layout.
215  *
216  * @param[out] audio_data array to be filled with the pointer for each channel
217  * @param[out] linesize aligned size for audio buffer(s), may be NULL
218  * @param nb_channels number of audio channels
219  * @param nb_samples number of samples per channel
220  * @param align buffer size alignment (0 = default, 1 = no alignment)
221  * @return >=0 on success or a negative error code on failure
222  * @todo return the size of the allocated buffer in case of success at the next bump
223  * @see av_samples_fill_arrays()
224  * @see av_samples_alloc_array_and_samples()
225  */
226 int av_samples_alloc(uint8_t **audio_data, int *linesize, int nb_channels,
227  int nb_samples, enum AVSampleFormat sample_fmt, int align);
228 
229 /**
230  * Allocate a data pointers array, samples buffer for nb_samples
231  * samples, and fill data pointers and linesize accordingly.
232  *
233  * This is the same as av_samples_alloc(), but also allocates the data
234  * pointers array.
235  *
236  * @see av_samples_alloc()
237  */
238 int av_samples_alloc_array_and_samples(uint8_t ***audio_data, int *linesize, int nb_channels,
239  int nb_samples, enum AVSampleFormat sample_fmt, int align);
240 
241 /**
242  * Copy samples from src to dst.
243  *
244  * @param dst destination array of pointers to data planes
245  * @param src source array of pointers to data planes
246  * @param dst_offset offset in samples at which the data will be written to dst
247  * @param src_offset offset in samples at which the data will be read from src
248  * @param nb_samples number of samples to be copied
249  * @param nb_channels number of audio channels
250  * @param sample_fmt audio sample format
251  */
252 int av_samples_copy(uint8_t **dst, uint8_t * const *src, int dst_offset,
253  int src_offset, int nb_samples, int nb_channels,
254  enum AVSampleFormat sample_fmt);
255 
256 /**
257  * Fill an audio buffer with silence.
258  *
259  * @param audio_data array of pointers to data planes
260  * @param offset offset in samples at which to start filling
261  * @param nb_samples number of samples to fill
262  * @param nb_channels number of audio channels
263  * @param sample_fmt audio sample format
264  */
265 int av_samples_set_silence(uint8_t **audio_data, int offset, int nb_samples,
266  int nb_channels, enum AVSampleFormat sample_fmt);
267 
268 /**
269  * @}
270  * @}
271  */
272 #endif /* AVUTIL_SAMPLEFMT_H */
float, planar
Definition: samplefmt.h:69
Number of sample formats. DO NOT USE if linking dynamically.
Definition: samplefmt.h:74
char * av_get_sample_fmt_string(char *buf, int buf_size, enum AVSampleFormat sample_fmt)
Generate a string corresponding to the sample format with sample_fmt, or a header if sample_fmt is ne...
Definition: samplefmt.c:93
double, planar
Definition: samplefmt.h:70
Convenience header that includes libavutil&#39;s core.
#define src
Definition: vp8dsp.c:254
int av_samples_alloc_array_and_samples(uint8_t ***audio_data, int *linesize, int nb_channels, int nb_samples, enum AVSampleFormat sample_fmt, int align)
Allocate a data pointers array, samples buffer for nb_samples samples, and fill data pointers and lin...
Definition: samplefmt.c:198
Macro definitions for various function/variable attributes.
uint8_t
it s the only field you need to keep assuming you have a context There is some magic you don t need to care about around this just let it vf offset
enum AVSampleFormat av_get_planar_sample_fmt(enum AVSampleFormat sample_fmt)
Get the planar alternative form of the given sample format.
Definition: samplefmt.c:84
signed 32 bits
Definition: samplefmt.h:62
int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt)
Check if the sample format is planar.
Definition: samplefmt.c:112
int av_samples_set_silence(uint8_t **audio_data, int offset, int nb_samples, int nb_channels, enum AVSampleFormat sample_fmt)
Fill an audio buffer with silence.
Definition: samplefmt.c:237
const char * av_get_sample_fmt_name(enum AVSampleFormat sample_fmt)
Return the name of sample_fmt, or NULL if sample_fmt is not recognized.
Definition: samplefmt.c:49
unsigned 8 bits
Definition: samplefmt.h:60
signed 32 bits, planar
Definition: samplefmt.h:68
signed 64 bits
Definition: samplefmt.h:71
unsigned 8 bits, planar
Definition: samplefmt.h:66
const AVS_VideoInfo int align
Definition: avisynth_c.h:887
int av_samples_alloc(uint8_t **audio_data, int *linesize, int nb_channels, int nb_samples, enum AVSampleFormat sample_fmt, int align)
Allocate a samples buffer for nb_samples samples, and fill data pointers and linesize accordingly...
Definition: samplefmt.c:173
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
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 const uint8_t **in ch off *out planar
Definition: audioconvert.c:56
int av_samples_copy(uint8_t **dst, uint8_t *const *src, int dst_offset, int src_offset, int nb_samples, int nb_channels, enum AVSampleFormat sample_fmt)
Copy samples from src to dst.
Definition: samplefmt.c:213
void * buf
Definition: avisynth_c.h:766
int av_samples_get_buffer_size(int *linesize, int nb_channels, int nb_samples, enum AVSampleFormat sample_fmt, int align)
Get the required buffer size for the given audio parameters.
Definition: samplefmt.c:119
enum AVSampleFormat av_get_sample_fmt(const char *name)
Return a sample format corresponding to name, or AV_SAMPLE_FMT_NONE on error.
Definition: samplefmt.c:56
int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)
Return number of bytes per sample.
Definition: samplefmt.c:106
enum AVSampleFormat av_get_alt_sample_fmt(enum AVSampleFormat sample_fmt, int planar)
Return the planar<->packed alternative form of the given sample format, or AV_SAMPLE_FMT_NONE on erro...
Definition: samplefmt.c:66
enum AVSampleFormat av_get_packed_sample_fmt(enum AVSampleFormat sample_fmt)
Get the packed alternative form of the given sample format.
Definition: samplefmt.c:75
signed 16 bits
Definition: samplefmt.h:61
int av_samples_fill_arrays(uint8_t **audio_data, int *linesize, const uint8_t *buf, int nb_channels, int nb_samples, enum AVSampleFormat sample_fmt, int align)
Fill plane data pointers and linesize for samples with sample format sample_fmt.
Definition: samplefmt.c:151
signed 64 bits, planar
Definition: samplefmt.h:72
signed 16 bits, planar
Definition: samplefmt.h:67
int nb_channels
const char * name
Definition: opengl_enc.c:102