FFmpeg  2.1.1
swresample.h
Go to the documentation of this file.
1 /*
2  * Copyright (C) 2011-2013 Michael Niedermayer (michaelni@gmx.at)
3  *
4  * This file is part of libswresample
5  *
6  * libswresample 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  * libswresample 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 libswresample; if not, write to the Free Software
18  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19  */
20 
21 #ifndef SWRESAMPLE_SWRESAMPLE_H
22 #define SWRESAMPLE_SWRESAMPLE_H
23 
24 /**
25  * @file
26  * @ingroup lswr
27  * libswresample public header
28  */
29 
30 /**
31  * @defgroup lswr Libswresample
32  * @{
33  *
34  * Libswresample (lswr) is a library that handles audio resampling, sample
35  * format conversion and mixing.
36  *
37  * Interaction with lswr is done through SwrContext, which is
38  * allocated with swr_alloc() or swr_alloc_set_opts(). It is opaque, so all parameters
39  * must be set with the @ref avoptions API.
40  *
41  * For example the following code will setup conversion from planar float sample
42  * format to interleaved signed 16-bit integer, downsampling from 48kHz to
43  * 44.1kHz and downmixing from 5.1 channels to stereo (using the default mixing
44  * matrix):
45  * @code
46  * SwrContext *swr = swr_alloc();
47  * av_opt_set_channel_layout(swr, "in_channel_layout", AV_CH_LAYOUT_5POINT1, 0);
48  * av_opt_set_channel_layout(swr, "out_channel_layout", AV_CH_LAYOUT_STEREO, 0);
49  * av_opt_set_int(swr, "in_sample_rate", 48000, 0);
50  * av_opt_set_int(swr, "out_sample_rate", 44100, 0);
51  * av_opt_set_sample_fmt(swr, "in_sample_fmt", AV_SAMPLE_FMT_FLTP, 0);
52  * av_opt_set_sample_fmt(swr, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0);
53  * @endcode
54  *
55  * Once all values have been set, it must be initialized with swr_init(). If
56  * you need to change the conversion parameters, you can change the parameters
57  * as described above, or by using swr_alloc_set_opts(), then call swr_init()
58  * again.
59  *
60  * The conversion itself is done by repeatedly calling swr_convert().
61  * Note that the samples may get buffered in swr if you provide insufficient
62  * output space or if sample rate conversion is done, which requires "future"
63  * samples. Samples that do not require future input can be retrieved at any
64  * time by using swr_convert() (in_count can be set to 0).
65  * At the end of conversion the resampling buffer can be flushed by calling
66  * swr_convert() with NULL in and 0 in_count.
67  *
68  * The delay between input and output, can at any time be found by using
69  * swr_get_delay().
70  *
71  * The following code demonstrates the conversion loop assuming the parameters
72  * from above and caller-defined functions get_input() and handle_output():
73  * @code
74  * uint8_t **input;
75  * int in_samples;
76  *
77  * while (get_input(&input, &in_samples)) {
78  * uint8_t *output;
79  * int out_samples = av_rescale_rnd(swr_get_delay(swr, 48000) +
80  * in_samples, 44100, 48000, AV_ROUND_UP);
81  * av_samples_alloc(&output, NULL, 2, out_samples,
82  * AV_SAMPLE_FMT_S16, 0);
83  * out_samples = swr_convert(swr, &output, out_samples,
84  * input, in_samples);
85  * handle_output(output, out_samples);
86  * av_freep(&output);
87  * }
88  * @endcode
89  *
90  * When the conversion is finished, the conversion
91  * context and everything associated with it must be freed with swr_free().
92  * There will be no memory leak if the data is not completely flushed before
93  * swr_free().
94  */
95 
96 #include <stdint.h>
97 #include "libavutil/samplefmt.h"
98 
99 #include "libswresample/version.h"
100 
101 #if LIBSWRESAMPLE_VERSION_MAJOR < 1
102 #define SWR_CH_MAX 32 ///< Maximum number of channels
103 #endif
104 
105 #define SWR_FLAG_RESAMPLE 1 ///< Force resampling even if equal sample rate
106 //TODO use int resample ?
107 //long term TODO can we enable this dynamically?
108 
114 
115  SWR_DITHER_NS = 64, ///< not part of API/ABI
123  SWR_DITHER_NB, ///< not part of API/ABI
124 };
125 
126 /** Resampling Engines */
127 enum SwrEngine {
128  SWR_ENGINE_SWR, /**< SW Resampler */
129  SWR_ENGINE_SOXR, /**< SoX Resampler */
130  SWR_ENGINE_NB, ///< not part of API/ABI
131 };
132 
133 /** Resampling Filter Types */
135  SWR_FILTER_TYPE_CUBIC, /**< Cubic */
136  SWR_FILTER_TYPE_BLACKMAN_NUTTALL, /**< Blackman Nuttall Windowed Sinc */
137  SWR_FILTER_TYPE_KAISER, /**< Kaiser Windowed Sinc */
138 };
139 
140 typedef struct SwrContext SwrContext;
141 
142 /**
143  * Get the AVClass for swrContext. It can be used in combination with
144  * AV_OPT_SEARCH_FAKE_OBJ for examining options.
145  *
146  * @see av_opt_find().
147  */
148 const AVClass *swr_get_class(void);
149 
150 /**
151  * Allocate SwrContext.
152  *
153  * If you use this function you will need to set the parameters (manually or
154  * with swr_alloc_set_opts()) before calling swr_init().
155  *
156  * @see swr_alloc_set_opts(), swr_init(), swr_free()
157  * @return NULL on error, allocated context otherwise
158  */
159 struct SwrContext *swr_alloc(void);
160 
161 /**
162  * Initialize context after user parameters have been set.
163  *
164  * @return AVERROR error code in case of failure.
165  */
166 int swr_init(struct SwrContext *s);
167 
168 /**
169  * Allocate SwrContext if needed and set/reset common parameters.
170  *
171  * This function does not require s to be allocated with swr_alloc(). On the
172  * other hand, swr_alloc() can use swr_alloc_set_opts() to set the parameters
173  * on the allocated context.
174  *
175  * @param s Swr context, can be NULL
176  * @param out_ch_layout output channel layout (AV_CH_LAYOUT_*)
177  * @param out_sample_fmt output sample format (AV_SAMPLE_FMT_*).
178  * @param out_sample_rate output sample rate (frequency in Hz)
179  * @param in_ch_layout input channel layout (AV_CH_LAYOUT_*)
180  * @param in_sample_fmt input sample format (AV_SAMPLE_FMT_*).
181  * @param in_sample_rate input sample rate (frequency in Hz)
182  * @param log_offset logging level offset
183  * @param log_ctx parent logging context, can be NULL
184  *
185  * @see swr_init(), swr_free()
186  * @return NULL on error, allocated context otherwise
187  */
188 struct SwrContext *swr_alloc_set_opts(struct SwrContext *s,
191  int log_offset, void *log_ctx);
192 
193 /**
194  * Free the given SwrContext and set the pointer to NULL.
195  */
196 void swr_free(struct SwrContext **s);
197 
198 /**
199  * Convert audio.
200  *
201  * in and in_count can be set to 0 to flush the last few samples out at the
202  * end.
203  *
204  * If more input is provided than output space then the input will be buffered.
205  * You can avoid this buffering by providing more output space than input.
206  * Convertion will run directly without copying whenever possible.
207  *
208  * @param s allocated Swr context, with parameters set
209  * @param out output buffers, only the first one need be set in case of packed audio
210  * @param out_count amount of space available for output in samples per channel
211  * @param in input buffers, only the first one need to be set in case of packed audio
212  * @param in_count number of input samples available in one channel
213  *
214  * @return number of samples output per channel, negative value on error
215  */
216 int swr_convert(struct SwrContext *s, uint8_t **out, int out_count,
217  const uint8_t **in , int in_count);
218 
219 /**
220  * Convert the next timestamp from input to output
221  * timestamps are in 1/(in_sample_rate * out_sample_rate) units.
222  *
223  * @note There are 2 slightly differently behaving modes.
224  * First is when automatic timestamp compensation is not used, (min_compensation >= FLT_MAX)
225  * in this case timestamps will be passed through with delays compensated
226  * Second is when automatic timestamp compensation is used, (min_compensation < FLT_MAX)
227  * in this case the output timestamps will match output sample numbers
228  *
229  * @param pts timestamp for the next input sample, INT64_MIN if unknown
230  * @return the output timestamp for the next output sample
231  */
232 int64_t swr_next_pts(struct SwrContext *s, int64_t pts);
233 
234 /**
235  * Activate resampling compensation.
236  */
237 int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance);
238 
239 /**
240  * Set a customized input channel mapping.
241  *
242  * @param s allocated Swr context, not yet initialized
243  * @param channel_map customized input channel mapping (array of channel
244  * indexes, -1 for a muted channel)
245  * @return AVERROR error code in case of failure.
246  */
247 int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map);
248 
249 /**
250  * Set a customized remix matrix.
251  *
252  * @param s allocated Swr context, not yet initialized
253  * @param matrix remix coefficients; matrix[i + stride * o] is
254  * the weight of input channel i in output channel o
255  * @param stride offset between lines of the matrix
256  * @return AVERROR error code in case of failure.
257  */
258 int swr_set_matrix(struct SwrContext *s, const double *matrix, int stride);
259 
260 /**
261  * Drops the specified number of output samples.
262  */
263 int swr_drop_output(struct SwrContext *s, int count);
264 
265 /**
266  * Injects the specified number of silence samples.
267  */
268 int swr_inject_silence(struct SwrContext *s, int count);
269 
270 /**
271  * Gets the delay the next input sample will experience relative to the next output sample.
272  *
273  * Swresample can buffer data if more input has been provided than available
274  * output space, also converting between sample rates needs a delay.
275  * This function returns the sum of all such delays.
276  * The exact delay is not necessarily an integer value in either input or
277  * output sample rate. Especially when downsampling by a large value, the
278  * output sample rate may be a poor choice to represent the delay, similarly
279  * for upsampling and the input sample rate.
280  *
281  * @param s swr context
282  * @param base timebase in which the returned delay will be
283  * if its set to 1 the returned delay is in seconds
284  * if its set to 1000 the returned delay is in milli seconds
285  * if its set to the input sample rate then the returned delay is in input samples
286  * if its set to the output sample rate then the returned delay is in output samples
287  * an exact rounding free delay can be found by using LCM(in_sample_rate, out_sample_rate)
288  * @returns the delay in 1/base units.
289  */
290 int64_t swr_get_delay(struct SwrContext *s, int64_t base);
291 
292 /**
293  * Return the LIBSWRESAMPLE_VERSION_INT constant.
294  */
295 unsigned swresample_version(void);
296 
297 /**
298  * Return the swr build-time configuration.
299  */
300 const char *swresample_configuration(void);
301 
302 /**
303  * Return the swr license.
304  */
305 const char *swresample_license(void);
306 
307 /**
308  * @}
309  */
310 
311 #endif /* SWRESAMPLE_SWRESAMPLE_H */
const char * s
Definition: avisynth_c.h:668
int out_sample_rate
output sample rate
SoX Resampler.
Definition: swresample.h:129
struct SwrContext * swr_alloc_set_opts(struct SwrContext *s, int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate, int64_t in_ch_layout, enum AVSampleFormat in_sample_fmt, int in_sample_rate, int log_offset, void *log_ctx)
Allocate SwrContext if needed and set/reset common parameters.
Definition: swresample.c:187
SwrFilterType
Resampling Filter Types.
Definition: swresample.h:134
int64_t swr_next_pts(struct SwrContext *s, int64_t pts)
Convert the next timestamp from input to output timestamps are in 1/(in_sample_rate * out_sample_rate...
Definition: swresample.c:902
struct SwrContext * swr_alloc(void)
Allocate SwrContext.
Definition: swresample.c:178
int stride
Definition: mace.c:144
const char * swresample_license(void)
Return the swr license.
Definition: swresample.c:160
const int * channel_map
channel index (or -1 if muted channel) map
int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance)
Activate resampling compensation.
Definition: swresample.c:882
uint8_t
SwrDitherType
Definition: swresample.h:109
void * log_ctx
parent logging context
int swr_convert(struct SwrContext *s, uint8_t **out, int out_count, const uint8_t **in, int in_count)
Convert audio.
Kaiser Windowed Sinc.
Definition: swresample.h:137
enum AVSampleFormat out_sample_fmt
output sample format
SwrEngine
Resampling Engines.
Definition: swresample.h:127
Blackman Nuttall Windowed Sinc.
Definition: swresample.h:136
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){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));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=av_sample_fmt_is_planar(in_fmt);out_planar=av_sample_fmt_is_planar(out_fmt);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);ff_audio_convert_init_arm(ac);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-> in
int64_t swr_get_delay(struct SwrContext *s, int64_t base)
Gets the delay the next input sample will experience relative to the next output sample.
Definition: swresample.c:874
not part of API/ABI
Definition: swresample.h:130
not part of API/ABI
Definition: swresample.h:123
int swr_drop_output(struct SwrContext *s, int count)
Drops the specified number of output samples.
Definition: swresample.c:836
int swr_set_matrix(struct SwrContext *s, const double *matrix, int stride)
Set a customized remix matrix.
Definition: rematrix.c:60
int64_t out_ch_layout
output channel layout
not part of API/ABI
Definition: swresample.h:115
int in_sample_rate
input sample rate
AVSampleFormat
Audio Sample Formats.
Definition: samplefmt.h:49
void swr_free(struct SwrContext **s)
Free the given SwrContext and set the pointer to NULL.
Definition: swresample.c:221
int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map)
Set a customized input channel mapping.
Definition: swresample.c:166
Describe the class of an AVClass context structure.
Definition: log.h:50
enum AVSampleFormat in_sample_fmt
input sample format
SW Resampler.
Definition: swresample.h:128
int64_t in_ch_layout
input channel layout
const char * swresample_configuration(void)
Return the swr build-time configuration.
Definition: swresample.c:155
unsigned swresample_version(void)
Return the LIBSWRESAMPLE_VERSION_INT constant.
Definition: swresample.c:149
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){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));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=av_sample_fmt_is_planar(in_fmt);out_planar=av_sample_fmt_is_planar(out_fmt);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);ff_audio_convert_init_arm(ac);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
float matrix[SWR_CH_MAX][SWR_CH_MAX]
floating point rematrixing coefficients
void INT64 INT64 count
Definition: avisynth_c.h:594
const AVClass * swr_get_class(void)
Get the AVClass for swrContext.
Definition: swresample.c:173
int swr_inject_silence(struct SwrContext *s, int count)
Injects the specified number of silence samples.
Definition: swresample.c:846
int swr_init(struct SwrContext *s)
Initialize context after user parameters have been set.
Definition: swresample.c:243