2 * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
4 * This file is part of FFmpeg.
6 * FFmpeg 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.
11 * FFmpeg 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.
16 * You should have received a copy of the GNU Lesser General Public
17 * License along with FFmpeg; if not, write to the Free Software
18 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
23 #include "libavutil/common.h"
24 #include "libavutil/libm.h"
25 #include "libavutil/samplefmt.h"
26 #include "avresample.h"
28 #include "audio_data.h"
29 #include "audio_mix.h"
31 static const char * const coeff_type_names
[] = { "q8", "q15", "flt" };
34 AVAudioResampleContext
*avr
;
35 enum AVSampleFormat fmt
;
36 enum AVMixCoeffType coeff_type
;
44 int has_optimized_func
;
45 const char *func_descr
;
46 const char *func_descr_generic
;
48 mix_func
*mix_generic
;
50 int in_matrix_channels
;
51 int out_matrix_channels
;
52 int output_zero
[AVRESAMPLE_MAX_CHANNELS
];
53 int input_skip
[AVRESAMPLE_MAX_CHANNELS
];
54 int output_skip
[AVRESAMPLE_MAX_CHANNELS
];
55 int16_t *matrix_q8
[AVRESAMPLE_MAX_CHANNELS
];
56 int32_t *matrix_q15
[AVRESAMPLE_MAX_CHANNELS
];
57 float *matrix_flt
[AVRESAMPLE_MAX_CHANNELS
];
61 void ff_audio_mix_set_func(AudioMix
*am
, enum AVSampleFormat fmt
,
62 enum AVMixCoeffType coeff_type
, int in_channels
,
63 int out_channels
, int ptr_align
, int samples_align
,
64 const char *descr
, void *mix_func
)
66 if (fmt
== am
->fmt
&& coeff_type
== am
->coeff_type
&&
67 ( in_channels
== am
->in_matrix_channels
|| in_channels
== 0) &&
68 (out_channels
== am
->out_matrix_channels
|| out_channels
== 0)) {
71 am
->func_descr
= descr
;
72 am
->ptr_align
= ptr_align
;
73 am
->samples_align
= samples_align
;
74 if (ptr_align
== 1 && samples_align
== 1) {
75 am
->mix_generic
= mix_func
;
76 am
->func_descr_generic
= descr
;
78 am
->has_optimized_func
= 1;
82 snprintf(chan_str
, sizeof(chan_str
), "[%d to %d] ",
83 in_channels
, out_channels
);
85 snprintf(chan_str
, sizeof(chan_str
), "[%d to any] ",
87 } else if (out_channels
) {
88 snprintf(chan_str
, sizeof(chan_str
), "[any to %d] ",
91 snprintf(chan_str
, sizeof(chan_str
), "[any to any] ");
93 av_log(am
->avr
, AV_LOG_DEBUG
, "audio_mix: found function: [fmt=%s] "
94 "[c=%s] %s(%s)\n", av_get_sample_fmt_name(fmt
),
95 coeff_type_names
[coeff_type
], chan_str
, descr
);
99 #define MIX_FUNC_NAME(fmt, cfmt) mix_any_ ## fmt ##_## cfmt ##_c
101 #define MIX_FUNC_GENERIC(fmt, cfmt, stype, ctype, sumtype, expr) \
102 static void MIX_FUNC_NAME(fmt, cfmt)(stype **samples, ctype **matrix, \
103 int len, int out_ch, int in_ch) \
106 stype temp[AVRESAMPLE_MAX_CHANNELS]; \
107 for (i = 0; i < len; i++) { \
108 for (out = 0; out < out_ch; out++) { \
110 for (in = 0; in < in_ch; in++) \
111 sum += samples[in][i] * matrix[out][in]; \
114 for (out = 0; out < out_ch; out++) \
115 samples[out][i] = temp[out]; \
119 MIX_FUNC_GENERIC(FLTP
, FLT
, float, float, float, sum
)
120 MIX_FUNC_GENERIC(S16P
, FLT
, int16_t, float, float, av_clip_int16(lrintf(sum
)))
121 MIX_FUNC_GENERIC(S16P
, Q15
, int16_t, int32_t, int64_t, av_clip_int16(sum
>> 15))
122 MIX_FUNC_GENERIC(S16P
, Q8
, int16_t, int16_t, int32_t, av_clip_int16(sum
>> 8))
124 /* TODO: templatize the channel-specific C functions */
126 static void mix_2_to_1_fltp_flt_c(float **samples
, float **matrix
, int len
,
127 int out_ch
, int in_ch
)
129 float *src0
= samples
[0];
130 float *src1
= samples
[1];
132 float m0
= matrix
[0][0];
133 float m1
= matrix
[0][1];
136 *dst
++ = *src0
++ * m0
+ *src1
++ * m1
;
137 *dst
++ = *src0
++ * m0
+ *src1
++ * m1
;
138 *dst
++ = *src0
++ * m0
+ *src1
++ * m1
;
139 *dst
++ = *src0
++ * m0
+ *src1
++ * m1
;
143 *dst
++ = *src0
++ * m0
+ *src1
++ * m1
;
148 static void mix_2_to_1_s16p_flt_c(int16_t **samples
, float **matrix
, int len
,
149 int out_ch
, int in_ch
)
151 int16_t *src0
= samples
[0];
152 int16_t *src1
= samples
[1];
154 float m0
= matrix
[0][0];
155 float m1
= matrix
[0][1];
158 *dst
++ = av_clip_int16(lrintf(*src0
++ * m0
+ *src1
++ * m1
));
159 *dst
++ = av_clip_int16(lrintf(*src0
++ * m0
+ *src1
++ * m1
));
160 *dst
++ = av_clip_int16(lrintf(*src0
++ * m0
+ *src1
++ * m1
));
161 *dst
++ = av_clip_int16(lrintf(*src0
++ * m0
+ *src1
++ * m1
));
165 *dst
++ = av_clip_int16(lrintf(*src0
++ * m0
+ *src1
++ * m1
));
170 static void mix_2_to_1_s16p_q8_c(int16_t **samples
, int16_t **matrix
, int len
,
171 int out_ch
, int in_ch
)
173 int16_t *src0
= samples
[0];
174 int16_t *src1
= samples
[1];
176 int16_t m0
= matrix
[0][0];
177 int16_t m1
= matrix
[0][1];
180 *dst
++ = (*src0
++ * m0
+ *src1
++ * m1
) >> 8;
181 *dst
++ = (*src0
++ * m0
+ *src1
++ * m1
) >> 8;
182 *dst
++ = (*src0
++ * m0
+ *src1
++ * m1
) >> 8;
183 *dst
++ = (*src0
++ * m0
+ *src1
++ * m1
) >> 8;
187 *dst
++ = (*src0
++ * m0
+ *src1
++ * m1
) >> 8;
192 static void mix_1_to_2_fltp_flt_c(float **samples
, float **matrix
, int len
,
193 int out_ch
, int in_ch
)
196 float *dst0
= samples
[0];
197 float *dst1
= samples
[1];
199 float m0
= matrix
[0][0];
200 float m1
= matrix
[1][0];
225 static void mix_6_to_2_fltp_flt_c(float **samples
, float **matrix
, int len
,
226 int out_ch
, int in_ch
)
229 float *src0
= samples
[0];
230 float *src1
= samples
[1];
231 float *src2
= samples
[2];
232 float *src3
= samples
[3];
233 float *src4
= samples
[4];
234 float *src5
= samples
[5];
237 float *m0
= matrix
[0];
238 float *m1
= matrix
[1];
243 *dst0
++ = v0
* m0
[0] +
249 *dst1
++ = v0
* m1
[0] +
259 static void mix_2_to_6_fltp_flt_c(float **samples
, float **matrix
, int len
,
260 int out_ch
, int in_ch
)
263 float *dst0
= samples
[0];
264 float *dst1
= samples
[1];
265 float *dst2
= samples
[2];
266 float *dst3
= samples
[3];
267 float *dst4
= samples
[4];
268 float *dst5
= samples
[5];
275 *dst0
++ = v0
* matrix
[0][0] + v1
* matrix
[0][1];
276 *dst1
++ = v0
* matrix
[1][0] + v1
* matrix
[1][1];
277 *dst2
++ = v0
* matrix
[2][0] + v1
* matrix
[2][1];
278 *dst3
++ = v0
* matrix
[3][0] + v1
* matrix
[3][1];
279 *dst4
++ = v0
* matrix
[4][0] + v1
* matrix
[4][1];
280 *dst5
++ = v0
* matrix
[5][0] + v1
* matrix
[5][1];
285 static av_cold
int mix_function_init(AudioMix
*am
)
287 am
->func_descr
= am
->func_descr_generic
= "n/a";
288 am
->mix
= am
->mix_generic
= NULL
;
290 /* no need to set a mix function when we're skipping mixing */
291 if (!am
->in_matrix_channels
|| !am
->out_matrix_channels
)
294 /* any-to-any C versions */
296 ff_audio_mix_set_func(am
, AV_SAMPLE_FMT_FLTP
, AV_MIX_COEFF_TYPE_FLT
,
297 0, 0, 1, 1, "C", MIX_FUNC_NAME(FLTP
, FLT
));
299 ff_audio_mix_set_func(am
, AV_SAMPLE_FMT_S16P
, AV_MIX_COEFF_TYPE_FLT
,
300 0, 0, 1, 1, "C", MIX_FUNC_NAME(S16P
, FLT
));
302 ff_audio_mix_set_func(am
, AV_SAMPLE_FMT_S16P
, AV_MIX_COEFF_TYPE_Q15
,
303 0, 0, 1, 1, "C", MIX_FUNC_NAME(S16P
, Q15
));
305 ff_audio_mix_set_func(am
, AV_SAMPLE_FMT_S16P
, AV_MIX_COEFF_TYPE_Q8
,
306 0, 0, 1, 1, "C", MIX_FUNC_NAME(S16P
, Q8
));
308 /* channel-specific C versions */
310 ff_audio_mix_set_func(am
, AV_SAMPLE_FMT_FLTP
, AV_MIX_COEFF_TYPE_FLT
,
311 2, 1, 1, 1, "C", mix_2_to_1_fltp_flt_c
);
313 ff_audio_mix_set_func(am
, AV_SAMPLE_FMT_S16P
, AV_MIX_COEFF_TYPE_FLT
,
314 2, 1, 1, 1, "C", mix_2_to_1_s16p_flt_c
);
316 ff_audio_mix_set_func(am
, AV_SAMPLE_FMT_S16P
, AV_MIX_COEFF_TYPE_Q8
,
317 2, 1, 1, 1, "C", mix_2_to_1_s16p_q8_c
);
319 ff_audio_mix_set_func(am
, AV_SAMPLE_FMT_FLTP
, AV_MIX_COEFF_TYPE_FLT
,
320 1, 2, 1, 1, "C", mix_1_to_2_fltp_flt_c
);
322 ff_audio_mix_set_func(am
, AV_SAMPLE_FMT_FLTP
, AV_MIX_COEFF_TYPE_FLT
,
323 6, 2, 1, 1, "C", mix_6_to_2_fltp_flt_c
);
325 ff_audio_mix_set_func(am
, AV_SAMPLE_FMT_FLTP
, AV_MIX_COEFF_TYPE_FLT
,
326 2, 6, 1, 1, "C", mix_2_to_6_fltp_flt_c
);
329 ff_audio_mix_init_x86(am
);
332 av_log(am
->avr
, AV_LOG_ERROR
, "audio_mix: NO FUNCTION FOUND: [fmt=%s] "
333 "[c=%s] [%d to %d]\n", av_get_sample_fmt_name(am
->fmt
),
334 coeff_type_names
[am
->coeff_type
], am
->in_channels
,
336 return AVERROR_PATCHWELCOME
;
341 AudioMix
*ff_audio_mix_alloc(AVAudioResampleContext
*avr
)
346 am
= av_mallocz(sizeof(*am
));
351 if (avr
->internal_sample_fmt
!= AV_SAMPLE_FMT_S16P
&&
352 avr
->internal_sample_fmt
!= AV_SAMPLE_FMT_FLTP
) {
353 av_log(avr
, AV_LOG_ERROR
, "Unsupported internal format for "
355 av_get_sample_fmt_name(avr
->internal_sample_fmt
));
359 am
->fmt
= avr
->internal_sample_fmt
;
360 am
->coeff_type
= avr
->mix_coeff_type
;
361 am
->in_layout
= avr
->in_channel_layout
;
362 am
->out_layout
= avr
->out_channel_layout
;
363 am
->in_channels
= avr
->in_channels
;
364 am
->out_channels
= avr
->out_channels
;
366 /* build matrix if the user did not already set one */
367 if (avr
->mix_matrix
) {
368 ret
= ff_audio_mix_set_matrix(am
, avr
->mix_matrix
, avr
->in_channels
);
371 av_freep(&avr
->mix_matrix
);
373 double *matrix_dbl
= av_mallocz(avr
->out_channels
* avr
->in_channels
*
374 sizeof(*matrix_dbl
));
378 ret
= avresample_build_matrix(avr
->in_channel_layout
,
379 avr
->out_channel_layout
,
380 avr
->center_mix_level
,
381 avr
->surround_mix_level
,
383 avr
->normalize_mix_level
,
386 avr
->matrix_encoding
);
392 ret
= ff_audio_mix_set_matrix(am
, matrix_dbl
, avr
->in_channels
);
394 av_log(avr
, AV_LOG_ERROR
, "error setting mix matrix\n");
409 void ff_audio_mix_free(AudioMix
**am_p
)
418 av_free(am
->matrix
[0]);
421 memset(am
->matrix_q8
, 0, sizeof(am
->matrix_q8
));
422 memset(am
->matrix_q15
, 0, sizeof(am
->matrix_q15
));
423 memset(am
->matrix_flt
, 0, sizeof(am
->matrix_flt
));
428 int ff_audio_mix(AudioMix
*am
, AudioData
*src
)
431 int len
= src
->nb_samples
;
434 /* determine whether to use the optimized function based on pointer and
435 samples alignment in both the input and output */
436 if (am
->has_optimized_func
) {
437 int aligned_len
= FFALIGN(len
, am
->samples_align
);
438 if (!(src
->ptr_align
% am
->ptr_align
) &&
439 src
->samples_align
>= aligned_len
) {
444 av_dlog(am
->avr
, "audio_mix: %d samples - %d to %d channels (%s)\n",
445 src
->nb_samples
, am
->in_channels
, am
->out_channels
,
446 use_generic
? am
->func_descr_generic
: am
->func_descr
);
448 if (am
->in_matrix_channels
&& am
->out_matrix_channels
) {
450 uint8_t *data0
[AVRESAMPLE_MAX_CHANNELS
] = { NULL
};
452 if (am
->out_matrix_channels
< am
->out_channels
||
453 am
->in_matrix_channels
< am
->in_channels
) {
454 for (i
= 0, j
= 0; i
< FFMAX(am
->in_channels
, am
->out_channels
); i
++) {
455 if (am
->input_skip
[i
] || am
->output_skip
[i
] || am
->output_zero
[i
])
457 data0
[j
++] = src
->data
[i
];
465 am
->mix_generic(data
, am
->matrix
, len
, am
->out_matrix_channels
,
466 am
->in_matrix_channels
);
468 am
->mix(data
, am
->matrix
, len
, am
->out_matrix_channels
,
469 am
->in_matrix_channels
);
472 if (am
->out_matrix_channels
< am
->out_channels
) {
473 for (i
= 0; i
< am
->out_channels
; i
++)
474 if (am
->output_zero
[i
])
475 av_samples_set_silence(&src
->data
[i
], 0, len
, 1, am
->fmt
);
478 ff_audio_data_set_channels(src
, am
->out_channels
);
483 int ff_audio_mix_get_matrix(AudioMix
*am
, double *matrix
, int stride
)
487 if ( am
->in_channels
<= 0 || am
->in_channels
> AVRESAMPLE_MAX_CHANNELS
||
488 am
->out_channels
<= 0 || am
->out_channels
> AVRESAMPLE_MAX_CHANNELS
) {
489 av_log(am
->avr
, AV_LOG_ERROR
, "Invalid channel counts\n");
490 return AVERROR(EINVAL
);
493 #define GET_MATRIX_CONVERT(suffix, scale) \
494 if (!am->matrix_ ## suffix[0]) { \
495 av_log(am->avr, AV_LOG_ERROR, "matrix is not set\n"); \
496 return AVERROR(EINVAL); \
498 for (o = 0, o0 = 0; o < am->out_channels; o++) { \
499 for (i = 0, i0 = 0; i < am->in_channels; i++) { \
500 if (am->input_skip[i] || am->output_zero[o]) \
501 matrix[o * stride + i] = 0.0; \
503 matrix[o * stride + i] = am->matrix_ ## suffix[o0][i0] * \
505 if (!am->input_skip[i]) \
508 if (!am->output_zero[o]) \
512 switch (am
->coeff_type
) {
513 case AV_MIX_COEFF_TYPE_Q8
:
514 GET_MATRIX_CONVERT(q8
, 1.0 / 256.0);
516 case AV_MIX_COEFF_TYPE_Q15
:
517 GET_MATRIX_CONVERT(q15
, 1.0 / 32768.0);
519 case AV_MIX_COEFF_TYPE_FLT
:
520 GET_MATRIX_CONVERT(flt
, 1.0);
523 av_log(am
->avr
, AV_LOG_ERROR
, "Invalid mix coeff type\n");
524 return AVERROR(EINVAL
);
530 static void reduce_matrix(AudioMix
*am
, const double *matrix
, int stride
)
534 memset(am
->output_zero
, 0, sizeof(am
->output_zero
));
535 memset(am
->input_skip
, 0, sizeof(am
->input_skip
));
536 memset(am
->output_skip
, 0, sizeof(am
->output_skip
));
538 /* exclude output channels if they can be zeroed instead of mixed */
539 for (o
= 0; o
< am
->out_channels
; o
++) {
542 /* check if the output is always silent */
543 for (i
= 0; i
< am
->in_channels
; i
++) {
544 if (matrix
[o
* stride
+ i
] != 0.0) {
549 /* check if the corresponding input channel makes a contribution to
550 any output channel */
551 if (o
< am
->in_channels
) {
552 for (i
= 0; i
< am
->out_channels
; i
++) {
553 if (matrix
[i
* stride
+ o
] != 0.0) {
560 am
->output_zero
[o
] = 1;
561 am
->out_matrix_channels
--;
562 if (o
< am
->in_channels
)
563 am
->in_matrix_channels
--;
566 if (am
->out_matrix_channels
== 0 || am
->in_matrix_channels
== 0) {
567 am
->out_matrix_channels
= 0;
568 am
->in_matrix_channels
= 0;
572 /* skip input channels that contribute fully only to the corresponding
574 for (i
= 0; i
< FFMIN(am
->in_channels
, am
->out_channels
); i
++) {
577 for (o
= 0; o
< am
->out_channels
; o
++) {
579 if ((o
!= i
&& matrix
[o
* stride
+ i
] != 0.0) ||
580 (o
== i
&& matrix
[o
* stride
+ i
] != 1.0)) {
584 /* if the input contributes fully to the output, also check that no
585 other inputs contribute to this output */
587 for (i0
= 0; i0
< am
->in_channels
; i0
++) {
588 if (i0
!= i
&& matrix
[o
* stride
+ i0
] != 0.0) {
596 am
->input_skip
[i
] = 1;
597 am
->in_matrix_channels
--;
600 /* skip input channels that do not contribute to any output channel */
601 for (; i
< am
->in_channels
; i
++) {
604 for (o
= 0; o
< am
->out_channels
; o
++) {
605 if (matrix
[o
* stride
+ i
] != 0.0) {
611 am
->input_skip
[i
] = 1;
612 am
->in_matrix_channels
--;
615 if (am
->in_matrix_channels
== 0) {
616 am
->out_matrix_channels
= 0;
620 /* skip output channels that only get full contribution from the
621 corresponding input channel */
622 for (o
= 0; o
< FFMIN(am
->in_channels
, am
->out_channels
); o
++) {
626 for (i
= 0; i
< am
->in_channels
; i
++) {
627 if ((o
!= i
&& matrix
[o
* stride
+ i
] != 0.0) ||
628 (o
== i
&& matrix
[o
* stride
+ i
] != 1.0)) {
633 /* check if the corresponding input channel makes a contribution to
634 any other output channel */
636 for (o0
= 0; o0
< am
->out_channels
; o0
++) {
637 if (o0
!= i
&& matrix
[o0
* stride
+ i
] != 0.0) {
643 am
->output_skip
[o
] = 1;
644 am
->out_matrix_channels
--;
647 if (am
->out_matrix_channels
== 0) {
648 am
->in_matrix_channels
= 0;
653 int ff_audio_mix_set_matrix(AudioMix
*am
, const double *matrix
, int stride
)
655 int i
, o
, i0
, o0
, ret
;
656 char in_layout_name
[128];
657 char out_layout_name
[128];
659 if ( am
->in_channels
<= 0 || am
->in_channels
> AVRESAMPLE_MAX_CHANNELS
||
660 am
->out_channels
<= 0 || am
->out_channels
> AVRESAMPLE_MAX_CHANNELS
) {
661 av_log(am
->avr
, AV_LOG_ERROR
, "Invalid channel counts\n");
662 return AVERROR(EINVAL
);
666 av_free(am
->matrix
[0]);
670 am
->in_matrix_channels
= am
->in_channels
;
671 am
->out_matrix_channels
= am
->out_channels
;
673 reduce_matrix(am
, matrix
, stride
);
675 #define CONVERT_MATRIX(type, expr) \
676 am->matrix_## type[0] = av_mallocz(am->out_matrix_channels * \
677 am->in_matrix_channels * \
678 sizeof(*am->matrix_## type[0])); \
679 if (!am->matrix_## type[0]) \
680 return AVERROR(ENOMEM); \
681 for (o = 0, o0 = 0; o < am->out_channels; o++) { \
682 if (am->output_zero[o] || am->output_skip[o]) \
685 am->matrix_## type[o0] = am->matrix_## type[o0 - 1] + \
686 am->in_matrix_channels; \
687 for (i = 0, i0 = 0; i < am->in_channels; i++) { \
689 if (am->input_skip[i] || am->output_zero[i]) \
691 v = matrix[o * stride + i]; \
692 am->matrix_## type[o0][i0] = expr; \
697 am->matrix = (void **)am->matrix_## type;
699 if (am
->in_matrix_channels
&& am
->out_matrix_channels
) {
700 switch (am
->coeff_type
) {
701 case AV_MIX_COEFF_TYPE_Q8
:
702 CONVERT_MATRIX(q8
, av_clip_int16(lrint(256.0 * v
)))
704 case AV_MIX_COEFF_TYPE_Q15
:
705 CONVERT_MATRIX(q15
, av_clipl_int32(llrint(32768.0 * v
)))
707 case AV_MIX_COEFF_TYPE_FLT
:
708 CONVERT_MATRIX(flt
, v
)
711 av_log(am
->avr
, AV_LOG_ERROR
, "Invalid mix coeff type\n");
712 return AVERROR(EINVAL
);
716 ret
= mix_function_init(am
);
720 av_get_channel_layout_string(in_layout_name
, sizeof(in_layout_name
),
721 am
->in_channels
, am
->in_layout
);
722 av_get_channel_layout_string(out_layout_name
, sizeof(out_layout_name
),
723 am
->out_channels
, am
->out_layout
);
724 av_log(am
->avr
, AV_LOG_DEBUG
, "audio_mix: %s to %s\n",
725 in_layout_name
, out_layout_name
);
726 av_log(am
->avr
, AV_LOG_DEBUG
, "matrix size: %d x %d\n",
727 am
->in_matrix_channels
, am
->out_matrix_channels
);
728 for (o
= 0; o
< am
->out_channels
; o
++) {
729 for (i
= 0; i
< am
->in_channels
; i
++) {
730 if (am
->output_zero
[o
])
731 av_log(am
->avr
, AV_LOG_DEBUG
, " (ZERO)");
732 else if (am
->input_skip
[i
] || am
->output_zero
[i
] || am
->output_skip
[o
])
733 av_log(am
->avr
, AV_LOG_DEBUG
, " (SKIP)");
735 av_log(am
->avr
, AV_LOG_DEBUG
, " %0.3f ",
736 matrix
[o
* am
->in_channels
+ i
]);
738 av_log(am
->avr
, AV_LOG_DEBUG
, "\n");