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-rw-r--r--src/plugins/previewopus_extractor.c1247
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diff --git a/src/plugins/previewopus_extractor.c b/src/plugins/previewopus_extractor.c
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index f137f38..0000000
--- a/src/plugins/previewopus_extractor.c
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1/*
2 This file is part of libextractor.
3 Copyright Copyright (C) 2008, 2013 Bruno Cabral and Christian Grothoff
4
5 libextractor is free software; you can redistribute it and/or modify
6 it under the terms of the GNU General Public License as published
7 by the Free Software Foundation; either version 3, or (at your
8 option) any later version.
9
10 libextractor is distributed in the hope that it will be useful, but
11 WITHOUT ANY WARRANTY; without even the implied warranty of
12 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
13 General Public License for more details.
14
15 You should have received a copy of the GNU General Public License
16 along with libextractor; see the file COPYING. If not, write to the
17 Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor,
18 Boston, MA 02110-1301, USA.
19 */
20/**
21 * @file previewopus_extractor.c
22 * @author Bruno Cabral
23 * @author Christian Grothoff
24 * @brief this extractor produces a binary encoded
25 * audio snippet of music/video files using ffmpeg libs.
26 *
27 * Based on ffmpeg samples.
28 *
29 * Note that ffmpeg has a few issues:
30 * (1) there are no recent official releases of the ffmpeg libs
31 * (2) ffmpeg has a history of having security issues (parser is not robust)
32 *
33 * So this plugin cannot be recommended for system with high security
34 *requirements.
35 */
36#include "platform.h"
37#include "extractor.h"
38#include <magic.h>
39
40#include <libavutil/avutil.h>
41#include <libavutil/audio_fifo.h>
42#include <libavutil/opt.h>
43#include <libavutil/mathematics.h>
44#include <libavformat/avformat.h>
45#include <libavcodec/avcodec.h>
46#include <libswscale/swscale.h>
47#include <libswresample/swresample.h>
48
49
50/**
51 * Set to 1 to enable debug output.
52 */
53#define DEBUG 0
54
55/**
56 * Set to 1 to enable a output file for testing.
57 */
58#define OUTPUT_FILE 0
59
60
61/**
62 * Maximum size in bytes for the preview.
63 */
64#define MAX_SIZE (28 * 1024)
65
66/**
67 * HardLimit for file
68 */
69#define HARD_LIMIT_SIZE (50 * 1024)
70
71
72/** The output bit rate in kbit/s */
73#define OUTPUT_BIT_RATE 28000
74/** The number of output channels */
75#define OUTPUT_CHANNELS 2
76/** The audio sample output format */
77#define OUTPUT_SAMPLE_FORMAT AV_SAMPLE_FMT_S16
78
79
80/** Our output buffer*/
81static unsigned char *buffer;
82
83/** Actual output buffer size */
84static int totalSize;
85
86/**
87 * Convert an error code into a text message.
88 * @param error Error code to be converted
89 * @return Corresponding error text (not thread-safe)
90 */
91static char *const
92get_error_text (const int error)
93{
94 static char error_buffer[255];
95 av_strerror (error, error_buffer, sizeof(error_buffer));
96 return error_buffer;
97}
98
99
100/**
101 * Read callback.
102 *
103 * @param opaque the 'struct EXTRACTOR_ExtractContext'
104 * @param buf where to write data
105 * @param buf_size how many bytes to read
106 * @return -1 on error (or for unknown file size)
107 */
108static int
109read_cb (void *opaque,
110 uint8_t *buf,
111 int buf_size)
112{
113 struct EXTRACTOR_ExtractContext *ec = opaque;
114 void *data;
115 ssize_t ret;
116
117 ret = ec->read (ec->cls, &data, buf_size);
118 if (ret <= 0)
119 return ret;
120 memcpy (buf, data, ret);
121 return ret;
122}
123
124
125/**
126 * Seek callback.
127 *
128 * @param opaque the 'struct EXTRACTOR_ExtractContext'
129 * @param offset where to seek
130 * @param whence how to seek; AVSEEK_SIZE to return file size without seeking
131 * @return -1 on error (or for unknown file size)
132 */
133static int64_t
134seek_cb (void *opaque,
135 int64_t offset,
136 int whence)
137{
138 struct EXTRACTOR_ExtractContext *ec = opaque;
139
140 if (AVSEEK_SIZE == whence)
141 return ec->get_size (ec->cls);
142 return ec->seek (ec->cls, offset, whence);
143}
144
145
146/**
147 * write callback.
148 *
149 * @param opaque NULL
150 * @param pBuffer to write
151 * @param pBufferSize , amount to write
152 * @return 0 on error
153 */
154static int
155writePacket (void *opaque,
156 unsigned char *pBuffer,
157 int pBufferSize)
158{
159 int sizeToCopy = pBufferSize;
160
161 if ( (totalSize + pBufferSize) > HARD_LIMIT_SIZE)
162 sizeToCopy = HARD_LIMIT_SIZE - totalSize;
163
164 memcpy (buffer + totalSize, pBuffer, sizeToCopy);
165 totalSize += sizeToCopy;
166 return sizeToCopy;
167}
168
169
170/**
171 * Open an output file and the required encoder.
172 * Also set some basic encoder parameters.
173 * Some of these parameters are based on the input file's parameters.
174 */
175static int
176open_output_file (
177 AVCodecContext *input_codec_context,
178 AVFormatContext **output_format_context,
179 AVCodecContext **output_codec_context)
180{
181 AVStream *stream = NULL;
182 AVCodec *output_codec = NULL;
183 AVIOContext *io_ctx;
184 int error;
185 unsigned char *iob;
186
187 if (NULL == (iob = av_malloc (16 * 1024)))
188 return AVERROR_EXIT;
189 if (NULL == (io_ctx = avio_alloc_context (iob, 16 * 1024,
190 AVIO_FLAG_WRITE, NULL,
191 NULL,
192 &writePacket /* no writing */,
193 NULL)))
194 {
195 av_free (iob);
196 return AVERROR_EXIT;
197 }
198 if (NULL == ((*output_format_context) = avformat_alloc_context ()))
199 {
200 av_free (io_ctx);
201 return AVERROR_EXIT;
202 }
203 (*output_format_context)->pb = io_ctx;
204
205 /** Guess the desired container format based on the file extension. */
206 if (! ((*output_format_context)->oformat = av_guess_format (NULL,
207 "file.ogg",
208 NULL)))
209 {
210#if DEBUG
211 fprintf (stderr, "Could not find output file format\n");
212#endif
213 error = AVERROR (ENOSYS);
214 goto cleanup;
215 }
216
217 /** Find the encoder to be used by its name. */
218 if (! (output_codec = avcodec_find_encoder (AV_CODEC_ID_OPUS)))
219 {
220#if DEBUG
221 fprintf (stderr, "Could not find an OPUS encoder.\n");
222#endif
223 error = AVERROR (ENOSYS);
224 goto cleanup;
225 }
226
227 /** Create a new audio stream in the output file container. */
228 if (! (stream = avformat_new_stream (*output_format_context, output_codec)))
229 {
230#if DEBUG
231 fprintf (stderr, "Could not create new stream\n");
232#endif
233 error = AVERROR (ENOMEM);
234 goto cleanup;
235 }
236
237 /** Save the encoder context for easiert access later. */
238 *output_codec_context = stream->codec;
239
240 /**
241 * Set the basic encoder parameters.
242 * The input file's sample rate is used to avoid a sample rate conversion.
243 */
244 (*output_codec_context)->channels = OUTPUT_CHANNELS;
245 (*output_codec_context)->channel_layout = av_get_default_channel_layout (
246 OUTPUT_CHANNELS);
247 (*output_codec_context)->sample_rate = 48000; // Opus need 48000
248 (*output_codec_context)->sample_fmt = AV_SAMPLE_FMT_S16;
249 (*output_codec_context)->bit_rate = OUTPUT_BIT_RATE;
250
251 /** Open the encoder for the audio stream to use it later. */
252 if ((error = avcodec_open2 (*output_codec_context, output_codec, NULL)) < 0)
253 {
254#if DEBUG
255 fprintf (stderr, "Could not open output codec (error '%s')\n",
256 get_error_text (error));
257#endif
258 goto cleanup;
259 }
260 return 0;
261
262cleanup:
263 av_free (io_ctx);
264 return error < 0 ? error : AVERROR_EXIT;
265}
266
267
268/** Initialize one data packet for reading or writing. */
269static void
270init_packet (AVPacket *packet)
271{
272 av_init_packet (packet);
273 /** Set the packet data and size so that it is recognized as being empty. */
274 packet->data = NULL;
275 packet->size = 0;
276}
277
278
279/** Initialize one audio frame for reading from the input file */
280static int
281init_input_frame (AVFrame **frame)
282{
283 *frame = av_frame_alloc ();
284 if (NULL == *frame)
285 {
286#if DEBUG
287 fprintf (stderr, "Could not allocate input frame\n");
288#endif
289 return AVERROR (ENOMEM);
290 }
291 return 0;
292}
293
294
295/**
296 * Initialize the audio resampler based on the input and output codec settings.
297 * If the input and output sample formats differ, a conversion is required
298 * libavresample takes care of this, but requires initialization.
299 */
300static int
301init_resampler (AVCodecContext *input_codec_context,
302 AVCodecContext *output_codec_context,
303 SwrContext **resample_context)
304{
305 /**
306 * Only initialize the resampler if it is necessary, i.e.,
307 * if and only if the sample formats differ.
308 */
309 if ((input_codec_context->sample_fmt != output_codec_context->sample_fmt) ||
310 (input_codec_context->channels != output_codec_context->channels) )
311 {
312 int error;
313
314 /** Create a resampler context for the conversion. */
315 if (! (*resample_context = swr_alloc ()))
316 {
317#if DEBUG
318 fprintf (stderr, "Could not allocate resample context\n");
319#endif
320 return AVERROR (ENOMEM);
321 }
322
323
324 /**
325 * Set the conversion parameters.
326 * Default channel layouts based on the number of channels
327 * are assumed for simplicity (they are sometimes not detected
328 * properly by the demuxer and/or decoder).
329 */
330 av_opt_set_int (*resample_context,
331 "in_channel_layout",
332 av_get_default_channel_layout (
333 input_codec_context->channels), 0);
334 av_opt_set_int (*resample_context,
335 "out_channel_layout",
336 av_get_default_channel_layout (
337 output_codec_context->channels), 0);
338 av_opt_set_int (*resample_context,
339 "in_sample_rate",
340 input_codec_context->sample_rate, 0);
341 av_opt_set_int (*resample_context,
342 "out_sample_rate",
343 output_codec_context->sample_rate, 0);
344 av_opt_set_int (*resample_context,
345 "in_sample_fmt",
346 input_codec_context->sample_fmt, 0);
347 av_opt_set_int (*resample_context,
348 "out_sample_fmt",
349 output_codec_context->sample_fmt, 0);
350
351 /** Open the resampler with the specified parameters. */
352 if ((error = swr_init (*resample_context)) < 0)
353 {
354#if DEBUG
355 fprintf (stderr, "Could not open resample context\n");
356#endif
357 swr_free (resample_context);
358 return error;
359 }
360 }
361 return 0;
362}
363
364
365/** Initialize a FIFO buffer for the audio samples to be encoded. */
366static int
367init_fifo (AVAudioFifo **fifo)
368{
369 /** Create the FIFO buffer based on the specified output sample format. */
370 if (! (*fifo = av_audio_fifo_alloc (OUTPUT_SAMPLE_FORMAT, OUTPUT_CHANNELS,
371 1)))
372 {
373#if DEBUG
374 fprintf (stderr, "Could not allocate FIFO\n");
375#endif
376 return AVERROR (ENOMEM);
377 }
378 return 0;
379}
380
381
382/** Write the header of the output file container. */
383static int
384write_output_file_header (AVFormatContext *output_format_context)
385{
386 int error;
387 if ((error = avformat_write_header (output_format_context, NULL)) < 0)
388 {
389#if DEBUG
390 fprintf (stderr, "Could not write output file header (error '%s')\n",
391 get_error_text (error));
392#endif
393 return error;
394 }
395 return 0;
396}
397
398
399/** Decode one audio frame from the input file. */
400static int
401decode_audio_frame (AVFrame *frame,
402 AVFormatContext *input_format_context,
403 AVCodecContext *input_codec_context, int audio_stream_index,
404 int *data_present, int *finished)
405{
406 /** Packet used for temporary storage. */
407 AVPacket input_packet;
408 int error;
409 init_packet (&input_packet);
410
411 /** Read one audio frame from the input file into a temporary packet. */
412 while (1)
413 {
414 if ((error = av_read_frame (input_format_context, &input_packet)) < 0)
415 {
416 /** If we are the the end of the file, flush the decoder below. */
417 if (error == AVERROR_EOF)
418 {
419#if DEBUG
420 fprintf (stderr, "EOF in decode_audio\n");
421#endif
422 *finished = 1;
423 }
424 else
425 {
426#if DEBUG
427 fprintf (stderr, "Could not read frame (error '%s')\n",
428 get_error_text (error));
429#endif
430 return error;
431 }
432 }
433
434 if (input_packet.stream_index == audio_stream_index)
435 break;
436 }
437
438 /**
439 * Decode the audio frame stored in the temporary packet.
440 * The input audio stream decoder is used to do this.
441 * If we are at the end of the file, pass an empty packet to the decoder
442 * to flush it.
443 */if ((error = avcodec_decode_audio4 (input_codec_context, frame,
444 data_present, &input_packet)) < 0)
445 {
446#if DEBUG
447 fprintf (stderr, "Could not decode frame (error '%s')\n",
448 get_error_text (error));
449#endif
450 av_packet_unref (&input_packet);
451 return error;
452 }
453
454 /**
455 * If the decoder has not been flushed completely, we are not finished,
456 * so that this function has to be called again.
457 */
458 if (*finished && *data_present)
459 *finished = 0;
460 av_packet_unref (&input_packet);
461 return 0;
462}
463
464
465/**
466 * Initialize a temporary storage for the specified number of audio samples.
467 * The conversion requires temporary storage due to the different format.
468 * The number of audio samples to be allocated is specified in frame_size.
469 */
470static int
471init_converted_samples (uint8_t ***converted_input_samples,
472 int*out_linesize,
473 AVCodecContext *output_codec_context,
474 int frame_size)
475{
476 int error;
477
478 /**
479 * Allocate as many pointers as there are audio channels.
480 * Each pointer will later point to the audio samples of the corresponding
481 * channels (although it may be NULL for interleaved formats).
482 */if (! (*converted_input_samples = calloc (output_codec_context->channels,
483 sizeof(**converted_input_samples))))
484 {
485#if DEBUG
486 fprintf (stderr, "Could not allocate converted input sample pointers\n");
487#endif
488 return AVERROR (ENOMEM);
489 }
490
491 /**
492 * Allocate memory for the samples of all channels in one consecutive
493 * block for convenience.
494 */
495 if ((error = av_samples_alloc (*converted_input_samples,
496 out_linesize,
497 output_codec_context->channels,
498 frame_size,
499 output_codec_context->sample_fmt, 0)) < 0)
500 {
501#if DEBUG
502 fprintf (stderr,
503 "Could not allocate converted input samples (error '%s')\n",
504 get_error_text (error));
505#endif
506 av_freep (&(*converted_input_samples)[0]);
507 free (*converted_input_samples);
508 return error;
509 }
510 return 0;
511}
512
513
514/**
515 * Convert the input audio samples into the output sample format.
516 * The conversion happens on a per-frame basis, the size of which is specified
517 * by frame_size.
518 */
519static int
520convert_samples (uint8_t **input_data,
521 uint8_t **converted_data,
522 int in_sample,
523 int out_sample,
524 int out_linesize,
525 SwrContext *resample_context)
526{
527 int error;
528
529 /** Convert the samples using the resampler. */
530 if ((error = swr_convert (resample_context,
531 converted_data,
532 out_linesize,
533 out_sample,
534 input_data,
535 0,
536 in_sample)) < 0)
537 {
538#if DEBUG
539 fprintf (stderr,
540 "Could not convert input samples (error '%s')\n",
541 get_error_text (error));
542#endif
543 return error;
544 }
545
546
547 /**
548 * Perform a sanity check so that the number of converted samples is
549 * not greater than the number of samples to be converted.
550 * If the sample rates differ, this case has to be handled differently
551 */
552 if (avresample_available (resample_context))
553 {
554#if DEBUG
555 fprintf (stderr,
556 "%i Converted samples left over\n",
557 avresample_available (resample_context));
558#endif
559 }
560
561
562 return 0;
563}
564
565
566/** Add converted input audio samples to the FIFO buffer for later processing. */
567static int
568add_samples_to_fifo (AVAudioFifo *fifo,
569 uint8_t **converted_input_samples,
570 const int frame_size)
571{
572 int error;
573
574 /**
575 * Make the FIFO as large as it needs to be to hold both,
576 * the old and the new samples.
577 */
578 if ((error = av_audio_fifo_realloc (fifo, av_audio_fifo_size (fifo)
579 + frame_size)) < 0)
580 {
581#if DEBUG
582 fprintf (stderr, "Could not reallocate FIFO\n");
583#endif
584 return error;
585 }
586
587 /** Store the new samples in the FIFO buffer. */
588 if (av_audio_fifo_write (fifo, (void **) converted_input_samples,
589 frame_size) < frame_size)
590 {
591#if DEBUG
592 fprintf (stderr, "Could not write data to FIFO\n");
593#endif
594 return AVERROR_EXIT;
595 }
596 return 0;
597}
598
599
600/**
601 * Read one audio frame from the input file, decodes, converts and stores
602 * it in the FIFO buffer.
603 */
604static int
605read_decode_convert_and_store (AVAudioFifo *fifo,
606 AVFormatContext *input_format_context,
607 AVCodecContext *input_codec_context,
608 AVCodecContext *output_codec_context,
609 SwrContext *resampler_context, int
610 audio_stream_index,
611 int *finished)
612{
613 /** Temporary storage of the input samples of the frame read from the file. */
614 AVFrame *input_frame = NULL;
615 /** Temporary storage for the converted input samples. */
616 uint8_t **converted_input_samples = NULL;
617 int data_present;
618 int ret = AVERROR_EXIT;
619
620 /** Initialize temporary storage for one input frame. */
621 if (init_input_frame (&input_frame))
622 {
623#if DEBUG
624 fprintf (stderr, "Failed at init frame\n");
625#endif
626 goto cleanup;
627
628 }
629 /** Decode one frame worth of audio samples. */
630 if (decode_audio_frame (input_frame,
631 input_format_context,
632 input_codec_context,
633 audio_stream_index,
634 &data_present,
635 finished))
636 {
637#if DEBUG
638 fprintf (stderr, "Failed at decode audio\n");
639#endif
640
641 goto cleanup;
642 }
643 /**
644 * If we are at the end of the file and there are no more samples
645 * in the decoder which are delayed, we are actually finished.
646 * This must not be treated as an error.
647 */
648 if (*finished && ! data_present)
649 {
650 ret = 0;
651#if DEBUG
652 fprintf (stderr, "Failed at finished or no data\n");
653#endif
654 goto cleanup;
655 }
656 /** If there is decoded data, convert and store it */
657 if (data_present)
658 {
659 int out_linesize;
660 // FIXME: I'm losing samples, but can't get it to work.
661 int out_samples = avresample_available (resampler_context)
662 + avresample_get_delay (resampler_context)
663 + input_frame->nb_samples;
664
665
666 // fprintf(stderr, "Input nbsamples %i out_samples: %i \n",input_frame->nb_samples,out_samples);
667
668 /** Initialize the temporary storage for the converted input samples. */
669 if (init_converted_samples (&converted_input_samples,
670 &out_linesize,
671 output_codec_context,
672 out_samples))
673 {
674#if DEBUG
675 fprintf (stderr, "Failed at init_converted_samples\n");
676#endif
677 goto cleanup;
678 }
679
680 /**
681 * Convert the input samples to the desired output sample format.
682 * This requires a temporary storage provided by converted_input_samples.
683 */
684 if (convert_samples (input_frame->extended_data,
685 converted_input_samples,
686 input_frame->nb_samples,
687 out_samples,
688 out_linesize,
689 resampler_context))
690 {
691#if DEBUG
692 fprintf (stderr,
693 "Failed at convert_samples, input frame %i \n",
694 input_frame->nb_samples);
695#endif
696 goto cleanup;
697 }
698 /** Add the converted input samples to the FIFO buffer for later processing. */
699 if (add_samples_to_fifo (fifo,
700 converted_input_samples,
701 out_samples))
702 {
703#if DEBUG
704 fprintf (stderr, "Failed at add_samples_to_fifo\n");
705#endif
706 goto cleanup;
707 }
708 ret = 0;
709 }
710 ret = 0;
711
712cleanup:
713 if (converted_input_samples)
714 {
715 av_freep (&converted_input_samples[0]);
716 free (converted_input_samples);
717 }
718 av_frame_free (&input_frame);
719 return ret;
720}
721
722
723/**
724 * Initialize one input frame for writing to the output file.
725 * The frame will be exactly frame_size samples large.
726 */
727static int
728init_output_frame (AVFrame **frame,
729 AVCodecContext *output_codec_context,
730 int frame_size)
731{
732 int error;
733
734 /** Create a new frame to store the audio samples. */
735 *frame = av_frame_alloc ();
736 if (NULL == *frame)
737 {
738#if DEBUG
739 fprintf (stderr, "Could not allocate output frame\n");
740#endif
741 return AVERROR_EXIT;
742 }
743
744 /**
745 * Set the frame's parameters, especially its size and format.
746 * av_frame_get_buffer needs this to allocate memory for the
747 * audio samples of the frame.
748 * Default channel layouts based on the number of channels
749 * are assumed for simplicity.
750 */(*frame)->nb_samples = frame_size;
751 (*frame)->channel_layout = output_codec_context->channel_layout;
752 (*frame)->format = output_codec_context->sample_fmt;
753 (*frame)->sample_rate = output_codec_context->sample_rate;
754
755
756 // fprintf(stderr, "%i %i \n",frame_size , (*frame)->format,(*frame)->sample_rate);
757
758 /**
759 * Allocate the samples of the created frame. This call will make
760 * sure that the audio frame can hold as many samples as specified.
761 */
762 if ((error = av_frame_get_buffer (*frame, 0)) < 0)
763 {
764#if DEBUG
765 fprintf (stderr, "Could allocate output frame samples (error '%s')\n",
766 get_error_text (error));
767#endif
768 av_frame_free (frame);
769 return error;
770 }
771
772 return 0;
773}
774
775
776/** Encode one frame worth of audio to the output file. */
777static int
778encode_audio_frame (AVFrame *frame,
779 AVFormatContext *output_format_context,
780 AVCodecContext *output_codec_context,
781 int *data_present)
782{
783 /** Packet used for temporary storage. */
784 AVPacket output_packet;
785 int error;
786 init_packet (&output_packet);
787
788 /**
789 * Encode the audio frame and store it in the temporary packet.
790 * The output audio stream encoder is used to do this.
791 */
792 if ((error = avcodec_encode_audio2 (output_codec_context, &output_packet,
793 frame, data_present)) < 0)
794 {
795#if DEBUG
796 fprintf (stderr, "Could not encode frame (error '%s')\n",
797 get_error_text (error));
798#endif
799 av_packet_unref (&output_packet);
800 return error;
801 }
802
803 /** Write one audio frame from the temporary packet to the output file. */
804 if (*data_present)
805 {
806 if ((error = av_write_frame (output_format_context, &output_packet)) < 0)
807 {
808#if DEBUG
809 fprintf (stderr, "Could not write frame (error '%s')\n",
810 get_error_text (error));
811#endif
812
813 av_packet_unref (&output_packet);
814 return error;
815 }
816
817 av_packet_unref (&output_packet);
818 }
819
820 return 0;
821}
822
823
824/**
825 * Load one audio frame from the FIFO buffer, encode and write it to the
826 * output file.
827 */
828static int
829load_encode_and_write (AVAudioFifo *fifo,
830 AVFormatContext *output_format_context,
831 AVCodecContext *output_codec_context)
832{
833 /** Temporary storage of the output samples of the frame written to the file. */
834 AVFrame *output_frame;
835 /**
836 * Use the maximum number of possible samples per frame.
837 * If there is less than the maximum possible frame size in the FIFO
838 * buffer use this number. Otherwise, use the maximum possible frame size
839 */const int frame_size = FFMIN (av_audio_fifo_size (fifo),
840 output_codec_context->frame_size);
841 int data_written;
842
843 /** Initialize temporary storage for one output frame. */
844 if (init_output_frame (&output_frame, output_codec_context, frame_size))
845 return AVERROR_EXIT;
846
847 /**
848 * Read as many samples from the FIFO buffer as required to fill the frame.
849 * The samples are stored in the frame temporarily.
850 */
851 if (av_audio_fifo_read (fifo, (void **) output_frame->data, frame_size) <
852 frame_size)
853 {
854#if DEBUG
855 fprintf (stderr, "Could not read data from FIFO\n");
856#endif
857 av_frame_free (&output_frame);
858 return AVERROR_EXIT;
859 }
860
861 /** Encode one frame worth of audio samples. */
862 if (encode_audio_frame (output_frame, output_format_context,
863 output_codec_context, &data_written))
864 {
865 av_frame_free (&output_frame);
866 return AVERROR_EXIT;
867 }
868 av_frame_free (&output_frame);
869 return 0;
870}
871
872
873/** Write the trailer of the output file container. */
874static int
875write_output_file_trailer (AVFormatContext *output_format_context)
876{
877 int error;
878 if ((error = av_write_trailer (output_format_context)) < 0)
879 {
880#if DEBUG
881 fprintf (stderr, "Could not write output file trailer (error '%s')\n",
882 get_error_text (error));
883#endif
884 return error;
885 }
886 return 0;
887}
888
889
890#define ENUM_CODEC_ID enum AVCodecID
891
892
893/**
894 * Perform the audio snippet extraction
895 *
896 * @param ec extraction context to use
897 */
898static void
899extract_audio (struct EXTRACTOR_ExtractContext *ec)
900{
901 AVIOContext *io_ctx;
902 struct AVFormatContext *format_ctx;
903 AVCodecContext *codec_ctx;
904 AVFormatContext *output_format_context = NULL;
905 AVCodec *codec;
906 AVDictionary *options;
907 AVFrame *frame;
908 AVCodecContext*output_codec_context = NULL;
909 SwrContext *resample_context = NULL;
910 AVAudioFifo *fifo = NULL;
911
912 int audio_stream_index;
913 int i;
914 int err;
915 int duration;
916 unsigned char *iob;
917
918
919 totalSize = 0;
920 if (NULL == (iob = av_malloc (16 * 1024)))
921 return;
922 if (NULL == (io_ctx = avio_alloc_context (iob,
923 16 * 1024,
924 0, ec,
925 &read_cb,
926 NULL /* no writing */,
927 &seek_cb)))
928 {
929 av_free (iob);
930 return;
931 }
932 if (NULL == (format_ctx = avformat_alloc_context ()))
933 {
934 av_free (io_ctx);
935 return;
936 }
937 format_ctx->pb = io_ctx;
938 options = NULL;
939 if (0 != avformat_open_input (&format_ctx, "<no file>", NULL, &options))
940 {
941 av_free (io_ctx);
942 return;
943 }
944 av_dict_free (&options);
945 if (0 > avformat_find_stream_info (format_ctx, NULL))
946 {
947#if DEBUG
948 fprintf (stderr,
949 "Failed to read stream info\n");
950#endif
951 avformat_close_input (&format_ctx);
952 av_free (io_ctx);
953 return;
954 }
955 codec = NULL;
956 codec_ctx = NULL;
957 audio_stream_index = -1;
958 for (i = 0; i<format_ctx->nb_streams; i++)
959 {
960 codec_ctx = format_ctx->streams[i]->codec;
961 if (AVMEDIA_TYPE_AUDIO != codec_ctx->codec_type)
962 continue;
963 if (NULL == (codec = avcodec_find_decoder (codec_ctx->codec_id)))
964 continue;
965 options = NULL;
966 if (0 != (err = avcodec_open2 (codec_ctx, codec, &options)))
967 {
968 codec = NULL;
969 continue;
970 }
971 av_dict_free (&options);
972 audio_stream_index = i;
973 break;
974 }
975 if ( (-1 == audio_stream_index) ||
976 (0 == codec_ctx->channels) )
977 {
978#if DEBUG
979 fprintf (stderr,
980 "No audio streams or no suitable codec found\n");
981#endif
982 if (NULL != codec)
983 avcodec_close (codec_ctx);
984 avformat_close_input (&format_ctx);
985 av_free (io_ctx);
986 return;
987 }
988
989 frame = av_frame_alloc ();
990 if (NULL == frame)
991 {
992#if DEBUG
993 fprintf (stderr,
994 "Failed to allocate frame\n");
995#endif
996 avcodec_close (codec_ctx);
997 avformat_close_input (&format_ctx);
998 av_free (io_ctx);
999 return;
1000 }
1001
1002
1003 if (! (buffer = malloc (HARD_LIMIT_SIZE)))
1004 goto cleanup;
1005
1006
1007 /** Open the output file for writing. */
1008 if (open_output_file (codec_ctx,
1009 &output_format_context,
1010 &output_codec_context))
1011 goto cleanup;
1012 /** Initialize the resampler to be able to convert audio sample formats. */
1013 if (init_resampler (codec_ctx,
1014 output_codec_context,
1015 &resample_context))
1016 goto cleanup;
1017 /** Initialize the FIFO buffer to store audio samples to be encoded. */
1018 if (init_fifo (&fifo))
1019 goto cleanup;
1020
1021 /** Write the header of the output file container. */
1022 if (write_output_file_header (output_format_context))
1023 goto cleanup;
1024
1025
1026 if (format_ctx->duration == AV_NOPTS_VALUE)
1027 {
1028 duration = -1;
1029#if DEBUG
1030 fprintf (stderr,
1031 "Duration unknown\n");
1032#endif
1033 }
1034 else
1035 {
1036 duration = format_ctx->duration;
1037#if DEBUG
1038 fprintf (stderr,
1039 "Duration: %lld\n",
1040 format_ctx->duration);
1041#endif
1042 }
1043
1044 /* if duration is known, seek to first tried,
1045 * else use 10 sec into stream */
1046
1047 if (-1 != duration)
1048 err = av_seek_frame (format_ctx, -1, (duration / 3), 0);
1049 else
1050 err = av_seek_frame (format_ctx, -1, 10 * AV_TIME_BASE, 0);
1051
1052
1053 if (err >= 0)
1054 avcodec_flush_buffers (codec_ctx);
1055
1056
1057 /**
1058 * Loop as long as we have input samples to read or output samples
1059 * to write; abort as soon as we have neither.
1060 */
1061 while (1)
1062 {
1063 /** Use the encoder's desired frame size for processing. */
1064 const int output_frame_size = output_codec_context->frame_size;
1065 int finished = 0;
1066
1067 /**
1068 * Make sure that there is one frame worth of samples in the FIFO
1069 * buffer so that the encoder can do its work.
1070 * Since the decoder's and the encoder's frame size may differ, we
1071 * need to FIFO buffer to store as many frames worth of input samples
1072 * that they make up at least one frame worth of output samples.
1073 */
1074 while ((av_audio_fifo_size (fifo) < output_frame_size))
1075 {
1076 /**
1077 * Decode one frame worth of audio samples, convert it to the
1078 * output sample format and put it into the FIFO buffer.
1079 */
1080 if (read_decode_convert_and_store (fifo,
1081 format_ctx,
1082 codec_ctx,
1083 output_codec_context,
1084 resample_context,
1085 audio_stream_index,
1086 &finished))
1087 {
1088 goto cleanup;
1089 }
1090
1091 /**
1092 * If we are at the end of the input file, we continue
1093 * encoding the remaining audio samples to the output file.
1094 */
1095 if (finished)
1096 break;
1097 }
1098
1099 /* Already over our limit*/
1100 if (totalSize >= MAX_SIZE)
1101 finished = 1;
1102
1103 /**
1104 * If we have enough samples for the encoder, we encode them.
1105 * At the end of the file, we pass the remaining samples to
1106 * the encoder.
1107 *///
1108 while (av_audio_fifo_size (fifo) >= output_frame_size ||
1109 (finished && av_audio_fifo_size (fifo) > 0))
1110 {
1111 /**
1112 * Take one frame worth of audio samples from the FIFO buffer,
1113 * encode it and write it to the output file.
1114 */
1115 if (load_encode_and_write (fifo,
1116 output_format_context,
1117 output_codec_context))
1118 goto cleanup;
1119 }
1120 /**
1121 * If we are at the end of the input file and have encoded
1122 * all remaining samples, we can exit this loop and finish.
1123 */
1124 if (finished)
1125 {
1126 int data_written;
1127 /** Flush the encoder as it may have delayed frames. */
1128 do {
1129 encode_audio_frame (NULL,
1130 output_format_context,
1131 output_codec_context,
1132 &data_written);
1133 } while (data_written);
1134 break;
1135 }
1136 }
1137
1138 /** Write the trailer of the output file container. */
1139 if (write_output_file_trailer (output_format_context))
1140 goto cleanup;
1141 ec->proc (ec->cls,
1142 "previewopus",
1143 EXTRACTOR_METATYPE_AUDIO_PREVIEW,
1144 EXTRACTOR_METAFORMAT_BINARY,
1145 "audio/opus",
1146 buffer,
1147 totalSize);
1148
1149#if OUTPUT_FILE
1150 {
1151 FILE *f;
1152
1153 f = fopen ("example.opus", "wb");
1154 if (! f)
1155 {
1156 fprintf (stderr, "Could not open %s\n", "file");
1157 exit (1);
1158 }
1159 fwrite (buffer, 1, totalSize, f);
1160 fclose (f);
1161 }
1162#endif
1163
1164cleanup:
1165 av_free (frame);
1166 free (buffer);
1167
1168 if (fifo)
1169 av_audio_fifo_free (fifo);
1170 if (resample_context)
1171 {
1172 swr_close (resample_context);
1173 swr_free (&resample_context);
1174 }
1175 if (output_codec_context)
1176 avcodec_close (output_codec_context);
1177
1178 avcodec_close (codec_ctx);
1179 avformat_close_input (&format_ctx);
1180 av_free (io_ctx);
1181}
1182
1183
1184/**
1185 * Main method for the opus-preview plugin.
1186 *
1187 * @param ec extraction context
1188 */
1189void
1190EXTRACTOR_previewopus_extract_method (struct EXTRACTOR_ExtractContext *ec)
1191{
1192 ssize_t iret;
1193 void *data;
1194
1195 if (-1 == (iret = ec->read (ec->cls,
1196 &data,
1197 16 * 1024)))
1198 return;
1199
1200 if (0 != ec->seek (ec->cls, 0, SEEK_SET))
1201 return;
1202
1203 extract_audio (ec);
1204}
1205
1206
1207/**
1208 * Log callback. Does nothing.
1209 *
1210 * @param ptr NULL
1211 * @param level log level
1212 * @param format format string
1213 * @param ap arguments for format
1214 */
1215static void
1216previewopus_av_log_callback (void*ptr,
1217 int level,
1218 const char *format,
1219 va_list ap)
1220{
1221#if DEBUG
1222 vfprintf (stderr, format, ap);
1223#endif
1224}
1225
1226
1227/**
1228 * Initialize av-libs
1229 */
1230void __attribute__ ((constructor))
1231previewopus_lib_init (void)
1232{
1233 av_log_set_callback (&previewopus_av_log_callback);
1234}
1235
1236
1237/**
1238 * Destructor for the library, cleans up.
1239 */
1240void __attribute__ ((destructor))
1241previewopus_ltdl_fini ()
1242{
1243
1244}
1245
1246
1247/* end of previewopus_extractor.c */
generated by cgit v1.2.3 (git 2.39.1) at 2024-04-26 12:51:32 +0000