diff options
Diffstat (limited to 'src/plugins/previewopus_extractor.c')
| -rw-r--r-- | src/plugins/previewopus_extractor.c | 1247 |
1 files changed, 0 insertions, 1247 deletions
diff --git a/src/plugins/previewopus_extractor.c b/src/plugins/previewopus_extractor.c deleted file mode 100644 index f137f38..0000000 --- a/src/plugins/previewopus_extractor.c +++ /dev/null @@ -1,1247 +0,0 @@ -/* - This file is part of libextractor. - Copyright Copyright (C) 2008, 2013 Bruno Cabral and Christian Grothoff - - libextractor is free software; you can redistribute it and/or modify - it under the terms of the GNU General Public License as published - by the Free Software Foundation; either version 3, or (at your - option) any later version. - - libextractor is distributed in the hope that it will be useful, but - WITHOUT ANY WARRANTY; without even the implied warranty of - MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - General Public License for more details. - - You should have received a copy of the GNU General Public License - along with libextractor; see the file COPYING. If not, write to the - Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor, - Boston, MA 02110-1301, USA. - */ -/** - * @file previewopus_extractor.c - * @author Bruno Cabral - * @author Christian Grothoff - * @brief this extractor produces a binary encoded - * audio snippet of music/video files using ffmpeg libs. - * - * Based on ffmpeg samples. - * - * Note that ffmpeg has a few issues: - * (1) there are no recent official releases of the ffmpeg libs - * (2) ffmpeg has a history of having security issues (parser is not robust) - * - * So this plugin cannot be recommended for system with high security - *requirements. - */ -#include "platform.h" -#include "extractor.h" -#include <magic.h> - -#include <libavutil/avutil.h> -#include <libavutil/audio_fifo.h> -#include <libavutil/opt.h> -#include <libavutil/mathematics.h> -#include <libavformat/avformat.h> -#include <libavcodec/avcodec.h> -#include <libswscale/swscale.h> -#include <libswresample/swresample.h> - - -/** - * Set to 1 to enable debug output. - */ -#define DEBUG 0 - -/** - * Set to 1 to enable a output file for testing. - */ -#define OUTPUT_FILE 0 - - -/** - * Maximum size in bytes for the preview. - */ -#define MAX_SIZE (28 * 1024) - -/** - * HardLimit for file - */ -#define HARD_LIMIT_SIZE (50 * 1024) - - -/** The output bit rate in kbit/s */ -#define OUTPUT_BIT_RATE 28000 -/** The number of output channels */ -#define OUTPUT_CHANNELS 2 -/** The audio sample output format */ -#define OUTPUT_SAMPLE_FORMAT AV_SAMPLE_FMT_S16 - - -/** Our output buffer*/ -static unsigned char *buffer; - -/** Actual output buffer size */ -static int totalSize; - -/** - * Convert an error code into a text message. - * @param error Error code to be converted - * @return Corresponding error text (not thread-safe) - */ -static char *const -get_error_text (const int error) -{ - static char error_buffer[255]; - av_strerror (error, error_buffer, sizeof(error_buffer)); - return error_buffer; -} - - -/** - * Read callback. - * - * @param opaque the 'struct EXTRACTOR_ExtractContext' - * @param buf where to write data - * @param buf_size how many bytes to read - * @return -1 on error (or for unknown file size) - */ -static int -read_cb (void *opaque, - uint8_t *buf, - int buf_size) -{ - struct EXTRACTOR_ExtractContext *ec = opaque; - void *data; - ssize_t ret; - - ret = ec->read (ec->cls, &data, buf_size); - if (ret <= 0) - return ret; - memcpy (buf, data, ret); - return ret; -} - - -/** - * Seek callback. - * - * @param opaque the 'struct EXTRACTOR_ExtractContext' - * @param offset where to seek - * @param whence how to seek; AVSEEK_SIZE to return file size without seeking - * @return -1 on error (or for unknown file size) - */ -static int64_t -seek_cb (void *opaque, - int64_t offset, - int whence) -{ - struct EXTRACTOR_ExtractContext *ec = opaque; - - if (AVSEEK_SIZE == whence) - return ec->get_size (ec->cls); - return ec->seek (ec->cls, offset, whence); -} - - -/** - * write callback. - * - * @param opaque NULL - * @param pBuffer to write - * @param pBufferSize , amount to write - * @return 0 on error - */ -static int -writePacket (void *opaque, - unsigned char *pBuffer, - int pBufferSize) -{ - int sizeToCopy = pBufferSize; - - if ( (totalSize + pBufferSize) > HARD_LIMIT_SIZE) - sizeToCopy = HARD_LIMIT_SIZE - totalSize; - - memcpy (buffer + totalSize, pBuffer, sizeToCopy); - totalSize += sizeToCopy; - return sizeToCopy; -} - - -/** - * Open an output file and the required encoder. - * Also set some basic encoder parameters. - * Some of these parameters are based on the input file's parameters. - */ -static int -open_output_file ( - AVCodecContext *input_codec_context, - AVFormatContext **output_format_context, - AVCodecContext **output_codec_context) -{ - AVStream *stream = NULL; - AVCodec *output_codec = NULL; - AVIOContext *io_ctx; - int error; - unsigned char *iob; - - if (NULL == (iob = av_malloc (16 * 1024))) - return AVERROR_EXIT; - if (NULL == (io_ctx = avio_alloc_context (iob, 16 * 1024, - AVIO_FLAG_WRITE, NULL, - NULL, - &writePacket /* no writing */, - NULL))) - { - av_free (iob); - return AVERROR_EXIT; - } - if (NULL == ((*output_format_context) = avformat_alloc_context ())) - { - av_free (io_ctx); - return AVERROR_EXIT; - } - (*output_format_context)->pb = io_ctx; - - /** Guess the desired container format based on the file extension. */ - if (! ((*output_format_context)->oformat = av_guess_format (NULL, - "file.ogg", - NULL))) - { -#if DEBUG - fprintf (stderr, "Could not find output file format\n"); -#endif - error = AVERROR (ENOSYS); - goto cleanup; - } - - /** Find the encoder to be used by its name. */ - if (! (output_codec = avcodec_find_encoder (AV_CODEC_ID_OPUS))) - { -#if DEBUG - fprintf (stderr, "Could not find an OPUS encoder.\n"); -#endif - error = AVERROR (ENOSYS); - goto cleanup; - } - - /** Create a new audio stream in the output file container. */ - if (! (stream = avformat_new_stream (*output_format_context, output_codec))) - { -#if DEBUG - fprintf (stderr, "Could not create new stream\n"); -#endif - error = AVERROR (ENOMEM); - goto cleanup; - } - - /** Save the encoder context for easiert access later. */ - *output_codec_context = stream->codec; - - /** - * Set the basic encoder parameters. - * The input file's sample rate is used to avoid a sample rate conversion. - */ - (*output_codec_context)->channels = OUTPUT_CHANNELS; - (*output_codec_context)->channel_layout = av_get_default_channel_layout ( - OUTPUT_CHANNELS); - (*output_codec_context)->sample_rate = 48000; // Opus need 48000 - (*output_codec_context)->sample_fmt = AV_SAMPLE_FMT_S16; - (*output_codec_context)->bit_rate = OUTPUT_BIT_RATE; - - /** Open the encoder for the audio stream to use it later. */ - if ((error = avcodec_open2 (*output_codec_context, output_codec, NULL)) < 0) - { -#if DEBUG - fprintf (stderr, "Could not open output codec (error '%s')\n", - get_error_text (error)); -#endif - goto cleanup; - } - return 0; - -cleanup: - av_free (io_ctx); - return error < 0 ? error : AVERROR_EXIT; -} - - -/** Initialize one data packet for reading or writing. */ -static void -init_packet (AVPacket *packet) -{ - av_init_packet (packet); - /** Set the packet data and size so that it is recognized as being empty. */ - packet->data = NULL; - packet->size = 0; -} - - -/** Initialize one audio frame for reading from the input file */ -static int -init_input_frame (AVFrame **frame) -{ - *frame = av_frame_alloc (); - if (NULL == *frame) - { -#if DEBUG - fprintf (stderr, "Could not allocate input frame\n"); -#endif - return AVERROR (ENOMEM); - } - return 0; -} - - -/** - * Initialize the audio resampler based on the input and output codec settings. - * If the input and output sample formats differ, a conversion is required - * libavresample takes care of this, but requires initialization. - */ -static int -init_resampler (AVCodecContext *input_codec_context, - AVCodecContext *output_codec_context, - SwrContext **resample_context) -{ - /** - * Only initialize the resampler if it is necessary, i.e., - * if and only if the sample formats differ. - */ - if ((input_codec_context->sample_fmt != output_codec_context->sample_fmt) || - (input_codec_context->channels != output_codec_context->channels) ) - { - int error; - - /** Create a resampler context for the conversion. */ - if (! (*resample_context = swr_alloc ())) - { -#if DEBUG - fprintf (stderr, "Could not allocate resample context\n"); -#endif - return AVERROR (ENOMEM); - } - - - /** - * Set the conversion parameters. - * Default channel layouts based on the number of channels - * are assumed for simplicity (they are sometimes not detected - * properly by the demuxer and/or decoder). - */ - av_opt_set_int (*resample_context, - "in_channel_layout", - av_get_default_channel_layout ( - input_codec_context->channels), 0); - av_opt_set_int (*resample_context, - "out_channel_layout", - av_get_default_channel_layout ( - output_codec_context->channels), 0); - av_opt_set_int (*resample_context, - "in_sample_rate", - input_codec_context->sample_rate, 0); - av_opt_set_int (*resample_context, - "out_sample_rate", - output_codec_context->sample_rate, 0); - av_opt_set_int (*resample_context, - "in_sample_fmt", - input_codec_context->sample_fmt, 0); - av_opt_set_int (*resample_context, - "out_sample_fmt", - output_codec_context->sample_fmt, 0); - - /** Open the resampler with the specified parameters. */ - if ((error = swr_init (*resample_context)) < 0) - { -#if DEBUG - fprintf (stderr, "Could not open resample context\n"); -#endif - swr_free (resample_context); - return error; - } - } - return 0; -} - - -/** Initialize a FIFO buffer for the audio samples to be encoded. */ -static int -init_fifo (AVAudioFifo **fifo) -{ - /** Create the FIFO buffer based on the specified output sample format. */ - if (! (*fifo = av_audio_fifo_alloc (OUTPUT_SAMPLE_FORMAT, OUTPUT_CHANNELS, - 1))) - { -#if DEBUG - fprintf (stderr, "Could not allocate FIFO\n"); -#endif - return AVERROR (ENOMEM); - } - return 0; -} - - -/** Write the header of the output file container. */ -static int -write_output_file_header (AVFormatContext *output_format_context) -{ - int error; - if ((error = avformat_write_header (output_format_context, NULL)) < 0) - { -#if DEBUG - fprintf (stderr, "Could not write output file header (error '%s')\n", - get_error_text (error)); -#endif - return error; - } - return 0; -} - - -/** Decode one audio frame from the input file. */ -static int -decode_audio_frame (AVFrame *frame, - AVFormatContext *input_format_context, - AVCodecContext *input_codec_context, int audio_stream_index, - int *data_present, int *finished) -{ - /** Packet used for temporary storage. */ - AVPacket input_packet; - int error; - init_packet (&input_packet); - - /** Read one audio frame from the input file into a temporary packet. */ - while (1) - { - if ((error = av_read_frame (input_format_context, &input_packet)) < 0) - { - /** If we are the the end of the file, flush the decoder below. */ - if (error == AVERROR_EOF) - { -#if DEBUG - fprintf (stderr, "EOF in decode_audio\n"); -#endif - *finished = 1; - } - else - { -#if DEBUG - fprintf (stderr, "Could not read frame (error '%s')\n", - get_error_text (error)); -#endif - return error; - } - } - - if (input_packet.stream_index == audio_stream_index) - break; - } - - /** - * Decode the audio frame stored in the temporary packet. - * The input audio stream decoder is used to do this. - * If we are at the end of the file, pass an empty packet to the decoder - * to flush it. - */if ((error = avcodec_decode_audio4 (input_codec_context, frame, - data_present, &input_packet)) < 0) - { -#if DEBUG - fprintf (stderr, "Could not decode frame (error '%s')\n", - get_error_text (error)); -#endif - av_packet_unref (&input_packet); - return error; - } - - /** - * If the decoder has not been flushed completely, we are not finished, - * so that this function has to be called again. - */ - if (*finished && *data_present) - *finished = 0; - av_packet_unref (&input_packet); - return 0; -} - - -/** - * Initialize a temporary storage for the specified number of audio samples. - * The conversion requires temporary storage due to the different format. - * The number of audio samples to be allocated is specified in frame_size. - */ -static int -init_converted_samples (uint8_t ***converted_input_samples, - int*out_linesize, - AVCodecContext *output_codec_context, - int frame_size) -{ - int error; - - /** - * Allocate as many pointers as there are audio channels. - * Each pointer will later point to the audio samples of the corresponding - * channels (although it may be NULL for interleaved formats). - */if (! (*converted_input_samples = calloc (output_codec_context->channels, - sizeof(**converted_input_samples)))) - { -#if DEBUG - fprintf (stderr, "Could not allocate converted input sample pointers\n"); -#endif - return AVERROR (ENOMEM); - } - - /** - * Allocate memory for the samples of all channels in one consecutive - * block for convenience. - */ - if ((error = av_samples_alloc (*converted_input_samples, - out_linesize, - output_codec_context->channels, - frame_size, - output_codec_context->sample_fmt, 0)) < 0) - { -#if DEBUG - fprintf (stderr, - "Could not allocate converted input samples (error '%s')\n", - get_error_text (error)); -#endif - av_freep (&(*converted_input_samples)[0]); - free (*converted_input_samples); - return error; - } - return 0; -} - - -/** - * Convert the input audio samples into the output sample format. - * The conversion happens on a per-frame basis, the size of which is specified - * by frame_size. - */ -static int -convert_samples (uint8_t **input_data, - uint8_t **converted_data, - int in_sample, - int out_sample, - int out_linesize, - SwrContext *resample_context) -{ - int error; - - /** Convert the samples using the resampler. */ - if ((error = swr_convert (resample_context, - converted_data, - out_linesize, - out_sample, - input_data, - 0, - in_sample)) < 0) - { -#if DEBUG - fprintf (stderr, - "Could not convert input samples (error '%s')\n", - get_error_text (error)); -#endif - return error; - } - - - /** - * Perform a sanity check so that the number of converted samples is - * not greater than the number of samples to be converted. - * If the sample rates differ, this case has to be handled differently - */ - if (avresample_available (resample_context)) - { -#if DEBUG - fprintf (stderr, - "%i Converted samples left over\n", - avresample_available (resample_context)); -#endif - } - - - return 0; -} - - -/** Add converted input audio samples to the FIFO buffer for later processing. */ -static int -add_samples_to_fifo (AVAudioFifo *fifo, - uint8_t **converted_input_samples, - const int frame_size) -{ - int error; - - /** - * Make the FIFO as large as it needs to be to hold both, - * the old and the new samples. - */ - if ((error = av_audio_fifo_realloc (fifo, av_audio_fifo_size (fifo) - + frame_size)) < 0) - { -#if DEBUG - fprintf (stderr, "Could not reallocate FIFO\n"); -#endif - return error; - } - - /** Store the new samples in the FIFO buffer. */ - if (av_audio_fifo_write (fifo, (void **) converted_input_samples, - frame_size) < frame_size) - { -#if DEBUG - fprintf (stderr, "Could not write data to FIFO\n"); -#endif - return AVERROR_EXIT; - } - return 0; -} - - -/** - * Read one audio frame from the input file, decodes, converts and stores - * it in the FIFO buffer. - */ -static int -read_decode_convert_and_store (AVAudioFifo *fifo, - AVFormatContext *input_format_context, - AVCodecContext *input_codec_context, - AVCodecContext *output_codec_context, - SwrContext *resampler_context, int - audio_stream_index, - int *finished) -{ - /** Temporary storage of the input samples of the frame read from the file. */ - AVFrame *input_frame = NULL; - /** Temporary storage for the converted input samples. */ - uint8_t **converted_input_samples = NULL; - int data_present; - int ret = AVERROR_EXIT; - - /** Initialize temporary storage for one input frame. */ - if (init_input_frame (&input_frame)) - { -#if DEBUG - fprintf (stderr, "Failed at init frame\n"); -#endif - goto cleanup; - - } - /** Decode one frame worth of audio samples. */ - if (decode_audio_frame (input_frame, - input_format_context, - input_codec_context, - audio_stream_index, - &data_present, - finished)) - { -#if DEBUG - fprintf (stderr, "Failed at decode audio\n"); -#endif - - goto cleanup; - } - /** - * If we are at the end of the file and there are no more samples - * in the decoder which are delayed, we are actually finished. - * This must not be treated as an error. - */ - if (*finished && ! data_present) - { - ret = 0; -#if DEBUG - fprintf (stderr, "Failed at finished or no data\n"); -#endif - goto cleanup; - } - /** If there is decoded data, convert and store it */ - if (data_present) - { - int out_linesize; - // FIXME: I'm losing samples, but can't get it to work. - int out_samples = avresample_available (resampler_context) - + avresample_get_delay (resampler_context) - + input_frame->nb_samples; - - - // fprintf(stderr, "Input nbsamples %i out_samples: %i \n",input_frame->nb_samples,out_samples); - - /** Initialize the temporary storage for the converted input samples. */ - if (init_converted_samples (&converted_input_samples, - &out_linesize, - output_codec_context, - out_samples)) - { -#if DEBUG - fprintf (stderr, "Failed at init_converted_samples\n"); -#endif - goto cleanup; - } - - /** - * Convert the input samples to the desired output sample format. - * This requires a temporary storage provided by converted_input_samples. - */ - if (convert_samples (input_frame->extended_data, - converted_input_samples, - input_frame->nb_samples, - out_samples, - out_linesize, - resampler_context)) - { -#if DEBUG - fprintf (stderr, - "Failed at convert_samples, input frame %i \n", - input_frame->nb_samples); -#endif - goto cleanup; - } - /** Add the converted input samples to the FIFO buffer for later processing. */ - if (add_samples_to_fifo (fifo, - converted_input_samples, - out_samples)) - { -#if DEBUG - fprintf (stderr, "Failed at add_samples_to_fifo\n"); -#endif - goto cleanup; - } - ret = 0; - } - ret = 0; - -cleanup: - if (converted_input_samples) - { - av_freep (&converted_input_samples[0]); - free (converted_input_samples); - } - av_frame_free (&input_frame); - return ret; -} - - -/** - * Initialize one input frame for writing to the output file. - * The frame will be exactly frame_size samples large. - */ -static int -init_output_frame (AVFrame **frame, - AVCodecContext *output_codec_context, - int frame_size) -{ - int error; - - /** Create a new frame to store the audio samples. */ - *frame = av_frame_alloc (); - if (NULL == *frame) - { -#if DEBUG - fprintf (stderr, "Could not allocate output frame\n"); -#endif - return AVERROR_EXIT; - } - - /** - * Set the frame's parameters, especially its size and format. - * av_frame_get_buffer needs this to allocate memory for the - * audio samples of the frame. - * Default channel layouts based on the number of channels - * are assumed for simplicity. - */(*frame)->nb_samples = frame_size; - (*frame)->channel_layout = output_codec_context->channel_layout; - (*frame)->format = output_codec_context->sample_fmt; - (*frame)->sample_rate = output_codec_context->sample_rate; - - - // fprintf(stderr, "%i %i \n",frame_size , (*frame)->format,(*frame)->sample_rate); - - /** - * Allocate the samples of the created frame. This call will make - * sure that the audio frame can hold as many samples as specified. - */ - if ((error = av_frame_get_buffer (*frame, 0)) < 0) - { -#if DEBUG - fprintf (stderr, "Could allocate output frame samples (error '%s')\n", - get_error_text (error)); -#endif - av_frame_free (frame); - return error; - } - - return 0; -} - - -/** Encode one frame worth of audio to the output file. */ -static int -encode_audio_frame (AVFrame *frame, - AVFormatContext *output_format_context, - AVCodecContext *output_codec_context, - int *data_present) -{ - /** Packet used for temporary storage. */ - AVPacket output_packet; - int error; - init_packet (&output_packet); - - /** - * Encode the audio frame and store it in the temporary packet. - * The output audio stream encoder is used to do this. - */ - if ((error = avcodec_encode_audio2 (output_codec_context, &output_packet, - frame, data_present)) < 0) - { -#if DEBUG - fprintf (stderr, "Could not encode frame (error '%s')\n", - get_error_text (error)); -#endif - av_packet_unref (&output_packet); - return error; - } - - /** Write one audio frame from the temporary packet to the output file. */ - if (*data_present) - { - if ((error = av_write_frame (output_format_context, &output_packet)) < 0) - { -#if DEBUG - fprintf (stderr, "Could not write frame (error '%s')\n", - get_error_text (error)); -#endif - - av_packet_unref (&output_packet); - return error; - } - - av_packet_unref (&output_packet); - } - - return 0; -} - - -/** - * Load one audio frame from the FIFO buffer, encode and write it to the - * output file. - */ -static int -load_encode_and_write (AVAudioFifo *fifo, - AVFormatContext *output_format_context, - AVCodecContext *output_codec_context) -{ - /** Temporary storage of the output samples of the frame written to the file. */ - AVFrame *output_frame; - /** - * Use the maximum number of possible samples per frame. - * If there is less than the maximum possible frame size in the FIFO - * buffer use this number. Otherwise, use the maximum possible frame size - */const int frame_size = FFMIN (av_audio_fifo_size (fifo), - output_codec_context->frame_size); - int data_written; - - /** Initialize temporary storage for one output frame. */ - if (init_output_frame (&output_frame, output_codec_context, frame_size)) - return AVERROR_EXIT; - - /** - * Read as many samples from the FIFO buffer as required to fill the frame. - * The samples are stored in the frame temporarily. - */ - if (av_audio_fifo_read (fifo, (void **) output_frame->data, frame_size) < - frame_size) - { -#if DEBUG - fprintf (stderr, "Could not read data from FIFO\n"); -#endif - av_frame_free (&output_frame); - return AVERROR_EXIT; - } - - /** Encode one frame worth of audio samples. */ - if (encode_audio_frame (output_frame, output_format_context, - output_codec_context, &data_written)) - { - av_frame_free (&output_frame); - return AVERROR_EXIT; - } - av_frame_free (&output_frame); - return 0; -} - - -/** Write the trailer of the output file container. */ -static int -write_output_file_trailer (AVFormatContext *output_format_context) -{ - int error; - if ((error = av_write_trailer (output_format_context)) < 0) - { -#if DEBUG - fprintf (stderr, "Could not write output file trailer (error '%s')\n", - get_error_text (error)); -#endif - return error; - } - return 0; -} - - -#define ENUM_CODEC_ID enum AVCodecID - - -/** - * Perform the audio snippet extraction - * - * @param ec extraction context to use - */ -static void -extract_audio (struct EXTRACTOR_ExtractContext *ec) -{ - AVIOContext *io_ctx; - struct AVFormatContext *format_ctx; - AVCodecContext *codec_ctx; - AVFormatContext *output_format_context = NULL; - AVCodec *codec; - AVDictionary *options; - AVFrame *frame; - AVCodecContext*output_codec_context = NULL; - SwrContext *resample_context = NULL; - AVAudioFifo *fifo = NULL; - - int audio_stream_index; - int i; - int err; - int duration; - unsigned char *iob; - - - totalSize = 0; - if (NULL == (iob = av_malloc (16 * 1024))) - return; - if (NULL == (io_ctx = avio_alloc_context (iob, - 16 * 1024, - 0, ec, - &read_cb, - NULL /* no writing */, - &seek_cb))) - { - av_free (iob); - return; - } - if (NULL == (format_ctx = avformat_alloc_context ())) - { - av_free (io_ctx); - return; - } - format_ctx->pb = io_ctx; - options = NULL; - if (0 != avformat_open_input (&format_ctx, "<no file>", NULL, &options)) - { - av_free (io_ctx); - return; - } - av_dict_free (&options); - if (0 > avformat_find_stream_info (format_ctx, NULL)) - { -#if DEBUG - fprintf (stderr, - "Failed to read stream info\n"); -#endif - avformat_close_input (&format_ctx); - av_free (io_ctx); - return; - } - codec = NULL; - codec_ctx = NULL; - audio_stream_index = -1; - for (i = 0; i<format_ctx->nb_streams; i++) - { - codec_ctx = format_ctx->streams[i]->codec; - if (AVMEDIA_TYPE_AUDIO != codec_ctx->codec_type) - continue; - if (NULL == (codec = avcodec_find_decoder (codec_ctx->codec_id))) - continue; - options = NULL; - if (0 != (err = avcodec_open2 (codec_ctx, codec, &options))) - { - codec = NULL; - continue; - } - av_dict_free (&options); - audio_stream_index = i; - break; - } - if ( (-1 == audio_stream_index) || - (0 == codec_ctx->channels) ) - { -#if DEBUG - fprintf (stderr, - "No audio streams or no suitable codec found\n"); -#endif - if (NULL != codec) - avcodec_close (codec_ctx); - avformat_close_input (&format_ctx); - av_free (io_ctx); - return; - } - - frame = av_frame_alloc (); - if (NULL == frame) - { -#if DEBUG - fprintf (stderr, - "Failed to allocate frame\n"); -#endif - avcodec_close (codec_ctx); - avformat_close_input (&format_ctx); - av_free (io_ctx); - return; - } - - - if (! (buffer = malloc (HARD_LIMIT_SIZE))) - goto cleanup; - - - /** Open the output file for writing. */ - if (open_output_file (codec_ctx, - &output_format_context, - &output_codec_context)) - goto cleanup; - /** Initialize the resampler to be able to convert audio sample formats. */ - if (init_resampler (codec_ctx, - output_codec_context, - &resample_context)) - goto cleanup; - /** Initialize the FIFO buffer to store audio samples to be encoded. */ - if (init_fifo (&fifo)) - goto cleanup; - - /** Write the header of the output file container. */ - if (write_output_file_header (output_format_context)) - goto cleanup; - - - if (format_ctx->duration == AV_NOPTS_VALUE) - { - duration = -1; -#if DEBUG - fprintf (stderr, - "Duration unknown\n"); -#endif - } - else - { - duration = format_ctx->duration; -#if DEBUG - fprintf (stderr, - "Duration: %lld\n", - format_ctx->duration); -#endif - } - - /* if duration is known, seek to first tried, - * else use 10 sec into stream */ - - if (-1 != duration) - err = av_seek_frame (format_ctx, -1, (duration / 3), 0); - else - err = av_seek_frame (format_ctx, -1, 10 * AV_TIME_BASE, 0); - - - if (err >= 0) - avcodec_flush_buffers (codec_ctx); - - - /** - * Loop as long as we have input samples to read or output samples - * to write; abort as soon as we have neither. - */ - while (1) - { - /** Use the encoder's desired frame size for processing. */ - const int output_frame_size = output_codec_context->frame_size; - int finished = 0; - - /** - * Make sure that there is one frame worth of samples in the FIFO - * buffer so that the encoder can do its work. - * Since the decoder's and the encoder's frame size may differ, we - * need to FIFO buffer to store as many frames worth of input samples - * that they make up at least one frame worth of output samples. - */ - while ((av_audio_fifo_size (fifo) < output_frame_size)) - { - /** - * Decode one frame worth of audio samples, convert it to the - * output sample format and put it into the FIFO buffer. - */ - if (read_decode_convert_and_store (fifo, - format_ctx, - codec_ctx, - output_codec_context, - resample_context, - audio_stream_index, - &finished)) - { - goto cleanup; - } - - /** - * If we are at the end of the input file, we continue - * encoding the remaining audio samples to the output file. - */ - if (finished) - break; - } - - /* Already over our limit*/ - if (totalSize >= MAX_SIZE) - finished = 1; - - /** - * If we have enough samples for the encoder, we encode them. - * At the end of the file, we pass the remaining samples to - * the encoder. - */// - while (av_audio_fifo_size (fifo) >= output_frame_size || - (finished && av_audio_fifo_size (fifo) > 0)) - { - /** - * Take one frame worth of audio samples from the FIFO buffer, - * encode it and write it to the output file. - */ - if (load_encode_and_write (fifo, - output_format_context, - output_codec_context)) - goto cleanup; - } - /** - * If we are at the end of the input file and have encoded - * all remaining samples, we can exit this loop and finish. - */ - if (finished) - { - int data_written; - /** Flush the encoder as it may have delayed frames. */ - do { - encode_audio_frame (NULL, - output_format_context, - output_codec_context, - &data_written); - } while (data_written); - break; - } - } - - /** Write the trailer of the output file container. */ - if (write_output_file_trailer (output_format_context)) - goto cleanup; - ec->proc (ec->cls, - "previewopus", - EXTRACTOR_METATYPE_AUDIO_PREVIEW, - EXTRACTOR_METAFORMAT_BINARY, - "audio/opus", - buffer, - totalSize); - -#if OUTPUT_FILE - { - FILE *f; - - f = fopen ("example.opus", "wb"); - if (! f) - { - fprintf (stderr, "Could not open %s\n", "file"); - exit (1); - } - fwrite (buffer, 1, totalSize, f); - fclose (f); - } -#endif - -cleanup: - av_free (frame); - free (buffer); - - if (fifo) - av_audio_fifo_free (fifo); - if (resample_context) - { - swr_close (resample_context); - swr_free (&resample_context); - } - if (output_codec_context) - avcodec_close (output_codec_context); - - avcodec_close (codec_ctx); - avformat_close_input (&format_ctx); - av_free (io_ctx); -} - - -/** - * Main method for the opus-preview plugin. - * - * @param ec extraction context - */ -void -EXTRACTOR_previewopus_extract_method (struct EXTRACTOR_ExtractContext *ec) -{ - ssize_t iret; - void *data; - - if (-1 == (iret = ec->read (ec->cls, - &data, - 16 * 1024))) - return; - - if (0 != ec->seek (ec->cls, 0, SEEK_SET)) - return; - - extract_audio (ec); -} - - -/** - * Log callback. Does nothing. - * - * @param ptr NULL - * @param level log level - * @param format format string - * @param ap arguments for format - */ -static void -previewopus_av_log_callback (void*ptr, - int level, - const char *format, - va_list ap) -{ -#if DEBUG - vfprintf (stderr, format, ap); -#endif -} - - -/** - * Initialize av-libs - */ -void __attribute__ ((constructor)) -previewopus_lib_init (void) -{ - av_log_set_callback (&previewopus_av_log_callback); -} - - -/** - * Destructor for the library, cleans up. - */ -void __attribute__ ((destructor)) -previewopus_ltdl_fini () -{ - -} - - -/* end of previewopus_extractor.c */ |