commit 99c9532182880ffc92abadc7b2d6a2a5e2f9492b
parent 1581f16bd8621240a5fab9f8b13ddafdba5afe7d
Author: Bruno Cabral <brataoufba@gmail.com>
Date: Thu, 19 Dec 2013 02:28:37 +0000
Introducing Opus previewer. It should create a opus/ogg 30KB preview if the file have a audio stream.
Requires a recent libav.
Diffstat:
5 files changed, 1233 insertions(+), 2 deletions(-)
diff --git a/configure.ac b/configure.ac
@@ -654,6 +654,7 @@ AC_CHECK_PROG([HAVE_ZZUF],[zzuf], 1, 0)
AM_CONDITIONAL(HAVE_ZZUF, test 0 != $HAVE_ZZUF)
AC_MSG_CHECKING([whether to enable the FFmpeg thumbnail extractor])
+new_ffmpeg=0
AC_ARG_ENABLE(ffmpeg,
[AC_HELP_STRING([--enable-ffmpeg],[Enable FFmpeg support])
AC_HELP_STRING([--disable-ffmpeg],[Disable FFmpeg support])],
@@ -670,12 +671,15 @@ AC_ARG_ENABLE(ffmpeg,
if test x$ffmpeg_enabled = x1
then
ffmpeg_enabled=0
+ new_ffmpeg=0
+ AC_CHECK_HEADERS([libavutil/frame.h],new_ffmpeg=1)
AC_CHECK_LIB(swscale, sws_getContext,
AC_CHECK_LIB(avcodec, avcodec_alloc_context3,
ffmpeg_enabled=1))
- AC_CHECK_HEADERS([libavutil/avutil.h ffmpeg/avutil.h libavformat/avformat.h ffmpeg/avformat.h libavcodec/avcodec.h ffmpeg/avcodec.h libswscale/swscale.h ffmpeg/swscale.h])
+ AC_CHECK_HEADERS([libavutil/avutil.h ffmpeg/avutil.h libavformat/avformat.h ffmpeg/avformat.h libavcodec/avcodec.h ffmpeg/avcodec.h libswscale/swscale.h ffmpeg/swscale.h libavresample/avresample.h ffmpeg/avresample.h])
fi
AM_CONDITIONAL(HAVE_FFMPEG, test x$ffmpeg_enabled != x0)
+AM_CONDITIONAL(HAVE_FFMPEG_NEW, test x$new_ffmpeg != x0)
LE_INTLINCL=""
@@ -799,6 +803,11 @@ then
AC_MSG_NOTICE([NOTICE: FFmpeg thumbnailer plugin disabled])
fi
+if test "x$new_ffmpeg" = "x0"
+then
+ AC_MSG_NOTICE([NOTICE: FFmpeg/opus audio preview plugin disabled])
+fi
+
if test "x$without_gtk" = "xtrue"
then
AC_MSG_NOTICE([NOTICE: libgtk3+ not found, gtk thumbnail support disabled])
diff --git a/src/include/extractor.h b/src/include/extractor.h
@@ -382,8 +382,10 @@ enum EXTRACTOR_MetaType
EXTRACTOR_METATYPE_VIDEO_DURATION = 225,
EXTRACTOR_METATYPE_AUDIO_DURATION = 226,
EXTRACTOR_METATYPE_SUBTITLE_DURATION = 227,
+
+ EXTRACTOR_METATYPE_AUDIO_PREVIEW = 228,
- EXTRACTOR_METATYPE_LAST = 228
+ EXTRACTOR_METATYPE_LAST = 229
};
/** @} */ /* end of meta data types */
diff --git a/src/main/extractor_metatypes.c b/src/main/extractor_metatypes.c
@@ -548,6 +548,9 @@ static const struct MetaTypeDescription meta_type_descriptions[] = {
{ gettext_noop ("subtitle duration"),
gettext_noop ("duration of a subtitle stream") },
+ { gettext_noop ("audio preview"),
+ gettext_noop ("a preview of the file audio stream") },
+
{ gettext_noop ("last"),
gettext_noop ("last") }
};
diff --git a/src/plugins/Makefile.am b/src/plugins/Makefile.am
@@ -70,6 +70,13 @@ if HAVE_FFMPEG
# FFmpeg-thumbnailer requires MAGIC and FFMPEG
PLUGIN_FFMPEG=libextractor_thumbnailffmpeg.la
TEST_FFMPEG=test_thumbnailffmpeg
+
+
+if HAVE_FFMPEG_NEW
+PLUGIN_PREVIEWOPUS=libextractor_previewopus.la
+TEST_PREVIEWOPUS=test_previewopus
+endif
+
endif
if HAVE_GTK
@@ -153,6 +160,7 @@ PLUGIN_OGG=libextractor_ogg.la
TEST_OGG=test_ogg
endif
+
if HAVE_ZLIB
PLUGIN_ZLIB= \
libextractor_deb.la \
@@ -190,6 +198,7 @@ plugin_LTLIBRARIES = \
$(PLUGIN_MP4) \
$(PLUGIN_MPEG) \
$(PLUGIN_OGG) \
+ $(PLUGIN_PREVIEWOPUS) \
$(PLUGIN_RPM) \
$(PLUGIN_TIFF) \
$(PLUGIN_ZLIB)
@@ -216,6 +225,7 @@ check_PROGRAMS = \
$(TEST_ARCHIVE) \
$(TEST_EXIV2) \
$(TEST_FFMPEG) \
+ $(TEST_PREVIEWOPUS) \
$(TEST_FLAC) \
$(TEST_GIF) \
$(TEST_GSF) \
@@ -620,7 +630,19 @@ test_thumbnailgtk_SOURCES = \
test_thumbnailgtk_LDADD = \
$(top_builddir)/src/plugins/libtest.la
+libextractor_previewopus_la_SOURCES = \
+ previewopus_extractor.c
+libextractor_previewopus_la_LDFLAGS = \
+ $(PLUGINFLAGS)
+libextractor_previewopus_la_LIBADD = \
+ -lavutil -lavformat -lavcodec -lswscale -lavresample -lmagic $(XLIB)
+
+test_previewopus_SOURCES = \
+ test_previewopus.c
+test_previewopus_LDADD = \
+ $(top_builddir)/src/plugins/libtest.la
+
libextractor_tiff_la_SOURCES = \
tiff_extractor.c
libextractor_tiff_la_LDFLAGS = \
diff --git a/src/plugins/previewopus_extractor.c b/src/plugins/previewopus_extractor.c
@@ -0,0 +1,1195 @@
+/*
+ This file is part of libextractor.
+ 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., 59 Temple Place - Suite 330,
+ Boston, MA 02111-1307, 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>
+
+#if HAVE_LIBAVUTIL_AVUTIL_H
+#include <libavutil/avutil.h>
+#include <libavutil/audio_fifo.h>
+#include <libavutil/opt.h>
+#include <libavutil/mathematics.h>
+
+#elif HAVE_FFMPEG_AVUTIL_H
+#include <ffmpeg/avutil.h>
+#include <ffmpeg/audio_fifo.h>
+#include <ffmpeg/opt.h>
+#include <ffmpeg/mathematics.h>
+#endif
+#if HAVE_LIBAVFORMAT_AVFORMAT_H
+#include <libavformat/avformat.h>
+#elif HAVE_FFMPEG_AVFORMAT_H
+#include <ffmpeg/avformat.h>
+#endif
+#if HAVE_LIBAVCODEC_AVCODEC_H
+#include <libavcodec/avcodec.h>
+#elif HAVE_FFMPEG_AVCODEC_H
+#include <ffmpeg/avcodec.h>
+#endif
+#if HAVE_LIBSWSCALE_SWSCALE_H
+#include <libswscale/swscale.h>
+#elif HAVE_FFMPEG_SWSCALE_H
+#include <ffmpeg/swscale.h>
+#endif
+
+//TODO: Check for ffmpeg
+#if HAVE_LIBAVRESAMPLE_AVRESAMPLE_H
+#include <libavresample/avresample.h>
+#elif HAVE_FFMPEG_AVRESAMPLE_H
+#include <ffmpeg/avresample.h>
+#endif
+
+
+
+
+/**
+ * Set to 1 to enable debug output.
+ */
+#define DEBUG 1
+
+/**
+ * Set to 1 to enable a output file for testing.
+ */
+#define OUTPUT_FILE 1
+
+
+
+/**
+ * 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
+
+
+/**
+ * Global handle to MAGIC data.
+ */
+static magic_t magic;
+
+static unsigned char *buffer;
+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)
+{
+ AVIOContext *output_io_context = NULL;
+ AVStream *stream = NULL;
+ AVCodec *output_codec = NULL;
+ AVIOContext *io_ctx;
+ int error;
+
+
+
+ AVDictionary *options;
+ unsigned char *iob;
+
+ if (NULL == (iob = av_malloc (16 * 1024)))
+ return;
+ if (NULL == (io_ctx = avio_alloc_context (iob, 16 * 1024,
+ AVIO_FLAG_WRITE, NULL,
+ NULL,
+ &writePacket /* no writing */,
+ NULL)))
+ {
+ av_free (iob);
+ return;
+ }
+ if (NULL == ((*output_format_context) = avformat_alloc_context ()))
+ {
+ av_free (io_ctx);
+ return;
+ }
+ (*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
+ 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
+ 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:
+ 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)
+{
+ if (!(*frame = av_frame_alloc())) {
+ #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,
+ AVAudioResampleContext **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 = avresample_alloc_context())) {
+ #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 = avresample_open(*resample_context)) < 0) {
+ #if DEBUG
+ fprintf(stderr, "Could not open resample context\n");
+ #endif
+ avresample_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_free_packet(&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_free_packet(&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, const int in_sample, const int out_sample, const int out_linesize,
+ AVAudioResampleContext *resample_context)
+{
+ int error;
+
+ /** Convert the samples using the resampler. */
+ if ((error = avresample_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,
+ AVAudioResampleContext *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;
+ //FIX ME: 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. */
+ if (!(*frame = av_frame_alloc())) {
+ #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_free_packet(&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_free_packet(&output_packet);
+ return error;
+ }
+
+ av_free_packet(&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)
+{
+ AVPacket packet;
+ 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;
+
+
+ AVAudioResampleContext *resample_context = NULL;
+ AVAudioFifo *fifo = NULL;
+
+
+ int audio_stream_index;
+ int thumb_width;
+ int thumb_height;
+ int i;
+ int err;
+ int frame_finished;
+ 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))
+ 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;
+ }
+
+ if (NULL == (frame = avcodec_alloc_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
+ {
+ #if DEBUG
+ duration = format_ctx->duration;
+ 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);
+ frame_finished = 0;
+
+
+
+ /**
+ * 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) {
+ avresample_close(resample_context);
+ avresample_free(&resample_context);
+ }
+ if (output_codec_context)
+ avcodec_close(output_codec_context);
+
+ if (codec_ctx)
+ avcodec_close(codec_ctx);
+ if (format_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)
+{
+ unsigned int i;
+ ssize_t iret;
+ void *data;
+ const char *mime;
+
+ if (-1 == (iret = ec->read (ec->cls,
+ &data,
+ 16 * 1024)))
+ return;
+ if (NULL == (mime = magic_buffer (magic, data, iret)))
+ 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 and load magic file.
+ */
+void __attribute__ ((constructor))
+previewopus_lib_init (void)
+{
+ av_log_set_callback (&previewopus_av_log_callback);
+ av_register_all ();
+ magic = magic_open (MAGIC_MIME_TYPE);
+ if (0 != magic_load (magic, NULL))
+ {
+ /* FIXME: how to deal with errors? */
+ }
+}
+
+
+/**
+ * Destructor for the library, cleans up.
+ */
+void __attribute__ ((destructor))
+previewopus_ltdl_fini ()
+{
+ if (NULL != magic)
+ {
+ magic_close (magic);
+ magic = NULL;
+ }
+}
+
+
+/* end of previewopus_extractor.c */
