Initial commit

19 files changed, 5844 insertions, 0 deletions

diff --git a/Music/ffmpeg/doc/examples/Makefile b/Music/ffmpeg/doc/examples/Makefile
new file mode 100755
index 0000000..af38159
--- /dev/null
+++ b/Music/ffmpeg/doc/examples/Makefile
@@ -0,0 +1,46 @@
+# use pkg-config for getting CFLAGS and LDLIBS
+FFMPEG_LIBS=    libavdevice                        \
+                libavformat                        \
+                libavfilter                        \
+                libavcodec                         \
+                libswresample                      \
+                libswscale                         \
+                libavutil                          \
+
+CFLAGS += -Wall -g
+CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
+LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)
+
+EXAMPLES=       avio_dir_cmd                       \
+                avio_reading                       \
+                decoding_encoding                  \
+                demuxing_decoding                  \
+                extract_mvs                        \
+                filtering_video                    \
+                filtering_audio                    \
+                http_multiclient                   \
+                metadata                           \
+                muxing                             \
+                remuxing                           \
+                resampling_audio                   \
+                scaling_video                      \
+                transcode_aac                      \
+                transcoding                        \
+
+OBJS=$(addsuffix .o,$(EXAMPLES))
+
+# the following examples make explicit use of the math library
+avcodec:           LDLIBS += -lm
+decoding_encoding: LDLIBS += -lm
+muxing:            LDLIBS += -lm
+resampling_audio:  LDLIBS += -lm
+
+.phony: all clean-test clean
+
+all: $(OBJS) $(EXAMPLES)
+
+clean-test:
+	$(RM) test*.pgm test.h264 test.mp2 test.sw test.mpg
+
+clean: clean-test
+	$(RM) $(EXAMPLES) $(OBJS)
diff --git a/Music/ffmpeg/doc/examples/README b/Music/ffmpeg/doc/examples/README
new file mode 100755
index 0000000..c1ce619
--- /dev/null
+++ b/Music/ffmpeg/doc/examples/README
@@ -0,0 +1,23 @@
+FFmpeg examples README
+----------------------
+
+Both following use cases rely on pkg-config and make, thus make sure
+that you have them installed and working on your system.
+
+
+Method 1: build the installed examples in a generic read/write user directory
+
+Copy to a read/write user directory and just use "make", it will link
+to the libraries on your system, assuming the PKG_CONFIG_PATH is
+correctly configured.
+
+Method 2: build the examples in-tree
+
+Assuming you are in the source FFmpeg checkout directory, you need to build
+FFmpeg (no need to make install in any prefix). Then just run "make examples".
+This will build the examples using the FFmpeg build system. You can clean those
+examples using "make examplesclean"
+
+If you want to try the dedicated Makefile examples (to emulate the first
+method), go into doc/examples and run a command such as
+PKG_CONFIG_PATH=pc-uninstalled make.
diff --git a/Music/ffmpeg/doc/examples/avio_dir_cmd.c b/Music/ffmpeg/doc/examples/avio_dir_cmd.c
new file mode 100755
index 0000000..50c435c
--- /dev/null
+++ b/Music/ffmpeg/doc/examples/avio_dir_cmd.c
@@ -0,0 +1,180 @@
+/*
+ * Copyright (c) 2014 Lukasz Marek
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+#include <libavcodec/avcodec.h>
+#include <libavformat/avformat.h>
+#include <libavformat/avio.h>
+
+static const char *type_string(int type)
+{
+    switch (type) {
+    case AVIO_ENTRY_DIRECTORY:
+        return "<DIR>";
+    case AVIO_ENTRY_FILE:
+        return "<FILE>";
+    case AVIO_ENTRY_BLOCK_DEVICE:
+        return "<BLOCK DEVICE>";
+    case AVIO_ENTRY_CHARACTER_DEVICE:
+        return "<CHARACTER DEVICE>";
+    case AVIO_ENTRY_NAMED_PIPE:
+        return "<PIPE>";
+    case AVIO_ENTRY_SYMBOLIC_LINK:
+        return "<LINK>";
+    case AVIO_ENTRY_SOCKET:
+        return "<SOCKET>";
+    case AVIO_ENTRY_SERVER:
+        return "<SERVER>";
+    case AVIO_ENTRY_SHARE:
+        return "<SHARE>";
+    case AVIO_ENTRY_WORKGROUP:
+        return "<WORKGROUP>";
+    case AVIO_ENTRY_UNKNOWN:
+    default:
+        break;
+    }
+    return "<UNKNOWN>";
+}
+
+static int list_op(const char *input_dir)
+{
+    AVIODirEntry *entry = NULL;
+    AVIODirContext *ctx = NULL;
+    int cnt, ret;
+    char filemode[4], uid_and_gid[20];
+
+    if ((ret = avio_open_dir(&ctx, input_dir, NULL)) < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot open directory: %s.\n", av_err2str(ret));
+        goto fail;
+    }
+
+    cnt = 0;
+    for (;;) {
+        if ((ret = avio_read_dir(ctx, &entry)) < 0) {
+            av_log(NULL, AV_LOG_ERROR, "Cannot list directory: %s.\n", av_err2str(ret));
+            goto fail;
+        }
+        if (!entry)
+            break;
+        if (entry->filemode == -1) {
+            snprintf(filemode, 4, "???");
+        } else {
+            snprintf(filemode, 4, "%3"PRIo64, entry->filemode);
+        }
+        snprintf(uid_and_gid, 20, "%"PRId64"(%"PRId64")", entry->user_id, entry->group_id);
+        if (cnt == 0)
+            av_log(NULL, AV_LOG_INFO, "%-9s %12s %30s %10s %s %16s %16s %16s\n",
+                   "TYPE", "SIZE", "NAME", "UID(GID)", "UGO", "MODIFIED",
+                   "ACCESSED", "STATUS_CHANGED");
+        av_log(NULL, AV_LOG_INFO, "%-9s %12"PRId64" %30s %10s %s %16"PRId64" %16"PRId64" %16"PRId64"\n",
+               type_string(entry->type),
+               entry->size,
+               entry->name,
+               uid_and_gid,
+               filemode,
+               entry->modification_timestamp,
+               entry->access_timestamp,
+               entry->status_change_timestamp);
+        avio_free_directory_entry(&entry);
+        cnt++;
+    };
+
+  fail:
+    avio_close_dir(&ctx);
+    return ret;
+}
+
+static int del_op(const char *url)
+{
+    int ret = avpriv_io_delete(url);
+    if (ret < 0)
+        av_log(NULL, AV_LOG_ERROR, "Cannot delete '%s': %s.\n", url, av_err2str(ret));
+    return ret;
+}
+
+static int move_op(const char *src, const char *dst)
+{
+    int ret = avpriv_io_move(src, dst);
+    if (ret < 0)
+        av_log(NULL, AV_LOG_ERROR, "Cannot move '%s' into '%s': %s.\n", src, dst, av_err2str(ret));
+    return ret;
+}
+
+
+static void usage(const char *program_name)
+{
+    fprintf(stderr, "usage: %s OPERATION entry1 [entry2]\n"
+            "API example program to show how to manipulate resources "
+            "accessed through AVIOContext.\n"
+            "OPERATIONS:\n"
+            "list      list content of the directory\n"
+            "move      rename content in directory\n"
+            "del       delete content in directory\n",
+            program_name);
+}
+
+int main(int argc, char *argv[])
+{
+    const char *op = NULL;
+    int ret;
+
+    av_log_set_level(AV_LOG_DEBUG);
+
+    if (argc < 2) {
+        usage(argv[0]);
+        return 1;
+    }
+
+    /* register codecs and formats and other lavf/lavc components*/
+    av_register_all();
+    avformat_network_init();
+
+    op = argv[1];
+    if (strcmp(op, "list") == 0) {
+        if (argc < 3) {
+            av_log(NULL, AV_LOG_INFO, "Missing argument for list operation.\n");
+            ret = AVERROR(EINVAL);
+        } else {
+            ret = list_op(argv[2]);
+        }
+    } else if (strcmp(op, "del") == 0) {
+        if (argc < 3) {
+            av_log(NULL, AV_LOG_INFO, "Missing argument for del operation.\n");
+            ret = AVERROR(EINVAL);
+        } else {
+            ret = del_op(argv[2]);
+        }
+    } else if (strcmp(op, "move") == 0) {
+        if (argc < 4) {
+            av_log(NULL, AV_LOG_INFO, "Missing argument for move operation.\n");
+            ret = AVERROR(EINVAL);
+        } else {
+            ret = move_op(argv[2], argv[3]);
+        }
+    } else {
+        av_log(NULL, AV_LOG_INFO, "Invalid operation %s\n", op);
+        ret = AVERROR(EINVAL);
+    }
+
+    avformat_network_deinit();
+
+    return ret < 0 ? 1 : 0;
+}
diff --git a/Music/ffmpeg/doc/examples/avio_reading.c b/Music/ffmpeg/doc/examples/avio_reading.c
new file mode 100755
index 0000000..02474e9
--- /dev/null
+++ b/Music/ffmpeg/doc/examples/avio_reading.c
@@ -0,0 +1,134 @@
+/*
+ * Copyright (c) 2014 Stefano Sabatini
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+/**
+ * @file
+ * libavformat AVIOContext API example.
+ *
+ * Make libavformat demuxer access media content through a custom
+ * AVIOContext read callback.
+ * @example avio_reading.c
+ */
+
+#include <libavcodec/avcodec.h>
+#include <libavformat/avformat.h>
+#include <libavformat/avio.h>
+#include <libavutil/file.h>
+
+struct buffer_data {
+    uint8_t *ptr;
+    size_t size; ///< size left in the buffer
+};
+
+static int read_packet(void *opaque, uint8_t *buf, int buf_size)
+{
+    struct buffer_data *bd = (struct buffer_data *)opaque;
+    buf_size = FFMIN(buf_size, bd->size);
+
+    printf("ptr:%p size:%zu\n", bd->ptr, bd->size);
+
+    /* copy internal buffer data to buf */
+    memcpy(buf, bd->ptr, buf_size);
+    bd->ptr  += buf_size;
+    bd->size -= buf_size;
+
+    return buf_size;
+}
+
+int main(int argc, char *argv[])
+{
+    AVFormatContext *fmt_ctx = NULL;
+    AVIOContext *avio_ctx = NULL;
+    uint8_t *buffer = NULL, *avio_ctx_buffer = NULL;
+    size_t buffer_size, avio_ctx_buffer_size = 4096;
+    char *input_filename = NULL;
+    int ret = 0;
+    struct buffer_data bd = { 0 };
+
+    if (argc != 2) {
+        fprintf(stderr, "usage: %s input_file\n"
+                "API example program to show how to read from a custom buffer "
+                "accessed through AVIOContext.\n", argv[0]);
+        return 1;
+    }
+    input_filename = argv[1];
+
+    /* register codecs and formats and other lavf/lavc components*/
+    av_register_all();
+
+    /* slurp file content into buffer */
+    ret = av_file_map(input_filename, &buffer, &buffer_size, 0, NULL);
+    if (ret < 0)
+        goto end;
+
+    /* fill opaque structure used by the AVIOContext read callback */
+    bd.ptr  = buffer;
+    bd.size = buffer_size;
+
+    if (!(fmt_ctx = avformat_alloc_context())) {
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }
+
+    avio_ctx_buffer = av_malloc(avio_ctx_buffer_size);
+    if (!avio_ctx_buffer) {
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }
+    avio_ctx = avio_alloc_context(avio_ctx_buffer, avio_ctx_buffer_size,
+                                  0, &bd, &read_packet, NULL, NULL);
+    if (!avio_ctx) {
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }
+    fmt_ctx->pb = avio_ctx;
+
+    ret = avformat_open_input(&fmt_ctx, NULL, NULL, NULL);
+    if (ret < 0) {
+        fprintf(stderr, "Could not open input\n");
+        goto end;
+    }
+
+    ret = avformat_find_stream_info(fmt_ctx, NULL);
+    if (ret < 0) {
+        fprintf(stderr, "Could not find stream information\n");
+        goto end;
+    }
+
+    av_dump_format(fmt_ctx, 0, input_filename, 0);
+
+end:
+    avformat_close_input(&fmt_ctx);
+    /* note: the internal buffer could have changed, and be != avio_ctx_buffer */
+    if (avio_ctx) {
+        av_freep(&avio_ctx->buffer);
+        av_freep(&avio_ctx);
+    }
+    av_file_unmap(buffer, buffer_size);
+
+    if (ret < 0) {
+        fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
+        return 1;
+    }
+
+    return 0;
+}
diff --git a/Music/ffmpeg/doc/examples/decoding_encoding.c b/Music/ffmpeg/doc/examples/decoding_encoding.c
new file mode 100755
index 0000000..43a64c2
--- /dev/null
+++ b/Music/ffmpeg/doc/examples/decoding_encoding.c
@@ -0,0 +1,665 @@
+/*
+ * Copyright (c) 2001 Fabrice Bellard
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+/**
+ * @file
+ * libavcodec API use example.
+ *
+ * @example decoding_encoding.c
+ * Note that libavcodec only handles codecs (MPEG, MPEG-4, etc...),
+ * not file formats (AVI, VOB, MP4, MOV, MKV, MXF, FLV, MPEG-TS, MPEG-PS, etc...).
+ * See library 'libavformat' for the format handling
+ */
+
+#include <math.h>
+
+#include <libavutil/opt.h>
+#include <libavcodec/avcodec.h>
+#include <libavutil/channel_layout.h>
+#include <libavutil/common.h>
+#include <libavutil/imgutils.h>
+#include <libavutil/mathematics.h>
+#include <libavutil/samplefmt.h>
+
+#define INBUF_SIZE 4096
+#define AUDIO_INBUF_SIZE 20480
+#define AUDIO_REFILL_THRESH 4096
+
+/* check that a given sample format is supported by the encoder */
+static int check_sample_fmt(AVCodec *codec, enum AVSampleFormat sample_fmt)
+{
+    const enum AVSampleFormat *p = codec->sample_fmts;
+
+    while (*p != AV_SAMPLE_FMT_NONE) {
+        if (*p == sample_fmt)
+            return 1;
+        p++;
+    }
+    return 0;
+}
+
+/* just pick the highest supported samplerate */
+static int select_sample_rate(AVCodec *codec)
+{
+    const int *p;
+    int best_samplerate = 0;
+
+    if (!codec->supported_samplerates)
+        return 44100;
+
+    p = codec->supported_samplerates;
+    while (*p) {
+        best_samplerate = FFMAX(*p, best_samplerate);
+        p++;
+    }
+    return best_samplerate;
+}
+
+/* select layout with the highest channel count */
+static int select_channel_layout(AVCodec *codec)
+{
+    const uint64_t *p;
+    uint64_t best_ch_layout = 0;
+    int best_nb_channels   = 0;
+
+    if (!codec->channel_layouts)
+        return AV_CH_LAYOUT_STEREO;
+
+    p = codec->channel_layouts;
+    while (*p) {
+        int nb_channels = av_get_channel_layout_nb_channels(*p);
+
+        if (nb_channels > best_nb_channels) {
+            best_ch_layout    = *p;
+            best_nb_channels = nb_channels;
+        }
+        p++;
+    }
+    return best_ch_layout;
+}
+
+/*
+ * Audio encoding example
+ */
+static void audio_encode_example(const char *filename)
+{
+    AVCodec *codec;
+    AVCodecContext *c= NULL;
+    AVFrame *frame;
+    AVPacket pkt;
+    int i, j, k, ret, got_output;
+    int buffer_size;
+    FILE *f;
+    uint16_t *samples;
+    float t, tincr;
+
+    printf("Encode audio file %s\n", filename);
+
+    /* find the MP2 encoder */
+    codec = avcodec_find_encoder(AV_CODEC_ID_MP2);
+    if (!codec) {
+        fprintf(stderr, "Codec not found\n");
+        exit(1);
+    }
+
+    c = avcodec_alloc_context3(codec);
+    if (!c) {
+        fprintf(stderr, "Could not allocate audio codec context\n");
+        exit(1);
+    }
+
+    /* put sample parameters */
+    c->bit_rate = 64000;
+
+    /* check that the encoder supports s16 pcm input */
+    c->sample_fmt = AV_SAMPLE_FMT_S16;
+    if (!check_sample_fmt(codec, c->sample_fmt)) {
+        fprintf(stderr, "Encoder does not support sample format %s",
+                av_get_sample_fmt_name(c->sample_fmt));
+        exit(1);
+    }
+
+    /* select other audio parameters supported by the encoder */
+    c->sample_rate    = select_sample_rate(codec);
+    c->channel_layout = select_channel_layout(codec);
+    c->channels       = av_get_channel_layout_nb_channels(c->channel_layout);
+
+    /* open it */
+    if (avcodec_open2(c, codec, NULL) < 0) {
+        fprintf(stderr, "Could not open codec\n");
+        exit(1);
+    }
+
+    f = fopen(filename, "wb");
+    if (!f) {
+        fprintf(stderr, "Could not open %s\n", filename);
+        exit(1);
+    }
+
+    /* frame containing input raw audio */
+    frame = av_frame_alloc();
+    if (!frame) {
+        fprintf(stderr, "Could not allocate audio frame\n");
+        exit(1);
+    }
+
+    frame->nb_samples     = c->frame_size;
+    frame->format         = c->sample_fmt;
+    frame->channel_layout = c->channel_layout;
+
+    /* the codec gives us the frame size, in samples,
+     * we calculate the size of the samples buffer in bytes */
+    buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size,
+                                             c->sample_fmt, 0);
+    if (buffer_size < 0) {
+        fprintf(stderr, "Could not get sample buffer size\n");
+        exit(1);
+    }
+    samples = av_malloc(buffer_size);
+    if (!samples) {
+        fprintf(stderr, "Could not allocate %d bytes for samples buffer\n",
+                buffer_size);
+        exit(1);
+    }
+    /* setup the data pointers in the AVFrame */
+    ret = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
+                                   (const uint8_t*)samples, buffer_size, 0);
+    if (ret < 0) {
+        fprintf(stderr, "Could not setup audio frame\n");
+        exit(1);
+    }
+
+    /* encode a single tone sound */
+    t = 0;
+    tincr = 2 * M_PI * 440.0 / c->sample_rate;
+    for (i = 0; i < 200; i++) {
+        av_init_packet(&pkt);
+        pkt.data = NULL; // packet data will be allocated by the encoder
+        pkt.size = 0;
+
+        for (j = 0; j < c->frame_size; j++) {
+            samples[2*j] = (int)(sin(t) * 10000);
+
+            for (k = 1; k < c->channels; k++)
+                samples[2*j + k] = samples[2*j];
+            t += tincr;
+        }
+        /* encode the samples */
+        ret = avcodec_encode_audio2(c, &pkt, frame, &got_output);
+        if (ret < 0) {
+            fprintf(stderr, "Error encoding audio frame\n");
+            exit(1);
+        }
+        if (got_output) {
+            fwrite(pkt.data, 1, pkt.size, f);
+            av_packet_unref(&pkt);
+        }
+    }
+
+    /* get the delayed frames */
+    for (got_output = 1; got_output; i++) {
+        ret = avcodec_encode_audio2(c, &pkt, NULL, &got_output);
+        if (ret < 0) {
+            fprintf(stderr, "Error encoding frame\n");
+            exit(1);
+        }
+
+        if (got_output) {
+            fwrite(pkt.data, 1, pkt.size, f);
+            av_packet_unref(&pkt);
+        }
+    }
+    fclose(f);
+
+    av_freep(&samples);
+    av_frame_free(&frame);
+    avcodec_close(c);
+    av_free(c);
+}
+
+/*
+ * Audio decoding.
+ */
+static void audio_decode_example(const char *outfilename, const char *filename)
+{
+    AVCodec *codec;
+    AVCodecContext *c= NULL;
+    int len;
+    FILE *f, *outfile;
+    uint8_t inbuf[AUDIO_INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
+    AVPacket avpkt;
+    AVFrame *decoded_frame = NULL;
+
+    av_init_packet(&avpkt);
+
+    printf("Decode audio file %s to %s\n", filename, outfilename);
+
+    /* find the MPEG audio decoder */
+    codec = avcodec_find_decoder(AV_CODEC_ID_MP2);
+    if (!codec) {
+        fprintf(stderr, "Codec not found\n");
+        exit(1);
+    }
+
+    c = avcodec_alloc_context3(codec);
+    if (!c) {
+        fprintf(stderr, "Could not allocate audio codec context\n");
+        exit(1);
+    }
+
+    /* open it */
+    if (avcodec_open2(c, codec, NULL) < 0) {
+        fprintf(stderr, "Could not open codec\n");
+        exit(1);
+    }
+
+    f = fopen(filename, "rb");
+    if (!f) {
+        fprintf(stderr, "Could not open %s\n", filename);
+        exit(1);
+    }
+    outfile = fopen(outfilename, "wb");
+    if (!outfile) {
+        av_free(c);
+        exit(1);
+    }
+
+    /* decode until eof */
+    avpkt.data = inbuf;
+    avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
+
+    while (avpkt.size > 0) {
+        int i, ch;
+        int got_frame = 0;
+
+        if (!decoded_frame) {
+            if (!(decoded_frame = av_frame_alloc())) {
+                fprintf(stderr, "Could not allocate audio frame\n");
+                exit(1);
+            }
+        }
+
+        len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt);
+        if (len < 0) {
+            fprintf(stderr, "Error while decoding\n");
+            exit(1);
+        }
+        if (got_frame) {
+            /* if a frame has been decoded, output it */
+            int data_size = av_get_bytes_per_sample(c->sample_fmt);
+            if (data_size < 0) {
+                /* This should not occur, checking just for paranoia */
+                fprintf(stderr, "Failed to calculate data size\n");
+                exit(1);
+            }
+            for (i=0; i<decoded_frame->nb_samples; i++)
+                for (ch=0; ch<c->channels; ch++)
+                    fwrite(decoded_frame->data[ch] + data_size*i, 1, data_size, outfile);
+        }
+        avpkt.size -= len;
+        avpkt.data += len;
+        avpkt.dts =
+        avpkt.pts = AV_NOPTS_VALUE;
+        if (avpkt.size < AUDIO_REFILL_THRESH) {
+            /* Refill the input buffer, to avoid trying to decode
+             * incomplete frames. Instead of this, one could also use
+             * a parser, or use a proper container format through
+             * libavformat. */
+            memmove(inbuf, avpkt.data, avpkt.size);
+            avpkt.data = inbuf;
+            len = fread(avpkt.data + avpkt.size, 1,
+                        AUDIO_INBUF_SIZE - avpkt.size, f);
+            if (len > 0)
+                avpkt.size += len;
+        }
+    }
+
+    fclose(outfile);
+    fclose(f);
+
+    avcodec_close(c);
+    av_free(c);
+    av_frame_free(&decoded_frame);
+}
+
+/*
+ * Video encoding example
+ */
+static void video_encode_example(const char *filename, int codec_id)
+{
+    AVCodec *codec;
+    AVCodecContext *c= NULL;
+    int i, ret, x, y, got_output;
+    FILE *f;
+    AVFrame *frame;
+    AVPacket pkt;
+    uint8_t endcode[] = { 0, 0, 1, 0xb7 };
+
+    printf("Encode video file %s\n", filename);
+
+    /* find the video encoder */
+    codec = avcodec_find_encoder(codec_id);
+    if (!codec) {
+        fprintf(stderr, "Codec not found\n");
+        exit(1);
+    }
+
+    c = avcodec_alloc_context3(codec);
+    if (!c) {
+        fprintf(stderr, "Could not allocate video codec context\n");
+        exit(1);
+    }
+
+    /* put sample parameters */
+    c->bit_rate = 400000;
+    /* resolution must be a multiple of two */
+    c->width = 352;
+    c->height = 288;
+    /* frames per second */
+    c->time_base = (AVRational){1,25};
+    /* emit one intra frame every ten frames
+     * check frame pict_type before passing frame
+     * to encoder, if frame->pict_type is AV_PICTURE_TYPE_I
+     * then gop_size is ignored and the output of encoder
+     * will always be I frame irrespective to gop_size
+     */
+    c->gop_size = 10;
+    c->max_b_frames = 1;
+    c->pix_fmt = AV_PIX_FMT_YUV420P;
+
+    if (codec_id == AV_CODEC_ID_H264)
+        av_opt_set(c->priv_data, "preset", "slow", 0);
+
+    /* open it */
+    if (avcodec_open2(c, codec, NULL) < 0) {
+        fprintf(stderr, "Could not open codec\n");
+        exit(1);
+    }
+
+    f = fopen(filename, "wb");
+    if (!f) {
+        fprintf(stderr, "Could not open %s\n", filename);
+        exit(1);
+    }
+
+    frame = av_frame_alloc();
+    if (!frame) {
+        fprintf(stderr, "Could not allocate video frame\n");
+        exit(1);
+    }
+    frame->format = c->pix_fmt;
+    frame->width  = c->width;
+    frame->height = c->height;
+
+    /* the image can be allocated by any means and av_image_alloc() is
+     * just the most convenient way if av_malloc() is to be used */
+    ret = av_image_alloc(frame->data, frame->linesize, c->width, c->height,
+                         c->pix_fmt, 32);
+    if (ret < 0) {
+        fprintf(stderr, "Could not allocate raw picture buffer\n");
+        exit(1);
+    }
+
+    /* encode 1 second of video */
+    for (i = 0; i < 25; i++) {
+        av_init_packet(&pkt);
+        pkt.data = NULL;    // packet data will be allocated by the encoder
+        pkt.size = 0;
+
+        fflush(stdout);
+        /* prepare a dummy image */
+        /* Y */
+        for (y = 0; y < c->height; y++) {
+            for (x = 0; x < c->width; x++) {
+                frame->data[0][y * frame->linesize[0] + x] = x + y + i * 3;
+            }
+        }
+
+        /* Cb and Cr */
+        for (y = 0; y < c->height/2; y++) {
+            for (x = 0; x < c->width/2; x++) {
+                frame->data[1][y * frame->linesize[1] + x] = 128 + y + i * 2;
+                frame->data[2][y * frame->linesize[2] + x] = 64 + x + i * 5;
+            }
+        }
+
+        frame->pts = i;
+
+        /* encode the image */
+        ret = avcodec_encode_video2(c, &pkt, frame, &got_output);
+        if (ret < 0) {
+            fprintf(stderr, "Error encoding frame\n");
+            exit(1);
+        }
+
+        if (got_output) {
+            printf("Write frame %3d (size=%5d)\n", i, pkt.size);
+            fwrite(pkt.data, 1, pkt.size, f);
+            av_packet_unref(&pkt);
+        }
+    }
+
+    /* get the delayed frames */
+    for (got_output = 1; got_output; i++) {
+        fflush(stdout);
+
+        ret = avcodec_encode_video2(c, &pkt, NULL, &got_output);
+        if (ret < 0) {
+            fprintf(stderr, "Error encoding frame\n");
+            exit(1);
+        }
+
+        if (got_output) {
+            printf("Write frame %3d (size=%5d)\n", i, pkt.size);
+            fwrite(pkt.data, 1, pkt.size, f);
+            av_packet_unref(&pkt);
+        }
+    }
+
+    /* add sequence end code to have a real MPEG file */
+    fwrite(endcode, 1, sizeof(endcode), f);
+    fclose(f);
+
+    avcodec_close(c);
+    av_free(c);
+    av_freep(&frame->data[0]);
+    av_frame_free(&frame);
+    printf("\n");
+}
+
+/*
+ * Video decoding example
+ */
+
+static void pgm_save(unsigned char *buf, int wrap, int xsize, int ysize,
+                     char *filename)
+{
+    FILE *f;
+    int i;
+
+    f = fopen(filename,"w");
+    fprintf(f, "P5\n%d %d\n%d\n", xsize, ysize, 255);
+    for (i = 0; i < ysize; i++)
+        fwrite(buf + i * wrap, 1, xsize, f);
+    fclose(f);
+}
+
+static int decode_write_frame(const char *outfilename, AVCodecContext *avctx,
+                              AVFrame *frame, int *frame_count, AVPacket *pkt, int last)
+{
+    int len, got_frame;
+    char buf[1024];
+
+    len = avcodec_decode_video2(avctx, frame, &got_frame, pkt);
+    if (len < 0) {
+        fprintf(stderr, "Error while decoding frame %d\n", *frame_count);
+        return len;
+    }
+    if (got_frame) {
+        printf("Saving %sframe %3d\n", last ? "last " : "", *frame_count);
+        fflush(stdout);
+
+        /* the picture is allocated by the decoder, no need to free it */
+        snprintf(buf, sizeof(buf), outfilename, *frame_count);
+        pgm_save(frame->data[0], frame->linesize[0],
+                 frame->width, frame->height, buf);
+        (*frame_count)++;
+    }
+    if (pkt->data) {
+        pkt->size -= len;
+        pkt->data += len;
+    }
+    return 0;
+}
+
+static void video_decode_example(const char *outfilename, const char *filename)
+{
+    AVCodec *codec;
+    AVCodecContext *c= NULL;
+    int frame_count;
+    FILE *f;
+    AVFrame *frame;
+    uint8_t inbuf[INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
+    AVPacket avpkt;
+
+    av_init_packet(&avpkt);
+
+    /* set end of buffer to 0 (this ensures that no overreading happens for damaged MPEG streams) */
+    memset(inbuf + INBUF_SIZE, 0, AV_INPUT_BUFFER_PADDING_SIZE);
+
+    printf("Decode video file %s to %s\n", filename, outfilename);
+
+    /* find the MPEG-1 video decoder */
+    codec = avcodec_find_decoder(AV_CODEC_ID_MPEG1VIDEO);
+    if (!codec) {
+        fprintf(stderr, "Codec not found\n");
+        exit(1);
+    }
+
+    c = avcodec_alloc_context3(codec);
+    if (!c) {
+        fprintf(stderr, "Could not allocate video codec context\n");
+        exit(1);
+    }
+
+    if (codec->capabilities & AV_CODEC_CAP_TRUNCATED)
+        c->flags |= AV_CODEC_FLAG_TRUNCATED; // we do not send complete frames
+
+    /* For some codecs, such as msmpeg4 and mpeg4, width and height
+       MUST be initialized there because this information is not
+       available in the bitstream. */
+
+    /* open it */
+    if (avcodec_open2(c, codec, NULL) < 0) {
+        fprintf(stderr, "Could not open codec\n");
+        exit(1);
+    }
+
+    f = fopen(filename, "rb");
+    if (!f) {
+        fprintf(stderr, "Could not open %s\n", filename);
+        exit(1);
+    }
+
+    frame = av_frame_alloc();
+    if (!frame) {
+        fprintf(stderr, "Could not allocate video frame\n");
+        exit(1);
+    }
+
+    frame_count = 0;
+    for (;;) {
+        avpkt.size = fread(inbuf, 1, INBUF_SIZE, f);
+        if (avpkt.size == 0)
+            break;
+
+        /* NOTE1: some codecs are stream based (mpegvideo, mpegaudio)
+           and this is the only method to use them because you cannot
+           know the compressed data size before analysing it.
+
+           BUT some other codecs (msmpeg4, mpeg4) are inherently frame
+           based, so you must call them with all the data for one
+           frame exactly. You must also initialize 'width' and
+           'height' before initializing them. */
+
+        /* NOTE2: some codecs allow the raw parameters (frame size,
+           sample rate) to be changed at any frame. We handle this, so
+           you should also take care of it */
+
+        /* here, we use a stream based decoder (mpeg1video), so we
+           feed decoder and see if it could decode a frame */
+        avpkt.data = inbuf;
+        while (avpkt.size > 0)
+            if (decode_write_frame(outfilename, c, frame, &frame_count, &avpkt, 0) < 0)
+                exit(1);
+    }
+
+    /* Some codecs, such as MPEG, transmit the I- and P-frame with a
+       latency of one frame. You must do the following to have a
+       chance to get the last frame of the video. */
+    avpkt.data = NULL;
+    avpkt.size = 0;
+    decode_write_frame(outfilename, c, frame, &frame_count, &avpkt, 1);
+
+    fclose(f);
+
+    avcodec_close(c);
+    av_free(c);
+    av_frame_free(&frame);
+    printf("\n");
+}
+
+int main(int argc, char **argv)
+{
+    const char *output_type;
+
+    /* register all the codecs */
+    avcodec_register_all();
+
+    if (argc < 2) {
+        printf("usage: %s output_type\n"
+               "API example program to decode/encode a media stream with libavcodec.\n"
+               "This program generates a synthetic stream and encodes it to a file\n"
+               "named test.h264, test.mp2 or test.mpg depending on output_type.\n"
+               "The encoded stream is then decoded and written to a raw data output.\n"
+               "output_type must be chosen between 'h264', 'mp2', 'mpg'.\n",
+               argv[0]);
+        return 1;
+    }
+    output_type = argv[1];
+
+    if (!strcmp(output_type, "h264")) {
+        video_encode_example("test.h264", AV_CODEC_ID_H264);
+    } else if (!strcmp(output_type, "mp2")) {
+        audio_encode_example("test.mp2");
+        audio_decode_example("test.pcm", "test.mp2");
+    } else if (!strcmp(output_type, "mpg")) {
+        video_encode_example("test.mpg", AV_CODEC_ID_MPEG1VIDEO);
+        video_decode_example("test%02d.pgm", "test.mpg");
+    } else {
+        fprintf(stderr, "Invalid output type '%s', choose between 'h264', 'mp2', or 'mpg'\n",
+                output_type);
+        return 1;
+    }
+
+    return 0;
+}
diff --git a/Music/ffmpeg/doc/examples/demuxing_decoding.c b/Music/ffmpeg/doc/examples/demuxing_decoding.c
new file mode 100755
index 0000000..49fb6af
--- /dev/null
+++ b/Music/ffmpeg/doc/examples/demuxing_decoding.c
@@ -0,0 +1,394 @@
+/*
+ * Copyright (c) 2012 Stefano Sabatini
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+/**
+ * @file
+ * Demuxing and decoding example.
+ *
+ * Show how to use the libavformat and libavcodec API to demux and
+ * decode audio and video data.
+ * @example demuxing_decoding.c
+ */
+
+#include <libavutil/imgutils.h>
+#include <libavutil/samplefmt.h>
+#include <libavutil/timestamp.h>
+#include <libavformat/avformat.h>
+
+static AVFormatContext *fmt_ctx = NULL;
+static AVCodecContext *video_dec_ctx = NULL, *audio_dec_ctx;
+static int width, height;
+static enum AVPixelFormat pix_fmt;
+static AVStream *video_stream = NULL, *audio_stream = NULL;
+static const char *src_filename = NULL;
+static const char *video_dst_filename = NULL;
+static const char *audio_dst_filename = NULL;
+static FILE *video_dst_file = NULL;
+static FILE *audio_dst_file = NULL;
+
+static uint8_t *video_dst_data[4] = {NULL};
+static int      video_dst_linesize[4];
+static int video_dst_bufsize;
+
+static int video_stream_idx = -1, audio_stream_idx = -1;
+static AVFrame *frame = NULL;
+static AVPacket pkt;
+static int video_frame_count = 0;
+static int audio_frame_count = 0;
+
+/* Enable or disable frame reference counting. You are not supposed to support
+ * both paths in your application but pick the one most appropriate to your
+ * needs. Look for the use of refcount in this example to see what are the
+ * differences of API usage between them. */
+static int refcount = 0;
+
+static int decode_packet(int *got_frame, int cached)
+{
+    int ret = 0;
+    int decoded = pkt.size;
+
+    *got_frame = 0;
+
+    if (pkt.stream_index == video_stream_idx) {
+        /* decode video frame */
+        ret = avcodec_decode_video2(video_dec_ctx, frame, got_frame, &pkt);
+        if (ret < 0) {
+            fprintf(stderr, "Error decoding video frame (%s)\n", av_err2str(ret));
+            return ret;
+        }
+
+        if (*got_frame) {
+
+            if (frame->width != width || frame->height != height ||
+                frame->format != pix_fmt) {
+                /* To handle this change, one could call av_image_alloc again and
+                 * decode the following frames into another rawvideo file. */
+                fprintf(stderr, "Error: Width, height and pixel format have to be "
+                        "constant in a rawvideo file, but the width, height or "
+                        "pixel format of the input video changed:\n"
+                        "old: width = %d, height = %d, format = %s\n"
+                        "new: width = %d, height = %d, format = %s\n",
+                        width, height, av_get_pix_fmt_name(pix_fmt),
+                        frame->width, frame->height,
+                        av_get_pix_fmt_name(frame->format));
+                return -1;
+            }
+
+            printf("video_frame%s n:%d coded_n:%d pts:%s\n",
+                   cached ? "(cached)" : "",
+                   video_frame_count++, frame->coded_picture_number,
+                   av_ts2timestr(frame->pts, &video_dec_ctx->time_base));
+
+            /* copy decoded frame to destination buffer:
+             * this is required since rawvideo expects non aligned data */
+            av_image_copy(video_dst_data, video_dst_linesize,
+                          (const uint8_t **)(frame->data), frame->linesize,
+                          pix_fmt, width, height);
+
+            /* write to rawvideo file */
+            fwrite(video_dst_data[0], 1, video_dst_bufsize, video_dst_file);
+        }
+    } else if (pkt.stream_index == audio_stream_idx) {
+        /* decode audio frame */
+        ret = avcodec_decode_audio4(audio_dec_ctx, frame, got_frame, &pkt);
+        if (ret < 0) {
+            fprintf(stderr, "Error decoding audio frame (%s)\n", av_err2str(ret));
+            return ret;
+        }
+        /* Some audio decoders decode only part of the packet, and have to be
+         * called again with the remainder of the packet data.
+         * Sample: fate-suite/lossless-audio/luckynight-partial.shn
+         * Also, some decoders might over-read the packet. */
+        decoded = FFMIN(ret, pkt.size);
+
+        if (*got_frame) {
+            size_t unpadded_linesize = frame->nb_samples * av_get_bytes_per_sample(frame->format);
+            printf("audio_frame%s n:%d nb_samples:%d pts:%s\n",
+                   cached ? "(cached)" : "",
+                   audio_frame_count++, frame->nb_samples,
+                   av_ts2timestr(frame->pts, &audio_dec_ctx->time_base));
+
+            /* Write the raw audio data samples of the first plane. This works
+             * fine for packed formats (e.g. AV_SAMPLE_FMT_S16). However,
+             * most audio decoders output planar audio, which uses a separate
+             * plane of audio samples for each channel (e.g. AV_SAMPLE_FMT_S16P).
+             * In other words, this code will write only the first audio channel
+             * in these cases.
+             * You should use libswresample or libavfilter to convert the frame
+             * to packed data. */
+            fwrite(frame->extended_data[0], 1, unpadded_linesize, audio_dst_file);
+        }
+    }
+
+    /* If we use frame reference counting, we own the data and need
+     * to de-reference it when we don't use it anymore */
+    if (*got_frame && refcount)
+        av_frame_unref(frame);
+
+    return decoded;
+}
+
+static int open_codec_context(int *stream_idx,
+                              AVCodecContext **dec_ctx, AVFormatContext *fmt_ctx, enum AVMediaType type)
+{
+    int ret, stream_index;
+    AVStream *st;
+    AVCodec *dec = NULL;
+    AVDictionary *opts = NULL;
+
+    ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0);
+    if (ret < 0) {
+        fprintf(stderr, "Could not find %s stream in input file '%s'\n",
+                av_get_media_type_string(type), src_filename);
+        return ret;
+    } else {
+        stream_index = ret;
+        st = fmt_ctx->streams[stream_index];
+
+        /* find decoder for the stream */
+        dec = avcodec_find_decoder(st->codecpar->codec_id);
+        if (!dec) {
+            fprintf(stderr, "Failed to find %s codec\n",
+                    av_get_media_type_string(type));
+            return AVERROR(EINVAL);
+        }
+
+        /* Allocate a codec context for the decoder */
+        *dec_ctx = avcodec_alloc_context3(dec);
+        if (!*dec_ctx) {
+            fprintf(stderr, "Failed to allocate the %s codec context\n",
+                    av_get_media_type_string(type));
+            return AVERROR(ENOMEM);
+        }
+
+        /* Copy codec parameters from input stream to output codec context */
+        if ((ret = avcodec_parameters_to_context(*dec_ctx, st->codecpar)) < 0) {
+            fprintf(stderr, "Failed to copy %s codec parameters to decoder context\n",
+                    av_get_media_type_string(type));
+            return ret;
+        }
+
+        /* Init the decoders, with or without reference counting */
+        av_dict_set(&opts, "refcounted_frames", refcount ? "1" : "0", 0);
+        if ((ret = avcodec_open2(*dec_ctx, dec, &opts)) < 0) {
+            fprintf(stderr, "Failed to open %s codec\n",
+                    av_get_media_type_string(type));
+            return ret;
+        }
+        *stream_idx = stream_index;
+    }
+
+    return 0;
+}
+
+static int get_format_from_sample_fmt(const char **fmt,
+                                      enum AVSampleFormat sample_fmt)
+{
+    int i;
+    struct sample_fmt_entry {
+        enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
+    } sample_fmt_entries[] = {
+        { AV_SAMPLE_FMT_U8,  "u8",    "u8"    },
+        { AV_SAMPLE_FMT_S16, "s16be", "s16le" },
+        { AV_SAMPLE_FMT_S32, "s32be", "s32le" },
+        { AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
+        { AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
+    };
+    *fmt = NULL;
+
+    for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
+        struct sample_fmt_entry *entry = &sample_fmt_entries[i];
+        if (sample_fmt == entry->sample_fmt) {
+            *fmt = AV_NE(entry->fmt_be, entry->fmt_le);
+            return 0;
+        }
+    }
+
+    fprintf(stderr,
+            "sample format %s is not supported as output format\n",
+            av_get_sample_fmt_name(sample_fmt));
+    return -1;
+}
+
+int main (int argc, char **argv)
+{
+    int ret = 0, got_frame;
+
+    if (argc != 4 && argc != 5) {
+        fprintf(stderr, "usage: %s [-refcount] input_file video_output_file audio_output_file\n"
+                "API example program to show how to read frames from an input file.\n"
+                "This program reads frames from a file, decodes them, and writes decoded\n"
+                "video frames to a rawvideo file named video_output_file, and decoded\n"
+                "audio frames to a rawaudio file named audio_output_file.\n\n"
+                "If the -refcount option is specified, the program use the\n"
+                "reference counting frame system which allows keeping a copy of\n"
+                "the data for longer than one decode call.\n"
+                "\n", argv[0]);
+        exit(1);
+    }
+    if (argc == 5 && !strcmp(argv[1], "-refcount")) {
+        refcount = 1;
+        argv++;
+    }
+    src_filename = argv[1];
+    video_dst_filename = argv[2];
+    audio_dst_filename = argv[3];
+
+    /* register all formats and codecs */
+    av_register_all();
+
+    /* open input file, and allocate format context */
+    if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0) {
+        fprintf(stderr, "Could not open source file %s\n", src_filename);
+        exit(1);
+    }
+
+    /* retrieve stream information */
+    if (avformat_find_stream_info(fmt_ctx, NULL) < 0) {
+        fprintf(stderr, "Could not find stream information\n");
+        exit(1);
+    }
+
+    if (open_codec_context(&video_stream_idx, &video_dec_ctx, fmt_ctx, AVMEDIA_TYPE_VIDEO) >= 0) {
+        video_stream = fmt_ctx->streams[video_stream_idx];
+
+        video_dst_file = fopen(video_dst_filename, "wb");
+        if (!video_dst_file) {
+            fprintf(stderr, "Could not open destination file %s\n", video_dst_filename);
+            ret = 1;
+            goto end;
+        }
+
+        /* allocate image where the decoded image will be put */
+        width = video_dec_ctx->width;
+        height = video_dec_ctx->height;
+        pix_fmt = video_dec_ctx->pix_fmt;
+        ret = av_image_alloc(video_dst_data, video_dst_linesize,
+                             width, height, pix_fmt, 1);
+        if (ret < 0) {
+            fprintf(stderr, "Could not allocate raw video buffer\n");
+            goto end;
+        }
+        video_dst_bufsize = ret;
+    }
+
+    if (open_codec_context(&audio_stream_idx, &audio_dec_ctx, fmt_ctx, AVMEDIA_TYPE_AUDIO) >= 0) {
+        audio_stream = fmt_ctx->streams[audio_stream_idx];
+        audio_dst_file = fopen(audio_dst_filename, "wb");
+        if (!audio_dst_file) {
+            fprintf(stderr, "Could not open destination file %s\n", audio_dst_filename);
+            ret = 1;
+            goto end;
+        }
+    }
+
+    /* dump input information to stderr */
+    av_dump_format(fmt_ctx, 0, src_filename, 0);
+
+    if (!audio_stream && !video_stream) {
+        fprintf(stderr, "Could not find audio or video stream in the input, aborting\n");
+        ret = 1;
+        goto end;
+    }
+
+    frame = av_frame_alloc();
+    if (!frame) {
+        fprintf(stderr, "Could not allocate frame\n");
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }
+
+    /* initialize packet, set data to NULL, let the demuxer fill it */
+    av_init_packet(&pkt);
+    pkt.data = NULL;
+    pkt.size = 0;
+
+    if (video_stream)
+        printf("Demuxing video from file '%s' into '%s'\n", src_filename, video_dst_filename);
+    if (audio_stream)
+        printf("Demuxing audio from file '%s' into '%s'\n", src_filename, audio_dst_filename);
+
+    /* read frames from the file */
+    while (av_read_frame(fmt_ctx, &pkt) >= 0) {
+        AVPacket orig_pkt = pkt;
+        do {
+            ret = decode_packet(&got_frame, 0);
+            if (ret < 0)
+                break;
+            pkt.data += ret;
+            pkt.size -= ret;
+        } while (pkt.size > 0);
+        av_packet_unref(&orig_pkt);
+    }
+
+    /* flush cached frames */
+    pkt.data = NULL;
+    pkt.size = 0;
+    do {
+        decode_packet(&got_frame, 1);
+    } while (got_frame);
+
+    printf("Demuxing succeeded.\n");
+
+    if (video_stream) {
+        printf("Play the output video file with the command:\n"
+               "ffplay -f rawvideo -pix_fmt %s -video_size %dx%d %s\n",
+               av_get_pix_fmt_name(pix_fmt), width, height,
+               video_dst_filename);
+    }
+
+    if (audio_stream) {
+        enum AVSampleFormat sfmt = audio_dec_ctx->sample_fmt;
+        int n_channels = audio_dec_ctx->channels;
+        const char *fmt;
+
+        if (av_sample_fmt_is_planar(sfmt)) {
+            const char *packed = av_get_sample_fmt_name(sfmt);
+            printf("Warning: the sample format the decoder produced is planar "
+                   "(%s). This example will output the first channel only.\n",
+                   packed ? packed : "?");
+            sfmt = av_get_packed_sample_fmt(sfmt);
+            n_channels = 1;
+        }
+
+        if ((ret = get_format_from_sample_fmt(&fmt, sfmt)) < 0)
+            goto end;
+
+        printf("Play the output audio file with the command:\n"
+               "ffplay -f %s -ac %d -ar %d %s\n",
+               fmt, n_channels, audio_dec_ctx->sample_rate,
+               audio_dst_filename);
+    }
+
+end:
+    avcodec_free_context(&video_dec_ctx);
+    avcodec_free_context(&audio_dec_ctx);
+    avformat_close_input(&fmt_ctx);
+    if (video_dst_file)
+        fclose(video_dst_file);
+    if (audio_dst_file)
+        fclose(audio_dst_file);
+    av_frame_free(&frame);
+    av_free(video_dst_data[0]);
+
+    return ret < 0;
+}
diff --git a/Music/ffmpeg/doc/examples/extract_mvs.c b/Music/ffmpeg/doc/examples/extract_mvs.c
new file mode 100755
index 0000000..975189c
--- /dev/null
+++ b/Music/ffmpeg/doc/examples/extract_mvs.c
@@ -0,0 +1,185 @@
+/*
+ * Copyright (c) 2012 Stefano Sabatini
+ * Copyright (c) 2014 Clément Bœsch
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+#include <libavutil/motion_vector.h>
+#include <libavformat/avformat.h>
+
+static AVFormatContext *fmt_ctx = NULL;
+static AVCodecContext *video_dec_ctx = NULL;
+static AVStream *video_stream = NULL;
+static const char *src_filename = NULL;
+
+static int video_stream_idx = -1;
+static AVFrame *frame = NULL;
+static AVPacket pkt;
+static int video_frame_count = 0;
+
+static int decode_packet(int *got_frame, int cached)
+{
+    int decoded = pkt.size;
+
+    *got_frame = 0;
+
+    if (pkt.stream_index == video_stream_idx) {
+        int ret = avcodec_decode_video2(video_dec_ctx, frame, got_frame, &pkt);
+        if (ret < 0) {
+            fprintf(stderr, "Error decoding video frame (%s)\n", av_err2str(ret));
+            return ret;
+        }
+
+        if (*got_frame) {
+            int i;
+            AVFrameSideData *sd;
+
+            video_frame_count++;
+            sd = av_frame_get_side_data(frame, AV_FRAME_DATA_MOTION_VECTORS);
+            if (sd) {
+                const AVMotionVector *mvs = (const AVMotionVector *)sd->data;
+                for (i = 0; i < sd->size / sizeof(*mvs); i++) {
+                    const AVMotionVector *mv = &mvs[i];
+                    printf("%d,%2d,%2d,%2d,%4d,%4d,%4d,%4d,0x%"PRIx64"\n",
+                           video_frame_count, mv->source,
+                           mv->w, mv->h, mv->src_x, mv->src_y,
+                           mv->dst_x, mv->dst_y, mv->flags);
+                }
+            }
+        }
+    }
+
+    return decoded;
+}
+
+static int open_codec_context(int *stream_idx,
+                              AVFormatContext *fmt_ctx, enum AVMediaType type)
+{
+    int ret;
+    AVStream *st;
+    AVCodecContext *dec_ctx = NULL;
+    AVCodec *dec = NULL;
+    AVDictionary *opts = NULL;
+
+    ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0);
+    if (ret < 0) {
+        fprintf(stderr, "Could not find %s stream in input file '%s'\n",
+                av_get_media_type_string(type), src_filename);
+        return ret;
+    } else {
+        *stream_idx = ret;
+        st = fmt_ctx->streams[*stream_idx];
+
+        /* find decoder for the stream */
+        dec_ctx = st->codec;
+        dec = avcodec_find_decoder(dec_ctx->codec_id);
+        if (!dec) {
+            fprintf(stderr, "Failed to find %s codec\n",
+                    av_get_media_type_string(type));
+            return AVERROR(EINVAL);
+        }
+
+        /* Init the video decoder */
+        av_dict_set(&opts, "flags2", "+export_mvs", 0);
+        if ((ret = avcodec_open2(dec_ctx, dec, &opts)) < 0) {
+            fprintf(stderr, "Failed to open %s codec\n",
+                    av_get_media_type_string(type));
+            return ret;
+        }
+    }
+
+    return 0;
+}
+
+int main(int argc, char **argv)
+{
+    int ret = 0, got_frame;
+
+    if (argc != 2) {
+        fprintf(stderr, "Usage: %s <video>\n", argv[0]);
+        exit(1);
+    }
+    src_filename = argv[1];
+
+    av_register_all();
+
+    if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0) {
+        fprintf(stderr, "Could not open source file %s\n", src_filename);
+        exit(1);
+    }
+
+    if (avformat_find_stream_info(fmt_ctx, NULL) < 0) {
+        fprintf(stderr, "Could not find stream information\n");
+        exit(1);
+    }
+
+    if (open_codec_context(&video_stream_idx, fmt_ctx, AVMEDIA_TYPE_VIDEO) >= 0) {
+        video_stream = fmt_ctx->streams[video_stream_idx];
+        video_dec_ctx = video_stream->codec;
+    }
+
+    av_dump_format(fmt_ctx, 0, src_filename, 0);
+
+    if (!video_stream) {
+        fprintf(stderr, "Could not find video stream in the input, aborting\n");
+        ret = 1;
+        goto end;
+    }
+
+    frame = av_frame_alloc();
+    if (!frame) {
+        fprintf(stderr, "Could not allocate frame\n");
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }
+
+    printf("framenum,source,blockw,blockh,srcx,srcy,dstx,dsty,flags\n");
+
+    /* initialize packet, set data to NULL, let the demuxer fill it */
+    av_init_packet(&pkt);
+    pkt.data = NULL;
+    pkt.size = 0;
+
+    /* read frames from the file */
+    while (av_read_frame(fmt_ctx, &pkt) >= 0) {
+        AVPacket orig_pkt = pkt;
+        do {
+            ret = decode_packet(&got_frame, 0);
+            if (ret < 0)
+                break;
+            pkt.data += ret;
+            pkt.size -= ret;
+        } while (pkt.size > 0);
+        av_packet_unref(&orig_pkt);
+    }
+
+    /* flush cached frames */
+    pkt.data = NULL;
+    pkt.size = 0;
+    do {
+        decode_packet(&got_frame, 1);
+    } while (got_frame);
+
+end:
+    avcodec_close(video_dec_ctx);
+    avformat_close_input(&fmt_ctx);
+    av_frame_free(&frame);
+    return ret < 0;
+}
diff --git a/Music/ffmpeg/doc/examples/filter_audio.c b/Music/ffmpeg/doc/examples/filter_audio.c
new file mode 100755
index 0000000..01761dc
--- /dev/null
+++ b/Music/ffmpeg/doc/examples/filter_audio.c
@@ -0,0 +1,365 @@
+/*
+ * copyright (c) 2013 Andrew Kelley
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg 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
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * libavfilter API usage example.
+ *
+ * @example filter_audio.c
+ * This example will generate a sine wave audio,
+ * pass it through a simple filter chain, and then compute the MD5 checksum of
+ * the output data.
+ *
+ * The filter chain it uses is:
+ * (input) -> abuffer -> volume -> aformat -> abuffersink -> (output)
+ *
+ * abuffer: This provides the endpoint where you can feed the decoded samples.
+ * volume: In this example we hardcode it to 0.90.
+ * aformat: This converts the samples to the samplefreq, channel layout,
+ *          and sample format required by the audio device.
+ * abuffersink: This provides the endpoint where you can read the samples after
+ *              they have passed through the filter chain.
+ */
+
+#include <inttypes.h>
+#include <math.h>
+#include <stdio.h>
+#include <stdlib.h>
+
+#include "libavutil/channel_layout.h"
+#include "libavutil/md5.h"
+#include "libavutil/mem.h"
+#include "libavutil/opt.h"
+#include "libavutil/samplefmt.h"
+
+#include "libavfilter/avfilter.h"
+#include "libavfilter/buffersink.h"
+#include "libavfilter/buffersrc.h"
+
+#define INPUT_SAMPLERATE     48000
+#define INPUT_FORMAT         AV_SAMPLE_FMT_FLTP
+#define INPUT_CHANNEL_LAYOUT AV_CH_LAYOUT_5POINT0
+
+#define VOLUME_VAL 0.90
+
+static int init_filter_graph(AVFilterGraph **graph, AVFilterContext **src,
+                             AVFilterContext **sink)
+{
+    AVFilterGraph *filter_graph;
+    AVFilterContext *abuffer_ctx;
+    AVFilter        *abuffer;
+    AVFilterContext *volume_ctx;
+    AVFilter        *volume;
+    AVFilterContext *aformat_ctx;
+    AVFilter        *aformat;
+    AVFilterContext *abuffersink_ctx;
+    AVFilter        *abuffersink;
+
+    AVDictionary *options_dict = NULL;
+    uint8_t options_str[1024];
+    uint8_t ch_layout[64];
+
+    int err;
+
+    /* Create a new filtergraph, which will contain all the filters. */
+    filter_graph = avfilter_graph_alloc();
+    if (!filter_graph) {
+        fprintf(stderr, "Unable to create filter graph.\n");
+        return AVERROR(ENOMEM);
+    }
+
+    /* Create the abuffer filter;
+     * it will be used for feeding the data into the graph. */
+    abuffer = avfilter_get_by_name("abuffer");
+    if (!abuffer) {
+        fprintf(stderr, "Could not find the abuffer filter.\n");
+        return AVERROR_FILTER_NOT_FOUND;
+    }
+
+    abuffer_ctx = avfilter_graph_alloc_filter(filter_graph, abuffer, "src");
+    if (!abuffer_ctx) {
+        fprintf(stderr, "Could not allocate the abuffer instance.\n");
+        return AVERROR(ENOMEM);
+    }
+
+    /* Set the filter options through the AVOptions API. */
+    av_get_channel_layout_string(ch_layout, sizeof(ch_layout), 0, INPUT_CHANNEL_LAYOUT);
+    av_opt_set    (abuffer_ctx, "channel_layout", ch_layout,                            AV_OPT_SEARCH_CHILDREN);
+    av_opt_set    (abuffer_ctx, "sample_fmt",     av_get_sample_fmt_name(INPUT_FORMAT), AV_OPT_SEARCH_CHILDREN);
+    av_opt_set_q  (abuffer_ctx, "time_base",      (AVRational){ 1, INPUT_SAMPLERATE },  AV_OPT_SEARCH_CHILDREN);
+    av_opt_set_int(abuffer_ctx, "sample_rate",    INPUT_SAMPLERATE,                     AV_OPT_SEARCH_CHILDREN);
+
+    /* Now initialize the filter; we pass NULL options, since we have already
+     * set all the options above. */
+    err = avfilter_init_str(abuffer_ctx, NULL);
+    if (err < 0) {
+        fprintf(stderr, "Could not initialize the abuffer filter.\n");
+        return err;
+    }
+
+    /* Create volume filter. */
+    volume = avfilter_get_by_name("volume");
+    if (!volume) {
+        fprintf(stderr, "Could not find the volume filter.\n");
+        return AVERROR_FILTER_NOT_FOUND;
+    }
+
+    volume_ctx = avfilter_graph_alloc_filter(filter_graph, volume, "volume");
+    if (!volume_ctx) {
+        fprintf(stderr, "Could not allocate the volume instance.\n");
+        return AVERROR(ENOMEM);
+    }
+
+    /* A different way of passing the options is as key/value pairs in a
+     * dictionary. */
+    av_dict_set(&options_dict, "volume", AV_STRINGIFY(VOLUME_VAL), 0);
+    err = avfilter_init_dict(volume_ctx, &options_dict);
+    av_dict_free(&options_dict);
+    if (err < 0) {
+        fprintf(stderr, "Could not initialize the volume filter.\n");
+        return err;
+    }
+
+    /* Create the aformat filter;
+     * it ensures that the output is of the format we want. */
+    aformat = avfilter_get_by_name("aformat");
+    if (!aformat) {
+        fprintf(stderr, "Could not find the aformat filter.\n");
+        return AVERROR_FILTER_NOT_FOUND;
+    }
+
+    aformat_ctx = avfilter_graph_alloc_filter(filter_graph, aformat, "aformat");
+    if (!aformat_ctx) {
+        fprintf(stderr, "Could not allocate the aformat instance.\n");
+        return AVERROR(ENOMEM);
+    }
+
+    /* A third way of passing the options is in a string of the form
+     * key1=value1:key2=value2.... */
+    snprintf(options_str, sizeof(options_str),
+             "sample_fmts=%s:sample_rates=%d:channel_layouts=0x%"PRIx64,
+             av_get_sample_fmt_name(AV_SAMPLE_FMT_S16), 44100,
+             (uint64_t)AV_CH_LAYOUT_STEREO);
+    err = avfilter_init_str(aformat_ctx, options_str);
+    if (err < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Could not initialize the aformat filter.\n");
+        return err;
+    }
+
+    /* Finally create the abuffersink filter;
+     * it will be used to get the filtered data out of the graph. */
+    abuffersink = avfilter_get_by_name("abuffersink");
+    if (!abuffersink) {
+        fprintf(stderr, "Could not find the abuffersink filter.\n");
+        return AVERROR_FILTER_NOT_FOUND;
+    }
+
+    abuffersink_ctx = avfilter_graph_alloc_filter(filter_graph, abuffersink, "sink");
+    if (!abuffersink_ctx) {
+        fprintf(stderr, "Could not allocate the abuffersink instance.\n");
+        return AVERROR(ENOMEM);
+    }
+
+    /* This filter takes no options. */
+    err = avfilter_init_str(abuffersink_ctx, NULL);
+    if (err < 0) {
+        fprintf(stderr, "Could not initialize the abuffersink instance.\n");
+        return err;
+    }
+
+    /* Connect the filters;
+     * in this simple case the filters just form a linear chain. */
+    err = avfilter_link(abuffer_ctx, 0, volume_ctx, 0);
+    if (err >= 0)
+        err = avfilter_link(volume_ctx, 0, aformat_ctx, 0);
+    if (err >= 0)
+        err = avfilter_link(aformat_ctx, 0, abuffersink_ctx, 0);
+    if (err < 0) {
+        fprintf(stderr, "Error connecting filters\n");
+        return err;
+    }
+
+    /* Configure the graph. */
+    err = avfilter_graph_config(filter_graph, NULL);
+    if (err < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Error configuring the filter graph\n");
+        return err;
+    }
+
+    *graph = filter_graph;
+    *src   = abuffer_ctx;
+    *sink  = abuffersink_ctx;
+
+    return 0;
+}
+
+/* Do something useful with the filtered data: this simple
+ * example just prints the MD5 checksum of each plane to stdout. */
+static int process_output(struct AVMD5 *md5, AVFrame *frame)
+{
+    int planar     = av_sample_fmt_is_planar(frame->format);
+    int channels   = av_get_channel_layout_nb_channels(frame->channel_layout);
+    int planes     = planar ? channels : 1;
+    int bps        = av_get_bytes_per_sample(frame->format);
+    int plane_size = bps * frame->nb_samples * (planar ? 1 : channels);
+    int i, j;
+
+    for (i = 0; i < planes; i++) {
+        uint8_t checksum[16];
+
+        av_md5_init(md5);
+        av_md5_sum(checksum, frame->extended_data[i], plane_size);
+
+        fprintf(stdout, "plane %d: 0x", i);
+        for (j = 0; j < sizeof(checksum); j++)
+            fprintf(stdout, "%02X", checksum[j]);
+        fprintf(stdout, "\n");
+    }
+    fprintf(stdout, "\n");
+
+    return 0;
+}
+
+/* Construct a frame of audio data to be filtered;
+ * this simple example just synthesizes a sine wave. */
+static int get_input(AVFrame *frame, int frame_num)
+{
+    int err, i, j;
+
+#define FRAME_SIZE 1024
+
+    /* Set up the frame properties and allocate the buffer for the data. */
+    frame->sample_rate    = INPUT_SAMPLERATE;
+    frame->format         = INPUT_FORMAT;
+    frame->channel_layout = INPUT_CHANNEL_LAYOUT;
+    frame->nb_samples     = FRAME_SIZE;
+    frame->pts            = frame_num * FRAME_SIZE;
+
+    err = av_frame_get_buffer(frame, 0);
+    if (err < 0)
+        return err;
+
+    /* Fill the data for each channel. */
+    for (i = 0; i < 5; i++) {
+        float *data = (float*)frame->extended_data[i];
+
+        for (j = 0; j < frame->nb_samples; j++)
+            data[j] = sin(2 * M_PI * (frame_num + j) * (i + 1) / FRAME_SIZE);
+    }
+
+    return 0;
+}
+
+int main(int argc, char *argv[])
+{
+    struct AVMD5 *md5;
+    AVFilterGraph *graph;
+    AVFilterContext *src, *sink;
+    AVFrame *frame;
+    uint8_t errstr[1024];
+    float duration;
+    int err, nb_frames, i;
+
+    if (argc < 2) {
+        fprintf(stderr, "Usage: %s <duration>\n", argv[0]);
+        return 1;
+    }
+
+    duration  = atof(argv[1]);
+    nb_frames = duration * INPUT_SAMPLERATE / FRAME_SIZE;
+    if (nb_frames <= 0) {
+        fprintf(stderr, "Invalid duration: %s\n", argv[1]);
+        return 1;
+    }
+
+    avfilter_register_all();
+
+    /* Allocate the frame we will be using to store the data. */
+    frame  = av_frame_alloc();
+    if (!frame) {
+        fprintf(stderr, "Error allocating the frame\n");
+        return 1;
+    }
+
+    md5 = av_md5_alloc();
+    if (!md5) {
+        fprintf(stderr, "Error allocating the MD5 context\n");
+        return 1;
+    }
+
+    /* Set up the filtergraph. */
+    err = init_filter_graph(&graph, &src, &sink);
+    if (err < 0) {
+        fprintf(stderr, "Unable to init filter graph:");
+        goto fail;
+    }
+
+    /* the main filtering loop */
+    for (i = 0; i < nb_frames; i++) {
+        /* get an input frame to be filtered */
+        err = get_input(frame, i);
+        if (err < 0) {
+            fprintf(stderr, "Error generating input frame:");
+            goto fail;
+        }
+
+        /* Send the frame to the input of the filtergraph. */
+        err = av_buffersrc_add_frame(src, frame);
+        if (err < 0) {
+            av_frame_unref(frame);
+            fprintf(stderr, "Error submitting the frame to the filtergraph:");
+            goto fail;
+        }
+
+        /* Get all the filtered output that is available. */
+        while ((err = av_buffersink_get_frame(sink, frame)) >= 0) {
+            /* now do something with our filtered frame */
+            err = process_output(md5, frame);
+            if (err < 0) {
+                fprintf(stderr, "Error processing the filtered frame:");
+                goto fail;
+            }
+            av_frame_unref(frame);
+        }
+
+        if (err == AVERROR(EAGAIN)) {
+            /* Need to feed more frames in. */
+            continue;
+        } else if (err == AVERROR_EOF) {
+            /* Nothing more to do, finish. */
+            break;
+        } else if (err < 0) {
+            /* An error occurred. */
+            fprintf(stderr, "Error filtering the data:");
+            goto fail;
+        }
+    }
+
+    avfilter_graph_free(&graph);
+    av_frame_free(&frame);
+    av_freep(&md5);
+
+    return 0;
+
+fail:
+    av_strerror(err, errstr, sizeof(errstr));
+    fprintf(stderr, "%s\n", errstr);
+    return 1;
+}
diff --git a/Music/ffmpeg/doc/examples/filtering_audio.c b/Music/ffmpeg/doc/examples/filtering_audio.c
new file mode 100755
index 0000000..6bb24a4
--- /dev/null
+++ b/Music/ffmpeg/doc/examples/filtering_audio.c
@@ -0,0 +1,295 @@
+/*
+ * Copyright (c) 2010 Nicolas George
+ * Copyright (c) 2011 Stefano Sabatini
+ * Copyright (c) 2012 Clément Bœsch
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+/**
+ * @file
+ * API example for audio decoding and filtering
+ * @example filtering_audio.c
+ */
+
+#include <unistd.h>
+
+#include <libavcodec/avcodec.h>
+#include <libavformat/avformat.h>
+#include <libavfilter/avfiltergraph.h>
+#include <libavfilter/buffersink.h>
+#include <libavfilter/buffersrc.h>
+#include <libavutil/opt.h>
+
+static const char *filter_descr = "aresample=8000,aformat=sample_fmts=s16:channel_layouts=mono";
+static const char *player       = "ffplay -f s16le -ar 8000 -ac 1 -";
+
+static AVFormatContext *fmt_ctx;
+static AVCodecContext *dec_ctx;
+AVFilterContext *buffersink_ctx;
+AVFilterContext *buffersrc_ctx;
+AVFilterGraph *filter_graph;
+static int audio_stream_index = -1;
+
+static int open_input_file(const char *filename)
+{
+    int ret;
+    AVCodec *dec;
+
+    if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
+        return ret;
+    }
+
+    if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
+        return ret;
+    }
+
+    /* select the audio stream */
+    ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_AUDIO, -1, -1, &dec, 0);
+    if (ret < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot find an audio stream in the input file\n");
+        return ret;
+    }
+    audio_stream_index = ret;
+    dec_ctx = fmt_ctx->streams[audio_stream_index]->codec;
+    av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0);
+
+    /* init the audio decoder */
+    if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot open audio decoder\n");
+        return ret;
+    }
+
+    return 0;
+}
+
+static int init_filters(const char *filters_descr)
+{
+    char args[512];
+    int ret = 0;
+    AVFilter *abuffersrc  = avfilter_get_by_name("abuffer");
+    AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
+    AVFilterInOut *outputs = avfilter_inout_alloc();
+    AVFilterInOut *inputs  = avfilter_inout_alloc();
+    static const enum AVSampleFormat out_sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
+    static const int64_t out_channel_layouts[] = { AV_CH_LAYOUT_MONO, -1 };
+    static const int out_sample_rates[] = { 8000, -1 };
+    const AVFilterLink *outlink;
+    AVRational time_base = fmt_ctx->streams[audio_stream_index]->time_base;
+
+    filter_graph = avfilter_graph_alloc();
+    if (!outputs || !inputs || !filter_graph) {
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }
+
+    /* buffer audio source: the decoded frames from the decoder will be inserted here. */
+    if (!dec_ctx->channel_layout)
+        dec_ctx->channel_layout = av_get_default_channel_layout(dec_ctx->channels);
+    snprintf(args, sizeof(args),
+            "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,
+             time_base.num, time_base.den, dec_ctx->sample_rate,
+             av_get_sample_fmt_name(dec_ctx->sample_fmt), dec_ctx->channel_layout);
+    ret = avfilter_graph_create_filter(&buffersrc_ctx, abuffersrc, "in",
+                                       args, NULL, filter_graph);
+    if (ret < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
+        goto end;
+    }
+
+    /* buffer audio sink: to terminate the filter chain. */
+    ret = avfilter_graph_create_filter(&buffersink_ctx, abuffersink, "out",
+                                       NULL, NULL, filter_graph);
+    if (ret < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
+        goto end;
+    }
+
+    ret = av_opt_set_int_list(buffersink_ctx, "sample_fmts", out_sample_fmts, -1,
+                              AV_OPT_SEARCH_CHILDREN);
+    if (ret < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n");
+        goto end;
+    }
+
+    ret = av_opt_set_int_list(buffersink_ctx, "channel_layouts", out_channel_layouts, -1,
+                              AV_OPT_SEARCH_CHILDREN);
+    if (ret < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
+        goto end;
+    }
+
+    ret = av_opt_set_int_list(buffersink_ctx, "sample_rates", out_sample_rates, -1,
+                              AV_OPT_SEARCH_CHILDREN);
+    if (ret < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
+        goto end;
+    }
+
+    /*
+     * Set the endpoints for the filter graph. The filter_graph will
+     * be linked to the graph described by filters_descr.
+     */
+
+    /*
+     * The buffer source output must be connected to the input pad of
+     * the first filter described by filters_descr; since the first
+     * filter input label is not specified, it is set to "in" by
+     * default.
+     */
+    outputs->name       = av_strdup("in");
+    outputs->filter_ctx = buffersrc_ctx;
+    outputs->pad_idx    = 0;
+    outputs->next       = NULL;
+
+    /*
+     * The buffer sink input must be connected to the output pad of
+     * the last filter described by filters_descr; since the last
+     * filter output label is not specified, it is set to "out" by
+     * default.
+     */
+    inputs->name       = av_strdup("out");
+    inputs->filter_ctx = buffersink_ctx;
+    inputs->pad_idx    = 0;
+    inputs->next       = NULL;
+
+    if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
+                                        &inputs, &outputs, NULL)) < 0)
+        goto end;
+
+    if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
+        goto end;
+
+    /* Print summary of the sink buffer
+     * Note: args buffer is reused to store channel layout string */
+    outlink = buffersink_ctx->inputs[0];
+    av_get_channel_layout_string(args, sizeof(args), -1, outlink->channel_layout);
+    av_log(NULL, AV_LOG_INFO, "Output: srate:%dHz fmt:%s chlayout:%s\n",
+           (int)outlink->sample_rate,
+           (char *)av_x_if_null(av_get_sample_fmt_name(outlink->format), "?"),
+           args);
+
+end:
+    avfilter_inout_free(&inputs);
+    avfilter_inout_free(&outputs);
+
+    return ret;
+}
+
+static void print_frame(const AVFrame *frame)
+{
+    const int n = frame->nb_samples * av_get_channel_layout_nb_channels(av_frame_get_channel_layout(frame));
+    const uint16_t *p     = (uint16_t*)frame->data[0];
+    const uint16_t *p_end = p + n;
+
+    while (p < p_end) {
+        fputc(*p    & 0xff, stdout);
+        fputc(*p>>8 & 0xff, stdout);
+        p++;
+    }
+    fflush(stdout);
+}
+
+int main(int argc, char **argv)
+{
+    int ret;
+    AVPacket packet0, packet;
+    AVFrame *frame = av_frame_alloc();
+    AVFrame *filt_frame = av_frame_alloc();
+    int got_frame;
+
+    if (!frame || !filt_frame) {
+        perror("Could not allocate frame");
+        exit(1);
+    }
+    if (argc != 2) {
+        fprintf(stderr, "Usage: %s file | %s\n", argv[0], player);
+        exit(1);
+    }
+
+    av_register_all();
+    avfilter_register_all();
+
+    if ((ret = open_input_file(argv[1])) < 0)
+        goto end;
+    if ((ret = init_filters(filter_descr)) < 0)
+        goto end;
+
+    /* read all packets */
+    packet0.data = NULL;
+    packet.data = NULL;
+    while (1) {
+        if (!packet0.data) {
+            if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
+                break;
+            packet0 = packet;
+        }
+
+        if (packet.stream_index == audio_stream_index) {
+            got_frame = 0;
+            ret = avcodec_decode_audio4(dec_ctx, frame, &got_frame, &packet);
+            if (ret < 0) {
+                av_log(NULL, AV_LOG_ERROR, "Error decoding audio\n");
+                continue;
+            }
+            packet.size -= ret;
+            packet.data += ret;
+
+            if (got_frame) {
+                /* push the audio data from decoded frame into the filtergraph */
+                if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, 0) < 0) {
+                    av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n");
+                    break;
+                }
+
+                /* pull filtered audio from the filtergraph */
+                while (1) {
+                    ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
+                    if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
+                        break;
+                    if (ret < 0)
+                        goto end;
+                    print_frame(filt_frame);
+                    av_frame_unref(filt_frame);
+                }
+            }
+
+            if (packet.size <= 0)
+                av_packet_unref(&packet0);
+        } else {
+            /* discard non-wanted packets */
+            av_packet_unref(&packet0);
+        }
+    }
+end:
+    avfilter_graph_free(&filter_graph);
+    avcodec_close(dec_ctx);
+    avformat_close_input(&fmt_ctx);
+    av_frame_free(&frame);
+    av_frame_free(&filt_frame);
+
+    if (ret < 0 && ret != AVERROR_EOF) {
+        fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
+        exit(1);
+    }
+
+    exit(0);
+}
diff --git a/Music/ffmpeg/doc/examples/filtering_video.c b/Music/ffmpeg/doc/examples/filtering_video.c
new file mode 100755
index 0000000..3dabf13
--- /dev/null
+++ b/Music/ffmpeg/doc/examples/filtering_video.c
@@ -0,0 +1,280 @@
+/*
+ * Copyright (c) 2010 Nicolas George
+ * Copyright (c) 2011 Stefano Sabatini
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+/**
+ * @file
+ * API example for decoding and filtering
+ * @example filtering_video.c
+ */
+
+#define _XOPEN_SOURCE 600 /* for usleep */
+#include <unistd.h>
+
+#include <libavcodec/avcodec.h>
+#include <libavformat/avformat.h>
+#include <libavfilter/avfiltergraph.h>
+#include <libavfilter/buffersink.h>
+#include <libavfilter/buffersrc.h>
+#include <libavutil/opt.h>
+
+const char *filter_descr = "scale=78:24,transpose=cclock";
+/* other way:
+   scale=78:24 [scl]; [scl] transpose=cclock // assumes "[in]" and "[out]" to be input output pads respectively
+ */
+
+static AVFormatContext *fmt_ctx;
+static AVCodecContext *dec_ctx;
+AVFilterContext *buffersink_ctx;
+AVFilterContext *buffersrc_ctx;
+AVFilterGraph *filter_graph;
+static int video_stream_index = -1;
+static int64_t last_pts = AV_NOPTS_VALUE;
+
+static int open_input_file(const char *filename)
+{
+    int ret;
+    AVCodec *dec;
+
+    if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
+        return ret;
+    }
+
+    if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
+        return ret;
+    }
+
+    /* select the video stream */
+    ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_VIDEO, -1, -1, &dec, 0);
+    if (ret < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot find a video stream in the input file\n");
+        return ret;
+    }
+    video_stream_index = ret;
+    dec_ctx = fmt_ctx->streams[video_stream_index]->codec;
+    av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0);
+
+    /* init the video decoder */
+    if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot open video decoder\n");
+        return ret;
+    }
+
+    return 0;
+}
+
+static int init_filters(const char *filters_descr)
+{
+    char args[512];
+    int ret = 0;
+    AVFilter *buffersrc  = avfilter_get_by_name("buffer");
+    AVFilter *buffersink = avfilter_get_by_name("buffersink");
+    AVFilterInOut *outputs = avfilter_inout_alloc();
+    AVFilterInOut *inputs  = avfilter_inout_alloc();
+    AVRational time_base = fmt_ctx->streams[video_stream_index]->time_base;
+    enum AVPixelFormat pix_fmts[] = { AV_PIX_FMT_GRAY8, AV_PIX_FMT_NONE };
+
+    filter_graph = avfilter_graph_alloc();
+    if (!outputs || !inputs || !filter_graph) {
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }
+
+    /* buffer video source: the decoded frames from the decoder will be inserted here. */
+    snprintf(args, sizeof(args),
+            "video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:pixel_aspect=%d/%d",
+            dec_ctx->width, dec_ctx->height, dec_ctx->pix_fmt,
+            time_base.num, time_base.den,
+            dec_ctx->sample_aspect_ratio.num, dec_ctx->sample_aspect_ratio.den);
+
+    ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
+                                       args, NULL, filter_graph);
+    if (ret < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot create buffer source\n");
+        goto end;
+    }
+
+    /* buffer video sink: to terminate the filter chain. */
+    ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
+                                       NULL, NULL, filter_graph);
+    if (ret < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n");
+        goto end;
+    }
+
+    ret = av_opt_set_int_list(buffersink_ctx, "pix_fmts", pix_fmts,
+                              AV_PIX_FMT_NONE, AV_OPT_SEARCH_CHILDREN);
+    if (ret < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot set output pixel format\n");
+        goto end;
+    }
+
+    /*
+     * Set the endpoints for the filter graph. The filter_graph will
+     * be linked to the graph described by filters_descr.
+     */
+
+    /*
+     * The buffer source output must be connected to the input pad of
+     * the first filter described by filters_descr; since the first
+     * filter input label is not specified, it is set to "in" by
+     * default.
+     */
+    outputs->name       = av_strdup("in");
+    outputs->filter_ctx = buffersrc_ctx;
+    outputs->pad_idx    = 0;
+    outputs->next       = NULL;
+
+    /*
+     * The buffer sink input must be connected to the output pad of
+     * the last filter described by filters_descr; since the last
+     * filter output label is not specified, it is set to "out" by
+     * default.
+     */
+    inputs->name       = av_strdup("out");
+    inputs->filter_ctx = buffersink_ctx;
+    inputs->pad_idx    = 0;
+    inputs->next       = NULL;
+
+    if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
+                                    &inputs, &outputs, NULL)) < 0)
+        goto end;
+
+    if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
+        goto end;
+
+end:
+    avfilter_inout_free(&inputs);
+    avfilter_inout_free(&outputs);
+
+    return ret;
+}
+
+static void display_frame(const AVFrame *frame, AVRational time_base)
+{
+    int x, y;
+    uint8_t *p0, *p;
+    int64_t delay;
+
+    if (frame->pts != AV_NOPTS_VALUE) {
+        if (last_pts != AV_NOPTS_VALUE) {
+            /* sleep roughly the right amount of time;
+             * usleep is in microseconds, just like AV_TIME_BASE. */
+            delay = av_rescale_q(frame->pts - last_pts,
+                                 time_base, AV_TIME_BASE_Q);
+            if (delay > 0 && delay < 1000000)
+                usleep(delay);
+        }
+        last_pts = frame->pts;
+    }
+
+    /* Trivial ASCII grayscale display. */
+    p0 = frame->data[0];
+    puts("\033c");
+    for (y = 0; y < frame->height; y++) {
+        p = p0;
+        for (x = 0; x < frame->width; x++)
+            putchar(" .-+#"[*(p++) / 52]);
+        putchar('\n');
+        p0 += frame->linesize[0];
+    }
+    fflush(stdout);
+}
+
+int main(int argc, char **argv)
+{
+    int ret;
+    AVPacket packet;
+    AVFrame *frame = av_frame_alloc();
+    AVFrame *filt_frame = av_frame_alloc();
+    int got_frame;
+
+    if (!frame || !filt_frame) {
+        perror("Could not allocate frame");
+        exit(1);
+    }
+    if (argc != 2) {
+        fprintf(stderr, "Usage: %s file\n", argv[0]);
+        exit(1);
+    }
+
+    av_register_all();
+    avfilter_register_all();
+
+    if ((ret = open_input_file(argv[1])) < 0)
+        goto end;
+    if ((ret = init_filters(filter_descr)) < 0)
+        goto end;
+
+    /* read all packets */
+    while (1) {
+        if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
+            break;
+
+        if (packet.stream_index == video_stream_index) {
+            got_frame = 0;
+            ret = avcodec_decode_video2(dec_ctx, frame, &got_frame, &packet);
+            if (ret < 0) {
+                av_log(NULL, AV_LOG_ERROR, "Error decoding video\n");
+                break;
+            }
+
+            if (got_frame) {
+                frame->pts = av_frame_get_best_effort_timestamp(frame);
+
+                /* push the decoded frame into the filtergraph */
+                if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) {
+                    av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
+                    break;
+                }
+
+                /* pull filtered frames from the filtergraph */
+                while (1) {
+                    ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
+                    if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
+                        break;
+                    if (ret < 0)
+                        goto end;
+                    display_frame(filt_frame, buffersink_ctx->inputs[0]->time_base);
+                    av_frame_unref(filt_frame);
+                }
+                av_frame_unref(frame);
+            }
+        }
+        av_packet_unref(&packet);
+    }
+end:
+    avfilter_graph_free(&filter_graph);
+    avcodec_close(dec_ctx);
+    avformat_close_input(&fmt_ctx);
+    av_frame_free(&frame);
+    av_frame_free(&filt_frame);
+
+    if (ret < 0 && ret != AVERROR_EOF) {
+        fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
+        exit(1);
+    }
+
+    exit(0);
+}
diff --git a/Music/ffmpeg/doc/examples/http_multiclient.c b/Music/ffmpeg/doc/examples/http_multiclient.c
new file mode 100755
index 0000000..7173c4d
--- /dev/null
+++ b/Music/ffmpeg/doc/examples/http_multiclient.c
@@ -0,0 +1,155 @@
+/*
+ * Copyright (c) 2015 Stephan Holljes
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+/**
+ * @file
+ * libavformat multi-client network API usage example.
+ *
+ * @example http_multiclient.c
+ * This example will serve a file without decoding or demuxing it over http.
+ * Multiple clients can connect and will receive the same file.
+ */
+
+#include <libavformat/avformat.h>
+#include <libavutil/opt.h>
+#include <unistd.h>
+
+static void process_client(AVIOContext *client, const char *in_uri)
+{
+    AVIOContext *input = NULL;
+    uint8_t buf[1024];
+    int ret, n, reply_code;
+    uint8_t *resource = NULL;
+    while ((ret = avio_handshake(client)) > 0) {
+        av_opt_get(client, "resource", AV_OPT_SEARCH_CHILDREN, &resource);
+        // check for strlen(resource) is necessary, because av_opt_get()
+        // may return empty string.
+        if (resource && strlen(resource))
+            break;
+    }
+    if (ret < 0)
+        goto end;
+    av_log(client, AV_LOG_TRACE, "resource=%p\n", resource);
+    if (resource && resource[0] == '/' && !strcmp((resource + 1), in_uri)) {
+        reply_code = 200;
+    } else {
+        reply_code = AVERROR_HTTP_NOT_FOUND;
+    }
+    if ((ret = av_opt_set_int(client, "reply_code", reply_code, AV_OPT_SEARCH_CHILDREN)) < 0) {
+        av_log(client, AV_LOG_ERROR, "Failed to set reply_code: %s.\n", av_err2str(ret));
+        goto end;
+    }
+    av_log(client, AV_LOG_TRACE, "Set reply code to %d\n", reply_code);
+
+    while ((ret = avio_handshake(client)) > 0);
+
+    if (ret < 0)
+        goto end;
+
+    fprintf(stderr, "Handshake performed.\n");
+    if (reply_code != 200)
+        goto end;
+    fprintf(stderr, "Opening input file.\n");
+    if ((ret = avio_open2(&input, in_uri, AVIO_FLAG_READ, NULL, NULL)) < 0) {
+        av_log(input, AV_LOG_ERROR, "Failed to open input: %s: %s.\n", in_uri,
+               av_err2str(ret));
+        goto end;
+    }
+    for(;;) {
+        n = avio_read(input, buf, sizeof(buf));
+        if (n < 0) {
+            if (n == AVERROR_EOF)
+                break;
+            av_log(input, AV_LOG_ERROR, "Error reading from input: %s.\n",
+                   av_err2str(n));
+            break;
+        }
+        avio_write(client, buf, n);
+        avio_flush(client);
+    }
+end:
+    fprintf(stderr, "Flushing client\n");
+    avio_flush(client);
+    fprintf(stderr, "Closing client\n");
+    avio_close(client);
+    fprintf(stderr, "Closing input\n");
+    avio_close(input);
+}
+
+int main(int argc, char **argv)
+{
+    AVDictionary *options = NULL;
+    AVIOContext *client = NULL, *server = NULL;
+    const char *in_uri, *out_uri;
+    int ret, pid;
+    av_log_set_level(AV_LOG_TRACE);
+    if (argc < 3) {
+        printf("usage: %s input http://hostname[:port]\n"
+               "API example program to serve http to multiple clients.\n"
+               "\n", argv[0]);
+        return 1;
+    }
+
+    in_uri = argv[1];
+    out_uri = argv[2];
+
+    av_register_all();
+    avformat_network_init();
+
+    if ((ret = av_dict_set(&options, "listen", "2", 0)) < 0) {
+        fprintf(stderr, "Failed to set listen mode for server: %s\n", av_err2str(ret));
+        return ret;
+    }
+    if ((ret = avio_open2(&server, out_uri, AVIO_FLAG_WRITE, NULL, &options)) < 0) {
+        fprintf(stderr, "Failed to open server: %s\n", av_err2str(ret));
+        return ret;
+    }
+    fprintf(stderr, "Entering main loop.\n");
+    for(;;) {
+        if ((ret = avio_accept(server, &client)) < 0)
+            goto end;
+        fprintf(stderr, "Accepted client, forking process.\n");
+        // XXX: Since we don't reap our children and don't ignore signals
+        //      this produces zombie processes.
+        pid = fork();
+        if (pid < 0) {
+            perror("Fork failed");
+            ret = AVERROR(errno);
+            goto end;
+        }
+        if (pid == 0) {
+            fprintf(stderr, "In child.\n");
+            process_client(client, in_uri);
+            avio_close(server);
+            exit(0);
+        }
+        if (pid > 0)
+            avio_close(client);
+    }
+end:
+    avio_close(server);
+    if (ret < 0 && ret != AVERROR_EOF) {
+        fprintf(stderr, "Some errors occurred: %s\n", av_err2str(ret));
+        return 1;
+    }
+    return 0;
+}
diff --git a/Music/ffmpeg/doc/examples/metadata.c b/Music/ffmpeg/doc/examples/metadata.c
new file mode 100755
index 0000000..f73c267
--- /dev/null
+++ b/Music/ffmpeg/doc/examples/metadata.c
@@ -0,0 +1,56 @@
+/*
+ * Copyright (c) 2011 Reinhard Tartler
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+/**
+ * @file
+ * Shows how the metadata API can be used in application programs.
+ * @example metadata.c
+ */
+
+#include <stdio.h>
+
+#include <libavformat/avformat.h>
+#include <libavutil/dict.h>
+
+int main (int argc, char **argv)
+{
+    AVFormatContext *fmt_ctx = NULL;
+    AVDictionaryEntry *tag = NULL;
+    int ret;
+
+    if (argc != 2) {
+        printf("usage: %s <input_file>\n"
+               "example program to demonstrate the use of the libavformat metadata API.\n"
+               "\n", argv[0]);
+        return 1;
+    }
+
+    av_register_all();
+    if ((ret = avformat_open_input(&fmt_ctx, argv[1], NULL, NULL)))
+        return ret;
+
+    while ((tag = av_dict_get(fmt_ctx->metadata, "", tag, AV_DICT_IGNORE_SUFFIX)))
+        printf("%s=%s\n", tag->key, tag->value);
+
+    avformat_close_input(&fmt_ctx);
+    return 0;
+}
diff --git a/Music/ffmpeg/doc/examples/muxing.c b/Music/ffmpeg/doc/examples/muxing.c
new file mode 100755
index 0000000..f1f5bb8
--- /dev/null
+++ b/Music/ffmpeg/doc/examples/muxing.c
@@ -0,0 +1,673 @@
+/*
+ * Copyright (c) 2003 Fabrice Bellard
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+/**
+ * @file
+ * libavformat API example.
+ *
+ * Output a media file in any supported libavformat format. The default
+ * codecs are used.
+ * @example muxing.c
+ */
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <math.h>
+
+#include <libavutil/avassert.h>
+#include <libavutil/channel_layout.h>
+#include <libavutil/opt.h>
+#include <libavutil/mathematics.h>
+#include <libavutil/timestamp.h>
+#include <libavformat/avformat.h>
+#include <libswscale/swscale.h>
+#include <libswresample/swresample.h>
+
+#define STREAM_DURATION   10.0
+#define STREAM_FRAME_RATE 25 /* 25 images/s */
+#define STREAM_PIX_FMT    AV_PIX_FMT_YUV420P /* default pix_fmt */
+
+#define SCALE_FLAGS SWS_BICUBIC
+
+// a wrapper around a single output AVStream
+typedef struct OutputStream {
+    AVStream *st;
+    AVCodecContext *enc;
+
+    /* pts of the next frame that will be generated */
+    int64_t next_pts;
+    int samples_count;
+
+    AVFrame *frame;
+    AVFrame *tmp_frame;
+
+    float t, tincr, tincr2;
+
+    struct SwsContext *sws_ctx;
+    struct SwrContext *swr_ctx;
+} OutputStream;
+
+static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt)
+{
+    AVRational *time_base = &fmt_ctx->streams[pkt->stream_index]->time_base;
+
+    printf("pts:%s pts_time:%s dts:%s dts_time:%s duration:%s duration_time:%s stream_index:%d\n",
+           av_ts2str(pkt->pts), av_ts2timestr(pkt->pts, time_base),
+           av_ts2str(pkt->dts), av_ts2timestr(pkt->dts, time_base),
+           av_ts2str(pkt->duration), av_ts2timestr(pkt->duration, time_base),
+           pkt->stream_index);
+}
+
+static int write_frame(AVFormatContext *fmt_ctx, const AVRational *time_base, AVStream *st, AVPacket *pkt)
+{
+    /* rescale output packet timestamp values from codec to stream timebase */
+    av_packet_rescale_ts(pkt, *time_base, st->time_base);
+    pkt->stream_index = st->index;
+
+    /* Write the compressed frame to the media file. */
+    log_packet(fmt_ctx, pkt);
+    return av_interleaved_write_frame(fmt_ctx, pkt);
+}
+
+/* Add an output stream. */
+static void add_stream(OutputStream *ost, AVFormatContext *oc,
+                       AVCodec **codec,
+                       enum AVCodecID codec_id)
+{
+    AVCodecContext *c;
+    int i;
+
+    /* find the encoder */
+    *codec = avcodec_find_encoder(codec_id);
+    if (!(*codec)) {
+        fprintf(stderr, "Could not find encoder for '%s'\n",
+                avcodec_get_name(codec_id));
+        exit(1);
+    }
+
+    ost->st = avformat_new_stream(oc, NULL);
+    if (!ost->st) {
+        fprintf(stderr, "Could not allocate stream\n");
+        exit(1);
+    }
+    ost->st->id = oc->nb_streams-1;
+    c = avcodec_alloc_context3(*codec);
+    if (!c) {
+        fprintf(stderr, "Could not alloc an encoding context\n");
+        exit(1);
+    }
+    ost->enc = c;
+
+    switch ((*codec)->type) {
+    case AVMEDIA_TYPE_AUDIO:
+        c->sample_fmt  = (*codec)->sample_fmts ?
+            (*codec)->sample_fmts[0] : AV_SAMPLE_FMT_FLTP;
+        c->bit_rate    = 64000;
+        c->sample_rate = 44100;
+        if ((*codec)->supported_samplerates) {
+            c->sample_rate = (*codec)->supported_samplerates[0];
+            for (i = 0; (*codec)->supported_samplerates[i]; i++) {
+                if ((*codec)->supported_samplerates[i] == 44100)
+                    c->sample_rate = 44100;
+            }
+        }
+        c->channels        = av_get_channel_layout_nb_channels(c->channel_layout);
+        c->channel_layout = AV_CH_LAYOUT_STEREO;
+        if ((*codec)->channel_layouts) {
+            c->channel_layout = (*codec)->channel_layouts[0];
+            for (i = 0; (*codec)->channel_layouts[i]; i++) {
+                if ((*codec)->channel_layouts[i] == AV_CH_LAYOUT_STEREO)
+                    c->channel_layout = AV_CH_LAYOUT_STEREO;
+            }
+        }
+        c->channels        = av_get_channel_layout_nb_channels(c->channel_layout);
+        ost->st->time_base = (AVRational){ 1, c->sample_rate };
+        break;
+
+    case AVMEDIA_TYPE_VIDEO:
+        c->codec_id = codec_id;
+
+        c->bit_rate = 400000;
+        /* Resolution must be a multiple of two. */
+        c->width    = 352;
+        c->height   = 288;
+        /* timebase: This is the fundamental unit of time (in seconds) in terms
+         * of which frame timestamps are represented. For fixed-fps content,
+         * timebase should be 1/framerate and timestamp increments should be
+         * identical to 1. */
+        ost->st->time_base = (AVRational){ 1, STREAM_FRAME_RATE };
+        c->time_base       = ost->st->time_base;
+
+        c->gop_size      = 12; /* emit one intra frame every twelve frames at most */
+        c->pix_fmt       = STREAM_PIX_FMT;
+        if (c->codec_id == AV_CODEC_ID_MPEG2VIDEO) {
+            /* just for testing, we also add B-frames */
+            c->max_b_frames = 2;
+        }
+        if (c->codec_id == AV_CODEC_ID_MPEG1VIDEO) {
+            /* Needed to avoid using macroblocks in which some coeffs overflow.
+             * This does not happen with normal video, it just happens here as
+             * the motion of the chroma plane does not match the luma plane. */
+            c->mb_decision = 2;
+        }
+    break;
+
+    default:
+        break;
+    }
+
+    /* Some formats want stream headers to be separate. */
+    if (oc->oformat->flags & AVFMT_GLOBALHEADER)
+        c->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
+}
+
+/**************************************************************/
+/* audio output */
+
+static AVFrame *alloc_audio_frame(enum AVSampleFormat sample_fmt,
+                                  uint64_t channel_layout,
+                                  int sample_rate, int nb_samples)
+{
+    AVFrame *frame = av_frame_alloc();
+    int ret;
+
+    if (!frame) {
+        fprintf(stderr, "Error allocating an audio frame\n");
+        exit(1);
+    }
+
+    frame->format = sample_fmt;
+    frame->channel_layout = channel_layout;
+    frame->sample_rate = sample_rate;
+    frame->nb_samples = nb_samples;
+
+    if (nb_samples) {
+        ret = av_frame_get_buffer(frame, 0);
+        if (ret < 0) {
+            fprintf(stderr, "Error allocating an audio buffer\n");
+            exit(1);
+        }
+    }
+
+    return frame;
+}
+
+static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg)
+{
+    AVCodecContext *c;
+    int nb_samples;
+    int ret;
+    AVDictionary *opt = NULL;
+
+    c = ost->enc;
+
+    /* open it */
+    av_dict_copy(&opt, opt_arg, 0);
+    ret = avcodec_open2(c, codec, &opt);
+    av_dict_free(&opt);
+    if (ret < 0) {
+        fprintf(stderr, "Could not open audio codec: %s\n", av_err2str(ret));
+        exit(1);
+    }
+
+    /* init signal generator */
+    ost->t     = 0;
+    ost->tincr = 2 * M_PI * 110.0 / c->sample_rate;
+    /* increment frequency by 110 Hz per second */
+    ost->tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;
+
+    if (c->codec->capabilities & AV_CODEC_CAP_VARIABLE_FRAME_SIZE)
+        nb_samples = 10000;
+    else
+        nb_samples = c->frame_size;
+
+    ost->frame     = alloc_audio_frame(c->sample_fmt, c->channel_layout,
+                                       c->sample_rate, nb_samples);
+    ost->tmp_frame = alloc_audio_frame(AV_SAMPLE_FMT_S16, c->channel_layout,
+                                       c->sample_rate, nb_samples);
+
+    /* copy the stream parameters to the muxer */
+    ret = avcodec_parameters_from_context(ost->st->codecpar, c);
+    if (ret < 0) {
+        fprintf(stderr, "Could not copy the stream parameters\n");
+        exit(1);
+    }
+
+    /* create resampler context */
+        ost->swr_ctx = swr_alloc();
+        if (!ost->swr_ctx) {
+            fprintf(stderr, "Could not allocate resampler context\n");
+            exit(1);
+        }
+
+        /* set options */
+        av_opt_set_int       (ost->swr_ctx, "in_channel_count",   c->channels,       0);
+        av_opt_set_int       (ost->swr_ctx, "in_sample_rate",     c->sample_rate,    0);
+        av_opt_set_sample_fmt(ost->swr_ctx, "in_sample_fmt",      AV_SAMPLE_FMT_S16, 0);
+        av_opt_set_int       (ost->swr_ctx, "out_channel_count",  c->channels,       0);
+        av_opt_set_int       (ost->swr_ctx, "out_sample_rate",    c->sample_rate,    0);
+        av_opt_set_sample_fmt(ost->swr_ctx, "out_sample_fmt",     c->sample_fmt,     0);
+
+        /* initialize the resampling context */
+        if ((ret = swr_init(ost->swr_ctx)) < 0) {
+            fprintf(stderr, "Failed to initialize the resampling context\n");
+            exit(1);
+        }
+}
+
+/* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
+ * 'nb_channels' channels. */
+static AVFrame *get_audio_frame(OutputStream *ost)
+{
+    AVFrame *frame = ost->tmp_frame;
+    int j, i, v;
+    int16_t *q = (int16_t*)frame->data[0];
+
+    /* check if we want to generate more frames */
+    if (av_compare_ts(ost->next_pts, ost->enc->time_base,
+                      STREAM_DURATION, (AVRational){ 1, 1 }) >= 0)
+        return NULL;
+
+    for (j = 0; j <frame->nb_samples; j++) {
+        v = (int)(sin(ost->t) * 10000);
+        for (i = 0; i < ost->enc->channels; i++)
+            *q++ = v;
+        ost->t     += ost->tincr;
+        ost->tincr += ost->tincr2;
+    }
+
+    frame->pts = ost->next_pts;
+    ost->next_pts  += frame->nb_samples;
+
+    return frame;
+}
+
+/*
+ * encode one audio frame and send it to the muxer
+ * return 1 when encoding is finished, 0 otherwise
+ */
+static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
+{
+    AVCodecContext *c;
+    AVPacket pkt = { 0 }; // data and size must be 0;
+    AVFrame *frame;
+    int ret;
+    int got_packet;
+    int dst_nb_samples;
+
+    av_init_packet(&pkt);
+    c = ost->enc;
+
+    frame = get_audio_frame(ost);
+
+    if (frame) {
+        /* convert samples from native format to destination codec format, using the resampler */
+            /* compute destination number of samples */
+            dst_nb_samples = av_rescale_rnd(swr_get_delay(ost->swr_ctx, c->sample_rate) + frame->nb_samples,
+                                            c->sample_rate, c->sample_rate, AV_ROUND_UP);
+            av_assert0(dst_nb_samples == frame->nb_samples);
+
+        /* when we pass a frame to the encoder, it may keep a reference to it
+         * internally;
+         * make sure we do not overwrite it here
+         */
+        ret = av_frame_make_writable(ost->frame);
+        if (ret < 0)
+            exit(1);
+
+            /* convert to destination format */
+            ret = swr_convert(ost->swr_ctx,
+                              ost->frame->data, dst_nb_samples,
+                              (const uint8_t **)frame->data, frame->nb_samples);
+            if (ret < 0) {
+                fprintf(stderr, "Error while converting\n");
+                exit(1);
+            }
+            frame = ost->frame;
+
+        frame->pts = av_rescale_q(ost->samples_count, (AVRational){1, c->sample_rate}, c->time_base);
+        ost->samples_count += dst_nb_samples;
+    }
+
+    ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet);
+    if (ret < 0) {
+        fprintf(stderr, "Error encoding audio frame: %s\n", av_err2str(ret));
+        exit(1);
+    }
+
+    if (got_packet) {
+        ret = write_frame(oc, &c->time_base, ost->st, &pkt);
+        if (ret < 0) {
+            fprintf(stderr, "Error while writing audio frame: %s\n",
+                    av_err2str(ret));
+            exit(1);
+        }
+    }
+
+    return (frame || got_packet) ? 0 : 1;
+}
+
+/**************************************************************/
+/* video output */
+
+static AVFrame *alloc_picture(enum AVPixelFormat pix_fmt, int width, int height)
+{
+    AVFrame *picture;
+    int ret;
+
+    picture = av_frame_alloc();
+    if (!picture)
+        return NULL;
+
+    picture->format = pix_fmt;
+    picture->width  = width;
+    picture->height = height;
+
+    /* allocate the buffers for the frame data */
+    ret = av_frame_get_buffer(picture, 32);
+    if (ret < 0) {
+        fprintf(stderr, "Could not allocate frame data.\n");
+        exit(1);
+    }
+
+    return picture;
+}
+
+static void open_video(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg)
+{
+    int ret;
+    AVCodecContext *c = ost->enc;
+    AVDictionary *opt = NULL;
+
+    av_dict_copy(&opt, opt_arg, 0);
+
+    /* open the codec */
+    ret = avcodec_open2(c, codec, &opt);
+    av_dict_free(&opt);
+    if (ret < 0) {
+        fprintf(stderr, "Could not open video codec: %s\n", av_err2str(ret));
+        exit(1);
+    }
+
+    /* allocate and init a re-usable frame */
+    ost->frame = alloc_picture(c->pix_fmt, c->width, c->height);
+    if (!ost->frame) {
+        fprintf(stderr, "Could not allocate video frame\n");
+        exit(1);
+    }
+
+    /* If the output format is not YUV420P, then a temporary YUV420P
+     * picture is needed too. It is then converted to the required
+     * output format. */
+    ost->tmp_frame = NULL;
+    if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
+        ost->tmp_frame = alloc_picture(AV_PIX_FMT_YUV420P, c->width, c->height);
+        if (!ost->tmp_frame) {
+            fprintf(stderr, "Could not allocate temporary picture\n");
+            exit(1);
+        }
+    }
+
+    /* copy the stream parameters to the muxer */
+    ret = avcodec_parameters_from_context(ost->st->codecpar, c);
+    if (ret < 0) {
+        fprintf(stderr, "Could not copy the stream parameters\n");
+        exit(1);
+    }
+}
+
+/* Prepare a dummy image. */
+static void fill_yuv_image(AVFrame *pict, int frame_index,
+                           int width, int height)
+{
+    int x, y, i, ret;
+
+    /* when we pass a frame to the encoder, it may keep a reference to it
+     * internally;
+     * make sure we do not overwrite it here
+     */
+    ret = av_frame_make_writable(pict);
+    if (ret < 0)
+        exit(1);
+
+    i = frame_index;
+
+    /* Y */
+    for (y = 0; y < height; y++)
+        for (x = 0; x < width; x++)
+            pict->data[0][y * pict->linesize[0] + x] = x + y + i * 3;
+
+    /* Cb and Cr */
+    for (y = 0; y < height / 2; y++) {
+        for (x = 0; x < width / 2; x++) {
+            pict->data[1][y * pict->linesize[1] + x] = 128 + y + i * 2;
+            pict->data[2][y * pict->linesize[2] + x] = 64 + x + i * 5;
+        }
+    }
+}
+
+static AVFrame *get_video_frame(OutputStream *ost)
+{
+    AVCodecContext *c = ost->enc;
+
+    /* check if we want to generate more frames */
+    if (av_compare_ts(ost->next_pts, c->time_base,
+                      STREAM_DURATION, (AVRational){ 1, 1 }) >= 0)
+        return NULL;
+
+    if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
+        /* as we only generate a YUV420P picture, we must convert it
+         * to the codec pixel format if needed */
+        if (!ost->sws_ctx) {
+            ost->sws_ctx = sws_getContext(c->width, c->height,
+                                          AV_PIX_FMT_YUV420P,
+                                          c->width, c->height,
+                                          c->pix_fmt,
+                                          SCALE_FLAGS, NULL, NULL, NULL);
+            if (!ost->sws_ctx) {
+                fprintf(stderr,
+                        "Could not initialize the conversion context\n");
+                exit(1);
+            }
+        }
+        fill_yuv_image(ost->tmp_frame, ost->next_pts, c->width, c->height);
+        sws_scale(ost->sws_ctx,
+                  (const uint8_t * const *)ost->tmp_frame->data, ost->tmp_frame->linesize,
+                  0, c->height, ost->frame->data, ost->frame->linesize);
+    } else {
+        fill_yuv_image(ost->frame, ost->next_pts, c->width, c->height);
+    }
+
+    ost->frame->pts = ost->next_pts++;
+
+    return ost->frame;
+}
+
+/*
+ * encode one video frame and send it to the muxer
+ * return 1 when encoding is finished, 0 otherwise
+ */
+static int write_video_frame(AVFormatContext *oc, OutputStream *ost)
+{
+    int ret;
+    AVCodecContext *c;
+    AVFrame *frame;
+    int got_packet = 0;
+    AVPacket pkt = { 0 };
+
+    c = ost->enc;
+
+    frame = get_video_frame(ost);
+
+    av_init_packet(&pkt);
+
+    /* encode the image */
+    ret = avcodec_encode_video2(c, &pkt, frame, &got_packet);
+    if (ret < 0) {
+        fprintf(stderr, "Error encoding video frame: %s\n", av_err2str(ret));
+        exit(1);
+    }
+
+    if (got_packet) {
+        ret = write_frame(oc, &c->time_base, ost->st, &pkt);
+    } else {
+        ret = 0;
+    }
+
+    if (ret < 0) {
+        fprintf(stderr, "Error while writing video frame: %s\n", av_err2str(ret));
+        exit(1);
+    }
+
+    return (frame || got_packet) ? 0 : 1;
+}
+
+static void close_stream(AVFormatContext *oc, OutputStream *ost)
+{
+    avcodec_free_context(&ost->enc);
+    av_frame_free(&ost->frame);
+    av_frame_free(&ost->tmp_frame);
+    sws_freeContext(ost->sws_ctx);
+    swr_free(&ost->swr_ctx);
+}
+
+/**************************************************************/
+/* media file output */
+
+int main(int argc, char **argv)
+{
+    OutputStream video_st = { 0 }, audio_st = { 0 };
+    const char *filename;
+    AVOutputFormat *fmt;
+    AVFormatContext *oc;
+    AVCodec *audio_codec, *video_codec;
+    int ret;
+    int have_video = 0, have_audio = 0;
+    int encode_video = 0, encode_audio = 0;
+    AVDictionary *opt = NULL;
+    int i;
+
+    /* Initialize libavcodec, and register all codecs and formats. */
+    av_register_all();
+
+    if (argc < 2) {
+        printf("usage: %s output_file\n"
+               "API example program to output a media file with libavformat.\n"
+               "This program generates a synthetic audio and video stream, encodes and\n"
+               "muxes them into a file named output_file.\n"
+               "The output format is automatically guessed according to the file extension.\n"
+               "Raw images can also be output by using '%%d' in the filename.\n"
+               "\n", argv[0]);
+        return 1;
+    }
+
+    filename = argv[1];
+    for (i = 2; i+1 < argc; i+=2) {
+        if (!strcmp(argv[i], "-flags") || !strcmp(argv[i], "-fflags"))
+            av_dict_set(&opt, argv[i]+1, argv[i+1], 0);
+    }
+
+    /* allocate the output media context */
+    avformat_alloc_output_context2(&oc, NULL, NULL, filename);
+    if (!oc) {
+        printf("Could not deduce output format from file extension: using MPEG.\n");
+        avformat_alloc_output_context2(&oc, NULL, "mpeg", filename);
+    }
+    if (!oc)
+        return 1;
+
+    fmt = oc->oformat;
+
+    /* Add the audio and video streams using the default format codecs
+     * and initialize the codecs. */
+    if (fmt->video_codec != AV_CODEC_ID_NONE) {
+        add_stream(&video_st, oc, &video_codec, fmt->video_codec);
+        have_video = 1;
+        encode_video = 1;
+    }
+    if (fmt->audio_codec != AV_CODEC_ID_NONE) {
+        add_stream(&audio_st, oc, &audio_codec, fmt->audio_codec);
+        have_audio = 1;
+        encode_audio = 1;
+    }
+
+    /* Now that all the parameters are set, we can open the audio and
+     * video codecs and allocate the necessary encode buffers. */
+    if (have_video)
+        open_video(oc, video_codec, &video_st, opt);
+
+    if (have_audio)
+        open_audio(oc, audio_codec, &audio_st, opt);
+
+    av_dump_format(oc, 0, filename, 1);
+
+    /* open the output file, if needed */
+    if (!(fmt->flags & AVFMT_NOFILE)) {
+        ret = avio_open(&oc->pb, filename, AVIO_FLAG_WRITE);
+        if (ret < 0) {
+            fprintf(stderr, "Could not open '%s': %s\n", filename,
+                    av_err2str(ret));
+            return 1;
+        }
+    }
+
+    /* Write the stream header, if any. */
+    ret = avformat_write_header(oc, &opt);
+    if (ret < 0) {
+        fprintf(stderr, "Error occurred when opening output file: %s\n",
+                av_err2str(ret));
+        return 1;
+    }
+
+    while (encode_video || encode_audio) {
+        /* select the stream to encode */
+        if (encode_video &&
+            (!encode_audio || av_compare_ts(video_st.next_pts, video_st.enc->time_base,
+                                            audio_st.next_pts, audio_st.enc->time_base) <= 0)) {
+            encode_video = !write_video_frame(oc, &video_st);
+        } else {
+            encode_audio = !write_audio_frame(oc, &audio_st);
+        }
+    }
+
+    /* Write the trailer, if any. The trailer must be written before you
+     * close the CodecContexts open when you wrote the header; otherwise
+     * av_write_trailer() may try to use memory that was freed on
+     * av_codec_close(). */
+    av_write_trailer(oc);
+
+    /* Close each codec. */
+    if (have_video)
+        close_stream(oc, &video_st);
+    if (have_audio)
+        close_stream(oc, &audio_st);
+
+    if (!(fmt->flags & AVFMT_NOFILE))
+        /* Close the output file. */
+        avio_closep(&oc->pb);
+
+    /* free the stream */
+    avformat_free_context(oc);
+
+    return 0;
+}
diff --git a/Music/ffmpeg/doc/examples/qsvdec.c b/Music/ffmpeg/doc/examples/qsvdec.c
new file mode 100755
index 0000000..aaecd81
--- /dev/null
+++ b/Music/ffmpeg/doc/examples/qsvdec.c
@@ -0,0 +1,487 @@
+/*
+ * Copyright (c) 2015 Anton Khirnov
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+/**
+ * @file
+ * Intel QSV-accelerated H.264 decoding example.
+ *
+ * @example qsvdec.c
+ * This example shows how to do QSV-accelerated H.264 decoding with output
+ * frames in the VA-API video surfaces.
+ */
+
+#include "config.h"
+
+#include <stdio.h>
+
+#include <mfx/mfxvideo.h>
+
+#include <va/va.h>
+#include <va/va_x11.h>
+#include <X11/Xlib.h>
+
+#include "libavformat/avformat.h"
+#include "libavformat/avio.h"
+
+#include "libavcodec/avcodec.h"
+#include "libavcodec/qsv.h"
+
+#include "libavutil/error.h"
+#include "libavutil/mem.h"
+
+typedef struct DecodeContext {
+    mfxSession mfx_session;
+    VADisplay va_dpy;
+
+    VASurfaceID *surfaces;
+    mfxMemId    *surface_ids;
+    int         *surface_used;
+    int       nb_surfaces;
+
+    mfxFrameInfo frame_info;
+} DecodeContext;
+
+static mfxStatus frame_alloc(mfxHDL pthis, mfxFrameAllocRequest *req,
+                             mfxFrameAllocResponse *resp)
+{
+    DecodeContext *decode = pthis;
+    int err, i;
+
+    if (decode->surfaces) {
+        fprintf(stderr, "Multiple allocation requests.\n");
+        return MFX_ERR_MEMORY_ALLOC;
+    }
+    if (!(req->Type & MFX_MEMTYPE_VIDEO_MEMORY_DECODER_TARGET)) {
+        fprintf(stderr, "Unsupported surface type: %d\n", req->Type);
+        return MFX_ERR_UNSUPPORTED;
+    }
+    if (req->Info.BitDepthLuma != 8 || req->Info.BitDepthChroma != 8 ||
+        req->Info.Shift || req->Info.FourCC != MFX_FOURCC_NV12 ||
+        req->Info.ChromaFormat != MFX_CHROMAFORMAT_YUV420) {
+        fprintf(stderr, "Unsupported surface properties.\n");
+        return MFX_ERR_UNSUPPORTED;
+    }
+
+    decode->surfaces     = av_malloc_array (req->NumFrameSuggested, sizeof(*decode->surfaces));
+    decode->surface_ids  = av_malloc_array (req->NumFrameSuggested, sizeof(*decode->surface_ids));
+    decode->surface_used = av_mallocz_array(req->NumFrameSuggested, sizeof(*decode->surface_used));
+    if (!decode->surfaces || !decode->surface_ids || !decode->surface_used)
+        goto fail;
+
+    err = vaCreateSurfaces(decode->va_dpy, VA_RT_FORMAT_YUV420,
+                           req->Info.Width, req->Info.Height,
+                           decode->surfaces, req->NumFrameSuggested,
+                           NULL, 0);
+    if (err != VA_STATUS_SUCCESS) {
+        fprintf(stderr, "Error allocating VA surfaces\n");
+        goto fail;
+    }
+    decode->nb_surfaces = req->NumFrameSuggested;
+
+    for (i = 0; i < decode->nb_surfaces; i++)
+        decode->surface_ids[i] = &decode->surfaces[i];
+
+    resp->mids           = decode->surface_ids;
+    resp->NumFrameActual = decode->nb_surfaces;
+
+    decode->frame_info = req->Info;
+
+    return MFX_ERR_NONE;
+fail:
+    av_freep(&decode->surfaces);
+    av_freep(&decode->surface_ids);
+    av_freep(&decode->surface_used);
+
+    return MFX_ERR_MEMORY_ALLOC;
+}
+
+static mfxStatus frame_free(mfxHDL pthis, mfxFrameAllocResponse *resp)
+{
+    return MFX_ERR_NONE;
+}
+
+static mfxStatus frame_lock(mfxHDL pthis, mfxMemId mid, mfxFrameData *ptr)
+{
+    return MFX_ERR_UNSUPPORTED;
+}
+
+static mfxStatus frame_unlock(mfxHDL pthis, mfxMemId mid, mfxFrameData *ptr)
+{
+    return MFX_ERR_UNSUPPORTED;
+}
+
+static mfxStatus frame_get_hdl(mfxHDL pthis, mfxMemId mid, mfxHDL *hdl)
+{
+    *hdl = mid;
+    return MFX_ERR_NONE;
+}
+
+static void free_surfaces(DecodeContext *decode)
+{
+    if (decode->surfaces)
+        vaDestroySurfaces(decode->va_dpy, decode->surfaces, decode->nb_surfaces);
+    av_freep(&decode->surfaces);
+    av_freep(&decode->surface_ids);
+    av_freep(&decode->surface_used);
+    decode->nb_surfaces = 0;
+}
+
+static void free_buffer(void *opaque, uint8_t *data)
+{
+    int *used = opaque;
+    *used = 0;
+    av_freep(&data);
+}
+
+static int get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
+{
+    DecodeContext *decode = avctx->opaque;
+
+    mfxFrameSurface1 *surf;
+    AVBufferRef *surf_buf;
+    int idx;
+
+    for (idx = 0; idx < decode->nb_surfaces; idx++) {
+        if (!decode->surface_used[idx])
+            break;
+    }
+    if (idx == decode->nb_surfaces) {
+        fprintf(stderr, "No free surfaces\n");
+        return AVERROR(ENOMEM);
+    }
+
+    surf = av_mallocz(sizeof(*surf));
+    if (!surf)
+        return AVERROR(ENOMEM);
+    surf_buf = av_buffer_create((uint8_t*)surf, sizeof(*surf), free_buffer,
+                                &decode->surface_used[idx], AV_BUFFER_FLAG_READONLY);
+    if (!surf_buf) {
+        av_freep(&surf);
+        return AVERROR(ENOMEM);
+    }
+
+    surf->Info       = decode->frame_info;
+    surf->Data.MemId = &decode->surfaces[idx];
+
+    frame->buf[0]  = surf_buf;
+    frame->data[3] = (uint8_t*)surf;
+
+    decode->surface_used[idx] = 1;
+
+    return 0;
+}
+
+static int get_format(AVCodecContext *avctx, const enum AVPixelFormat *pix_fmts)
+{
+    while (*pix_fmts != AV_PIX_FMT_NONE) {
+        if (*pix_fmts == AV_PIX_FMT_QSV) {
+            if (!avctx->hwaccel_context) {
+                DecodeContext *decode = avctx->opaque;
+                AVQSVContext *qsv = av_qsv_alloc_context();
+                if (!qsv)
+                    return AV_PIX_FMT_NONE;
+
+                qsv->session   = decode->mfx_session;
+                qsv->iopattern = MFX_IOPATTERN_OUT_VIDEO_MEMORY;
+
+                avctx->hwaccel_context = qsv;
+            }
+
+            return AV_PIX_FMT_QSV;
+        }
+
+        pix_fmts++;
+    }
+
+    fprintf(stderr, "The QSV pixel format not offered in get_format()\n");
+
+    return AV_PIX_FMT_NONE;
+}
+
+static int decode_packet(DecodeContext *decode, AVCodecContext *decoder_ctx,
+                         AVFrame *frame, AVPacket *pkt,
+                         AVIOContext *output_ctx)
+{
+    int ret = 0;
+    int got_frame = 1;
+
+    while (pkt->size > 0 || (!pkt->data && got_frame)) {
+        ret = avcodec_decode_video2(decoder_ctx, frame, &got_frame, pkt);
+        if (ret < 0) {
+            fprintf(stderr, "Error during decoding\n");
+            return ret;
+        }
+
+        pkt->data += ret;
+        pkt->size -= ret;
+
+        /* A real program would do something useful with the decoded frame here.
+         * We just retrieve the raw data and write it to a file, which is rather
+         * useless but pedagogic. */
+        if (got_frame) {
+            mfxFrameSurface1 *surf = (mfxFrameSurface1*)frame->data[3];
+            VASurfaceID    surface = *(VASurfaceID*)surf->Data.MemId;
+
+            VAImageFormat img_fmt = {
+                .fourcc         = VA_FOURCC_NV12,
+                .byte_order     = VA_LSB_FIRST,
+                .bits_per_pixel = 8,
+                .depth          = 8,
+            };
+
+            VAImage img;
+
+            VAStatus err;
+            uint8_t *data;
+            int i, j;
+
+            img.buf      = VA_INVALID_ID;
+            img.image_id = VA_INVALID_ID;
+
+            err = vaCreateImage(decode->va_dpy, &img_fmt,
+                                frame->width, frame->height, &img);
+            if (err != VA_STATUS_SUCCESS) {
+                fprintf(stderr, "Error creating an image: %s\n",
+                        vaErrorStr(err));
+                ret = AVERROR_UNKNOWN;
+                goto fail;
+            }
+
+            err = vaGetImage(decode->va_dpy, surface, 0, 0,
+                             frame->width, frame->height,
+                             img.image_id);
+            if (err != VA_STATUS_SUCCESS) {
+                fprintf(stderr, "Error getting an image: %s\n",
+                        vaErrorStr(err));
+                ret = AVERROR_UNKNOWN;
+                goto fail;
+            }
+
+            err = vaMapBuffer(decode->va_dpy, img.buf, (void**)&data);
+            if (err != VA_STATUS_SUCCESS) {
+                fprintf(stderr, "Error mapping the image buffer: %s\n",
+                        vaErrorStr(err));
+                ret = AVERROR_UNKNOWN;
+                goto fail;
+            }
+
+            for (i = 0; i < img.num_planes; i++)
+                for (j = 0; j < (img.height >> (i > 0)); j++)
+                    avio_write(output_ctx, data + img.offsets[i] + j * img.pitches[i], img.width);
+
+fail:
+            if (img.buf != VA_INVALID_ID)
+                vaUnmapBuffer(decode->va_dpy, img.buf);
+            if (img.image_id != VA_INVALID_ID)
+                vaDestroyImage(decode->va_dpy, img.image_id);
+            av_frame_unref(frame);
+
+            if (ret < 0)
+                return ret;
+        }
+    }
+
+    return 0;
+}
+
+int main(int argc, char **argv)
+{
+    AVFormatContext *input_ctx = NULL;
+    AVStream *video_st = NULL;
+    AVCodecContext *decoder_ctx = NULL;
+    const AVCodec *decoder;
+
+    AVPacket pkt = { 0 };
+    AVFrame *frame = NULL;
+
+    DecodeContext decode = { NULL };
+
+    Display *dpy = NULL;
+    int va_ver_major, va_ver_minor;
+
+    mfxIMPL mfx_impl = MFX_IMPL_AUTO_ANY;
+    mfxVersion mfx_ver = { { 1, 1 } };
+
+    mfxFrameAllocator frame_allocator = {
+        .pthis = &decode,
+        .Alloc = frame_alloc,
+        .Lock  = frame_lock,
+        .Unlock = frame_unlock,
+        .GetHDL = frame_get_hdl,
+        .Free   = frame_free,
+    };
+
+    AVIOContext *output_ctx = NULL;
+
+    int ret, i, err;
+
+    av_register_all();
+
+    if (argc < 3) {
+        fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
+        return 1;
+    }
+
+    /* open the input file */
+    ret = avformat_open_input(&input_ctx, argv[1], NULL, NULL);
+    if (ret < 0) {
+        fprintf(stderr, "Cannot open input file '%s': ", argv[1]);
+        goto finish;
+    }
+
+    /* find the first H.264 video stream */
+    for (i = 0; i < input_ctx->nb_streams; i++) {
+        AVStream *st = input_ctx->streams[i];
+
+        if (st->codecpar->codec_id == AV_CODEC_ID_H264 && !video_st)
+            video_st = st;
+        else
+            st->discard = AVDISCARD_ALL;
+    }
+    if (!video_st) {
+        fprintf(stderr, "No H.264 video stream in the input file\n");
+        goto finish;
+    }
+
+    /* initialize VA-API */
+    dpy = XOpenDisplay(NULL);
+    if (!dpy) {
+        fprintf(stderr, "Cannot open the X display\n");
+        goto finish;
+    }
+    decode.va_dpy = vaGetDisplay(dpy);
+    if (!decode.va_dpy) {
+        fprintf(stderr, "Cannot open the VA display\n");
+        goto finish;
+    }
+
+    err = vaInitialize(decode.va_dpy, &va_ver_major, &va_ver_minor);
+    if (err != VA_STATUS_SUCCESS) {
+        fprintf(stderr, "Cannot initialize VA: %s\n", vaErrorStr(err));
+        goto finish;
+    }
+    fprintf(stderr, "Initialized VA v%d.%d\n", va_ver_major, va_ver_minor);
+
+    /* initialize an MFX session */
+    err = MFXInit(mfx_impl, &mfx_ver, &decode.mfx_session);
+    if (err != MFX_ERR_NONE) {
+        fprintf(stderr, "Error initializing an MFX session\n");
+        goto finish;
+    }
+
+    MFXVideoCORE_SetHandle(decode.mfx_session, MFX_HANDLE_VA_DISPLAY, decode.va_dpy);
+    MFXVideoCORE_SetFrameAllocator(decode.mfx_session, &frame_allocator);
+
+    /* initialize the decoder */
+    decoder = avcodec_find_decoder_by_name("h264_qsv");
+    if (!decoder) {
+        fprintf(stderr, "The QSV decoder is not present in libavcodec\n");
+        goto finish;
+    }
+
+    decoder_ctx = avcodec_alloc_context3(decoder);
+    if (!decoder_ctx) {
+        ret = AVERROR(ENOMEM);
+        goto finish;
+    }
+    decoder_ctx->codec_id = AV_CODEC_ID_H264;
+    if (video_st->codecpar->extradata_size) {
+        decoder_ctx->extradata = av_mallocz(video_st->codecpar->extradata_size +
+                                            AV_INPUT_BUFFER_PADDING_SIZE);
+        if (!decoder_ctx->extradata) {
+            ret = AVERROR(ENOMEM);
+            goto finish;
+        }
+        memcpy(decoder_ctx->extradata, video_st->codecpar->extradata,
+               video_st->codecpar->extradata_size);
+        decoder_ctx->extradata_size = video_st->codecpar->extradata_size;
+    }
+    decoder_ctx->refcounted_frames = 1;
+
+    decoder_ctx->opaque      = &decode;
+    decoder_ctx->get_buffer2 = get_buffer;
+    decoder_ctx->get_format  = get_format;
+
+    ret = avcodec_open2(decoder_ctx, NULL, NULL);
+    if (ret < 0) {
+        fprintf(stderr, "Error opening the decoder: ");
+        goto finish;
+    }
+
+    /* open the output stream */
+    ret = avio_open(&output_ctx, argv[2], AVIO_FLAG_WRITE);
+    if (ret < 0) {
+        fprintf(stderr, "Error opening the output context: ");
+        goto finish;
+    }
+
+    frame = av_frame_alloc();
+    if (!frame) {
+        ret = AVERROR(ENOMEM);
+        goto finish;
+    }
+
+    /* actual decoding */
+    while (ret >= 0) {
+        ret = av_read_frame(input_ctx, &pkt);
+        if (ret < 0)
+            break;
+
+        if (pkt.stream_index == video_st->index)
+            ret = decode_packet(&decode, decoder_ctx, frame, &pkt, output_ctx);
+
+        av_packet_unref(&pkt);
+    }
+
+    /* flush the decoder */
+    pkt.data = NULL;
+    pkt.size = 0;
+    ret = decode_packet(&decode, decoder_ctx, frame, &pkt, output_ctx);
+
+finish:
+    if (ret < 0) {
+        char buf[1024];
+        av_strerror(ret, buf, sizeof(buf));
+        fprintf(stderr, "%s\n", buf);
+    }
+
+    avformat_close_input(&input_ctx);
+
+    av_frame_free(&frame);
+
+    if (decoder_ctx)
+        av_freep(&decoder_ctx->hwaccel_context);
+    avcodec_free_context(&decoder_ctx);
+
+    free_surfaces(&decode);
+
+    if (decode.mfx_session)
+        MFXClose(decode.mfx_session);
+    if (decode.va_dpy)
+        vaTerminate(decode.va_dpy);
+    if (dpy)
+        XCloseDisplay(dpy);
+
+    avio_close(output_ctx);
+
+    return ret;
+}
diff --git a/Music/ffmpeg/doc/examples/remuxing.c b/Music/ffmpeg/doc/examples/remuxing.c
new file mode 100755
index 0000000..65437d9
--- /dev/null
+++ b/Music/ffmpeg/doc/examples/remuxing.c
@@ -0,0 +1,165 @@
+/*
+ * Copyright (c) 2013 Stefano Sabatini
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+/**
+ * @file
+ * libavformat/libavcodec demuxing and muxing API example.
+ *
+ * Remux streams from one container format to another.
+ * @example remuxing.c
+ */
+
+#include <libavutil/timestamp.h>
+#include <libavformat/avformat.h>
+
+static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt, const char *tag)
+{
+    AVRational *time_base = &fmt_ctx->streams[pkt->stream_index]->time_base;
+
+    printf("%s: pts:%s pts_time:%s dts:%s dts_time:%s duration:%s duration_time:%s stream_index:%d\n",
+           tag,
+           av_ts2str(pkt->pts), av_ts2timestr(pkt->pts, time_base),
+           av_ts2str(pkt->dts), av_ts2timestr(pkt->dts, time_base),
+           av_ts2str(pkt->duration), av_ts2timestr(pkt->duration, time_base),
+           pkt->stream_index);
+}
+
+int main(int argc, char **argv)
+{
+    AVOutputFormat *ofmt = NULL;
+    AVFormatContext *ifmt_ctx = NULL, *ofmt_ctx = NULL;
+    AVPacket pkt;
+    const char *in_filename, *out_filename;
+    int ret, i;
+
+    if (argc < 3) {
+        printf("usage: %s input output\n"
+               "API example program to remux a media file with libavformat and libavcodec.\n"
+               "The output format is guessed according to the file extension.\n"
+               "\n", argv[0]);
+        return 1;
+    }
+
+    in_filename  = argv[1];
+    out_filename = argv[2];
+
+    av_register_all();
+
+    if ((ret = avformat_open_input(&ifmt_ctx, in_filename, 0, 0)) < 0) {
+        fprintf(stderr, "Could not open input file '%s'", in_filename);
+        goto end;
+    }
+
+    if ((ret = avformat_find_stream_info(ifmt_ctx, 0)) < 0) {
+        fprintf(stderr, "Failed to retrieve input stream information");
+        goto end;
+    }
+
+    av_dump_format(ifmt_ctx, 0, in_filename, 0);
+
+    avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, out_filename);
+    if (!ofmt_ctx) {
+        fprintf(stderr, "Could not create output context\n");
+        ret = AVERROR_UNKNOWN;
+        goto end;
+    }
+
+    ofmt = ofmt_ctx->oformat;
+
+    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
+        AVStream *in_stream = ifmt_ctx->streams[i];
+        AVStream *out_stream = avformat_new_stream(ofmt_ctx, in_stream->codec->codec);
+        if (!out_stream) {
+            fprintf(stderr, "Failed allocating output stream\n");
+            ret = AVERROR_UNKNOWN;
+            goto end;
+        }
+
+        ret = avcodec_copy_context(out_stream->codec, in_stream->codec);
+        if (ret < 0) {
+            fprintf(stderr, "Failed to copy context from input to output stream codec context\n");
+            goto end;
+        }
+        out_stream->codec->codec_tag = 0;
+        if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
+            out_stream->codec->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
+    }
+    av_dump_format(ofmt_ctx, 0, out_filename, 1);
+
+    if (!(ofmt->flags & AVFMT_NOFILE)) {
+        ret = avio_open(&ofmt_ctx->pb, out_filename, AVIO_FLAG_WRITE);
+        if (ret < 0) {
+            fprintf(stderr, "Could not open output file '%s'", out_filename);
+            goto end;
+        }
+    }
+
+    ret = avformat_write_header(ofmt_ctx, NULL);
+    if (ret < 0) {
+        fprintf(stderr, "Error occurred when opening output file\n");
+        goto end;
+    }
+
+    while (1) {
+        AVStream *in_stream, *out_stream;
+
+        ret = av_read_frame(ifmt_ctx, &pkt);
+        if (ret < 0)
+            break;
+
+        in_stream  = ifmt_ctx->streams[pkt.stream_index];
+        out_stream = ofmt_ctx->streams[pkt.stream_index];
+
+        log_packet(ifmt_ctx, &pkt, "in");
+
+        /* copy packet */
+        pkt.pts = av_rescale_q_rnd(pkt.pts, in_stream->time_base, out_stream->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
+        pkt.dts = av_rescale_q_rnd(pkt.dts, in_stream->time_base, out_stream->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
+        pkt.duration = av_rescale_q(pkt.duration, in_stream->time_base, out_stream->time_base);
+        pkt.pos = -1;
+        log_packet(ofmt_ctx, &pkt, "out");
+
+        ret = av_interleaved_write_frame(ofmt_ctx, &pkt);
+        if (ret < 0) {
+            fprintf(stderr, "Error muxing packet\n");
+            break;
+        }
+        av_packet_unref(&pkt);
+    }
+
+    av_write_trailer(ofmt_ctx);
+end:
+
+    avformat_close_input(&ifmt_ctx);
+
+    /* close output */
+    if (ofmt_ctx && !(ofmt->flags & AVFMT_NOFILE))
+        avio_closep(&ofmt_ctx->pb);
+    avformat_free_context(ofmt_ctx);
+
+    if (ret < 0 && ret != AVERROR_EOF) {
+        fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
+        return 1;
+    }
+
+    return 0;
+}
diff --git a/Music/ffmpeg/doc/examples/resampling_audio.c b/Music/ffmpeg/doc/examples/resampling_audio.c
new file mode 100755
index 0000000..f35e7e1
--- /dev/null
+++ b/Music/ffmpeg/doc/examples/resampling_audio.c
@@ -0,0 +1,214 @@
+/*
+ * Copyright (c) 2012 Stefano Sabatini
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+/**
+ * @example resampling_audio.c
+ * libswresample API use example.
+ */
+
+#include <libavutil/opt.h>
+#include <libavutil/channel_layout.h>
+#include <libavutil/samplefmt.h>
+#include <libswresample/swresample.h>
+
+static int get_format_from_sample_fmt(const char **fmt,
+                                      enum AVSampleFormat sample_fmt)
+{
+    int i;
+    struct sample_fmt_entry {
+        enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
+    } sample_fmt_entries[] = {
+        { AV_SAMPLE_FMT_U8,  "u8",    "u8"    },
+        { AV_SAMPLE_FMT_S16, "s16be", "s16le" },
+        { AV_SAMPLE_FMT_S32, "s32be", "s32le" },
+        { AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
+        { AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
+    };
+    *fmt = NULL;
+
+    for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
+        struct sample_fmt_entry *entry = &sample_fmt_entries[i];
+        if (sample_fmt == entry->sample_fmt) {
+            *fmt = AV_NE(entry->fmt_be, entry->fmt_le);
+            return 0;
+        }
+    }
+
+    fprintf(stderr,
+            "Sample format %s not supported as output format\n",
+            av_get_sample_fmt_name(sample_fmt));
+    return AVERROR(EINVAL);
+}
+
+/**
+ * Fill dst buffer with nb_samples, generated starting from t.
+ */
+static void fill_samples(double *dst, int nb_samples, int nb_channels, int sample_rate, double *t)
+{
+    int i, j;
+    double tincr = 1.0 / sample_rate, *dstp = dst;
+    const double c = 2 * M_PI * 440.0;
+
+    /* generate sin tone with 440Hz frequency and duplicated channels */
+    for (i = 0; i < nb_samples; i++) {
+        *dstp = sin(c * *t);
+        for (j = 1; j < nb_channels; j++)
+            dstp[j] = dstp[0];
+        dstp += nb_channels;
+        *t += tincr;
+    }
+}
+
+int main(int argc, char **argv)
+{
+    int64_t src_ch_layout = AV_CH_LAYOUT_STEREO, dst_ch_layout = AV_CH_LAYOUT_SURROUND;
+    int src_rate = 48000, dst_rate = 44100;
+    uint8_t **src_data = NULL, **dst_data = NULL;
+    int src_nb_channels = 0, dst_nb_channels = 0;
+    int src_linesize, dst_linesize;
+    int src_nb_samples = 1024, dst_nb_samples, max_dst_nb_samples;
+    enum AVSampleFormat src_sample_fmt = AV_SAMPLE_FMT_DBL, dst_sample_fmt = AV_SAMPLE_FMT_S16;
+    const char *dst_filename = NULL;
+    FILE *dst_file;
+    int dst_bufsize;
+    const char *fmt;
+    struct SwrContext *swr_ctx;
+    double t;
+    int ret;
+
+    if (argc != 2) {
+        fprintf(stderr, "Usage: %s output_file\n"
+                "API example program to show how to resample an audio stream with libswresample.\n"
+                "This program generates a series of audio frames, resamples them to a specified "
+                "output format and rate and saves them to an output file named output_file.\n",
+            argv[0]);
+        exit(1);
+    }
+    dst_filename = argv[1];
+
+    dst_file = fopen(dst_filename, "wb");
+    if (!dst_file) {
+        fprintf(stderr, "Could not open destination file %s\n", dst_filename);
+        exit(1);
+    }
+
+    /* create resampler context */
+    swr_ctx = swr_alloc();
+    if (!swr_ctx) {
+        fprintf(stderr, "Could not allocate resampler context\n");
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }
+
+    /* set options */
+    av_opt_set_int(swr_ctx, "in_channel_layout",    src_ch_layout, 0);
+    av_opt_set_int(swr_ctx, "in_sample_rate",       src_rate, 0);
+    av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", src_sample_fmt, 0);
+
+    av_opt_set_int(swr_ctx, "out_channel_layout",    dst_ch_layout, 0);
+    av_opt_set_int(swr_ctx, "out_sample_rate",       dst_rate, 0);
+    av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", dst_sample_fmt, 0);
+
+    /* initialize the resampling context */
+    if ((ret = swr_init(swr_ctx)) < 0) {
+        fprintf(stderr, "Failed to initialize the resampling context\n");
+        goto end;
+    }
+
+    /* allocate source and destination samples buffers */
+
+    src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout);
+    ret = av_samples_alloc_array_and_samples(&src_data, &src_linesize, src_nb_channels,
+                                             src_nb_samples, src_sample_fmt, 0);
+    if (ret < 0) {
+        fprintf(stderr, "Could not allocate source samples\n");
+        goto end;
+    }
+
+    /* compute the number of converted samples: buffering is avoided
+     * ensuring that the output buffer will contain at least all the
+     * converted input samples */
+    max_dst_nb_samples = dst_nb_samples =
+        av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
+
+    /* buffer is going to be directly written to a rawaudio file, no alignment */
+    dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout);
+    ret = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize, dst_nb_channels,
+                                             dst_nb_samples, dst_sample_fmt, 0);
+    if (ret < 0) {
+        fprintf(stderr, "Could not allocate destination samples\n");
+        goto end;
+    }
+
+    t = 0;
+    do {
+        /* generate synthetic audio */
+        fill_samples((double *)src_data[0], src_nb_samples, src_nb_channels, src_rate, &t);
+
+        /* compute destination number of samples */
+        dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, src_rate) +
+                                        src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
+        if (dst_nb_samples > max_dst_nb_samples) {
+            av_freep(&dst_data[0]);
+            ret = av_samples_alloc(dst_data, &dst_linesize, dst_nb_channels,
+                                   dst_nb_samples, dst_sample_fmt, 1);
+            if (ret < 0)
+                break;
+            max_dst_nb_samples = dst_nb_samples;
+        }
+
+        /* convert to destination format */
+        ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, (const uint8_t **)src_data, src_nb_samples);
+        if (ret < 0) {
+            fprintf(stderr, "Error while converting\n");
+            goto end;
+        }
+        dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels,
+                                                 ret, dst_sample_fmt, 1);
+        if (dst_bufsize < 0) {
+            fprintf(stderr, "Could not get sample buffer size\n");
+            goto end;
+        }
+        printf("t:%f in:%d out:%d\n", t, src_nb_samples, ret);
+        fwrite(dst_data[0], 1, dst_bufsize, dst_file);
+    } while (t < 10);
+
+    if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt)) < 0)
+        goto end;
+    fprintf(stderr, "Resampling succeeded. Play the output file with the command:\n"
+            "ffplay -f %s -channel_layout %"PRId64" -channels %d -ar %d %s\n",
+            fmt, dst_ch_layout, dst_nb_channels, dst_rate, dst_filename);
+
+end:
+    fclose(dst_file);
+
+    if (src_data)
+        av_freep(&src_data[0]);
+    av_freep(&src_data);
+
+    if (dst_data)
+        av_freep(&dst_data[0]);
+    av_freep(&dst_data);
+
+    swr_free(&swr_ctx);
+    return ret < 0;
+}
diff --git a/Music/ffmpeg/doc/examples/scaling_video.c b/Music/ffmpeg/doc/examples/scaling_video.c
new file mode 100755
index 0000000..587f3ab
--- /dev/null
+++ b/Music/ffmpeg/doc/examples/scaling_video.c
@@ -0,0 +1,140 @@
+/*
+ * Copyright (c) 2012 Stefano Sabatini
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+/**
+ * @file
+ * libswscale API use example.
+ * @example scaling_video.c
+ */
+
+#include <libavutil/imgutils.h>
+#include <libavutil/parseutils.h>
+#include <libswscale/swscale.h>
+
+static void fill_yuv_image(uint8_t *data[4], int linesize[4],
+                           int width, int height, int frame_index)
+{
+    int x, y;
+
+    /* Y */
+    for (y = 0; y < height; y++)
+        for (x = 0; x < width; x++)
+            data[0][y * linesize[0] + x] = x + y + frame_index * 3;
+
+    /* Cb and Cr */
+    for (y = 0; y < height / 2; y++) {
+        for (x = 0; x < width / 2; x++) {
+            data[1][y * linesize[1] + x] = 128 + y + frame_index * 2;
+            data[2][y * linesize[2] + x] = 64 + x + frame_index * 5;
+        }
+    }
+}
+
+int main(int argc, char **argv)
+{
+    uint8_t *src_data[4], *dst_data[4];
+    int src_linesize[4], dst_linesize[4];
+    int src_w = 320, src_h = 240, dst_w, dst_h;
+    enum AVPixelFormat src_pix_fmt = AV_PIX_FMT_YUV420P, dst_pix_fmt = AV_PIX_FMT_RGB24;
+    const char *dst_size = NULL;
+    const char *dst_filename = NULL;
+    FILE *dst_file;
+    int dst_bufsize;
+    struct SwsContext *sws_ctx;
+    int i, ret;
+
+    if (argc != 3) {
+        fprintf(stderr, "Usage: %s output_file output_size\n"
+                "API example program to show how to scale an image with libswscale.\n"
+                "This program generates a series of pictures, rescales them to the given "
+                "output_size and saves them to an output file named output_file\n."
+                "\n", argv[0]);
+        exit(1);
+    }
+    dst_filename = argv[1];
+    dst_size     = argv[2];
+
+    if (av_parse_video_size(&dst_w, &dst_h, dst_size) < 0) {
+        fprintf(stderr,
+                "Invalid size '%s', must be in the form WxH or a valid size abbreviation\n",
+                dst_size);
+        exit(1);
+    }
+
+    dst_file = fopen(dst_filename, "wb");
+    if (!dst_file) {
+        fprintf(stderr, "Could not open destination file %s\n", dst_filename);
+        exit(1);
+    }
+
+    /* create scaling context */
+    sws_ctx = sws_getContext(src_w, src_h, src_pix_fmt,
+                             dst_w, dst_h, dst_pix_fmt,
+                             SWS_BILINEAR, NULL, NULL, NULL);
+    if (!sws_ctx) {
+        fprintf(stderr,
+                "Impossible to create scale context for the conversion "
+                "fmt:%s s:%dx%d -> fmt:%s s:%dx%d\n",
+                av_get_pix_fmt_name(src_pix_fmt), src_w, src_h,
+                av_get_pix_fmt_name(dst_pix_fmt), dst_w, dst_h);
+        ret = AVERROR(EINVAL);
+        goto end;
+    }
+
+    /* allocate source and destination image buffers */
+    if ((ret = av_image_alloc(src_data, src_linesize,
+                              src_w, src_h, src_pix_fmt, 16)) < 0) {
+        fprintf(stderr, "Could not allocate source image\n");
+        goto end;
+    }
+
+    /* buffer is going to be written to rawvideo file, no alignment */
+    if ((ret = av_image_alloc(dst_data, dst_linesize,
+                              dst_w, dst_h, dst_pix_fmt, 1)) < 0) {
+        fprintf(stderr, "Could not allocate destination image\n");
+        goto end;
+    }
+    dst_bufsize = ret;
+
+    for (i = 0; i < 100; i++) {
+        /* generate synthetic video */
+        fill_yuv_image(src_data, src_linesize, src_w, src_h, i);
+
+        /* convert to destination format */
+        sws_scale(sws_ctx, (const uint8_t * const*)src_data,
+                  src_linesize, 0, src_h, dst_data, dst_linesize);
+
+        /* write scaled image to file */
+        fwrite(dst_data[0], 1, dst_bufsize, dst_file);
+    }
+
+    fprintf(stderr, "Scaling succeeded. Play the output file with the command:\n"
+           "ffplay -f rawvideo -pix_fmt %s -video_size %dx%d %s\n",
+           av_get_pix_fmt_name(dst_pix_fmt), dst_w, dst_h, dst_filename);
+
+end:
+    fclose(dst_file);
+    av_freep(&src_data[0]);
+    av_freep(&dst_data[0]);
+    sws_freeContext(sws_ctx);
+    return ret < 0;
+}
diff --git a/Music/ffmpeg/doc/examples/transcode_aac.c b/Music/ffmpeg/doc/examples/transcode_aac.c
new file mode 100755
index 0000000..9b3ee67
--- /dev/null
+++ b/Music/ffmpeg/doc/examples/transcode_aac.c
@@ -0,0 +1,802 @@
+/*
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg 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
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * simple audio converter
+ *
+ * @example transcode_aac.c
+ * Convert an input audio file to AAC in an MP4 container using FFmpeg.
+ * @author Andreas Unterweger (dustsigns@gmail.com)
+ */
+
+#include <stdio.h>
+
+#include "libavformat/avformat.h"
+#include "libavformat/avio.h"
+
+#include "libavcodec/avcodec.h"
+
+#include "libavutil/audio_fifo.h"
+#include "libavutil/avassert.h"
+#include "libavutil/avstring.h"
+#include "libavutil/frame.h"
+#include "libavutil/opt.h"
+
+#include "libswresample/swresample.h"
+
+/** The output bit rate in kbit/s */
+#define OUTPUT_BIT_RATE 96000
+/** The number of output channels */
+#define OUTPUT_CHANNELS 2
+
+/**
+ * Convert an error code into a text message.
+ * @param error Error code to be converted
+ * @return Corresponding error text (not thread-safe)
+ */
+static const char *get_error_text(const int error)
+{
+    static char error_buffer[255];
+    av_strerror(error, error_buffer, sizeof(error_buffer));
+    return error_buffer;
+}
+
+/** Open an input file and the required decoder. */
+static int open_input_file(const char *filename,
+                           AVFormatContext **input_format_context,
+                           AVCodecContext **input_codec_context)
+{
+    AVCodecContext *avctx;
+    AVCodec *input_codec;
+    int error;
+
+    /** Open the input file to read from it. */
+    if ((error = avformat_open_input(input_format_context, filename, NULL,
+                                     NULL)) < 0) {
+        fprintf(stderr, "Could not open input file '%s' (error '%s')\n",
+                filename, get_error_text(error));
+        *input_format_context = NULL;
+        return error;
+    }
+
+    /** Get information on the input file (number of streams etc.). */
+    if ((error = avformat_find_stream_info(*input_format_context, NULL)) < 0) {
+        fprintf(stderr, "Could not open find stream info (error '%s')\n",
+                get_error_text(error));
+        avformat_close_input(input_format_context);
+        return error;
+    }
+
+    /** Make sure that there is only one stream in the input file. */
+    if ((*input_format_context)->nb_streams != 1) {
+        fprintf(stderr, "Expected one audio input stream, but found %d\n",
+                (*input_format_context)->nb_streams);
+        avformat_close_input(input_format_context);
+        return AVERROR_EXIT;
+    }
+
+    /** Find a decoder for the audio stream. */
+    if (!(input_codec = avcodec_find_decoder((*input_format_context)->streams[0]->codecpar->codec_id))) {
+        fprintf(stderr, "Could not find input codec\n");
+        avformat_close_input(input_format_context);
+        return AVERROR_EXIT;
+    }
+
+    /** allocate a new decoding context */
+    avctx = avcodec_alloc_context3(input_codec);
+    if (!avctx) {
+        fprintf(stderr, "Could not allocate a decoding context\n");
+        avformat_close_input(input_format_context);
+        return AVERROR(ENOMEM);
+    }
+
+    /** initialize the stream parameters with demuxer information */
+    error = avcodec_parameters_to_context(avctx, (*input_format_context)->streams[0]->codecpar);
+    if (error < 0) {
+        avformat_close_input(input_format_context);
+        avcodec_free_context(&avctx);
+        return error;
+    }
+
+    /** Open the decoder for the audio stream to use it later. */
+    if ((error = avcodec_open2(avctx, input_codec, NULL)) < 0) {
+        fprintf(stderr, "Could not open input codec (error '%s')\n",
+                get_error_text(error));
+        avcodec_free_context(&avctx);
+        avformat_close_input(input_format_context);
+        return error;
+    }
+
+    /** Save the decoder context for easier access later. */
+    *input_codec_context = avctx;
+
+    return 0;
+}
+
+/**
+ * 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(const char *filename,
+                            AVCodecContext *input_codec_context,
+                            AVFormatContext **output_format_context,
+                            AVCodecContext **output_codec_context)
+{
+    AVCodecContext *avctx          = NULL;
+    AVIOContext *output_io_context = NULL;
+    AVStream *stream               = NULL;
+    AVCodec *output_codec          = NULL;
+    int error;
+
+    /** Open the output file to write to it. */
+    if ((error = avio_open(&output_io_context, filename,
+                           AVIO_FLAG_WRITE)) < 0) {
+        fprintf(stderr, "Could not open output file '%s' (error '%s')\n",
+                filename, get_error_text(error));
+        return error;
+    }
+
+    /** Create a new format context for the output container format. */
+    if (!(*output_format_context = avformat_alloc_context())) {
+        fprintf(stderr, "Could not allocate output format context\n");
+        return AVERROR(ENOMEM);
+    }
+
+    /** Associate the output file (pointer) with the container format context. */
+    (*output_format_context)->pb = output_io_context;
+
+    /** Guess the desired container format based on the file extension. */
+    if (!((*output_format_context)->oformat = av_guess_format(NULL, filename,
+                                                              NULL))) {
+        fprintf(stderr, "Could not find output file format\n");
+        goto cleanup;
+    }
+
+    av_strlcpy((*output_format_context)->filename, filename,
+               sizeof((*output_format_context)->filename));
+
+    /** Find the encoder to be used by its name. */
+    if (!(output_codec = avcodec_find_encoder(AV_CODEC_ID_AAC))) {
+        fprintf(stderr, "Could not find an AAC encoder.\n");
+        goto cleanup;
+    }
+
+    /** Create a new audio stream in the output file container. */
+    if (!(stream = avformat_new_stream(*output_format_context, NULL))) {
+        fprintf(stderr, "Could not create new stream\n");
+        error = AVERROR(ENOMEM);
+        goto cleanup;
+    }
+
+    avctx = avcodec_alloc_context3(output_codec);
+    if (!avctx) {
+        fprintf(stderr, "Could not allocate an encoding context\n");
+        error = AVERROR(ENOMEM);
+        goto cleanup;
+    }
+
+    /**
+     * Set the basic encoder parameters.
+     * The input file's sample rate is used to avoid a sample rate conversion.
+     */
+    avctx->channels       = OUTPUT_CHANNELS;
+    avctx->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
+    avctx->sample_rate    = input_codec_context->sample_rate;
+    avctx->sample_fmt     = output_codec->sample_fmts[0];
+    avctx->bit_rate       = OUTPUT_BIT_RATE;
+
+    /** Allow the use of the experimental AAC encoder */
+    avctx->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
+
+    /** Set the sample rate for the container. */
+    stream->time_base.den = input_codec_context->sample_rate;
+    stream->time_base.num = 1;
+
+    /**
+     * Some container formats (like MP4) require global headers to be present
+     * Mark the encoder so that it behaves accordingly.
+     */
+    if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
+        avctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
+
+    /** Open the encoder for the audio stream to use it later. */
+    if ((error = avcodec_open2(avctx, output_codec, NULL)) < 0) {
+        fprintf(stderr, "Could not open output codec (error '%s')\n",
+                get_error_text(error));
+        goto cleanup;
+    }
+
+    error = avcodec_parameters_from_context(stream->codecpar, avctx);
+    if (error < 0) {
+        fprintf(stderr, "Could not initialize stream parameters\n");
+        goto cleanup;
+    }
+
+    /** Save the encoder context for easier access later. */
+    *output_codec_context = avctx;
+
+    return 0;
+
+cleanup:
+    avcodec_free_context(&avctx);
+    avio_closep(&(*output_format_context)->pb);
+    avformat_free_context(*output_format_context);
+    *output_format_context = NULL;
+    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())) {
+        fprintf(stderr, "Could not allocate input frame\n");
+        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
+ * libswresample takes care of this, but requires initialization.
+ */
+static int init_resampler(AVCodecContext *input_codec_context,
+                          AVCodecContext *output_codec_context,
+                          SwrContext **resample_context)
+{
+        int error;
+
+        /**
+         * Create a resampler context for the conversion.
+         * 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).
+         */
+        *resample_context = swr_alloc_set_opts(NULL,
+                                              av_get_default_channel_layout(output_codec_context->channels),
+                                              output_codec_context->sample_fmt,
+                                              output_codec_context->sample_rate,
+                                              av_get_default_channel_layout(input_codec_context->channels),
+                                              input_codec_context->sample_fmt,
+                                              input_codec_context->sample_rate,
+                                              0, NULL);
+        if (!*resample_context) {
+            fprintf(stderr, "Could not allocate resample context\n");
+            return AVERROR(ENOMEM);
+        }
+        /**
+        * 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
+        */
+        av_assert0(output_codec_context->sample_rate == input_codec_context->sample_rate);
+
+        /** Open the resampler with the specified parameters. */
+        if ((error = swr_init(*resample_context)) < 0) {
+            fprintf(stderr, "Could not open resample context\n");
+            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, AVCodecContext *output_codec_context)
+{
+    /** Create the FIFO buffer based on the specified output sample format. */
+    if (!(*fifo = av_audio_fifo_alloc(output_codec_context->sample_fmt,
+                                      output_codec_context->channels, 1))) {
+        fprintf(stderr, "Could not allocate FIFO\n");
+        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) {
+        fprintf(stderr, "Could not write output file header (error '%s')\n",
+                get_error_text(error));
+        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 *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. */
+    if ((error = av_read_frame(input_format_context, &input_packet)) < 0) {
+        /** If we are at the end of the file, flush the decoder below. */
+        if (error == AVERROR_EOF)
+            *finished = 1;
+        else {
+            fprintf(stderr, "Could not read frame (error '%s')\n",
+                    get_error_text(error));
+            return error;
+        }
+    }
+
+    /**
+     * 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) {
+        fprintf(stderr, "Could not decode frame (error '%s')\n",
+                get_error_text(error));
+        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,
+                                  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)))) {
+        fprintf(stderr, "Could not allocate converted input sample pointers\n");
+        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, NULL,
+                                  output_codec_context->channels,
+                                  frame_size,
+                                  output_codec_context->sample_fmt, 0)) < 0) {
+        fprintf(stderr,
+                "Could not allocate converted input samples (error '%s')\n",
+                get_error_text(error));
+        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(const uint8_t **input_data,
+                           uint8_t **converted_data, const int frame_size,
+                           SwrContext *resample_context)
+{
+    int error;
+
+    /** Convert the samples using the resampler. */
+    if ((error = swr_convert(resample_context,
+                             converted_data, frame_size,
+                             input_data    , frame_size)) < 0) {
+        fprintf(stderr, "Could not convert input samples (error '%s')\n",
+                get_error_text(error));
+        return error;
+    }
+
+    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) {
+        fprintf(stderr, "Could not reallocate FIFO\n");
+        return error;
+    }
+
+    /** Store the new samples in the FIFO buffer. */
+    if (av_audio_fifo_write(fifo, (void **)converted_input_samples,
+                            frame_size) < frame_size) {
+        fprintf(stderr, "Could not write data to FIFO\n");
+        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 *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))
+        goto cleanup;
+    /** Decode one frame worth of audio samples. */
+    if (decode_audio_frame(input_frame, input_format_context,
+                           input_codec_context, &data_present, finished))
+        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;
+        goto cleanup;
+    }
+    /** If there is decoded data, convert and store it */
+    if (data_present) {
+        /** Initialize the temporary storage for the converted input samples. */
+        if (init_converted_samples(&converted_input_samples, output_codec_context,
+                                   input_frame->nb_samples))
+            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((const uint8_t**)input_frame->extended_data, converted_input_samples,
+                            input_frame->nb_samples, resampler_context))
+            goto cleanup;
+
+        /** Add the converted input samples to the FIFO buffer for later processing. */
+        if (add_samples_to_fifo(fifo, converted_input_samples,
+                                input_frame->nb_samples))
+            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())) {
+        fprintf(stderr, "Could not allocate output frame\n");
+        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;
+
+    /**
+     * 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) {
+        fprintf(stderr, "Could allocate output frame samples (error '%s')\n",
+                get_error_text(error));
+        av_frame_free(frame);
+        return error;
+    }
+
+    return 0;
+}
+
+/** Global timestamp for the audio frames */
+static int64_t pts = 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);
+
+    /** Set a timestamp based on the sample rate for the container. */
+    if (frame) {
+        frame->pts = pts;
+        pts += frame->nb_samples;
+    }
+
+    /**
+     * 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) {
+        fprintf(stderr, "Could not encode frame (error '%s')\n",
+                get_error_text(error));
+        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) {
+            fprintf(stderr, "Could not write frame (error '%s')\n",
+                    get_error_text(error));
+            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) {
+        fprintf(stderr, "Could not read data from FIFO\n");
+        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) {
+        fprintf(stderr, "Could not write output file trailer (error '%s')\n",
+                get_error_text(error));
+        return error;
+    }
+    return 0;
+}
+
+/** Convert an audio file to an AAC file in an MP4 container. */
+int main(int argc, char **argv)
+{
+    AVFormatContext *input_format_context = NULL, *output_format_context = NULL;
+    AVCodecContext *input_codec_context = NULL, *output_codec_context = NULL;
+    SwrContext *resample_context = NULL;
+    AVAudioFifo *fifo = NULL;
+    int ret = AVERROR_EXIT;
+
+    if (argc < 3) {
+        fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
+        exit(1);
+    }
+
+    /** Register all codecs and formats so that they can be used. */
+    av_register_all();
+    /** Open the input file for reading. */
+    if (open_input_file(argv[1], &input_format_context,
+                        &input_codec_context))
+        goto cleanup;
+    /** Open the output file for writing. */
+    if (open_output_file(argv[2], input_codec_context,
+                         &output_format_context, &output_codec_context))
+        goto cleanup;
+    /** Initialize the resampler to be able to convert audio sample formats. */
+    if (init_resampler(input_codec_context, output_codec_context,
+                       &resample_context))
+        goto cleanup;
+    /** Initialize the FIFO buffer to store audio samples to be encoded. */
+    if (init_fifo(&fifo, output_codec_context))
+        goto cleanup;
+    /** Write the header of the output file container. */
+    if (write_output_file_header(output_format_context))
+        goto cleanup;
+
+    /**
+     * 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, input_format_context,
+                                              input_codec_context,
+                                              output_codec_context,
+                                              resample_context, &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;
+        }
+
+        /**
+         * 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 {
+                if (encode_audio_frame(NULL, output_format_context,
+                                       output_codec_context, &data_written))
+                    goto cleanup;
+            } while (data_written);
+            break;
+        }
+    }
+
+    /** Write the trailer of the output file container. */
+    if (write_output_file_trailer(output_format_context))
+        goto cleanup;
+    ret = 0;
+
+cleanup:
+    if (fifo)
+        av_audio_fifo_free(fifo);
+    swr_free(&resample_context);
+    if (output_codec_context)
+        avcodec_free_context(&output_codec_context);
+    if (output_format_context) {
+        avio_closep(&output_format_context->pb);
+        avformat_free_context(output_format_context);
+    }
+    if (input_codec_context)
+        avcodec_free_context(&input_codec_context);
+    if (input_format_context)
+        avformat_close_input(&input_format_context);
+
+    return ret;
+}
diff --git a/Music/ffmpeg/doc/examples/transcoding.c b/Music/ffmpeg/doc/examples/transcoding.c
new file mode 100755
index 0000000..8633362
--- /dev/null
+++ b/Music/ffmpeg/doc/examples/transcoding.c
@@ -0,0 +1,585 @@
+/*
+ * Copyright (c) 2010 Nicolas George
+ * Copyright (c) 2011 Stefano Sabatini
+ * Copyright (c) 2014 Andrey Utkin
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+/**
+ * @file
+ * API example for demuxing, decoding, filtering, encoding and muxing
+ * @example transcoding.c
+ */
+
+#include <libavcodec/avcodec.h>
+#include <libavformat/avformat.h>
+#include <libavfilter/avfiltergraph.h>
+#include <libavfilter/buffersink.h>
+#include <libavfilter/buffersrc.h>
+#include <libavutil/opt.h>
+#include <libavutil/pixdesc.h>
+
+static AVFormatContext *ifmt_ctx;
+static AVFormatContext *ofmt_ctx;
+typedef struct FilteringContext {
+    AVFilterContext *buffersink_ctx;
+    AVFilterContext *buffersrc_ctx;
+    AVFilterGraph *filter_graph;
+} FilteringContext;
+static FilteringContext *filter_ctx;
+
+static int open_input_file(const char *filename)
+{
+    int ret;
+    unsigned int i;
+
+    ifmt_ctx = NULL;
+    if ((ret = avformat_open_input(&ifmt_ctx, filename, NULL, NULL)) < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
+        return ret;
+    }
+
+    if ((ret = avformat_find_stream_info(ifmt_ctx, NULL)) < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
+        return ret;
+    }
+
+    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
+        AVStream *stream;
+        AVCodecContext *codec_ctx;
+        stream = ifmt_ctx->streams[i];
+        codec_ctx = stream->codec;
+        /* Reencode video & audio and remux subtitles etc. */
+        if (codec_ctx->codec_type == AVMEDIA_TYPE_VIDEO
+                || codec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
+            /* Open decoder */
+            ret = avcodec_open2(codec_ctx,
+                    avcodec_find_decoder(codec_ctx->codec_id), NULL);
+            if (ret < 0) {
+                av_log(NULL, AV_LOG_ERROR, "Failed to open decoder for stream #%u\n", i);
+                return ret;
+            }
+        }
+    }
+
+    av_dump_format(ifmt_ctx, 0, filename, 0);
+    return 0;
+}
+
+static int open_output_file(const char *filename)
+{
+    AVStream *out_stream;
+    AVStream *in_stream;
+    AVCodecContext *dec_ctx, *enc_ctx;
+    AVCodec *encoder;
+    int ret;
+    unsigned int i;
+
+    ofmt_ctx = NULL;
+    avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, filename);
+    if (!ofmt_ctx) {
+        av_log(NULL, AV_LOG_ERROR, "Could not create output context\n");
+        return AVERROR_UNKNOWN;
+    }
+
+
+    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
+        out_stream = avformat_new_stream(ofmt_ctx, NULL);
+        if (!out_stream) {
+            av_log(NULL, AV_LOG_ERROR, "Failed allocating output stream\n");
+            return AVERROR_UNKNOWN;
+        }
+
+        in_stream = ifmt_ctx->streams[i];
+        dec_ctx = in_stream->codec;
+        enc_ctx = out_stream->codec;
+
+        if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO
+                || dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
+            /* in this example, we choose transcoding to same codec */
+            encoder = avcodec_find_encoder(dec_ctx->codec_id);
+            if (!encoder) {
+                av_log(NULL, AV_LOG_FATAL, "Necessary encoder not found\n");
+                return AVERROR_INVALIDDATA;
+            }
+
+            /* In this example, we transcode to same properties (picture size,
+             * sample rate etc.). These properties can be changed for output
+             * streams easily using filters */
+            if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO) {
+                enc_ctx->height = dec_ctx->height;
+                enc_ctx->width = dec_ctx->width;
+                enc_ctx->sample_aspect_ratio = dec_ctx->sample_aspect_ratio;
+                /* take first format from list of supported formats */
+                if (encoder->pix_fmts)
+                    enc_ctx->pix_fmt = encoder->pix_fmts[0];
+                else
+                    enc_ctx->pix_fmt = dec_ctx->pix_fmt;
+                /* video time_base can be set to whatever is handy and supported by encoder */
+                enc_ctx->time_base = dec_ctx->time_base;
+            } else {
+                enc_ctx->sample_rate = dec_ctx->sample_rate;
+                enc_ctx->channel_layout = dec_ctx->channel_layout;
+                enc_ctx->channels = av_get_channel_layout_nb_channels(enc_ctx->channel_layout);
+                /* take first format from list of supported formats */
+                enc_ctx->sample_fmt = encoder->sample_fmts[0];
+                enc_ctx->time_base = (AVRational){1, enc_ctx->sample_rate};
+            }
+
+            /* Third parameter can be used to pass settings to encoder */
+            ret = avcodec_open2(enc_ctx, encoder, NULL);
+            if (ret < 0) {
+                av_log(NULL, AV_LOG_ERROR, "Cannot open video encoder for stream #%u\n", i);
+                return ret;
+            }
+        } else if (dec_ctx->codec_type == AVMEDIA_TYPE_UNKNOWN) {
+            av_log(NULL, AV_LOG_FATAL, "Elementary stream #%d is of unknown type, cannot proceed\n", i);
+            return AVERROR_INVALIDDATA;
+        } else {
+            /* if this stream must be remuxed */
+            ret = avcodec_copy_context(ofmt_ctx->streams[i]->codec,
+                    ifmt_ctx->streams[i]->codec);
+            if (ret < 0) {
+                av_log(NULL, AV_LOG_ERROR, "Copying stream context failed\n");
+                return ret;
+            }
+        }
+
+        if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
+            enc_ctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
+
+    }
+    av_dump_format(ofmt_ctx, 0, filename, 1);
+
+    if (!(ofmt_ctx->oformat->flags & AVFMT_NOFILE)) {
+        ret = avio_open(&ofmt_ctx->pb, filename, AVIO_FLAG_WRITE);
+        if (ret < 0) {
+            av_log(NULL, AV_LOG_ERROR, "Could not open output file '%s'", filename);
+            return ret;
+        }
+    }
+
+    /* init muxer, write output file header */
+    ret = avformat_write_header(ofmt_ctx, NULL);
+    if (ret < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Error occurred when opening output file\n");
+        return ret;
+    }
+
+    return 0;
+}
+
+static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx,
+        AVCodecContext *enc_ctx, const char *filter_spec)
+{
+    char args[512];
+    int ret = 0;
+    AVFilter *buffersrc = NULL;
+    AVFilter *buffersink = NULL;
+    AVFilterContext *buffersrc_ctx = NULL;
+    AVFilterContext *buffersink_ctx = NULL;
+    AVFilterInOut *outputs = avfilter_inout_alloc();
+    AVFilterInOut *inputs  = avfilter_inout_alloc();
+    AVFilterGraph *filter_graph = avfilter_graph_alloc();
+
+    if (!outputs || !inputs || !filter_graph) {
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }
+
+    if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO) {
+        buffersrc = avfilter_get_by_name("buffer");
+        buffersink = avfilter_get_by_name("buffersink");
+        if (!buffersrc || !buffersink) {
+            av_log(NULL, AV_LOG_ERROR, "filtering source or sink element not found\n");
+            ret = AVERROR_UNKNOWN;
+            goto end;
+        }
+
+        snprintf(args, sizeof(args),
+                "video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:pixel_aspect=%d/%d",
+                dec_ctx->width, dec_ctx->height, dec_ctx->pix_fmt,
+                dec_ctx->time_base.num, dec_ctx->time_base.den,
+                dec_ctx->sample_aspect_ratio.num,
+                dec_ctx->sample_aspect_ratio.den);
+
+        ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
+                args, NULL, filter_graph);
+        if (ret < 0) {
+            av_log(NULL, AV_LOG_ERROR, "Cannot create buffer source\n");
+            goto end;
+        }
+
+        ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
+                NULL, NULL, filter_graph);
+        if (ret < 0) {
+            av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n");
+            goto end;
+        }
+
+        ret = av_opt_set_bin(buffersink_ctx, "pix_fmts",
+                (uint8_t*)&enc_ctx->pix_fmt, sizeof(enc_ctx->pix_fmt),
+                AV_OPT_SEARCH_CHILDREN);
+        if (ret < 0) {
+            av_log(NULL, AV_LOG_ERROR, "Cannot set output pixel format\n");
+            goto end;
+        }
+    } else if (dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
+        buffersrc = avfilter_get_by_name("abuffer");
+        buffersink = avfilter_get_by_name("abuffersink");
+        if (!buffersrc || !buffersink) {
+            av_log(NULL, AV_LOG_ERROR, "filtering source or sink element not found\n");
+            ret = AVERROR_UNKNOWN;
+            goto end;
+        }
+
+        if (!dec_ctx->channel_layout)
+            dec_ctx->channel_layout =
+                av_get_default_channel_layout(dec_ctx->channels);
+        snprintf(args, sizeof(args),
+                "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,
+                dec_ctx->time_base.num, dec_ctx->time_base.den, dec_ctx->sample_rate,
+                av_get_sample_fmt_name(dec_ctx->sample_fmt),
+                dec_ctx->channel_layout);
+        ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
+                args, NULL, filter_graph);
+        if (ret < 0) {
+            av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
+            goto end;
+        }
+
+        ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
+                NULL, NULL, filter_graph);
+        if (ret < 0) {
+            av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
+            goto end;
+        }
+
+        ret = av_opt_set_bin(buffersink_ctx, "sample_fmts",
+                (uint8_t*)&enc_ctx->sample_fmt, sizeof(enc_ctx->sample_fmt),
+                AV_OPT_SEARCH_CHILDREN);
+        if (ret < 0) {
+            av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n");
+            goto end;
+        }
+
+        ret = av_opt_set_bin(buffersink_ctx, "channel_layouts",
+                (uint8_t*)&enc_ctx->channel_layout,
+                sizeof(enc_ctx->channel_layout), AV_OPT_SEARCH_CHILDREN);
+        if (ret < 0) {
+            av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
+            goto end;
+        }
+
+        ret = av_opt_set_bin(buffersink_ctx, "sample_rates",
+                (uint8_t*)&enc_ctx->sample_rate, sizeof(enc_ctx->sample_rate),
+                AV_OPT_SEARCH_CHILDREN);
+        if (ret < 0) {
+            av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
+            goto end;
+        }
+    } else {
+        ret = AVERROR_UNKNOWN;
+        goto end;
+    }
+
+    /* Endpoints for the filter graph. */
+    outputs->name       = av_strdup("in");
+    outputs->filter_ctx = buffersrc_ctx;
+    outputs->pad_idx    = 0;
+    outputs->next       = NULL;
+
+    inputs->name       = av_strdup("out");
+    inputs->filter_ctx = buffersink_ctx;
+    inputs->pad_idx    = 0;
+    inputs->next       = NULL;
+
+    if (!outputs->name || !inputs->name) {
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }
+
+    if ((ret = avfilter_graph_parse_ptr(filter_graph, filter_spec,
+                    &inputs, &outputs, NULL)) < 0)
+        goto end;
+
+    if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
+        goto end;
+
+    /* Fill FilteringContext */
+    fctx->buffersrc_ctx = buffersrc_ctx;
+    fctx->buffersink_ctx = buffersink_ctx;
+    fctx->filter_graph = filter_graph;
+
+end:
+    avfilter_inout_free(&inputs);
+    avfilter_inout_free(&outputs);
+
+    return ret;
+}
+
+static int init_filters(void)
+{
+    const char *filter_spec;
+    unsigned int i;
+    int ret;
+    filter_ctx = av_malloc_array(ifmt_ctx->nb_streams, sizeof(*filter_ctx));
+    if (!filter_ctx)
+        return AVERROR(ENOMEM);
+
+    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
+        filter_ctx[i].buffersrc_ctx  = NULL;
+        filter_ctx[i].buffersink_ctx = NULL;
+        filter_ctx[i].filter_graph   = NULL;
+        if (!(ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO
+                || ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO))
+            continue;
+
+
+        if (ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO)
+            filter_spec = "null"; /* passthrough (dummy) filter for video */
+        else
+            filter_spec = "anull"; /* passthrough (dummy) filter for audio */
+        ret = init_filter(&filter_ctx[i], ifmt_ctx->streams[i]->codec,
+                ofmt_ctx->streams[i]->codec, filter_spec);
+        if (ret)
+            return ret;
+    }
+    return 0;
+}
+
+static int encode_write_frame(AVFrame *filt_frame, unsigned int stream_index, int *got_frame) {
+    int ret;
+    int got_frame_local;
+    AVPacket enc_pkt;
+    int (*enc_func)(AVCodecContext *, AVPacket *, const AVFrame *, int *) =
+        (ifmt_ctx->streams[stream_index]->codec->codec_type ==
+         AVMEDIA_TYPE_VIDEO) ? avcodec_encode_video2 : avcodec_encode_audio2;
+
+    if (!got_frame)
+        got_frame = &got_frame_local;
+
+    av_log(NULL, AV_LOG_INFO, "Encoding frame\n");
+    /* encode filtered frame */
+    enc_pkt.data = NULL;
+    enc_pkt.size = 0;
+    av_init_packet(&enc_pkt);
+    ret = enc_func(ofmt_ctx->streams[stream_index]->codec, &enc_pkt,
+            filt_frame, got_frame);
+    av_frame_free(&filt_frame);
+    if (ret < 0)
+        return ret;
+    if (!(*got_frame))
+        return 0;
+
+    /* prepare packet for muxing */
+    enc_pkt.stream_index = stream_index;
+    av_packet_rescale_ts(&enc_pkt,
+                         ofmt_ctx->streams[stream_index]->codec->time_base,
+                         ofmt_ctx->streams[stream_index]->time_base);
+
+    av_log(NULL, AV_LOG_DEBUG, "Muxing frame\n");
+    /* mux encoded frame */
+    ret = av_interleaved_write_frame(ofmt_ctx, &enc_pkt);
+    return ret;
+}
+
+static int filter_encode_write_frame(AVFrame *frame, unsigned int stream_index)
+{
+    int ret;
+    AVFrame *filt_frame;
+
+    av_log(NULL, AV_LOG_INFO, "Pushing decoded frame to filters\n");
+    /* push the decoded frame into the filtergraph */
+    ret = av_buffersrc_add_frame_flags(filter_ctx[stream_index].buffersrc_ctx,
+            frame, 0);
+    if (ret < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
+        return ret;
+    }
+
+    /* pull filtered frames from the filtergraph */
+    while (1) {
+        filt_frame = av_frame_alloc();
+        if (!filt_frame) {
+            ret = AVERROR(ENOMEM);
+            break;
+        }
+        av_log(NULL, AV_LOG_INFO, "Pulling filtered frame from filters\n");
+        ret = av_buffersink_get_frame(filter_ctx[stream_index].buffersink_ctx,
+                filt_frame);
+        if (ret < 0) {
+            /* if no more frames for output - returns AVERROR(EAGAIN)
+             * if flushed and no more frames for output - returns AVERROR_EOF
+             * rewrite retcode to 0 to show it as normal procedure completion
+             */
+            if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
+                ret = 0;
+            av_frame_free(&filt_frame);
+            break;
+        }
+
+        filt_frame->pict_type = AV_PICTURE_TYPE_NONE;
+        ret = encode_write_frame(filt_frame, stream_index, NULL);
+        if (ret < 0)
+            break;
+    }
+
+    return ret;
+}
+
+static int flush_encoder(unsigned int stream_index)
+{
+    int ret;
+    int got_frame;
+
+    if (!(ofmt_ctx->streams[stream_index]->codec->codec->capabilities &
+                AV_CODEC_CAP_DELAY))
+        return 0;
+
+    while (1) {
+        av_log(NULL, AV_LOG_INFO, "Flushing stream #%u encoder\n", stream_index);
+        ret = encode_write_frame(NULL, stream_index, &got_frame);
+        if (ret < 0)
+            break;
+        if (!got_frame)
+            return 0;
+    }
+    return ret;
+}
+
+int main(int argc, char **argv)
+{
+    int ret;
+    AVPacket packet = { .data = NULL, .size = 0 };
+    AVFrame *frame = NULL;
+    enum AVMediaType type;
+    unsigned int stream_index;
+    unsigned int i;
+    int got_frame;
+    int (*dec_func)(AVCodecContext *, AVFrame *, int *, const AVPacket *);
+
+    if (argc != 3) {
+        av_log(NULL, AV_LOG_ERROR, "Usage: %s <input file> <output file>\n", argv[0]);
+        return 1;
+    }
+
+    av_register_all();
+    avfilter_register_all();
+
+    if ((ret = open_input_file(argv[1])) < 0)
+        goto end;
+    if ((ret = open_output_file(argv[2])) < 0)
+        goto end;
+    if ((ret = init_filters()) < 0)
+        goto end;
+
+    /* read all packets */
+    while (1) {
+        if ((ret = av_read_frame(ifmt_ctx, &packet)) < 0)
+            break;
+        stream_index = packet.stream_index;
+        type = ifmt_ctx->streams[packet.stream_index]->codec->codec_type;
+        av_log(NULL, AV_LOG_DEBUG, "Demuxer gave frame of stream_index %u\n",
+                stream_index);
+
+        if (filter_ctx[stream_index].filter_graph) {
+            av_log(NULL, AV_LOG_DEBUG, "Going to reencode&filter the frame\n");
+            frame = av_frame_alloc();
+            if (!frame) {
+                ret = AVERROR(ENOMEM);
+                break;
+            }
+            av_packet_rescale_ts(&packet,
+                                 ifmt_ctx->streams[stream_index]->time_base,
+                                 ifmt_ctx->streams[stream_index]->codec->time_base);
+            dec_func = (type == AVMEDIA_TYPE_VIDEO) ? avcodec_decode_video2 :
+                avcodec_decode_audio4;
+            ret = dec_func(ifmt_ctx->streams[stream_index]->codec, frame,
+                    &got_frame, &packet);
+            if (ret < 0) {
+                av_frame_free(&frame);
+                av_log(NULL, AV_LOG_ERROR, "Decoding failed\n");
+                break;
+            }
+
+            if (got_frame) {
+                frame->pts = av_frame_get_best_effort_timestamp(frame);
+                ret = filter_encode_write_frame(frame, stream_index);
+                av_frame_free(&frame);
+                if (ret < 0)
+                    goto end;
+            } else {
+                av_frame_free(&frame);
+            }
+        } else {
+            /* remux this frame without reencoding */
+            av_packet_rescale_ts(&packet,
+                                 ifmt_ctx->streams[stream_index]->time_base,
+                                 ofmt_ctx->streams[stream_index]->time_base);
+
+            ret = av_interleaved_write_frame(ofmt_ctx, &packet);
+            if (ret < 0)
+                goto end;
+        }
+        av_packet_unref(&packet);
+    }
+
+    /* flush filters and encoders */
+    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
+        /* flush filter */
+        if (!filter_ctx[i].filter_graph)
+            continue;
+        ret = filter_encode_write_frame(NULL, i);
+        if (ret < 0) {
+            av_log(NULL, AV_LOG_ERROR, "Flushing filter failed\n");
+            goto end;
+        }
+
+        /* flush encoder */
+        ret = flush_encoder(i);
+        if (ret < 0) {
+            av_log(NULL, AV_LOG_ERROR, "Flushing encoder failed\n");
+            goto end;
+        }
+    }
+
+    av_write_trailer(ofmt_ctx);
+end:
+    av_packet_unref(&packet);
+    av_frame_free(&frame);
+    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
+        avcodec_close(ifmt_ctx->streams[i]->codec);
+        if (ofmt_ctx && ofmt_ctx->nb_streams > i && ofmt_ctx->streams[i] && ofmt_ctx->streams[i]->codec)
+            avcodec_close(ofmt_ctx->streams[i]->codec);
+        if (filter_ctx && filter_ctx[i].filter_graph)
+            avfilter_graph_free(&filter_ctx[i].filter_graph);
+    }
+    av_free(filter_ctx);
+    avformat_close_input(&ifmt_ctx);
+    if (ofmt_ctx && !(ofmt_ctx->oformat->flags & AVFMT_NOFILE))
+        avio_closep(&ofmt_ctx->pb);
+    avformat_free_context(ofmt_ctx);
+
+    if (ret < 0)
+        av_log(NULL, AV_LOG_ERROR, "Error occurred: %s\n", av_err2str(ret));
+
+    return ret ? 1 : 0;
+}