basic type mechanism working

16 files changed, 267 insertions, 494 deletions

diff --git a/ffmpeg/doc/examples/Makefile b/ffmpeg/doc/examples/Makefile
index c849daa..f085532 100644
--- a/ffmpeg/doc/examples/Makefile
+++ b/ffmpeg/doc/examples/Makefile
@@ -7,24 +7,26 @@ FFMPEG_LIBS=    libavdevice                        \
                 libswscale                         \
                 libavutil                          \
 
-CFLAGS += -Wall -O2 -g
+CFLAGS += -Wall -g
 CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
 LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)
 
 EXAMPLES=       decoding_encoding                  \
-                demuxing                           \
+                demuxing_decoding                  \
                 filtering_video                    \
                 filtering_audio                    \
                 metadata                           \
                 muxing                             \
                 resampling_audio                   \
                 scaling_video                      \
+                transcode_aac                      \
 
 OBJS=$(addsuffix .o,$(EXAMPLES))
 
 # the following examples make explicit use of the math library
 decoding_encoding: LDLIBS += -lm
 muxing:            LDLIBS += -lm
+resampling_audio:  LDLIBS += -lm
 
 .phony: all clean-test clean
 
diff --git a/ffmpeg/doc/examples/README b/ffmpeg/doc/examples/README
index a461813..c1ce619 100644
--- a/ffmpeg/doc/examples/README
+++ b/ffmpeg/doc/examples/README
@@ -5,14 +5,19 @@ Both following use cases rely on pkg-config and make, thus make sure
 that you have them installed and working on your system.
 
 
-1) Build the installed examples in a generic read/write user directory
+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.
 
-2) Build the examples in-tree
+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 you can go into the
-doc/examples and run a command such as PKG_CONFIG_PATH=pc-uninstalled make.
+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/ffmpeg/doc/examples/decoding_encoding.c b/ffmpeg/doc/examples/decoding_encoding.c
index ae1057c..08e8b92 100644
--- a/ffmpeg/doc/examples/decoding_encoding.c
+++ b/ffmpeg/doc/examples/decoding_encoding.c
@@ -79,7 +79,7 @@ static int select_channel_layout(AVCodec *codec)
 {
     const uint64_t *p;
     uint64_t best_ch_layout = 0;
-    int best_nb_channells   = 0;
+    int best_nb_channels   = 0;
 
     if (!codec->channel_layouts)
         return AV_CH_LAYOUT_STEREO;
@@ -88,9 +88,9 @@ static int select_channel_layout(AVCodec *codec)
     while (*p) {
         int nb_channels = av_get_channel_layout_nb_channels(*p);
 
-        if (nb_channels > best_nb_channells) {
+        if (nb_channels > best_nb_channels) {
             best_ch_layout    = *p;
-            best_nb_channells = nb_channels;
+            best_nb_channels = nb_channels;
         }
         p++;
     }
@@ -156,7 +156,7 @@ static void audio_encode_example(const char *filename)
     }
 
     /* frame containing input raw audio */
-    frame = avcodec_alloc_frame();
+    frame = av_frame_alloc();
     if (!frame) {
         fprintf(stderr, "Could not allocate audio frame\n");
         exit(1);
@@ -170,6 +170,10 @@ static void audio_encode_example(const char *filename)
      * 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",
@@ -227,7 +231,7 @@ static void audio_encode_example(const char *filename)
     fclose(f);
 
     av_freep(&samples);
-    avcodec_free_frame(&frame);
+    av_frame_free(&frame);
     avcodec_close(c);
     av_free(c);
 }
@@ -287,12 +291,11 @@ static void audio_decode_example(const char *outfilename, const char *filename)
         int got_frame = 0;
 
         if (!decoded_frame) {
-            if (!(decoded_frame = avcodec_alloc_frame())) {
+            if (!(decoded_frame = av_frame_alloc())) {
                 fprintf(stderr, "Could not allocate audio frame\n");
                 exit(1);
             }
-        } else
-            avcodec_get_frame_defaults(decoded_frame);
+        }
 
         len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt);
         if (len < 0) {
@@ -329,7 +332,7 @@ static void audio_decode_example(const char *outfilename, const char *filename)
 
     avcodec_close(c);
     av_free(c);
-    avcodec_free_frame(&decoded_frame);
+    av_frame_free(&decoded_frame);
 }
 
 /*
@@ -386,7 +389,7 @@ static void video_encode_example(const char *filename, int codec_id)
         exit(1);
     }
 
-    frame = avcodec_alloc_frame();
+    frame = av_frame_alloc();
     if (!frame) {
         fprintf(stderr, "Could not allocate video frame\n");
         exit(1);
@@ -467,7 +470,7 @@ static void video_encode_example(const char *filename, int codec_id)
     avcodec_close(c);
     av_free(c);
     av_freep(&frame->data[0]);
-    avcodec_free_frame(&frame);
+    av_frame_free(&frame);
     printf("\n");
 }
 
@@ -565,7 +568,7 @@ static void video_decode_example(const char *outfilename, const char *filename)
         exit(1);
     }
 
-    frame = avcodec_alloc_frame();
+    frame = av_frame_alloc();
     if (!frame) {
         fprintf(stderr, "Could not allocate video frame\n");
         exit(1);
@@ -609,7 +612,7 @@ static void video_decode_example(const char *outfilename, const char *filename)
 
     avcodec_close(c);
     av_free(c);
-    avcodec_free_frame(&frame);
+    av_frame_free(&frame);
     printf("\n");
 }
 
diff --git a/ffmpeg/doc/examples/demuxing.c b/ffmpeg/doc/examples/demuxing.c
deleted file mode 100644
index 8a1b69b..0000000
--- a/ffmpeg/doc/examples/demuxing.c
+++ /dev/null
@@ -1,342 +0,0 @@
-/*
- * 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
- * libavformat demuxing API use example.
- *
- * Show how to use the libavformat and libavcodec API to demux and
- * decode audio and video data.
- * @example doc/examples/demuxing.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 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 uint8_t **audio_dst_data = NULL;
-static int       audio_dst_linesize;
-static int audio_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;
-
-static int decode_packet(int *got_frame, int cached)
-{
-    int ret = 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\n");
-            return ret;
-        }
-
-        if (*got_frame) {
-            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,
-                          video_dec_ctx->pix_fmt, video_dec_ctx->width, video_dec_ctx->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\n");
-            return ret;
-        }
-
-        if (*got_frame) {
-            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));
-
-            ret = av_samples_alloc(audio_dst_data, &audio_dst_linesize, av_frame_get_channels(frame),
-                                   frame->nb_samples, frame->format, 1);
-            if (ret < 0) {
-                fprintf(stderr, "Could not allocate audio buffer\n");
-                return AVERROR(ENOMEM);
-            }
-
-            /* TODO: extend return code of the av_samples_* functions so that this call is not needed */
-            audio_dst_bufsize =
-                av_samples_get_buffer_size(NULL, av_frame_get_channels(frame),
-                                           frame->nb_samples, frame->format, 1);
-
-            /* copy audio data to destination buffer:
-             * this is required since rawaudio expects non aligned data */
-            av_samples_copy(audio_dst_data, frame->data, 0, 0,
-                            frame->nb_samples, av_frame_get_channels(frame), frame->format);
-
-            /* write to rawaudio file */
-            fwrite(audio_dst_data[0], 1, audio_dst_bufsize, audio_dst_file);
-            av_freep(&audio_dst_data[0]);
-        }
-    }
-
-    return ret;
-}
-
-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;
-
-    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 ret;
-        }
-
-        if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
-            fprintf(stderr, "Failed to open %s codec\n",
-                    av_get_media_type_string(type));
-            return ret;
-        }
-    }
-
-    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) {
-        fprintf(stderr, "usage: %s 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", argv[0]);
-        exit(1);
-    }
-    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, fmt_ctx, AVMEDIA_TYPE_VIDEO) >= 0) {
-        video_stream = fmt_ctx->streams[video_stream_idx];
-        video_dec_ctx = video_stream->codec;
-
-        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 */
-        ret = av_image_alloc(video_dst_data, video_dst_linesize,
-                             video_dec_ctx->width, video_dec_ctx->height,
-                             video_dec_ctx->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, fmt_ctx, AVMEDIA_TYPE_AUDIO) >= 0) {
-        int nb_planes;
-
-        audio_stream = fmt_ctx->streams[audio_stream_idx];
-        audio_dec_ctx = audio_stream->codec;
-        audio_dst_file = fopen(audio_dst_filename, "wb");
-        if (!audio_dst_file) {
-            fprintf(stderr, "Could not open destination file %s\n", video_dst_filename);
-            ret = 1;
-            goto end;
-        }
-
-        nb_planes = av_sample_fmt_is_planar(audio_dec_ctx->sample_fmt) ?
-            audio_dec_ctx->channels : 1;
-        audio_dst_data = av_mallocz(sizeof(uint8_t *) * nb_planes);
-        if (!audio_dst_data) {
-            fprintf(stderr, "Could not allocate audio data buffers\n");
-            ret = AVERROR(ENOMEM);
-            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 = avcodec_alloc_frame();
-    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) {
-        decode_packet(&got_frame, 0);
-        av_free_packet(&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(video_dec_ctx->pix_fmt), video_dec_ctx->width, video_dec_ctx->height,
-               video_dst_filename);
-    }
-
-    if (audio_stream) {
-        const char *fmt;
-
-        if ((ret = get_format_from_sample_fmt(&fmt, audio_dec_ctx->sample_fmt)) < 0)
-            goto end;
-        printf("Play the output audio file with the command:\n"
-               "ffplay -f %s -ac %d -ar %d %s\n",
-               fmt, audio_dec_ctx->channels, audio_dec_ctx->sample_rate,
-               audio_dst_filename);
-    }
-
-end:
-    if (video_dec_ctx)
-        avcodec_close(video_dec_ctx);
-    if (audio_dec_ctx)
-        avcodec_close(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_free(frame);
-    av_free(video_dst_data[0]);
-    av_free(audio_dst_data);
-
-    return ret < 0;
-}
diff --git a/ffmpeg/doc/examples/filtering_audio.c b/ffmpeg/doc/examples/filtering_audio.c
index 456a1c9..1d66ca3 100644
--- a/ffmpeg/doc/examples/filtering_audio.c
+++ b/ffmpeg/doc/examples/filtering_audio.c
@@ -36,9 +36,10 @@
 #include <libavfilter/avcodec.h>
 #include <libavfilter/buffersink.h>
 #include <libavfilter/buffersrc.h>
+#include <libavutil/opt.h>
 
-const char *filter_descr = "aresample=8000,aconvert=s16:mono";
-const char *player       = "ffplay -f s16le -ar 8000 -ac 1 -";
+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;
@@ -70,6 +71,7 @@ static int open_input_file(const char *filename)
     }
     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) {
@@ -83,17 +85,22 @@ static int open_input_file(const char *filename)
 static int init_filters(const char *filters_descr)
 {
     char args[512];
-    int ret;
+    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();
-    const enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
-    AVABufferSinkParams *abuffersink_params;
+    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)
@@ -106,18 +113,36 @@ static int init_filters(const char *filters_descr)
                                        args, NULL, filter_graph);
     if (ret < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
-        return ret;
+        goto end;
     }
 
     /* buffer audio sink: to terminate the filter chain. */
-    abuffersink_params = av_abuffersink_params_alloc();
-    abuffersink_params->sample_fmts     = sample_fmts;
     ret = avfilter_graph_create_filter(&buffersink_ctx, abuffersink, "out",
-                                       NULL, abuffersink_params, filter_graph);
-    av_free(abuffersink_params);
+                                       NULL, NULL, filter_graph);
     if (ret < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
-        return ret;
+        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;
     }
 
     /* Endpoints for the filter graph. */
@@ -131,12 +156,12 @@ static int init_filters(const char *filters_descr)
     inputs->pad_idx    = 0;
     inputs->next       = NULL;
 
-    if ((ret = avfilter_graph_parse(filter_graph, filters_descr,
-                                    &inputs, &outputs, NULL)) < 0)
-        return ret;
+    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)
-        return ret;
+        goto end;
 
     /* Print summary of the sink buffer
      * Note: args buffer is reused to store channel layout string */
@@ -147,7 +172,11 @@ static int init_filters(const char *filters_descr)
            (char *)av_x_if_null(av_get_sample_fmt_name(outlink->format), "?"),
            args);
 
-    return 0;
+end:
+    avfilter_inout_free(&inputs);
+    avfilter_inout_free(&outputs);
+
+    return ret;
 }
 
 static void print_frame(const AVFrame *frame)
@@ -167,7 +196,7 @@ static void print_frame(const AVFrame *frame)
 int main(int argc, char **argv)
 {
     int ret;
-    AVPacket packet;
+    AVPacket packet0, packet;
     AVFrame *frame = av_frame_alloc();
     AVFrame *filt_frame = av_frame_alloc();
     int got_frame;
@@ -191,9 +220,14 @@ int main(int argc, char **argv)
         goto end;
 
     /* read all packets */
+    packet0.data = NULL;
+    packet.data = NULL;
     while (1) {
-        if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
-            break;
+        if (!packet0.data) {
+            if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
+                break;
+            packet0 = packet;
+        }
 
         if (packet.stream_index == audio_stream_index) {
             avcodec_get_frame_defaults(frame);
@@ -203,10 +237,12 @@ int main(int argc, char **argv)
                 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, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) {
+                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;
                 }
@@ -214,29 +250,31 @@ int main(int argc, char **argv)
                 /* pull filtered audio from the filtergraph */
                 while (1) {
                     ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
-                    if(ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
+                    if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
                         break;
-                    if(ret < 0)
+                    if (ret < 0)
                         goto end;
                     print_frame(filt_frame);
                     av_frame_unref(filt_frame);
                 }
             }
+
+            if (packet.size <= 0)
+                av_free_packet(&packet0);
+        } else {
+            /* discard non-wanted packets */
+            av_free_packet(&packet0);
         }
-        av_free_packet(&packet);
     }
 end:
     avfilter_graph_free(&filter_graph);
-    if (dec_ctx)
-        avcodec_close(dec_ctx);
+    avcodec_close(dec_ctx);
     avformat_close_input(&fmt_ctx);
     av_frame_free(&frame);
     av_frame_free(&filt_frame);
 
     if (ret < 0 && ret != AVERROR_EOF) {
-        char buf[1024];
-        av_strerror(ret, buf, sizeof(buf));
-        fprintf(stderr, "Error occurred: %s\n", buf);
+        fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
         exit(1);
     }
 
diff --git a/ffmpeg/doc/examples/filtering_video.c b/ffmpeg/doc/examples/filtering_video.c
index daa3966..790c641 100644
--- a/ffmpeg/doc/examples/filtering_video.c
+++ b/ffmpeg/doc/examples/filtering_video.c
@@ -36,6 +36,7 @@
 #include <libavfilter/avcodec.h>
 #include <libavfilter/buffersink.h>
 #include <libavfilter/buffersrc.h>
+#include <libavutil/opt.h>
 
 const char *filter_descr = "scale=78:24";
 
@@ -70,6 +71,7 @@ static int open_input_file(const char *filename)
     }
     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) {
@@ -83,15 +85,18 @@ static int open_input_file(const char *filename)
 static int init_filters(const char *filters_descr)
 {
     char args[512];
-    int ret;
+    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();
     enum AVPixelFormat pix_fmts[] = { AV_PIX_FMT_GRAY8, AV_PIX_FMT_NONE };
-    AVBufferSinkParams *buffersink_params;
 
     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),
@@ -104,18 +109,22 @@ static int init_filters(const char *filters_descr)
                                        args, NULL, filter_graph);
     if (ret < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot create buffer source\n");
-        return ret;
+        goto end;
     }
 
     /* buffer video sink: to terminate the filter chain. */
-    buffersink_params = av_buffersink_params_alloc();
-    buffersink_params->pixel_fmts = pix_fmts;
     ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
-                                       NULL, buffersink_params, filter_graph);
-    av_free(buffersink_params);
+                                       NULL, NULL, filter_graph);
     if (ret < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n");
-        return ret;
+        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;
     }
 
     /* Endpoints for the filter graph. */
@@ -129,13 +138,18 @@ static int init_filters(const char *filters_descr)
     inputs->pad_idx    = 0;
     inputs->next       = NULL;
 
-    if ((ret = avfilter_graph_parse(filter_graph, filters_descr,
+    if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
                                     &inputs, &outputs, NULL)) < 0)
-        return ret;
+        goto end;
 
     if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
-        return ret;
-    return 0;
+        goto end;
+
+end:
+    avfilter_inout_free(&inputs);
+    avfilter_inout_free(&outputs);
+
+    return ret;
 }
 
 static void display_frame(const AVFrame *frame, AVRational time_base)
@@ -228,22 +242,20 @@ int main(int argc, char **argv)
                     display_frame(filt_frame, buffersink_ctx->inputs[0]->time_base);
                     av_frame_unref(filt_frame);
                 }
+                av_frame_unref(frame);
             }
         }
         av_free_packet(&packet);
     }
 end:
     avfilter_graph_free(&filter_graph);
-    if (dec_ctx)
-        avcodec_close(dec_ctx);
+    avcodec_close(dec_ctx);
     avformat_close_input(&fmt_ctx);
     av_frame_free(&frame);
     av_frame_free(&filt_frame);
 
     if (ret < 0 && ret != AVERROR_EOF) {
-        char buf[1024];
-        av_strerror(ret, buf, sizeof(buf));
-        fprintf(stderr, "Error occurred: %s\n", buf);
+        fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
         exit(1);
     }
 
diff --git a/ffmpeg/doc/examples/muxing.c b/ffmpeg/doc/examples/muxing.c
index 7305cc6..4cd3f65 100644
--- a/ffmpeg/doc/examples/muxing.c
+++ b/ffmpeg/doc/examples/muxing.c
@@ -34,9 +34,11 @@
 #include <string.h>
 #include <math.h>
 
+#include <libavutil/opt.h>
 #include <libavutil/mathematics.h>
 #include <libavformat/avformat.h>
 #include <libswscale/swscale.h>
+#include <libswresample/swresample.h>
 
 /* 5 seconds stream duration */
 #define STREAM_DURATION   200.0
@@ -46,13 +48,6 @@
 
 static int sws_flags = SWS_BICUBIC;
 
-/**************************************************************/
-/* audio output */
-
-static float t, tincr, tincr2;
-static int16_t *samples;
-static int audio_input_frame_size;
-
 /* Add an output stream. */
 static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,
                             enum AVCodecID codec_id)
@@ -78,8 +73,7 @@ static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,
 
     switch ((*codec)->type) {
     case AVMEDIA_TYPE_AUDIO:
-        st->id = 1;
-        c->sample_fmt  = AV_SAMPLE_FMT_S16;
+        c->sample_fmt  = AV_SAMPLE_FMT_FLTP;
         c->bit_rate    = 64000;
         c->sample_rate = 44100;
         c->channels    = 2;
@@ -127,8 +121,17 @@ static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,
 /* audio output */
 
 static float t, tincr, tincr2;
-static int16_t *samples;
-static int audio_input_frame_size;
+
+static uint8_t **src_samples_data;
+static int       src_samples_linesize;
+static int       src_nb_samples;
+
+static int max_dst_nb_samples;
+uint8_t **dst_samples_data;
+int       dst_samples_linesize;
+int       dst_samples_size;
+
+struct SwrContext *swr_ctx = NULL;
 
 static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
 {
@@ -150,17 +153,54 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
     /* increment frequency by 110 Hz per second */
     tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;
 
-    if (c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)
-        audio_input_frame_size = 10000;
-    else
-        audio_input_frame_size = c->frame_size;
-    samples = av_malloc(audio_input_frame_size *
-                        av_get_bytes_per_sample(c->sample_fmt) *
-                        c->channels);
-    if (!samples) {
-        fprintf(stderr, "Could not allocate audio samples buffer\n");
+    src_nb_samples = c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE ?
+        10000 : c->frame_size;
+
+    ret = av_samples_alloc_array_and_samples(&src_samples_data, &src_samples_linesize, c->channels,
+                                             src_nb_samples, AV_SAMPLE_FMT_S16, 0);
+    if (ret < 0) {
+        fprintf(stderr, "Could not allocate source samples\n");
         exit(1);
     }
+
+    /* 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 = src_nb_samples;
+
+    /* create resampler context */
+    if (c->sample_fmt != AV_SAMPLE_FMT_S16) {
+        swr_ctx = swr_alloc();
+        if (!swr_ctx) {
+            fprintf(stderr, "Could not allocate resampler context\n");
+            exit(1);
+        }
+
+        /* set options */
+        av_opt_set_int       (swr_ctx, "in_channel_count",   c->channels,       0);
+        av_opt_set_int       (swr_ctx, "in_sample_rate",     c->sample_rate,    0);
+        av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt",      AV_SAMPLE_FMT_S16, 0);
+        av_opt_set_int       (swr_ctx, "out_channel_count",  c->channels,       0);
+        av_opt_set_int       (swr_ctx, "out_sample_rate",    c->sample_rate,    0);
+        av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt",     c->sample_fmt,     0);
+
+        /* initialize the resampling context */
+        if ((ret = swr_init(swr_ctx)) < 0) {
+            fprintf(stderr, "Failed to initialize the resampling context\n");
+            exit(1);
+        }
+
+        ret = av_samples_alloc_array_and_samples(&dst_samples_data, &dst_samples_linesize, c->channels,
+                                                 max_dst_nb_samples, c->sample_fmt, 0);
+        if (ret < 0) {
+            fprintf(stderr, "Could not allocate destination samples\n");
+            exit(1);
+        }
+    } else {
+        dst_samples_data = src_samples_data;
+    }
+    dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, max_dst_nb_samples,
+                                                  c->sample_fmt, 0);
 }
 
 /* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
@@ -184,19 +224,45 @@ static void write_audio_frame(AVFormatContext *oc, AVStream *st)
 {
     AVCodecContext *c;
     AVPacket pkt = { 0 }; // data and size must be 0;
-    AVFrame *frame = avcodec_alloc_frame();
-    int got_packet, ret;
+    AVFrame *frame = av_frame_alloc();
+    int got_packet, ret, dst_nb_samples;
 
     av_init_packet(&pkt);
     c = st->codec;
 
-    get_audio_frame(samples, audio_input_frame_size, c->channels);
-    frame->nb_samples = audio_input_frame_size;
+    get_audio_frame((int16_t *)src_samples_data[0], src_nb_samples, c->channels);
+
+    /* convert samples from native format to destination codec format, using the resampler */
+    if (swr_ctx) {
+        /* compute destination number of samples */
+        dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, c->sample_rate) + src_nb_samples,
+                                        c->sample_rate, c->sample_rate, AV_ROUND_UP);
+        if (dst_nb_samples > max_dst_nb_samples) {
+            av_free(dst_samples_data[0]);
+            ret = av_samples_alloc(dst_samples_data, &dst_samples_linesize, c->channels,
+                                   dst_nb_samples, c->sample_fmt, 0);
+            if (ret < 0)
+                exit(1);
+            max_dst_nb_samples = dst_nb_samples;
+            dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, dst_nb_samples,
+                                                          c->sample_fmt, 0);
+        }
+
+        /* convert to destination format */
+        ret = swr_convert(swr_ctx,
+                          dst_samples_data, dst_nb_samples,
+                          (const uint8_t **)src_samples_data, src_nb_samples);
+        if (ret < 0) {
+            fprintf(stderr, "Error while converting\n");
+            exit(1);
+        }
+    } else {
+        dst_nb_samples = src_nb_samples;
+    }
+
+    frame->nb_samples = dst_nb_samples;
     avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
-                             (uint8_t *)samples,
-                             audio_input_frame_size *
-                             av_get_bytes_per_sample(c->sample_fmt) *
-                             c->channels, 1);
+                             dst_samples_data[0], dst_samples_size, 0);
 
     ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet);
     if (ret < 0) {
@@ -205,7 +271,7 @@ static void write_audio_frame(AVFormatContext *oc, AVStream *st)
     }
 
     if (!got_packet)
-        return;
+        goto freeframe;
 
     pkt.stream_index = st->index;
 
@@ -216,14 +282,19 @@ static void write_audio_frame(AVFormatContext *oc, AVStream *st)
                 av_err2str(ret));
         exit(1);
     }
-    avcodec_free_frame(&frame);
+freeframe:
+    av_frame_free(&frame);
 }
 
 static void close_audio(AVFormatContext *oc, AVStream *st)
 {
     avcodec_close(st->codec);
-
-    av_free(samples);
+    if (dst_samples_data != src_samples_data) {
+        av_free(dst_samples_data[0]);
+        av_free(dst_samples_data);
+    }
+    av_free(src_samples_data[0]);
+    av_free(src_samples_data);
 }
 
 /**************************************************************/
@@ -246,7 +317,7 @@ static void open_video(AVFormatContext *oc, AVCodec *codec, AVStream *st)
     }
 
     /* allocate and init a re-usable frame */
-    frame = avcodec_alloc_frame();
+    frame = av_frame_alloc();
     if (!frame) {
         fprintf(stderr, "Could not allocate video frame\n");
         exit(1);
@@ -388,7 +459,7 @@ int main(int argc, char **argv)
     AVFormatContext *oc;
     AVStream *audio_st, *video_st;
     AVCodec *audio_codec, *video_codec;
-    double audio_pts, video_pts;
+    double audio_time, video_time;
     int ret;
 
     /* Initialize libavcodec, and register all codecs and formats. */
@@ -461,23 +532,15 @@ int main(int argc, char **argv)
         frame->pts = 0;
     for (;;) {
         /* Compute current audio and video time. */
-        if (audio_st)
-            audio_pts = (double)audio_st->pts.val * audio_st->time_base.num / audio_st->time_base.den;
-        else
-            audio_pts = 0.0;
-
-        if (video_st)
-            video_pts = (double)video_st->pts.val * video_st->time_base.num /
-                        video_st->time_base.den;
-        else
-            video_pts = 0.0;
-
-        if ((!audio_st || audio_pts >= STREAM_DURATION) &&
-            (!video_st || video_pts >= STREAM_DURATION))
+        audio_time = audio_st ? audio_st->pts.val * av_q2d(audio_st->time_base) : 0.0;
+        video_time = video_st ? video_st->pts.val * av_q2d(video_st->time_base) : 0.0;
+
+        if ((!audio_st || audio_time >= STREAM_DURATION) &&
+            (!video_st || video_time >= STREAM_DURATION))
             break;
 
         /* write interleaved audio and video frames */
-        if (!video_st || (video_st && audio_st && audio_pts < video_pts)) {
+        if (!video_st || (video_st && audio_st && audio_time < video_time)) {
             write_audio_frame(oc, audio_st);
         } else {
             write_video_frame(oc, video_st);
diff --git a/ffmpeg/doc/examples/pc-uninstalled/libavcodec.pc b/ffmpeg/doc/examples/pc-uninstalled/libavcodec.pc
index 787d687..a87ded7 100644
--- a/ffmpeg/doc/examples/pc-uninstalled/libavcodec.pc
+++ b/ffmpeg/doc/examples/pc-uninstalled/libavcodec.pc
@@ -5,8 +5,8 @@ includedir=${pcfiledir}/../../..
 
 Name: libavcodec
 Description: FFmpeg codec library
-Version: 55.1.100
-Requires: libavutil = 52.22.100
+Version: 55.46.100
+Requires: libavutil = 52.59.100
 Conflicts:
-Libs: -L${libdir} -lavcodec 
+Libs: -L${libdir} -Wl,-rpath,${libdir} -lavcodec 
 Cflags: -I${includedir}
diff --git a/ffmpeg/doc/examples/pc-uninstalled/libavdevice.pc b/ffmpeg/doc/examples/pc-uninstalled/libavdevice.pc
index 89ef046..7f05a29 100644
--- a/ffmpeg/doc/examples/pc-uninstalled/libavdevice.pc
+++ b/ffmpeg/doc/examples/pc-uninstalled/libavdevice.pc
@@ -5,8 +5,8 @@ includedir=${pcfiledir}/../../..
 
 Name: libavdevice
 Description: FFmpeg device handling library
-Version: 55.0.100
-Requires: libavfilter = 3.48.100, libavformat = 55.0.100
+Version: 55.5.102
+Requires: libavfilter = 4.0.103, libavformat = 55.22.100
 Conflicts:
-Libs: -L${libdir} -lavdevice 
+Libs: -L${libdir} -Wl,-rpath,${libdir} -lavdevice 
 Cflags: -I${includedir}
diff --git a/ffmpeg/doc/examples/pc-uninstalled/libavfilter.pc b/ffmpeg/doc/examples/pc-uninstalled/libavfilter.pc
index aacaf0a..b42f95d 100644
--- a/ffmpeg/doc/examples/pc-uninstalled/libavfilter.pc
+++ b/ffmpeg/doc/examples/pc-uninstalled/libavfilter.pc
@@ -5,8 +5,8 @@ includedir=${pcfiledir}/../../..
 
 Name: libavfilter
 Description: FFmpeg audio/video filtering library
-Version: 3.48.100
-Requires: libpostproc = 52.2.100, libswresample = 0.17.102, libswscale = 2.2.100, libavformat = 55.0.100, libavcodec = 55.1.100, libavutil = 52.22.100
+Version: 4.0.103
+Requires: libpostproc = 52.3.100, libswresample = 0.17.104, libswscale = 2.5.101, libavformat = 55.22.100, libavcodec = 55.46.100, libavutil = 52.59.100
 Conflicts:
-Libs: -L${libdir} -lavfilter 
+Libs: -L${libdir} -Wl,-rpath,${libdir} -lavfilter 
 Cflags: -I${includedir}
diff --git a/ffmpeg/doc/examples/pc-uninstalled/libavformat.pc b/ffmpeg/doc/examples/pc-uninstalled/libavformat.pc
index 8f27151..8bab324 100644
--- a/ffmpeg/doc/examples/pc-uninstalled/libavformat.pc
+++ b/ffmpeg/doc/examples/pc-uninstalled/libavformat.pc
@@ -5,8 +5,8 @@ includedir=${pcfiledir}/../../..
 
 Name: libavformat
 Description: FFmpeg container format library
-Version: 55.0.100
-Requires: libavcodec = 55.1.100
+Version: 55.22.100
+Requires: libavcodec = 55.46.100
 Conflicts:
-Libs: -L${libdir} -lavformat 
+Libs: -L${libdir} -Wl,-rpath,${libdir} -lavformat 
 Cflags: -I${includedir}
diff --git a/ffmpeg/doc/examples/pc-uninstalled/libavutil.pc b/ffmpeg/doc/examples/pc-uninstalled/libavutil.pc
index 8a95064..85df0f0 100644
--- a/ffmpeg/doc/examples/pc-uninstalled/libavutil.pc
+++ b/ffmpeg/doc/examples/pc-uninstalled/libavutil.pc
@@ -5,8 +5,8 @@ includedir=${pcfiledir}/../../..
 
 Name: libavutil
 Description: FFmpeg utility library
-Version: 52.22.100
+Version: 52.59.100
 Requires: 
 Conflicts:
-Libs: -L${libdir} -lavutil 
+Libs: -L${libdir} -Wl,-rpath,${libdir} -lavutil 
 Cflags: -I${includedir}
diff --git a/ffmpeg/doc/examples/pc-uninstalled/libpostproc.pc b/ffmpeg/doc/examples/pc-uninstalled/libpostproc.pc
index 5e87c13..94da503 100644
--- a/ffmpeg/doc/examples/pc-uninstalled/libpostproc.pc
+++ b/ffmpeg/doc/examples/pc-uninstalled/libpostproc.pc
@@ -5,8 +5,8 @@ includedir=${pcfiledir}/../../..
 
 Name: libpostproc
 Description: FFmpeg postprocessing library
-Version: 52.2.100
-Requires: libavutil = 52.22.100
+Version: 52.3.100
+Requires: libavutil = 52.59.100
 Conflicts:
-Libs: -L${libdir} -lpostproc 
+Libs: -L${libdir} -Wl,-rpath,${libdir} -lpostproc 
 Cflags: -I${includedir}
diff --git a/ffmpeg/doc/examples/pc-uninstalled/libswresample.pc b/ffmpeg/doc/examples/pc-uninstalled/libswresample.pc
index 873f39d..45bfa4a 100644
--- a/ffmpeg/doc/examples/pc-uninstalled/libswresample.pc
+++ b/ffmpeg/doc/examples/pc-uninstalled/libswresample.pc
@@ -5,8 +5,8 @@ includedir=${pcfiledir}/../../..
 
 Name: libswresample
 Description: FFmpeg audio resampling library
-Version: 0.17.102
-Requires: libavutil = 52.22.100
+Version: 0.17.104
+Requires: libavutil = 52.59.100
 Conflicts:
-Libs: -L${libdir} -lswresample 
+Libs: -L${libdir} -Wl,-rpath,${libdir} -lswresample 
 Cflags: -I${includedir}
diff --git a/ffmpeg/doc/examples/pc-uninstalled/libswscale.pc b/ffmpeg/doc/examples/pc-uninstalled/libswscale.pc
index 764a10c..8693580 100644
--- a/ffmpeg/doc/examples/pc-uninstalled/libswscale.pc
+++ b/ffmpeg/doc/examples/pc-uninstalled/libswscale.pc
@@ -5,8 +5,8 @@ includedir=${pcfiledir}/../../..
 
 Name: libswscale
 Description: FFmpeg image rescaling library
-Version: 2.2.100
-Requires: libavutil = 52.22.100
+Version: 2.5.101
+Requires: libavutil = 52.59.100
 Conflicts:
-Libs: -L${libdir} -lswscale 
+Libs: -L${libdir} -Wl,-rpath,${libdir} -lswscale 
 Cflags: -I${includedir}
diff --git a/ffmpeg/doc/examples/resampling_audio.c b/ffmpeg/doc/examples/resampling_audio.c
index dd128e8..a15e042 100644
--- a/ffmpeg/doc/examples/resampling_audio.c
+++ b/ffmpeg/doc/examples/resampling_audio.c
@@ -62,7 +62,7 @@ static int get_format_from_sample_fmt(const char **fmt,
 /**
  * Fill dst buffer with nb_samples, generated starting from t.
  */
-void fill_samples(double *dst, int nb_samples, int nb_channels, int sample_rate, double *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;
@@ -78,18 +78,6 @@ void fill_samples(double *dst, int nb_samples, int nb_channels, int sample_rate,
     }
 }
 
-int alloc_samples_array_and_data(uint8_t ***data, int *linesize, int nb_channels,
-                                    int nb_samples, enum AVSampleFormat sample_fmt, int align)
-{
-    int nb_planes = av_sample_fmt_is_planar(sample_fmt) ? nb_channels : 1;
-
-    *data = av_malloc(sizeof(*data) * nb_planes);
-    if (!*data)
-        return AVERROR(ENOMEM);
-    return av_samples_alloc(*data, linesize, nb_channels,
-                            nb_samples, sample_fmt, align);
-}
-
 int main(int argc, char **argv)
 {
     int64_t src_ch_layout = AV_CH_LAYOUT_STEREO, dst_ch_layout = AV_CH_LAYOUT_SURROUND;
@@ -149,8 +137,8 @@ int main(int argc, char **argv)
     /* allocate source and destination samples buffers */
 
     src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout);
-    ret = alloc_samples_array_and_data(&src_data, &src_linesize, src_nb_channels,
-                                       src_nb_samples, src_sample_fmt, 0);
+    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;
@@ -164,8 +152,8 @@ int main(int argc, char **argv)
 
     /* 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 = alloc_samples_array_and_data(&dst_data, &dst_linesize, dst_nb_channels,
-                                       dst_nb_samples, dst_sample_fmt, 0);
+    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;
@@ -196,6 +184,10 @@ int main(int argc, char **argv)
         }
         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);