updated for recent changes

README.md

@@ -0,0 +1,22 @@
+httpsegmenter
+=============
+
+I grabbed this off of Google Code a few years ago; it was useful at the time
+for streaming live audio to my iPhone (back when I was using an iPhone).  I
+came up with a workflow that I documented in [this StackOverflow post](https://stackoverflow.com/a/36227445), 
+and I put it to work in a live-stream timeshifter that converts radio
+stations' streams into podcasts.
+
+I just tried building it again, though, and the compile failed with some
+undeclared variables or something.  A bit of digging indicated that (1)
+something had changed in libavcodec that broke httpsegmenter and (2) ffmpeg
+has gained the ability to create the files for HTTP live streaming all by
+itself.
+
+I made a few tweaks to the source of this program to get it to build again. 
+They're in the latest commit, but are untested as of yet.  A better idea at
+this point might be to use ffmpeg's built-in segmenter.  I got a stream
+working pretty quickly with [this Medium post](https://medium.com/androvideo/convert-rtsp-to-hls-using-ffmpeg-2fe2cdf3a0de), 
+and got it dialed in further (mainly to keep it from filling the disk) with
+[this bit of ffmpeg documentation](https://www.ffmpeg.org/ffmpeg-formats.html#segment).
+

segmenter.c

@@ -34,6 +34,7 @@
 
 #include "libavformat/avformat.h"
 #include "libavutil/opt.h"
+#include "libavcodec/avcodec.h"
 
 #ifdef _WIN32
 //----------------------------------------------------------------
@@ -119,7 +120,7 @@ static AVStream *add_output_stream(AVFormatContext *output_format_context, AVStr
             output_codec_context->sample_rate = input_codec_context->sample_rate;
             output_codec_context->channels = input_codec_context->channels;
             output_codec_context->frame_size = input_codec_context->frame_size;
-            if ((input_codec_context->block_align == 1 && input_codec_context->codec_id == CODEC_ID_MP3) || input_codec_context->codec_id == CODEC_ID_AC3) {
+            if ((input_codec_context->block_align == 1 && input_codec_context->codec_id == AV_CODEC_ID_MP3) || input_codec_context->codec_id == AV_CODEC_ID_AC3) {
                 output_codec_context->block_align = 0;
             }
             else {
@@ -133,7 +134,7 @@ static AVStream *add_output_stream(AVFormatContext *output_format_context, AVStr
             output_codec_context->has_b_frames = input_codec_context->has_b_frames;
 
             if (output_format_context->oformat->flags & AVFMT_GLOBALHEADER) {
-                output_codec_context->flags |= CODEC_FLAG_GLOBAL_HEADER;
+                output_codec_context->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
             }
             break;
     default: