diff --git a/source/ffmpeg/libavformat/avformat.d b/source/ffmpeg/libavformat/avformat.d
index 6ae7722..a6caaf6 100755
--- a/source/ffmpeg/libavformat/avformat.d
+++ b/source/ffmpeg/libavformat/avformat.d
@@ -814,7 +814,7 @@ struct AVStream {
      * decoding: set by libavformat
      * encoding: set by the user, replaced by libavformat if left unset
      */
-    int avstream_id;
+    int id;
     /**
      * Codec context associated with this stream. Allocated and freed by
      * libavformat.
@@ -1862,7 +1862,7 @@ struct AVPacketList {
 /**
  * Return the LIBAVFORMAT_VERSION_INT constant.
  */
-ubyte avformat_version();
+uint avformat_version();
 
 /**
  * Return the libavformat build-time configuration.
diff --git a/source/ffmpeg/libswresample/audioconvert.d b/source/ffmpeg/libswresample/audioconvert.d
index 221f7e6..3b6b2cb 100644
--- a/source/ffmpeg/libswresample/audioconvert.d
+++ b/source/ffmpeg/libswresample/audioconvert.d
@@ -1,7 +1,7 @@
 module ffmpeg.libswresample.audioconvert;
 import std.stdint;
 import ffmpeg.libavutil.avutil;
-import ffmpeg.libavresample.audio_data;
+import ffmpeg.libswresample.swresample_internal;
 import ffmpeg.libswresample.swresample_version;
 
 @nogc nothrow extern(C):
@@ -47,4 +47,4 @@ void swri_audio_convert_free(AudioConvert **ctx);
  * @param[in] in array of input buffers for each channel
  * @param len length of audio frame size (measured in samples)
  */
-int swri_audio_convert(AudioConvert *ctx, AudioData *out_a_b, AudioData *in_a_b, int len);
\ No newline at end of file
+int swri_audio_convert(AudioConvert *ctx, AudioData *out_a_b, AudioData *in_a_b, int len);
diff --git a/source/ffmpeg/libswresample/swresample_internal.d b/source/ffmpeg/libswresample/swresample_internal.d
index 28ba197..fd1d46b 100644
--- a/source/ffmpeg/libswresample/swresample_internal.d
+++ b/source/ffmpeg/libswresample/swresample_internal.d
@@ -1,7 +1,11 @@
 module ffmpeg.libswresample.swresample_internal;
 
 import ffmpeg.libswresample.swresample_version;
+
+import std.stdint;
+
 import ffmpeg.libavutil.channel_layout;
+import ffmpeg.libavutil.samplefmt;
 
 @nogc nothrow extern(C):
 
@@ -11,7 +15,15 @@ enum SQRT3_2 = 1.22474487139158904909;  /* sqrt(3/2) */
 
 enum NS_TAPS = 20;
 
-
+struct AudioData{
+    uint8_t[SWR_CH_MAX] *ch;    ///< samples buffer per channel
+    uint8_t *data;              ///< samples buffer
+    int ch_count;               ///< number of channels
+    int bps;                    ///< bytes per sample
+    int count;                  ///< number of samples
+    int planar;                 ///< 1 if planar audio, 0 otherwise
+    AVSampleFormat fmt;         ///< sample format
+}