#380: correctly skip no-op transcodes

The format key is now the (lower-cased) format name string used by beets,
which means we can precisely detect which transcodes would be unnecessary. To
facilitate this, I added an ALIASES dict which allows more convenient names to
work for this (e.g., "wma" is easier to remember than "windows media").

beetsplug/convert.py

@@ -31,6 +31,12 @@ DEVNULL = open(os.devnull, 'wb')
 _fs_lock = threading.Lock()
 _temp_files = []  # Keep track of temporary transcoded files for deletion.
 
+# Some convenient alternate names for formats.
+ALIASES = {
+    u'wma': u'windows media',
+    u'vorbis': u'ogg',
+}
+
 
 def _destination(dest_dir, item, keep_new, path_formats):
     """Return the path under `dest_dir` where the file should be placed
@@ -50,7 +56,8 @@ def _destination(dest_dir, item, keep_new, path_formats):
 def get_format():
     """Get the currently configured format command and extension.
     """
-    format = config['convert']['format'].get(unicode)
+    format = config['convert']['format'].get(unicode).lower()
+    format = ALIASES.get(format, format)
     format_info = config['convert']['formats'][format].get(dict)
     try:
         return (format_info['command'].split(),
@@ -89,7 +96,9 @@ def encode(source, dest):
         return
 
     if not quiet:
-      log.info(u'Finished encoding {0}'.format(util.displayable_path(source)))
+        log.info(u'Finished encoding {0}'.format(
+            util.displayable_path(source))
+        )
 
 
 def should_transcode(item):
@@ -97,8 +106,9 @@ def should_transcode(item):
     conversion (i.e., its bitrate is high or it has the wrong format).
     """
     maxbr = config['convert']['max_bitrate'].get(int)
-
-    return item.format not in ['AAC', 'MP3', 'Opus', 'OGG', 'Windows Media'] or item.bitrate >= 1000 * maxbr
+    format_name = config['convert']['format'].get(unicode)
+    return format_name.lower() == item.format.lower() or \
+            item.bitrate >= 1000 * maxbr
 
 
 def convert_item(dest_dir, keep_new, path_formats):
@@ -235,11 +245,11 @@ class ConvertPlugin(BeetsPlugin):
                     u'command': u'ffmpeg -i $source -y -acodec libopus -vn -ab 96k $dest',
                     u'extension': u'opus',
                 },
-                u'vorbis': {
+                u'ogg': {
                     u'command': u'ffmpeg -i $source -y -acodec libvorbis -vn -aq 2 $dest',
                     u'extension': u'ogg',
                 },
-                u'wma': {
+                u'windows media': {
                     u'command': u'ffmpeg -i $source -y -acodec wmav2 -vn $dest',
                     u'extension': u'wma',
                 },

docs/plugins/convert.rst

@@ -1,13 +1,15 @@
 Convert Plugin
 ==============
 
-The ``convert`` plugin lets you convert parts of your collection to a directory
-of your choice. It converts all input formats supported by `FFmpeg`_ to MP3.
-It will skip files that are already present in the target directory. Converted
-files follow the same path formats as your library.
+The ``convert`` plugin lets you convert parts of your collection to a
+directory of your choice, transcoding audio and embedding album art along the
+way. It can transcode to and from any format using a configurable command
+line. It will skip files that are already present in the target directory.
+Converted files follow the same path formats as your library.
 
 .. _FFmpeg: http://ffmpeg.org
 
+
 Installation
 ------------
 
@@ -20,6 +22,7 @@ by the plugin. Otherwise, configure the plugin to locate the executable::
     convert:
         ffmpeg: /usr/bin/ffmpeg
 
+
 Usage
 -----
 
@@ -52,11 +55,12 @@ The plugin offers several configuration options, all of which live under the
   transcoded and those with a lower bitrate will simply be copied. Note that
   this does not guarantee that all converted files will have a lower
   bitrate---that depends on the encoder and its configuration.
-* ``format`` specify which format preset you would like to use. Default: mp3.
-* ``formats`` lets you specify additional formats to convert to. Presets for
-  AAC, ALAC, FLAC, MP3, Opus, Vorbis and Windows Meda are provided, however
-  support may vary depending on your ffmpeg library. Each format is defined as
-  a command and a file extension.
+* ``format`` is the name of the audio file format to transcode to. Files that
+  are already in the format (and are below the maximum bitrate) will not be
+  transcoded. Available formats include MP3, AAC, ALAC, FLAC, Opus, Vorbis,
+  and Windows Media; the default is MP3.
+* ``formats`` lets you specify additional formats to convert to. Each format
+  is defined as a command and a file extension.
 * ``auto`` gives you the option to import transcoded versions of your files
   automatically during the ``import`` command. With this option enabled, the
   importer will transcode all non-MP3 files over the maximum bitrate before
@@ -97,4 +101,4 @@ use ``$$`` to emit a dollars sign.
 
 In this example ``-aq <num>`` is equivalent to the LAME option ``-V num``. If
 you want to specify a bitrate, use ``-ab <bitrate>``. Refer to the `FFmpeg`_
-documentation for more details.
+documentation for more details.