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Another partial frame fix: the silence buffer was 1020 bytes, which
had room for 127.5 24 bit stereo frames. Don't send the partial last
frame in this case.
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24 bit output is as important as 16 bit output. Provide a
pcm_convert() implementation which can convert to 24 bit with as
little quality loss as possible.
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The old pcm_convert_size() ignored most of the destination format,
e.g. it did not check its sample size, and assumed it is 16 bit.
Simplify and universalize it by using audio_format_frame_size().
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pcm_convert() converted only to 16 bit. To be able to support other
sample sizes, move that 16 bit specific code to a separate function.
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The 24 bit implementation is mostly copy'n'paste of the 16 bit
version, except that the data type is int32_t instead of int16_t.
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Separate code from pcm_utils.c to keep it small and simple.
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Similar to pcm_resample_16(), implement pcm_resample_24(). The 24 bit
implementation is very similar, but it uses src_int_to_float_array()
instead of src_short_to_float_array() before sending data to
libsamplerate.
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A future patch will implement a 24 bit resampler. To unify code, move
code which can be shared to a separate function.
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Copy from source to destination buffer directly, don't use the
temporary variables "lsample" and "rsample".
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Added assertions which ensure that there are no partial samples in the
source buffer.
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Use sizeof(sample) instead of hard-coding "2". Although we're in 16
bit right now, this will make code sharing easier when we support
other sample sizes.
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Separate the resampling code from the rest of pcm_utils.c. Create two
sub-libraries: pcm_resample_libsamplerate.c and
pcm_resample_fallback.c.
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Due to a logic error, no value was valid for the boolean value
parser. Replace "||" with "&&".
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libmad produces samples of more than 24 bit. Rounding that down to 16
bits using dithering makes those people lose quality who have a 24 bit
capable sound device. Send 24 bit PCM data, and let the receiver
decide whether to apply 16 bit dithering.
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We are going to convert the code to 24 bit; don't hard-code a sample
size of 2 bytes.
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Copied and adapted code from the mp3 decoder plugin. This library now
replaces the old and low-quality function pcm_convert_24_to_16().
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I added 24 bit support a while ago, but it wasn't possible to force 24
bit output. Add 24 and 8 bit to the list of allowed sample sizes.
Although 8 bit audio isn't as widely used as 24 bit, there is no
reason to exclude it.
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Splitting a frame between two buffer chunks causes distortion in the
output. MPD used to assume that the chunk size 1020 would never cause
splitted frames, but that isn't the case for 24 bit stereo (127.5
frames), and even less for files with even more channels.
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Don't attempt to map paths which are already absolute with
map_song_fs(): check with song_in_database() instead of
song_is_file().
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The "rm" command did not send notifications to idle clients. Add it.
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The function deletePlaylist() shouldn't be in playlist.c.
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Don't use our deprecated linked list library, use GLib's GPtrArray
instead.
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The "long" result of strtol() was implicitly casted down to a 32 bit
integer. Add some range checking instead.
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Many command arguments must not be negative; add a separate
parser/checker function for that. For the same reason, add
check_bool(). This eliminates two strange special cases handlers from
check_int().
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Pass index arguments as unsigned integers. They must not be negative,
and even if some caller accidently passes -1, it won't pass the bound
checks (since it's now 2**32-1).
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There are some integers which have a "magic" -1 value which means
"undefined" or "nothing". All others can be converted to unsigned,
since they must not contain a negative number.
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Instead of returning 0 for success and -1 for failure, return true or
false. This seems more natural.
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Also add names for "error" and "ok". I don't like passing anonymous
integer codes around.
This is not yet complete: lots of functions (e.g. in playlist.c)
follow the same convention of -1/0, and these have to be adapted, too.
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http://xkcd.com/292/
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Eliminate CamelCase in all public and static functions.
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The typedef CommandHandlerFunction is only used once. Move its type
into the command struct.
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Send last modification timestamps to the client. This allows the
client to see when another client modifies a stored playlist.
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"listplaylists" returns a list of all stored playlists. This command
seems more elaborate than listing them below "lsinfo".
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The function print_spl_list() replaces the old function lsPlaylists()
from ls.c.
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spl_list() provides an interface for enumerating all stored playlists.
This separates the internal playlist logic from the protocol specific
function lsPlaylists().
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The return value of spl_append_uri() was somewhat buggy: some branches
returned ACK_* values, and some an enum playlist_result. Unify this.
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Rename addToStoredPlaylist() to spl_append_uri(), and remove the
clearStoredPlaylist() macro.
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The two functions clearStoredPlaylist() and addToStoredPlaylist()
don't belong into playlist.c. clearStoredPlaylist() was a wrapper for
spl_clear(), and is converted into a CPP macro for now.
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Renamed all public functions, prefix is "spl_".
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No CamelCase in file names.
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The list of commands is known at compile time. Instead of creating a
linked list on startup, we can just register all commands in a static
sorted array.
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The command pointers which are passed around aren't being modified -
in fact, no command pointer must be modified once it has been added to
the commandList.
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No CamelCase and no struct typedefs.
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Instead of manually calling memset(0) on the pcm_convert_state struct,
client code should use a library function from pcm_utils.c. This way,
we can change the semantics of the struct easily.
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Renamed all functions which were still in CamelCase.
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No CamelCase, and a struct instead of a typedef.
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Casting a pointer to some sort of integer and formatting it into a
string isn't valid. A pointer derived from this hex string won't work
reliably. Since ffmpeg doesn't provide a nice API for passing our
pointer, we have to think of a different hack: ffmpeg passes the exact
URL pointer to mpdurl_open(), and we can make this string part of a
struct. This reduces the problem to casting the string back to the
struct.
This is still a workaround, but this is "sort of portable", unless the
ffmpeg people start messing with the URL pointer (which would be valid
according to the API definition).
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Since ffmpeg svn r12865, you have to include libavcodec/avcodec.h
instead of avcodec.h. This cannot be checked at compile time, instead
we have to add a check to configure.ac. Viliam's original ffmpeg
plugin was based on the newer ffmpeg library, while my Debian
installation had the older version. My attempt to correct his include
statements wasn't correct after all.
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{song,dir}vec_for_each each failed to gracefully handle deleted
files when iterating through. While we were thread-safe, we
were not safe within the calling thread. If a callback we
passed caused sv->nr to shring, our index would still increment;
causing files to stay in the database.
A way to test this is to remove 10 or so contiguous songs from a
>10 song directory.
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