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off with white noise)
git-svn-id: https://svn.musicpd.org/mpd/trunk@6952 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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ogg_stream_type_detect may not be compiled correctly
when compiling FLAC (1.1.4+) without Vorbis
git-svn-id: https://svn.musicpd.org/mpd/trunk@6896 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@6888 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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Both mp4 and (ogg)flac inputPlugins got HTTP inputStream support
later in the game, so their calls to sendDataToOutputBuffer()
didn't get updated to support buffering while the outputBuffer
was full. This fixes it.
git-svn-id: https://svn.musicpd.org/mpd/trunk@6873 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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the force flag will issue FATAL() if an invalid value is
specified
git-svn-id: https://svn.musicpd.org/mpd/trunk@6857 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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For the default: case, just use the error message that libFLAC
provides instead of using something ambiguous. Also, this gets
rid of long lines in the code, making it easier to digest.
Of course, we save ~100 bytes of text space in the process :)
git-svn-id: https://svn.musicpd.org/mpd/trunk@6830 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@6826 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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Parse ReplayGain info in LAME tags and use it if no ID3v2 ReplayGain tags
are found. This is currently a bit unsafe, as apparently some LAME tags
have bogus ReplayGain values. But I'm finding a lot of MP3s with valid
LAME tags that fail the LAME tag CRC check. So until I figure out why
that's happening, it's an unreliable method for checking if the LAME tag is
valid.
A big thanks to tmz for writing the original patch.
git-svn-id: https://svn.musicpd.org/mpd/trunk@6798 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@6736 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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Turns out the fix was as simple as specifying the OPEN_TAGS flag when
opening the file. Thanks again to Kodest for figuring this one out.
git-svn-id: https://svn.musicpd.org/mpd/trunk@6657 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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This ReplayGain code is currently disabled because WavpackGetTagItem can't
seem to find replaygain_* fields in APEv2 tags (which is how wvgain stores
ReplayGain values). Additionally, because APEv2 tags are stored at the end
of the file, this code is only implemented for regular files and not HTTP
streams. Using HTTP seeking it *may* be possible to implement it for both.
git-svn-id: https://svn.musicpd.org/mpd/trunk@6656 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@6654 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@6651 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@6483 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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have any effect until the aac and mp4 input plugins actually support a
stream decoding API.
git-svn-id: https://svn.musicpd.org/mpd/trunk@6481 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@6468 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@6225 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@5894 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@5834 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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previous song played will be reused.
git-svn-id: https://svn.musicpd.org/mpd/trunk@5791 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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forever without this.
git-svn-id: https://svn.musicpd.org/mpd/trunk@5790 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@5492 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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size_t (1.1.3) makes a lot more sense, but older flac used unsigned
here...
git-svn-id: https://svn.musicpd.org/mpd/trunk@5258 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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Some compilers and linkers aren't smart enough to optimize this,
as global variables are implictly initialized to zero. As a
result, binaries are a bit smaller as more goes in the .bss and
less in the text section.
git-svn-id: https://svn.musicpd.org/mpd/trunk@5254 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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We'll be dealing with legacy server configurations for a long
time to come.
git-svn-id: https://svn.musicpd.org/mpd/trunk@5253 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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sendDataToOutputBuffer returns an int (and always has), and
using the existing 'ret' is fine in mp3Read().
git-svn-id: https://svn.musicpd.org/mpd/trunk@5246 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@5244 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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FLAC__{seekable_,}_stream_decoder_new() takes no arguments
git-svn-id: https://svn.musicpd.org/mpd/trunk@5241 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@5222 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@5165 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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MP3 playback, thus allowing songs that run longer than the Xing frame
claims (f.e., an MP3 created by catting two MP3s together) to continue
playing past the end.
git-svn-id: https://svn.musicpd.org/mpd/trunk@5157 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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assumption that non-seekable streams are live and any gapless info is
incorrect.
git-svn-id: https://svn.musicpd.org/mpd/trunk@5150 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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Instead, stop decoding as soon as we've found the frames/samples at the
"end" that we want drop.
git-svn-id: https://svn.musicpd.org/mpd/trunk@5149 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@5148 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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though in practice it should never matter.
git-svn-id: https://svn.musicpd.org/mpd/trunk@5147 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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This means that when using libFLAC as a shared object,
OggFLAC support is dependent on the compile-time options of
the libFLAC library loaded.
git-svn-id: https://svn.musicpd.org/mpd/trunk@5112 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@5111 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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We will restore compatibility with the old API in the
next few commits; along with OggFLAC support.
git-svn-id: https://svn.musicpd.org/mpd/trunk@5110 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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move flac_decode to the bottom, so we don't have to declare
all of our static functions.
git-svn-id: https://svn.musicpd.org/mpd/trunk@5109 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@4912 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@4876 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@4866 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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I'm checking for zero-size allocations and assert()-ing them,
so we can more easily get backtraces and debug problems, but we'll
also allow -DNDEBUG people to live on the edge if they wish.
We do not rely on errno when checking for OOM errors because
some implementations of malloc do not set it, and malloc
is commonly overridden by userspace wrappers.
I've spent some time looking through the source and didn't find any
obvious places where we would explicitly allocate 0 bytes, so we
shouldn't trip any of those assertions.
We also avoid allocating zero bytes because C libraries don't
handle this consistently (some return NULL, some not); and it's
dangerous either way.
git-svn-id: https://svn.musicpd.org/mpd/trunk@4690 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@4684 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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This finally fixes a bug from over two years ago playing a wave file
(oprah.wav) with the following characteristics (from sfinfo):
File Format Microsoft RIFF WAVE Format (wave)
Data Format 8-bit integer (unsigned, little endian)
Audio Data 986827 bytes begins at offset 58 (3a hex)
1 channel, 986827 frames
Sampling Rate 22050.00 Hz
Duration 44.754 seconds
Of course, this has been regression tested with all the files
that the previous commit got working. Thanks to Michael Pruett
(audiofile author) for the hint and shame on me for forgetting
about it for over two years :x
git-svn-id: https://svn.musicpd.org/mpd/trunk@4682 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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Use the 'Virtual' variants of afGetSampleFormat, afGetChannels,
afGetVirtualFrameSize in the audiofile library, since it already does
the necessary abstraction for us.
Of course, I've regression tested these changes against my
standard 44100Hz/2ch/16bit wave files and they continue to play
fine.
Files tested:
english.au (Linus Torvalds pronouncing 'Linux' in English)
B01.Red_Bright_Heart.au (Chinese opera, sounds correct to me even though
I don't actually understand the words)
git-svn-id: https://svn.musicpd.org/mpd/trunk@4681 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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apps write them in all caps.
git-svn-id: https://svn.musicpd.org/mpd/trunk@4672 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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the layer 1 frames looking for a layer 2 or 3 frame.
git-svn-id: https://svn.musicpd.org/mpd/trunk@4671 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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decode a frame (which will automatically decode the next header without allowing us to do some checks on it).
git-svn-id: https://svn.musicpd.org/mpd/trunk@4670 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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since it hasn't been tested.
git-svn-id: https://svn.musicpd.org/mpd/trunk@4669 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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