| Commit message (Collapse) | Author | Age | Files | Lines |
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The plugin code tried to force libavcodec to supply stereo samples.
That however has never actually worked. By removing this code, we are
able to play surround files for the first time.
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If both tags (stream and decoder) are present, we prefer the stream tag.
Fixes #2698, where ICY tag contained useful information, but was
overwritten with bogus decoder tag data.
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output plugin
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This includes some default values of #defined constants used in the
code; it won't compile without it.
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This patch prepares support for floating point samples (and probably
other formats). It changes the meaning of the "bits" attribute from a
bit count to a symbolic value.
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Copied sources from
http://beesbuzz.biz/code/audiocompress/AudioCompress-2.0.tar.gz
[mk: created this patch under fluffy's name and fixed some gcc
signed/unsigned comparison warnings]
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This fixes a regression due to a typo caused by "decoder: use
audio_format_init_checked()".
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Support 32 bit samples with software mixer.
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Conflicts:
NEWS
configure.ac
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Removed the "vtrack" local variable (which triggered a gcc warning
because it was after the newly introduced NULL check), and run
strtol() on the original parameter.
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The function flac_vtrack_tnum() was missing a strrchr()==NULL check.
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Allow RIFF/AIFF ID3 tags up to 4 MB (old limit was 256 kB). This
might still be too small for some users, and when somebody complains,
we might do something more clever (like streaming the data into
libid3tag?).
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On some platforms, libavcodec wants the output buffer aligned to 16
bytes (because it uses SSE/Altivec internally). It will segfault when
you don't obey this rule.
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Pass the audiofile_setup_sample_format() result to
audio_format_init_checked().
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Don't maintain the current time stamp in a floating point variable,
because this is subject to rounding errors.
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More exact total time.
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Negative return values are not documented here, but since the function
prototype is signed, let's be sure.
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Don't maintain the current time stamp in a floating point variable,
because this is subject to rounding errors.
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The new option "sample_rate" sets the sample rate for libmikmod.
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These functions are trivial, we don't need them separate.
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Don't allocate this object, put it on the stack.
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Unified function for converting an audio_format object to a string,
for log messages and for the "status" command.
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Let the audio_check library verify the audio format in all (relevant,
i.e. non-hardcoded) plugins.
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These functions are a wrapper for audio_valid_X(). On error, they
return a GError object.
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Internally, use only the integer time. When needed, convert it to a
floating point seconds value.
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At this point, the function may be called from the SEEK handler.
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When waiting for the decoder to provide more data, the player thread
generates silence chunks if needed. However, it forgot to initialize
the chunk.times attribute, which had now an undefined value. This
patch sets it to -1.0, meaning "value is undefined". Add a ">= 0.0"
check to audio_output_all_check(). This fixes spurious relative
seeking errors, because sometimes, the "elapsed" value falls back to
0.0.
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Temporary editor files.
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After we've been hit by Large File Support problems several times in
the past week (which only occur on 32 bit platforms, which I don't
have), this is yet another attempt to fix the issue.
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*sigh* another Large File breakage. ino_t/dev_t this time. We need
to include config.h in directory.h to get this straight.
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Remove the OPEN_2CH_MAX option. MPD's support for surround sound is
still clunky, but we're working on it.
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