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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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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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MPD has been supporting 32 bit samples since version 0.15. This patch
changes one check, and removes the 32->24 conversion code.
Note that WavPack floating point samples have 32 bits, and MPD doesn't
have a special check for floating point - therefore, this WavPack
plugin still returns 24 bit integer samples as before (until we have
float support in the MPD core).
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Call decoder_initialize() before entering the loop. We don't need to
call ov_read() before ov_info(). When the stream number changes,
check if the audio format is still the same.
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Use the struct name instead.
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This is done by audio_format_init().
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Return FLAC__STREAM_DECODER_SEEK_STATUS_UNSUPPORTED if this input
stream does not support seeking.
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Remove the audio_format attribute, add "frame_size" instead. The
audio_format initialization and check is moved both to
flac_data_get_audio_format().
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Use the sample rate stored in the stream_info struct instead of the
audio_format struct.
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When calculating the properties of the frame, use sample_rate and
other information from the frame header instead of the stored
audio_format object.
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