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oslibs, codecs: rebuilt opusfile for osx and windows from the latest 0.8 release from xiph.org.
git-svn-id: svn://svn.code.sf.net/p/quakespasm/code/trunk/quakespasm@1332 af15c1b1-3010-417e-b628-4374ebc0bcbd
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5 changed files with 42 additions and 24 deletions
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@ -1713,11 +1713,11 @@ int op_pcm_seek(OggOpusFile *_of,ogg_int64_t _pcm_offset) OP_ARG_NONNULL(1);
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These downmix multichannel files to two channels, so they can always return
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samples in the same format for every link in a chained file.
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If the rest of your audio processing chain can handle floating point, those
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routines should be preferred, as floating point output avoids introducing
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clipping and other issues which might be avoided entirely if, e.g., you
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scale down the volume at some other stage.
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However, if you intend to direct consume 16-bit samples, the conversion in
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If the rest of your audio processing chain can handle floating point, the
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floating-point routines should be preferred, as they prevent clipping and
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other issues which might be avoided entirely if, e.g., you scale down the
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volume at some other stage.
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However, if you intend to consume 16-bit samples directly, the conversion in
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<tt>libopusfile</tt> provides noise-shaping dithering and, if compiled
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against <tt>libopus</tt> 1.1 or later, soft-clipping prevention.
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@ -1770,26 +1770,35 @@ int op_pcm_seek(OggOpusFile *_of,ogg_int64_t _pcm_offset) OP_ARG_NONNULL(1);
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#OP_DEC_FORMAT_FLOAT.
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\param _li The index of the link from which this packet was decoded.
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\return A non-negative value on success, or a negative value on error.
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The error codes should be the same as those returned by
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Any error codes should be the same as those returned by
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opus_multistream_decode() or opus_multistream_decode_float().
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Success codes are as follows:
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\retval 0 Decoding was successful.
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The application has filled the buffer with
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exactly <code>\a _nsamples*\a
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_nchannels</code> samples in the requested
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format.
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\retval #OP_DEC_USE_DEFAULT No decoding was done.
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<tt>libopusfile</tt> should decode normally
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instead.*/
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<tt>libopusfile</tt> should do the decoding
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by itself instead.*/
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typedef int (*op_decode_cb_func)(void *_ctx,OpusMSDecoder *_decoder,void *_pcm,
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const ogg_packet *_op,int _nsamples,int _nchannels,int _format,int _li);
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/**Sets the packet decode callback function.
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This is called once for each packet that needs to be decoded.
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If set, this is called once for each packet that needs to be decoded.
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This can be used by advanced applications to do additional processing on the
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compressed or uncompressed data.
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For example, an application might save the final entropy coder state for
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debugging and testing purposes, or it might apply additional filters
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before the downmixing, dithering, or soft-clipping performed by
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<tt>libopusfile</tt>, so long as these filters do not introduce any
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latency.
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A call to this function is no guarantee that the audio will eventually be
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delivered to the application.
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Some or all of the data from the packet may be discarded (i.e., at the
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beginning or end of a link, or after a seek), however the callback is
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required to provide all of it.
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<tt>libopusfile</tt> may discard some or all of the decoded audio data
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(i.e., at the beginning or end of a link, or after a seek), however the
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callback is still required to provide all of it.
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\param _of The \c OggOpusFile on which to set the decode callback.
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\param _decode_cb The callback function to call.
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This may be <code>NULL</code> to disable calling the
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@ -1713,11 +1713,11 @@ int op_pcm_seek(OggOpusFile *_of,ogg_int64_t _pcm_offset) OP_ARG_NONNULL(1);
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These downmix multichannel files to two channels, so they can always return
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samples in the same format for every link in a chained file.
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If the rest of your audio processing chain can handle floating point, those
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routines should be preferred, as floating point output avoids introducing
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clipping and other issues which might be avoided entirely if, e.g., you
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scale down the volume at some other stage.
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However, if you intend to direct consume 16-bit samples, the conversion in
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If the rest of your audio processing chain can handle floating point, the
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floating-point routines should be preferred, as they prevent clipping and
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other issues which might be avoided entirely if, e.g., you scale down the
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volume at some other stage.
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However, if you intend to consume 16-bit samples directly, the conversion in
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<tt>libopusfile</tt> provides noise-shaping dithering and, if compiled
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against <tt>libopus</tt> 1.1 or later, soft-clipping prevention.
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@ -1770,26 +1770,35 @@ int op_pcm_seek(OggOpusFile *_of,ogg_int64_t _pcm_offset) OP_ARG_NONNULL(1);
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#OP_DEC_FORMAT_FLOAT.
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\param _li The index of the link from which this packet was decoded.
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\return A non-negative value on success, or a negative value on error.
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The error codes should be the same as those returned by
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Any error codes should be the same as those returned by
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opus_multistream_decode() or opus_multistream_decode_float().
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Success codes are as follows:
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\retval 0 Decoding was successful.
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The application has filled the buffer with
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exactly <code>\a _nsamples*\a
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_nchannels</code> samples in the requested
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format.
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\retval #OP_DEC_USE_DEFAULT No decoding was done.
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<tt>libopusfile</tt> should decode normally
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instead.*/
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<tt>libopusfile</tt> should do the decoding
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by itself instead.*/
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typedef int (*op_decode_cb_func)(void *_ctx,OpusMSDecoder *_decoder,void *_pcm,
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const ogg_packet *_op,int _nsamples,int _nchannels,int _format,int _li);
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/**Sets the packet decode callback function.
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This is called once for each packet that needs to be decoded.
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If set, this is called once for each packet that needs to be decoded.
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This can be used by advanced applications to do additional processing on the
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compressed or uncompressed data.
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For example, an application might save the final entropy coder state for
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debugging and testing purposes, or it might apply additional filters
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before the downmixing, dithering, or soft-clipping performed by
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<tt>libopusfile</tt>, so long as these filters do not introduce any
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latency.
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A call to this function is no guarantee that the audio will eventually be
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delivered to the application.
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Some or all of the data from the packet may be discarded (i.e., at the
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beginning or end of a link, or after a seek), however the callback is
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required to provide all of it.
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<tt>libopusfile</tt> may discard some or all of the decoded audio data
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(i.e., at the beginning or end of a link, or after a seek), however the
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callback is still required to provide all of it.
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\param _of The \c OggOpusFile on which to set the decode callback.
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\param _decode_cb The callback function to call.
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This may be <code>NULL</code> to disable calling the
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