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oslibs: macosx, windows: recompile libopusfile, using the latest

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from xiph.org git repo.

git-svn-id: svn+ssh://svn.code.sf.net/p/quakespasm/code/trunk@855 af15c1b1-3010-417e-b628-4374ebc0bcbd
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sezero 2013-07-17 12:51:34 +00:00
parent 683c9ec2f8
commit 82434dfbe6
9 changed files with 246 additions and 30 deletions
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138
quakespasm/MacOSX/codecs/include/opus/opusfile.h
View file

@ -49,8 +49,55 @@
Several additional sections are not tied to the main API.
- \ref stream_callbacks
- \ref header_info
- \ref error_codes*/
- \ref error_codes
\section Overview
The <tt>libopusfile</tt> API always decodes files to 48&nbsp;kHz.
The original sample rate is not preserved by the lossy compression, though
it is stored in the header to allow you to resample to it after decoding
(the <tt>libopusfile</tt> API does not currently provide a resampler,
but the
<a href="http://www.speex.org/docs/manual/speex-manual/node7.html#SECTION00760000000000000000">the
Speex resampler</a> is a good choice if you need one).
In general, if you are playing back the audio, you should leave it at
48&nbsp;kHz, provided your audio hardware supports it.
When decoding to a file, it may be worth resampling back to the original
sample rate, so as not to surprise users who might not expect the sample
rate to change after encoding to Opus and decoding.
Opus files can contain anywhere from 1 to 255 channels of audio.
The channel mappings for up to 8 channels are the same as the
<a href="http://www.xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-800004.3.9">Vorbis
mappings</a>.
A special stereo API can convert everything to 2 channels, making it simple
to support multichannel files in a application which only has stereo
output.
Although the <tt>libopusfile</tt> ABI provides support for the theoretical
maximum number of channels, the current implementation does not support
files with more than 8 channels, as they do not have well-defined channel
mappings.
Like all Ogg files, Opus files may be "chained".
That is, multiple Opus files may be combined into a single, longer file just
by concatenating the original files.
This is commonly done in internet radio streaming, as it allows the title
and artist to be updated each time the song changes, since each link in the
chain includes its own set of metadata.
<tt>libopusfile</tt> fully supports chained files.
It will decode the first Opus stream found in each link of a chained file
(ignoring any other streams that might be concurrently multiplexed with it,
such as a video stream).
The channel count can also change between links, but if your application is
not prepared to deal with this, it can use the stereo API to ensure the
audio from all links will always get decoded into a common format.
Since <tt>libopusfile</tt> always decodes to 48&nbsp;kHz, you do not have to
worry about the sample rate changing between links (as was possible with
Vorbis).
This makes application support for chained files with <tt>libopusfile</tt>
very easy.*/
# if defined(__cplusplus)
extern "C" {
@ -182,8 +229,9 @@ struct OpusHead{
opus_uint32 input_sample_rate;
/**The gain to apply to the decoded output, in dB, as a Q8 value in the range
-32768...32767.
The decoder will automatically scale the output by
pow(10,output_gain/(20.0*256)).*/
The <tt>libopusfile</tt> API will automatically apply this gain to the
decoded output before returning it, scaling it by
<code>pow(10,output_gain/(20.0*256))</code>.*/
int output_gain;
/**The channel mapping family, in the range 0...255.
Channel mapping family 0 covers mono or stereo in a single stream.
@ -385,6 +433,24 @@ const char *opus_tags_query(const OpusTags *_tags,const char *_tag,int _count)
int opus_tags_query_count(const OpusTags *_tags,const char *_tag)
OP_ARG_NONNULL(1) OP_ARG_NONNULL(2);
/**Get the track gain from an R128_TRACK_GAIN tag, if one was specified.
This searches for the first R128_TRACK_GAIN tag with a valid signed,
16-bit decimal integer value and returns the value.
This routine is exposed merely for convenience for applications which wish
to do something special with the track gain (i.e., display it).
If you simply wish to apply the track gain instead of the header gain, you
can use op_set_gain_offset() with an #OP_TRACK_GAIN type and no offset.
\param _tags An initialized #OpusTags structure.
\param[out] _gain_q8 The track gain, in 1/256ths of a dB.
This will lie in the range [-32768,32767], and should
be applied in <em>addition</em> to the header gain.
On error, no value is returned, and the previous
contents remain unchanged.
\return 0 on success, or a negative value on error.
\retval OP_EFALSE There was no track gain available in the given tags.*/
int opus_tags_get_track_gain(const OpusTags *_tags,int *_gain_q8)
OP_ARG_NONNULL(1) OP_ARG_NONNULL(2);
/**Clears the #OpusTags structure.
This should be called on an #OpusTags structure after it is no longer
needed.
@ -841,7 +907,11 @@ OP_WARN_UNUSED_RESULT OggOpusFile *op_open_url(const char *_url,
<dd>The first or last timestamp in a link failed
basic validity checks.</dd>
</dl>
\return A freshly opened \c OggOpusFile, or <code>NULL</code> on error.*/
\return A freshly opened \c OggOpusFile, or <code>NULL</code> on error.
<tt>libopusfile</tt> does <em>not</em> take ownership of the source
if the call fails.
The calling application is responsible for closing the source if
this call returns an error.*/
OP_WARN_UNUSED_RESULT OggOpusFile *op_open_callbacks(void *_source,
const OpusFileCallbacks *_cb,const unsigned char *_initial_data,
size_t _initial_bytes,int *_error) OP_ARG_NONNULL(2);
@ -974,7 +1044,11 @@ OP_WARN_UNUSED_RESULT OggOpusFile *op_test_url(const char *_url,
the failure code.
See op_open_callbacks() for a full list of failure
codes.
\return A partially opened \c OggOpusFile, or <code>NULL</code> on error.*/
\return A partially opened \c OggOpusFile, or <code>NULL</code> on error.
<tt>libopusfile</tt> does <em>not</em> take ownership of the source
if the call fails.
The calling application is responsible for closing the source if
this call returns an error.*/
OP_WARN_UNUSED_RESULT OggOpusFile *op_test_callbacks(void *_source,
const OpusFileCallbacks *_cb,const unsigned char *_initial_data,
size_t _initial_bytes,int *_error) OP_ARG_NONNULL(2);
@ -1303,11 +1377,12 @@ int op_pcm_seek(OggOpusFile *_of,ogg_int64_t _pcm_offset) OP_ARG_NONNULL(1);
clipping and other issues which might be avoided entirely if, e.g., you
scale down the volume at some other stage.
However, if you intend to direct consume 16-bit samples, the conversion in
<tt>libopusfile</tt> provides noise-shaping dithering API.
<tt>libopusfile</tt> provides noise-shaping dithering and, if compiled
against <tt>libopus</tt>&nbsp;1.1 or later, soft-clipping prevention.
<tt>libopusfile</tt> can also be configured at compile time to use the
fixed-point <tt>libopus</tt> API.
If so, the floating-point API may also be disabled.
If so, <tt>libopusfile</tt>'s floating-point API may also be disabled.
In that configuration, nothing in <tt>libopusfile</tt> will use any
floating-point operations, to simplify support on devices without an
adequate FPU.
@ -1323,13 +1398,46 @@ int op_pcm_seek(OggOpusFile *_of,ogg_int64_t _pcm_offset) OP_ARG_NONNULL(1);
appropriately.*/
/*@{*/
/**Gain offset type that indicates that the provided offset is relative to the
header gain.
This is the default.*/
#define OP_HEADER_GAIN (0)
/**Gain offset type that indicates that the provided offset is relative to the
R128_TRACK_GAIN value (if any), in addition to the header gain.*/
#define OP_TRACK_GAIN (3008)
/**Gain offset type that indicates that the provided offset should be used as
the gain directly, without applying any the header or track gains.*/
#define OP_ABSOLUTE_GAIN (3009)
/**Sets the gain to be used for decoded output.
By default, the gain in the header is applied with no additional offset.
The total gain (including header gain and/or track gain, if applicable, and
this offset), will be clamped to [-32768,32767]/256 dB.
This is more than enough to saturate or underflow 16-bit PCM.
\note The new gain will not be applied to any already buffered, decoded
output.
This means you cannot change it sample-by-sample, as at best it will be
updated packet-by-packet.
It is meant for setting a target volume level, rather than applying smooth
fades, etc.
\param _of The \c OggOpusFile on which to set the gain offset.
\param _gain_type One of #OP_HEADER_GAIN, #OP_TRACK_GAIN, or
#OP_ABSOLUTE_GAIN.
\param _gain_offset_q8 The gain offset to apply, in 1/256ths of a dB.
\return 0 on success or a negative value on error.
\retval #OP_EINVAL The \a _gain_type was unrecognized.*/
int op_set_gain_offset(OggOpusFile *_of,
int _gain_type,opus_int32 _gain_offset_q8);
/**Reads more samples from the stream.
\note Although \a _buf_size must indicate the total number of values that
can be stored in \a _pcm, the return value is the number of samples
<em>per channel</em>.
This is done because
<ol>
<li>The channel count cannot be known a prior (reading more samples might
<li>The channel count cannot be known a priori (reading more samples might
advance us into the next link, with a different channel count), so
\a _buf_size cannot also be in units of samples per channel,</li>
<li>Returning the samples per channel matches the <code>libopus</code> API
@ -1346,8 +1454,8 @@ int op_pcm_seek(OggOpusFile *_of,ogg_int64_t _pcm_offset) OP_ARG_NONNULL(1);
</ol>
\param _of The \c OggOpusFile from which to read.
\param[out] _pcm A buffer in which to store the output PCM samples, as
signed native-endian 16-bit values with a nominal
range of <code>[-32768,32767)</code>.
signed native-endian 16-bit values at 48&nbsp;kHz
with a nominal range of <code>[-32768,32767)</code>.
Multiple channels are interleaved using the
<a href="http://www.xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-800004.3.9">Vorbis
channel ordering</a>.
@ -1411,7 +1519,7 @@ OP_WARN_UNUSED_RESULT int op_read(OggOpusFile *_of,
can be stored in \a _pcm, the return value is the number of samples
<em>per channel</em>.
<ol>
<li>The channel count cannot be known a prior (reading more samples might
<li>The channel count cannot be known a priori (reading more samples might
advance us into the next link, with a different channel count), so
\a _buf_size cannot also be in units of samples per channel,</li>
<li>Returning the samples per channel matches the <code>libopus</code> API
@ -1428,7 +1536,7 @@ OP_WARN_UNUSED_RESULT int op_read(OggOpusFile *_of,
</ol>
\param _of The \c OggOpusFile from which to read.
\param[out] _pcm A buffer in which to store the output PCM samples as
signed floats with a nominal range of
signed floats at 48&nbsp;kHz with a nominal range of
<code>[-1.0,1.0]</code>.
Multiple channels are interleaved using the
<a href="http://www.xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-800004.3.9">Vorbis
@ -1497,8 +1605,8 @@ OP_WARN_UNUSED_RESULT int op_read_float(OggOpusFile *_of,
op_read().
\param _of The \c OggOpusFile from which to read.
\param[out] _pcm A buffer in which to store the output PCM samples, as
signed native-endian 16-bit values with a nominal
range of <code>[-32768,32767)</code>.
signed native-endian 16-bit values at 48&nbsp;kHz
with a nominal range of <code>[-32768,32767)</code>.
The left and right channels are interleaved in the
buffer.
This must have room for at least \a _buf_size values.
@ -1558,7 +1666,7 @@ OP_WARN_UNUSED_RESULT int op_read_stereo(OggOpusFile *_of,
op_read_float().
\param _of The \c OggOpusFile from which to read.
\param[out] _pcm A buffer in which to store the output PCM samples, as
signed floats with a nominal range of
signed floats at 48&nbsp;kHz with a nominal range of
<code>[-1.0,1.0]</code>.
The left and right channels are interleaved in the
buffer.

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quakespasm/MacOSX/codecs/lib/libopusfile.dylib
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138
quakespasm/Windows/codecs/include/opus/opusfile.h
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@ -49,8 +49,55 @@
Several additional sections are not tied to the main API.
- \ref stream_callbacks
- \ref header_info
- \ref error_codes*/
- \ref error_codes
\section Overview
The <tt>libopusfile</tt> API always decodes files to 48&nbsp;kHz.
The original sample rate is not preserved by the lossy compression, though
it is stored in the header to allow you to resample to it after decoding
(the <tt>libopusfile</tt> API does not currently provide a resampler,
but the
<a href="http://www.speex.org/docs/manual/speex-manual/node7.html#SECTION00760000000000000000">the
Speex resampler</a> is a good choice if you need one).
In general, if you are playing back the audio, you should leave it at
48&nbsp;kHz, provided your audio hardware supports it.
When decoding to a file, it may be worth resampling back to the original
sample rate, so as not to surprise users who might not expect the sample
rate to change after encoding to Opus and decoding.
Opus files can contain anywhere from 1 to 255 channels of audio.
The channel mappings for up to 8 channels are the same as the
<a href="http://www.xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-800004.3.9">Vorbis
mappings</a>.
A special stereo API can convert everything to 2 channels, making it simple
to support multichannel files in a application which only has stereo
output.
Although the <tt>libopusfile</tt> ABI provides support for the theoretical
maximum number of channels, the current implementation does not support
files with more than 8 channels, as they do not have well-defined channel
mappings.
Like all Ogg files, Opus files may be "chained".
That is, multiple Opus files may be combined into a single, longer file just
by concatenating the original files.
This is commonly done in internet radio streaming, as it allows the title
and artist to be updated each time the song changes, since each link in the
chain includes its own set of metadata.
<tt>libopusfile</tt> fully supports chained files.
It will decode the first Opus stream found in each link of a chained file
(ignoring any other streams that might be concurrently multiplexed with it,
such as a video stream).
The channel count can also change between links, but if your application is
not prepared to deal with this, it can use the stereo API to ensure the
audio from all links will always get decoded into a common format.
Since <tt>libopusfile</tt> always decodes to 48&nbsp;kHz, you do not have to
worry about the sample rate changing between links (as was possible with
Vorbis).
This makes application support for chained files with <tt>libopusfile</tt>
very easy.*/
# if defined(__cplusplus)
extern "C" {
@ -182,8 +229,9 @@ struct OpusHead{
opus_uint32 input_sample_rate;
/**The gain to apply to the decoded output, in dB, as a Q8 value in the range
-32768...32767.
The decoder will automatically scale the output by
pow(10,output_gain/(20.0*256)).*/
The <tt>libopusfile</tt> API will automatically apply this gain to the
decoded output before returning it, scaling it by
<code>pow(10,output_gain/(20.0*256))</code>.*/
int output_gain;
/**The channel mapping family, in the range 0...255.
Channel mapping family 0 covers mono or stereo in a single stream.
@ -385,6 +433,24 @@ const char *opus_tags_query(const OpusTags *_tags,const char *_tag,int _count)
int opus_tags_query_count(const OpusTags *_tags,const char *_tag)
OP_ARG_NONNULL(1) OP_ARG_NONNULL(2);
/**Get the track gain from an R128_TRACK_GAIN tag, if one was specified.
This searches for the first R128_TRACK_GAIN tag with a valid signed,
16-bit decimal integer value and returns the value.
This routine is exposed merely for convenience for applications which wish
to do something special with the track gain (i.e., display it).
If you simply wish to apply the track gain instead of the header gain, you
can use op_set_gain_offset() with an #OP_TRACK_GAIN type and no offset.
\param _tags An initialized #OpusTags structure.
\param[out] _gain_q8 The track gain, in 1/256ths of a dB.
This will lie in the range [-32768,32767], and should
be applied in <em>addition</em> to the header gain.
On error, no value is returned, and the previous
contents remain unchanged.
\return 0 on success, or a negative value on error.
\retval OP_EFALSE There was no track gain available in the given tags.*/
int opus_tags_get_track_gain(const OpusTags *_tags,int *_gain_q8)
OP_ARG_NONNULL(1) OP_ARG_NONNULL(2);
/**Clears the #OpusTags structure.
This should be called on an #OpusTags structure after it is no longer
needed.
@ -841,7 +907,11 @@ OP_WARN_UNUSED_RESULT OggOpusFile *op_open_url(const char *_url,
<dd>The first or last timestamp in a link failed
basic validity checks.</dd>
</dl>
\return A freshly opened \c OggOpusFile, or <code>NULL</code> on error.*/
\return A freshly opened \c OggOpusFile, or <code>NULL</code> on error.
<tt>libopusfile</tt> does <em>not</em> take ownership of the source
if the call fails.
The calling application is responsible for closing the source if
this call returns an error.*/
OP_WARN_UNUSED_RESULT OggOpusFile *op_open_callbacks(void *_source,
const OpusFileCallbacks *_cb,const unsigned char *_initial_data,
size_t _initial_bytes,int *_error) OP_ARG_NONNULL(2);
@ -974,7 +1044,11 @@ OP_WARN_UNUSED_RESULT OggOpusFile *op_test_url(const char *_url,
the failure code.
See op_open_callbacks() for a full list of failure
codes.
\return A partially opened \c OggOpusFile, or <code>NULL</code> on error.*/
\return A partially opened \c OggOpusFile, or <code>NULL</code> on error.
<tt>libopusfile</tt> does <em>not</em> take ownership of the source
if the call fails.
The calling application is responsible for closing the source if
this call returns an error.*/
OP_WARN_UNUSED_RESULT OggOpusFile *op_test_callbacks(void *_source,
const OpusFileCallbacks *_cb,const unsigned char *_initial_data,
size_t _initial_bytes,int *_error) OP_ARG_NONNULL(2);
@ -1303,11 +1377,12 @@ int op_pcm_seek(OggOpusFile *_of,ogg_int64_t _pcm_offset) OP_ARG_NONNULL(1);
clipping and other issues which might be avoided entirely if, e.g., you
scale down the volume at some other stage.
However, if you intend to direct consume 16-bit samples, the conversion in
<tt>libopusfile</tt> provides noise-shaping dithering API.
<tt>libopusfile</tt> provides noise-shaping dithering and, if compiled
against <tt>libopus</tt>&nbsp;1.1 or later, soft-clipping prevention.
<tt>libopusfile</tt> can also be configured at compile time to use the
fixed-point <tt>libopus</tt> API.
If so, the floating-point API may also be disabled.
If so, <tt>libopusfile</tt>'s floating-point API may also be disabled.
In that configuration, nothing in <tt>libopusfile</tt> will use any
floating-point operations, to simplify support on devices without an
adequate FPU.
@ -1323,13 +1398,46 @@ int op_pcm_seek(OggOpusFile *_of,ogg_int64_t _pcm_offset) OP_ARG_NONNULL(1);
appropriately.*/
/*@{*/
/**Gain offset type that indicates that the provided offset is relative to the
header gain.
This is the default.*/
#define OP_HEADER_GAIN (0)
/**Gain offset type that indicates that the provided offset is relative to the
R128_TRACK_GAIN value (if any), in addition to the header gain.*/
#define OP_TRACK_GAIN (3008)
/**Gain offset type that indicates that the provided offset should be used as
the gain directly, without applying any the header or track gains.*/
#define OP_ABSOLUTE_GAIN (3009)
/**Sets the gain to be used for decoded output.
By default, the gain in the header is applied with no additional offset.
The total gain (including header gain and/or track gain, if applicable, and
this offset), will be clamped to [-32768,32767]/256 dB.
This is more than enough to saturate or underflow 16-bit PCM.
\note The new gain will not be applied to any already buffered, decoded
output.
This means you cannot change it sample-by-sample, as at best it will be
updated packet-by-packet.
It is meant for setting a target volume level, rather than applying smooth
fades, etc.
\param _of The \c OggOpusFile on which to set the gain offset.
\param _gain_type One of #OP_HEADER_GAIN, #OP_TRACK_GAIN, or
#OP_ABSOLUTE_GAIN.
\param _gain_offset_q8 The gain offset to apply, in 1/256ths of a dB.
\return 0 on success or a negative value on error.
\retval #OP_EINVAL The \a _gain_type was unrecognized.*/
int op_set_gain_offset(OggOpusFile *_of,
int _gain_type,opus_int32 _gain_offset_q8);
/**Reads more samples from the stream.
\note Although \a _buf_size must indicate the total number of values that
can be stored in \a _pcm, the return value is the number of samples
<em>per channel</em>.
This is done because
<ol>
<li>The channel count cannot be known a prior (reading more samples might
<li>The channel count cannot be known a priori (reading more samples might
advance us into the next link, with a different channel count), so
\a _buf_size cannot also be in units of samples per channel,</li>
<li>Returning the samples per channel matches the <code>libopus</code> API
@ -1346,8 +1454,8 @@ int op_pcm_seek(OggOpusFile *_of,ogg_int64_t _pcm_offset) OP_ARG_NONNULL(1);
</ol>
\param _of The \c OggOpusFile from which to read.
\param[out] _pcm A buffer in which to store the output PCM samples, as
signed native-endian 16-bit values with a nominal
range of <code>[-32768,32767)</code>.
signed native-endian 16-bit values at 48&nbsp;kHz
with a nominal range of <code>[-32768,32767)</code>.
Multiple channels are interleaved using the
<a href="http://www.xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-800004.3.9">Vorbis
channel ordering</a>.
@ -1411,7 +1519,7 @@ OP_WARN_UNUSED_RESULT int op_read(OggOpusFile *_of,
can be stored in \a _pcm, the return value is the number of samples
<em>per channel</em>.
<ol>
<li>The channel count cannot be known a prior (reading more samples might
<li>The channel count cannot be known a priori (reading more samples might
advance us into the next link, with a different channel count), so
\a _buf_size cannot also be in units of samples per channel,</li>
<li>Returning the samples per channel matches the <code>libopus</code> API
@ -1428,7 +1536,7 @@ OP_WARN_UNUSED_RESULT int op_read(OggOpusFile *_of,
</ol>
\param _of The \c OggOpusFile from which to read.
\param[out] _pcm A buffer in which to store the output PCM samples as
signed floats with a nominal range of
signed floats at 48&nbsp;kHz with a nominal range of
<code>[-1.0,1.0]</code>.
Multiple channels are interleaved using the
<a href="http://www.xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-800004.3.9">Vorbis
@ -1497,8 +1605,8 @@ OP_WARN_UNUSED_RESULT int op_read_float(OggOpusFile *_of,
op_read().
\param _of The \c OggOpusFile from which to read.
\param[out] _pcm A buffer in which to store the output PCM samples, as
signed native-endian 16-bit values with a nominal
range of <code>[-32768,32767)</code>.
signed native-endian 16-bit values at 48&nbsp;kHz
with a nominal range of <code>[-32768,32767)</code>.
The left and right channels are interleaved in the
buffer.
This must have room for at least \a _buf_size values.
@ -1558,7 +1666,7 @@ OP_WARN_UNUSED_RESULT int op_read_stereo(OggOpusFile *_of,
op_read_float().
\param _of The \c OggOpusFile from which to read.
\param[out] _pcm A buffer in which to store the output PCM samples, as
signed floats with a nominal range of
signed floats at 48&nbsp;kHz with a nominal range of
<code>[-1.0,1.0]</code>.
The left and right channels are interleaved in the
buffer.

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