oslibs: macosx, windows: recompile libopusfile, using the latest
from xiph.org git repo. git-svn-id: svn+ssh://svn.code.sf.net/p/quakespasm/code/trunk@855 af15c1b1-3010-417e-b628-4374ebc0bcbd
This commit is contained in:
sezero
parent
683c9ec2f8
commit
82434dfbe6
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@ -49,8 +49,55 @@
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Several additional sections are not tied to the main API.
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- \ref stream_callbacks
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- \ref header_info
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- \ref error_codes*/
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- \ref error_codes
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\section Overview
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The <tt>libopusfile</tt> API always decodes files to 48 kHz.
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The original sample rate is not preserved by the lossy compression, though
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it is stored in the header to allow you to resample to it after decoding
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(the <tt>libopusfile</tt> API does not currently provide a resampler,
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but the
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<a href="http://www.speex.org/docs/manual/speex-manual/node7.html#SECTION00760000000000000000">the
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Speex resampler</a> is a good choice if you need one).
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In general, if you are playing back the audio, you should leave it at
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48 kHz, provided your audio hardware supports it.
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When decoding to a file, it may be worth resampling back to the original
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sample rate, so as not to surprise users who might not expect the sample
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rate to change after encoding to Opus and decoding.
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Opus files can contain anywhere from 1 to 255 channels of audio.
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The channel mappings for up to 8 channels are the same as the
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<a href="http://www.xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-800004.3.9">Vorbis
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mappings</a>.
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A special stereo API can convert everything to 2 channels, making it simple
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to support multichannel files in a application which only has stereo
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output.
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Although the <tt>libopusfile</tt> ABI provides support for the theoretical
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maximum number of channels, the current implementation does not support
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files with more than 8 channels, as they do not have well-defined channel
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mappings.
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Like all Ogg files, Opus files may be "chained".
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That is, multiple Opus files may be combined into a single, longer file just
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by concatenating the original files.
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This is commonly done in internet radio streaming, as it allows the title
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and artist to be updated each time the song changes, since each link in the
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chain includes its own set of metadata.
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<tt>libopusfile</tt> fully supports chained files.
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It will decode the first Opus stream found in each link of a chained file
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(ignoring any other streams that might be concurrently multiplexed with it,
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such as a video stream).
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The channel count can also change between links, but if your application is
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not prepared to deal with this, it can use the stereo API to ensure the
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audio from all links will always get decoded into a common format.
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Since <tt>libopusfile</tt> always decodes to 48 kHz, you do not have to
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worry about the sample rate changing between links (as was possible with
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Vorbis).
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This makes application support for chained files with <tt>libopusfile</tt>
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very easy.*/
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# if defined(__cplusplus)
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extern "C" {
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@ -182,8 +229,9 @@ struct OpusHead{
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opus_uint32 input_sample_rate;
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/**The gain to apply to the decoded output, in dB, as a Q8 value in the range
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-32768...32767.
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The decoder will automatically scale the output by
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pow(10,output_gain/(20.0*256)).*/
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The <tt>libopusfile</tt> API will automatically apply this gain to the
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decoded output before returning it, scaling it by
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<code>pow(10,output_gain/(20.0*256))</code>.*/
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int output_gain;
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/**The channel mapping family, in the range 0...255.
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Channel mapping family 0 covers mono or stereo in a single stream.
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@ -385,6 +433,24 @@ const char *opus_tags_query(const OpusTags *_tags,const char *_tag,int _count)
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int opus_tags_query_count(const OpusTags *_tags,const char *_tag)
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OP_ARG_NONNULL(1) OP_ARG_NONNULL(2);
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/**Get the track gain from an R128_TRACK_GAIN tag, if one was specified.
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This searches for the first R128_TRACK_GAIN tag with a valid signed,
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16-bit decimal integer value and returns the value.
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This routine is exposed merely for convenience for applications which wish
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to do something special with the track gain (i.e., display it).
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If you simply wish to apply the track gain instead of the header gain, you
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can use op_set_gain_offset() with an #OP_TRACK_GAIN type and no offset.
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\param _tags An initialized #OpusTags structure.
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\param[out] _gain_q8 The track gain, in 1/256ths of a dB.
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This will lie in the range [-32768,32767], and should
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be applied in <em>addition</em> to the header gain.
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On error, no value is returned, and the previous
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contents remain unchanged.
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\return 0 on success, or a negative value on error.
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\retval OP_EFALSE There was no track gain available in the given tags.*/
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int opus_tags_get_track_gain(const OpusTags *_tags,int *_gain_q8)
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OP_ARG_NONNULL(1) OP_ARG_NONNULL(2);
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/**Clears the #OpusTags structure.
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This should be called on an #OpusTags structure after it is no longer
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needed.
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@ -841,7 +907,11 @@ OP_WARN_UNUSED_RESULT OggOpusFile *op_open_url(const char *_url,
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<dd>The first or last timestamp in a link failed
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basic validity checks.</dd>
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</dl>
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\return A freshly opened \c OggOpusFile, or <code>NULL</code> on error.*/
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\return A freshly opened \c OggOpusFile, or <code>NULL</code> on error.
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<tt>libopusfile</tt> does <em>not</em> take ownership of the source
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if the call fails.
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The calling application is responsible for closing the source if
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this call returns an error.*/
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OP_WARN_UNUSED_RESULT OggOpusFile *op_open_callbacks(void *_source,
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const OpusFileCallbacks *_cb,const unsigned char *_initial_data,
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size_t _initial_bytes,int *_error) OP_ARG_NONNULL(2);
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@ -974,7 +1044,11 @@ OP_WARN_UNUSED_RESULT OggOpusFile *op_test_url(const char *_url,
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the failure code.
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See op_open_callbacks() for a full list of failure
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codes.
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\return A partially opened \c OggOpusFile, or <code>NULL</code> on error.*/
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\return A partially opened \c OggOpusFile, or <code>NULL</code> on error.
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<tt>libopusfile</tt> does <em>not</em> take ownership of the source
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if the call fails.
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The calling application is responsible for closing the source if
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this call returns an error.*/
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OP_WARN_UNUSED_RESULT OggOpusFile *op_test_callbacks(void *_source,
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const OpusFileCallbacks *_cb,const unsigned char *_initial_data,
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size_t _initial_bytes,int *_error) OP_ARG_NONNULL(2);
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@ -1303,11 +1377,12 @@ int op_pcm_seek(OggOpusFile *_of,ogg_int64_t _pcm_offset) OP_ARG_NONNULL(1);
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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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<tt>libopusfile</tt> provides noise-shaping dithering API.
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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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<tt>libopusfile</tt> can also be configured at compile time to use the
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fixed-point <tt>libopus</tt> API.
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If so, the floating-point API may also be disabled.
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If so, <tt>libopusfile</tt>'s floating-point API may also be disabled.
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In that configuration, nothing in <tt>libopusfile</tt> will use any
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floating-point operations, to simplify support on devices without an
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adequate FPU.
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@ -1323,13 +1398,46 @@ int op_pcm_seek(OggOpusFile *_of,ogg_int64_t _pcm_offset) OP_ARG_NONNULL(1);
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appropriately.*/
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/*@{*/
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/**Gain offset type that indicates that the provided offset is relative to the
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header gain.
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This is the default.*/
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#define OP_HEADER_GAIN (0)
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/**Gain offset type that indicates that the provided offset is relative to the
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R128_TRACK_GAIN value (if any), in addition to the header gain.*/
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#define OP_TRACK_GAIN (3008)
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/**Gain offset type that indicates that the provided offset should be used as
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the gain directly, without applying any the header or track gains.*/
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#define OP_ABSOLUTE_GAIN (3009)
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/**Sets the gain to be used for decoded output.
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By default, the gain in the header is applied with no additional offset.
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The total gain (including header gain and/or track gain, if applicable, and
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this offset), will be clamped to [-32768,32767]/256 dB.
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This is more than enough to saturate or underflow 16-bit PCM.
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\note The new gain will not be applied to any already buffered, decoded
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output.
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This means you cannot change it sample-by-sample, as at best it will be
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updated packet-by-packet.
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It is meant for setting a target volume level, rather than applying smooth
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fades, etc.
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\param _of The \c OggOpusFile on which to set the gain offset.
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\param _gain_type One of #OP_HEADER_GAIN, #OP_TRACK_GAIN, or
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#OP_ABSOLUTE_GAIN.
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\param _gain_offset_q8 The gain offset to apply, in 1/256ths of a dB.
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\return 0 on success or a negative value on error.
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\retval #OP_EINVAL The \a _gain_type was unrecognized.*/
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int op_set_gain_offset(OggOpusFile *_of,
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int _gain_type,opus_int32 _gain_offset_q8);
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/**Reads more samples from the stream.
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\note Although \a _buf_size must indicate the total number of values that
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can be stored in \a _pcm, the return value is the number of samples
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<em>per channel</em>.
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This is done because
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<ol>
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<li>The channel count cannot be known a prior (reading more samples might
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<li>The channel count cannot be known a priori (reading more samples might
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advance us into the next link, with a different channel count), so
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\a _buf_size cannot also be in units of samples per channel,</li>
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<li>Returning the samples per channel matches the <code>libopus</code> API
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@ -1346,8 +1454,8 @@ int op_pcm_seek(OggOpusFile *_of,ogg_int64_t _pcm_offset) OP_ARG_NONNULL(1);
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</ol>
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\param _of The \c OggOpusFile from which to read.
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\param[out] _pcm A buffer in which to store the output PCM samples, as
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signed native-endian 16-bit values with a nominal
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range of <code>[-32768,32767)</code>.
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signed native-endian 16-bit values at 48 kHz
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with a nominal range of <code>[-32768,32767)</code>.
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Multiple channels are interleaved using the
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<a href="http://www.xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-800004.3.9">Vorbis
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channel ordering</a>.
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@ -1411,7 +1519,7 @@ OP_WARN_UNUSED_RESULT int op_read(OggOpusFile *_of,
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can be stored in \a _pcm, the return value is the number of samples
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<em>per channel</em>.
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<ol>
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<li>The channel count cannot be known a prior (reading more samples might
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<li>The channel count cannot be known a priori (reading more samples might
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advance us into the next link, with a different channel count), so
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\a _buf_size cannot also be in units of samples per channel,</li>
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<li>Returning the samples per channel matches the <code>libopus</code> API
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@ -1428,7 +1536,7 @@ OP_WARN_UNUSED_RESULT int op_read(OggOpusFile *_of,
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</ol>
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\param _of The \c OggOpusFile from which to read.
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\param[out] _pcm A buffer in which to store the output PCM samples as
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signed floats with a nominal range of
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signed floats at 48 kHz with a nominal range of
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<code>[-1.0,1.0]</code>.
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Multiple channels are interleaved using the
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<a href="http://www.xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-800004.3.9">Vorbis
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@ -1497,8 +1605,8 @@ OP_WARN_UNUSED_RESULT int op_read_float(OggOpusFile *_of,
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op_read().
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\param _of The \c OggOpusFile from which to read.
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\param[out] _pcm A buffer in which to store the output PCM samples, as
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signed native-endian 16-bit values with a nominal
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range of <code>[-32768,32767)</code>.
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signed native-endian 16-bit values at 48 kHz
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with a nominal range of <code>[-32768,32767)</code>.
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The left and right channels are interleaved in the
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buffer.
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This must have room for at least \a _buf_size values.
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@ -1558,7 +1666,7 @@ OP_WARN_UNUSED_RESULT int op_read_stereo(OggOpusFile *_of,
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op_read_float().
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\param _of The \c OggOpusFile from which to read.
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\param[out] _pcm A buffer in which to store the output PCM samples, as
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signed floats with a nominal range of
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signed floats at 48 kHz with a nominal range of
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<code>[-1.0,1.0]</code>.
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The left and right channels are interleaved in the
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buffer.
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|
|
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Binary file not shown.
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@ -49,8 +49,55 @@
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Several additional sections are not tied to the main API.
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- \ref stream_callbacks
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- \ref header_info
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- \ref error_codes*/
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- \ref error_codes
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\section Overview
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The <tt>libopusfile</tt> API always decodes files to 48 kHz.
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The original sample rate is not preserved by the lossy compression, though
|
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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
|
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Speex resampler</a> is a good choice if you need one).
|
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In general, if you are playing back the audio, you should leave it at
|
||||
48 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.
|
||||
|
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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
|
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<a href="http://www.xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-800004.3.9">Vorbis
|
||||
mappings</a>.
|
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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.
|
||||
|
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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.
|
||||
|
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<tt>libopusfile</tt> fully supports chained files.
|
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It will decode the first Opus stream found in each link of a chained file
|
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(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 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)
|
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extern "C" {
|
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|
|
@ -182,8 +229,9 @@ struct OpusHead{
|
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opus_uint32 input_sample_rate;
|
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/**The gain to apply to the decoded output, in dB, as a Q8 value in the range
|
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-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.
|
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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)
|
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int opus_tags_query_count(const OpusTags *_tags,const char *_tag)
|
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OP_ARG_NONNULL(1) OP_ARG_NONNULL(2);
|
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|
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/**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.
|
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\retval OP_EFALSE There was no track gain available in the given tags.*/
|
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int opus_tags_get_track_gain(const OpusTags *_tags,int *_gain_q8)
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OP_ARG_NONNULL(1) OP_ARG_NONNULL(2);
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|
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/**Clears the #OpusTags structure.
|
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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,
|
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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> 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 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 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 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 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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Reference in New Issue