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Bug 2381 - Mac: Export to Opus (OggOpus) is not available on Mac - Opus import fails on Mac

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This commit is contained in:
Leland Lucius
2020-08-18 22:39:19 -05:00
parent 25ddfc7bb4
commit 743585fb4b
70 changed files with 5067 additions and 10533 deletions
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58
lib-src/ffmpeg/libavutil/samplefmt.h
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@@ -25,26 +25,35 @@
#include "attributes.h"
/**
* Audio Sample Formats
* @addtogroup lavu_audio
* @{
*
* @defgroup lavu_sampfmts Audio sample formats
*
* Audio sample format enumeration and related convenience functions.
* @{
*/
/**
* Audio sample formats
*
* - The data described by the sample format is always in native-endian order.
* Sample values can be expressed by native C types, hence the lack of a signed
* 24-bit sample format even though it is a common raw audio data format.
*
* - The floating-point formats are based on full volume being in the range
* [-1.0, 1.0]. Any values outside this range are beyond full volume level.
*
* - The data layout as used in av_samples_fill_arrays() and elsewhere in FFmpeg
* (such as AVFrame in libavcodec) is as follows:
*
* @par
* The data described by the sample format is always in native-endian order.
* Sample values can be expressed by native C types, hence the lack of a signed
* 24-bit sample format even though it is a common raw audio data format.
*
* @par
* The floating-point formats are based on full volume being in the range
* [-1.0, 1.0]. Any values outside this range are beyond full volume level.
*
* @par
* The data layout as used in av_samples_fill_arrays() and elsewhere in FFmpeg
* (such as AVFrame in libavcodec) is as follows:
*
* For planar sample formats, each audio channel is in a separate data plane,
* and linesize is the buffer size, in bytes, for a single plane. All data
* planes must be the same size. For packed sample formats, only the first data
* plane is used, and samples for each channel are interleaved. In this case,
* linesize is the buffer size, in bytes, for the 1 plane.
*
*/
enum AVSampleFormat {
AV_SAMPLE_FMT_NONE = -1,
@@ -59,6 +68,8 @@ enum AVSampleFormat {
AV_SAMPLE_FMT_S32P, ///< signed 32 bits, planar
AV_SAMPLE_FMT_FLTP, ///< float, planar
AV_SAMPLE_FMT_DBLP, ///< double, planar
AV_SAMPLE_FMT_S64, ///< signed 64 bits
AV_SAMPLE_FMT_S64P, ///< signed 64 bits, planar
AV_SAMPLE_FMT_NB ///< Number of sample formats. DO NOT USE if linking dynamically
};
@@ -119,14 +130,6 @@ enum AVSampleFormat av_get_planar_sample_fmt(enum AVSampleFormat sample_fmt);
*/
char *av_get_sample_fmt_string(char *buf, int buf_size, enum AVSampleFormat sample_fmt);
#if FF_API_GET_BITS_PER_SAMPLE_FMT
/**
* @deprecated Use av_get_bytes_per_sample() instead.
*/
attribute_deprecated
int av_get_bits_per_sample_fmt(enum AVSampleFormat sample_fmt);
#endif
/**
* Return number of bytes per sample.
*
@@ -157,6 +160,15 @@ int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt);
int av_samples_get_buffer_size(int *linesize, int nb_channels, int nb_samples,
enum AVSampleFormat sample_fmt, int align);
/**
* @}
*
* @defgroup lavu_sampmanip Samples manipulation
*
* Functions that manipulate audio samples
* @{
*/
/**
* Fill plane data pointers and linesize for samples with sample
* format sample_fmt.
@@ -253,4 +265,8 @@ int av_samples_copy(uint8_t **dst, uint8_t * const *src, int dst_offset,
int av_samples_set_silence(uint8_t **audio_data, int offset, int nb_samples,
int nb_channels, enum AVSampleFormat sample_fmt);
/**
* @}
* @}
*/
#endif /* AVUTIL_SAMPLEFMT_H */