react-native-audio-api 0.8.0-nightly-5a12f24-20250901 → 0.8.0

package/common/cpp/audioapi/external/ffmpeg_include/libavutil/ripemd.h

@@ -0,0 +1,83 @@
+/*
+ * Copyright (C) 2007 Michael Niedermayer <michaelni@gmx.at>
+ * Copyright (C) 2013 James Almer <jamrial@gmail.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * @ingroup lavu_ripemd
+ * Public header for RIPEMD hash function implementation.
+ */
+
+#ifndef AVUTIL_RIPEMD_H
+#define AVUTIL_RIPEMD_H
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "attributes.h"
+
+/**
+ * @defgroup lavu_ripemd RIPEMD
+ * @ingroup lavu_hash
+ * RIPEMD hash function implementation.
+ *
+ * @{
+ */
+
+extern const int av_ripemd_size;
+
+struct AVRIPEMD;
+
+/**
+ * Allocate an AVRIPEMD context.
+ */
+struct AVRIPEMD *av_ripemd_alloc(void);
+
+/**
+ * Initialize RIPEMD hashing.
+ *
+ * @param context pointer to the function context (of size av_ripemd_size)
+ * @param bits    number of bits in digest (128, 160, 256 or 320 bits)
+ * @return        zero if initialization succeeded, -1 otherwise
+ */
+int av_ripemd_init(struct AVRIPEMD* context, int bits);
+
+/**
+ * Update hash value.
+ *
+ * @param context hash function context
+ * @param data    input data to update hash with
+ * @param len     input data length
+ */
+void av_ripemd_update(struct AVRIPEMD* context, const uint8_t* data, size_t len);
+
+/**
+ * Finish hashing and output digest value.
+ *
+ * @param context hash function context
+ * @param digest  buffer where output digest value is stored
+ */
+void av_ripemd_final(struct AVRIPEMD* context, uint8_t *digest);
+
+/**
+ * @}
+ */
+
+#endif /* AVUTIL_RIPEMD_H */

package/common/cpp/audioapi/external/ffmpeg_include/libavutil/samplefmt.h

@@ -0,0 +1,269 @@
+/*
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVUTIL_SAMPLEFMT_H
+#define AVUTIL_SAMPLEFMT_H
+
+#include <stdint.h>
+
+/**
+ * @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
+ * 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,
+    AV_SAMPLE_FMT_U8,          ///< unsigned 8 bits
+    AV_SAMPLE_FMT_S16,         ///< signed 16 bits
+    AV_SAMPLE_FMT_S32,         ///< signed 32 bits
+    AV_SAMPLE_FMT_FLT,         ///< float
+    AV_SAMPLE_FMT_DBL,         ///< double
+
+    AV_SAMPLE_FMT_U8P,         ///< unsigned 8 bits, planar
+    AV_SAMPLE_FMT_S16P,        ///< signed 16 bits, planar
+    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
+};
+
+/**
+ * Return the name of sample_fmt, or NULL if sample_fmt is not
+ * recognized.
+ */
+const char *av_get_sample_fmt_name(enum AVSampleFormat sample_fmt);
+
+/**
+ * Return a sample format corresponding to name, or AV_SAMPLE_FMT_NONE
+ * on error.
+ */
+enum AVSampleFormat av_get_sample_fmt(const char *name);
+
+/**
+ * Return the planar<->packed alternative form of the given sample format, or
+ * AV_SAMPLE_FMT_NONE on error. If the passed sample_fmt is already in the
+ * requested planar/packed format, the format returned is the same as the
+ * input.
+ */
+enum AVSampleFormat av_get_alt_sample_fmt(enum AVSampleFormat sample_fmt, int planar);
+
+/**
+ * Get the packed alternative form of the given sample format.
+ *
+ * If the passed sample_fmt is already in packed format, the format returned is
+ * the same as the input.
+ *
+ * @return  the packed alternative form of the given sample format or
+            AV_SAMPLE_FMT_NONE on error.
+ */
+enum AVSampleFormat av_get_packed_sample_fmt(enum AVSampleFormat sample_fmt);
+
+/**
+ * Get the planar alternative form of the given sample format.
+ *
+ * If the passed sample_fmt is already in planar format, the format returned is
+ * the same as the input.
+ *
+ * @return  the planar alternative form of the given sample format or
+            AV_SAMPLE_FMT_NONE on error.
+ */
+enum AVSampleFormat av_get_planar_sample_fmt(enum AVSampleFormat sample_fmt);
+
+/**
+ * Generate a string corresponding to the sample format with
+ * sample_fmt, or a header if sample_fmt is negative.
+ *
+ * @param buf the buffer where to write the string
+ * @param buf_size the size of buf
+ * @param sample_fmt the number of the sample format to print the
+ * corresponding info string, or a negative value to print the
+ * corresponding header.
+ * @return the pointer to the filled buffer or NULL if sample_fmt is
+ * unknown or in case of other errors
+ */
+char *av_get_sample_fmt_string(char *buf, int buf_size, enum AVSampleFormat sample_fmt);
+
+/**
+ * Return number of bytes per sample.
+ *
+ * @param sample_fmt the sample format
+ * @return number of bytes per sample or zero if unknown for the given
+ * sample format
+ */
+int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt);
+
+/**
+ * Check if the sample format is planar.
+ *
+ * @param sample_fmt the sample format to inspect
+ * @return 1 if the sample format is planar, 0 if it is interleaved
+ */
+int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt);
+
+/**
+ * Get the required buffer size for the given audio parameters.
+ *
+ * @param[out] linesize calculated linesize, may be NULL
+ * @param nb_channels   the number of channels
+ * @param nb_samples    the number of samples in a single channel
+ * @param sample_fmt    the sample format
+ * @param align         buffer size alignment (0 = default, 1 = no alignment)
+ * @return              required buffer size, or negative error code on failure
+ */
+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.
+ *
+ * The audio_data array is filled with the pointers to the samples data planes:
+ * for planar, set the start point of each channel's data within the buffer,
+ * for packed, set the start point of the entire buffer only.
+ *
+ * The value pointed to by linesize is set to the aligned size of each
+ * channel's data buffer for planar layout, or to the aligned size of the
+ * buffer for all channels for packed layout.
+ *
+ * The buffer in buf must be big enough to contain all the samples
+ * (use av_samples_get_buffer_size() to compute its minimum size),
+ * otherwise the audio_data pointers will point to invalid data.
+ *
+ * @see enum AVSampleFormat
+ * The documentation for AVSampleFormat describes the data layout.
+ *
+ * @param[out] audio_data  array to be filled with the pointer for each channel
+ * @param[out] linesize    calculated linesize, may be NULL
+ * @param buf              the pointer to a buffer containing the samples
+ * @param nb_channels      the number of channels
+ * @param nb_samples       the number of samples in a single channel
+ * @param sample_fmt       the sample format
+ * @param align            buffer size alignment (0 = default, 1 = no alignment)
+ * @return                 minimum size in bytes required for the buffer on success,
+ *                         or a negative error code on failure
+ */
+int av_samples_fill_arrays(uint8_t **audio_data, int *linesize,
+                           const uint8_t *buf,
+                           int nb_channels, int nb_samples,
+                           enum AVSampleFormat sample_fmt, int align);
+
+/**
+ * Allocate a samples buffer for nb_samples samples, and fill data pointers and
+ * linesize accordingly.
+ * The allocated samples buffer can be freed by using av_freep(&audio_data[0])
+ * Allocated data will be initialized to silence.
+ *
+ * @see enum AVSampleFormat
+ * The documentation for AVSampleFormat describes the data layout.
+ *
+ * @param[out] audio_data  array to be filled with the pointer for each channel
+ * @param[out] linesize    aligned size for audio buffer(s), may be NULL
+ * @param nb_channels      number of audio channels
+ * @param nb_samples       number of samples per channel
+ * @param sample_fmt       the sample format
+ * @param align            buffer size alignment (0 = default, 1 = no alignment)
+ * @return                 >=0 on success or a negative error code on failure
+ * @todo return the size of the allocated buffer in case of success at the next bump
+ * @see av_samples_fill_arrays()
+ * @see av_samples_alloc_array_and_samples()
+ */
+int av_samples_alloc(uint8_t **audio_data, int *linesize, int nb_channels,
+                     int nb_samples, enum AVSampleFormat sample_fmt, int align);
+
+/**
+ * Allocate a data pointers array, samples buffer for nb_samples
+ * samples, and fill data pointers and linesize accordingly.
+ *
+ * This is the same as av_samples_alloc(), but also allocates the data
+ * pointers array.
+ *
+ * @see av_samples_alloc()
+ */
+int av_samples_alloc_array_and_samples(uint8_t ***audio_data, int *linesize, int nb_channels,
+                                       int nb_samples, enum AVSampleFormat sample_fmt, int align);
+
+/**
+ * Copy samples from src to dst.
+ *
+ * @param dst destination array of pointers to data planes
+ * @param src source array of pointers to data planes
+ * @param dst_offset offset in samples at which the data will be written to dst
+ * @param src_offset offset in samples at which the data will be read from src
+ * @param nb_samples number of samples to be copied
+ * @param nb_channels number of audio channels
+ * @param sample_fmt audio sample format
+ */
+int av_samples_copy(uint8_t * const *dst, uint8_t * const *src, int dst_offset,
+                    int src_offset, int nb_samples, int nb_channels,
+                    enum AVSampleFormat sample_fmt);
+
+/**
+ * Fill an audio buffer with silence.
+ *
+ * @param audio_data  array of pointers to data planes
+ * @param offset      offset in samples at which to start filling
+ * @param nb_samples  number of samples to fill
+ * @param nb_channels number of audio channels
+ * @param sample_fmt  audio sample format
+ */
+int av_samples_set_silence(uint8_t * const *audio_data, int offset, int nb_samples,
+                           int nb_channels, enum AVSampleFormat sample_fmt);
+
+/**
+ * @}
+ * @}
+ */
+#endif /* AVUTIL_SAMPLEFMT_H */

package/common/cpp/audioapi/external/ffmpeg_include/libavutil/sha.h

@@ -0,0 +1,90 @@
+/*
+ * Copyright (C) 2007 Michael Niedermayer <michaelni@gmx.at>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * @ingroup lavu_sha
+ * Public header for SHA-1 & SHA-256 hash function implementations.
+ */
+
+#ifndef AVUTIL_SHA_H
+#define AVUTIL_SHA_H
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "attributes.h"
+
+/**
+ * @defgroup lavu_sha SHA
+ * @ingroup lavu_hash
+ * SHA-1 and SHA-256 (Secure Hash Algorithm) hash function implementations.
+ *
+ * This module supports the following SHA hash functions:
+ *
+ * - SHA-1: 160 bits
+ * - SHA-224: 224 bits, as a variant of SHA-2
+ * - SHA-256: 256 bits, as a variant of SHA-2
+ *
+ * @see For SHA-384, SHA-512, and variants thereof, see @ref lavu_sha512.
+ *
+ * @{
+ */
+
+extern const int av_sha_size;
+
+struct AVSHA;
+
+/**
+ * Allocate an AVSHA context.
+ */
+struct AVSHA *av_sha_alloc(void);
+
+/**
+ * Initialize SHA-1 or SHA-2 hashing.
+ *
+ * @param context pointer to the function context (of size av_sha_size)
+ * @param bits    number of bits in digest (SHA-1 - 160 bits, SHA-2 224 or 256 bits)
+ * @return        zero if initialization succeeded, -1 otherwise
+ */
+int av_sha_init(struct AVSHA* context, int bits);
+
+/**
+ * Update hash value.
+ *
+ * @param ctx     hash function context
+ * @param data    input data to update hash with
+ * @param len     input data length
+ */
+void av_sha_update(struct AVSHA *ctx, const uint8_t *data, size_t len);
+
+/**
+ * Finish hashing and output digest value.
+ *
+ * @param context hash function context
+ * @param digest  buffer where output digest value is stored
+ */
+void av_sha_final(struct AVSHA* context, uint8_t *digest);
+
+/**
+ * @}
+ */
+
+#endif /* AVUTIL_SHA_H */

package/common/cpp/audioapi/external/ffmpeg_include/libavutil/sha512.h

@@ -0,0 +1,92 @@
+/*
+ * Copyright (C) 2007 Michael Niedermayer <michaelni@gmx.at>
+ * Copyright (C) 2013 James Almer <jamrial@gmail.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * @ingroup lavu_sha512
+ * Public header for SHA-512 implementation.
+ */
+
+#ifndef AVUTIL_SHA512_H
+#define AVUTIL_SHA512_H
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "attributes.h"
+
+/**
+ * @defgroup lavu_sha512 SHA-512
+ * @ingroup lavu_hash
+ * SHA-512 (Secure Hash Algorithm) hash function implementations.
+ *
+ * This module supports the following SHA-2 hash functions:
+ *
+ * - SHA-512/224: 224 bits
+ * - SHA-512/256: 256 bits
+ * - SHA-384: 384 bits
+ * - SHA-512: 512 bits
+ *
+ * @see For SHA-1, SHA-256, and variants thereof, see @ref lavu_sha.
+ *
+ * @{
+ */
+
+extern const int av_sha512_size;
+
+struct AVSHA512;
+
+/**
+ * Allocate an AVSHA512 context.
+ */
+struct AVSHA512 *av_sha512_alloc(void);
+
+/**
+ * Initialize SHA-2 512 hashing.
+ *
+ * @param context pointer to the function context (of size av_sha512_size)
+ * @param bits    number of bits in digest (224, 256, 384 or 512 bits)
+ * @return        zero if initialization succeeded, -1 otherwise
+ */
+int av_sha512_init(struct AVSHA512* context, int bits);
+
+/**
+ * Update hash value.
+ *
+ * @param context hash function context
+ * @param data    input data to update hash with
+ * @param len     input data length
+ */
+void av_sha512_update(struct AVSHA512* context, const uint8_t* data, size_t len);
+
+/**
+ * Finish hashing and output digest value.
+ *
+ * @param context hash function context
+ * @param digest  buffer where output digest value is stored
+ */
+void av_sha512_final(struct AVSHA512* context, uint8_t *digest);
+
+/**
+ * @}
+ */
+
+#endif /* AVUTIL_SHA512_H */