Fix libav/ffmpeg compilation across platforms: Move header back to 'stub_includes'

1 files changed, 220 insertions, 0 deletions

diff --git a/make/stub_includes/libav/lavc54.lavf54.lavu52.lavr01/libavutil/samplefmt.h b/make/stub_includes/libav/lavc54.lavf54.lavu52.lavr01/libavutil/samplefmt.h
new file mode 100644
index 000000000..33cbdedf5
--- /dev/null
+++ b/make/stub_includes/libav/lavc54.lavf54.lavu52.lavr01/libavutil/samplefmt.h
@@ -0,0 +1,220 @@
+/*
+ * This file is part of Libav.
+ *
+ * Libav 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.
+ *
+ * Libav 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 Libav; 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>
+
+#include "avutil.h"
+#include "attributes.h"
+
+/**
+ * Audio Sample Formats
+ *
+ * @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 Libav
+ * (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,
+    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_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);
+
+/**
+ * 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);
+
+/**
+ * Fill channel data pointers and linesize for samples with sample
+ * format sample_fmt.
+ *
+ * The pointers array is filled with the pointers to the samples data:
+ * 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 linesize array is filled with the aligned size of each channel's data
+ * buffer for planar layout, or the aligned size of the buffer for all channels
+ * for packed layout.
+ *
+ * @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                 0 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 align            buffer size alignment (0 = default, 1 = no alignment)
+ * @return                 0 on success or a negative error code on failure
+ * @see av_samples_fill_arrays()
+ */
+int av_samples_alloc(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 **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 **audio_data, int offset, int nb_samples,
+                           int nb_channels, enum AVSampleFormat sample_fmt);
+
+#endif /* AVUTIL_SAMPLEFMT_H */