* commit 'ab35ec29a4071871934856c00da7d6ebcc0c095b': vf_overlay: get rid of pointless messing with timebase. samplefmt: make av_samples_alloc() initialize the data to silence. libspeexdec: handle NULL return value from speex_packet_to_header() h264probe: Don't error out on bits that no longer are reserved mpegvideo: set extended_data in ff_update_duplicate_context() libspeexdec: properly handle DTX for multiple frames-per-packet libspeexdec: move the SpeexHeader from LibSpeexContext to where it is used libspeexdec: simplify setting of frame_size libspeexdec: set channel_layout Conflicts: libavfilter/vf_overlay.c libavformat/h264dec.c libavutil/version.h Merged-by: Michael Niedermayer <michaelni@gmx.at>
		
			
				
	
	
		
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			241 lines
		
	
	
		
			9.2 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
/*
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 * This file is part of FFmpeg.
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 *
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 * FFmpeg is free software; you can redistribute it and/or
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 * modify it under the terms of the GNU Lesser General Public
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 * License as published by the Free Software Foundation; either
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 * version 2.1 of the License, or (at your option) any later version.
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 *
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 * FFmpeg is distributed in the hope that it will be useful,
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 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
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 * Lesser General Public License for more details.
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 *
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 * You should have received a copy of the GNU Lesser General Public
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 * License along with FFmpeg; if not, write to the Free Software
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 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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 */
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#ifndef AVUTIL_SAMPLEFMT_H
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#define AVUTIL_SAMPLEFMT_H
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#include <stdint.h>
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#include "avutil.h"
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#include "attributes.h"
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/**
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 * Audio Sample Formats
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 *
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 * @par
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 * The data described by the sample format is always in native-endian order.
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 * Sample values can be expressed by native C types, hence the lack of a signed
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 * 24-bit sample format even though it is a common raw audio data format.
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 *
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 * @par
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 * The floating-point formats are based on full volume being in the range
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 * [-1.0, 1.0]. Any values outside this range are beyond full volume level.
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 *
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 * @par
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 * The data layout as used in av_samples_fill_arrays() and elsewhere in Libav
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 * (such as AVFrame in libavcodec) is as follows:
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 *
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 * For planar sample formats, each audio channel is in a separate data plane,
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 * and linesize is the buffer size, in bytes, for a single plane. All data
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 * planes must be the same size. For packed sample formats, only the first data
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 * plane is used, and samples for each channel are interleaved. In this case,
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 * linesize is the buffer size, in bytes, for the 1 plane.
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 */
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enum AVSampleFormat {
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    AV_SAMPLE_FMT_NONE = -1,
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    AV_SAMPLE_FMT_U8,          ///< unsigned 8 bits
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    AV_SAMPLE_FMT_S16,         ///< signed 16 bits
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    AV_SAMPLE_FMT_S32,         ///< signed 32 bits
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    AV_SAMPLE_FMT_FLT,         ///< float
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    AV_SAMPLE_FMT_DBL,         ///< double
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    AV_SAMPLE_FMT_U8P,         ///< unsigned 8 bits, planar
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    AV_SAMPLE_FMT_S16P,        ///< signed 16 bits, planar
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    AV_SAMPLE_FMT_S32P,        ///< signed 32 bits, planar
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    AV_SAMPLE_FMT_FLTP,        ///< float, planar
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    AV_SAMPLE_FMT_DBLP,        ///< double, planar
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    AV_SAMPLE_FMT_NB           ///< Number of sample formats. DO NOT USE if linking dynamically
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};
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/**
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 * Return the name of sample_fmt, or NULL if sample_fmt is not
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 * recognized.
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 */
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const char *av_get_sample_fmt_name(enum AVSampleFormat sample_fmt);
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/**
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 * Return a sample format corresponding to name, or AV_SAMPLE_FMT_NONE
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 * on error.
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 */
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enum AVSampleFormat av_get_sample_fmt(const char *name);
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/**
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 * Return the planar<->packed alternative form of the given sample format, or
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 * AV_SAMPLE_FMT_NONE on error. If the passed sample_fmt is already in the
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 * requested planar/packed format, the format returned is the same as the
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 * input.
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 */
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enum AVSampleFormat av_get_alt_sample_fmt(enum AVSampleFormat sample_fmt, int planar);
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/**
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 * Get the packed alternative form of the given sample format.
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 *
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 * If the passed sample_fmt is already in packed format, the format returned is
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 * the same as the input.
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 *
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 * @return  the packed alternative form of the given sample format or
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            AV_SAMPLE_FMT_NONE on error.
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 */
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enum AVSampleFormat av_get_packed_sample_fmt(enum AVSampleFormat sample_fmt);
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/**
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 * Get the planar alternative form of the given sample format.
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 *
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 * If the passed sample_fmt is already in planar format, the format returned is
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 * the same as the input.
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 *
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 * @return  the planar alternative form of the given sample format or
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            AV_SAMPLE_FMT_NONE on error.
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 */
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enum AVSampleFormat av_get_planar_sample_fmt(enum AVSampleFormat sample_fmt);
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/**
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 * Generate a string corresponding to the sample format with
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 * sample_fmt, or a header if sample_fmt is negative.
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 *
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 * @param buf the buffer where to write the string
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 * @param buf_size the size of buf
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 * @param sample_fmt the number of the sample format to print the
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 * corresponding info string, or a negative value to print the
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 * corresponding header.
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 * @return the pointer to the filled buffer or NULL if sample_fmt is
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 * unknown or in case of other errors
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 */
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char *av_get_sample_fmt_string(char *buf, int buf_size, enum AVSampleFormat sample_fmt);
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#if FF_API_GET_BITS_PER_SAMPLE_FMT
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/**
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 * @deprecated Use av_get_bytes_per_sample() instead.
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 */
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attribute_deprecated
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int av_get_bits_per_sample_fmt(enum AVSampleFormat sample_fmt);
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#endif
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/**
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 * Return number of bytes per sample.
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 *
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 * @param sample_fmt the sample format
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 * @return number of bytes per sample or zero if unknown for the given
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 * sample format
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 */
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int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt);
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/**
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 * Check if the sample format is planar.
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 *
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 * @param sample_fmt the sample format to inspect
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 * @return 1 if the sample format is planar, 0 if it is interleaved
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 */
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int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt);
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/**
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 * Get the required buffer size for the given audio parameters.
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 *
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 * @param[out] linesize calculated linesize, may be NULL
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 * @param nb_channels   the number of channels
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 * @param nb_samples    the number of samples in a single channel
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 * @param sample_fmt    the sample format
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 * @param align         buffer size alignment (0 = default, 1 = no alignment)
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 * @return              required buffer size, or negative error code on failure
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 */
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int av_samples_get_buffer_size(int *linesize, int nb_channels, int nb_samples,
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                               enum AVSampleFormat sample_fmt, int align);
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/**
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 * Fill plane data pointers and linesize for samples with sample
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 * format sample_fmt.
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 *
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 * The audio_data array is filled with the pointers to the samples data planes:
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 * for planar, set the start point of each channel's data within the buffer,
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 * for packed, set the start point of the entire buffer only.
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 *
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 * The value pointed to by linesize is set to the aligned size of each
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 * channel's data buffer for planar layout, or to the aligned size of the
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 * buffer for all channels for packed layout.
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 *
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 * The buffer in buf must be big enough to contain all the samples
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 * (use av_samples_get_buffer_size() to compute its minimum size),
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 * otherwise the audio_data pointers will point to invalid data.
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 *
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 * @see enum AVSampleFormat
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 * The documentation for AVSampleFormat describes the data layout.
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 *
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 * @param[out] audio_data  array to be filled with the pointer for each channel
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 * @param[out] linesize    calculated linesize, may be NULL
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 * @param buf              the pointer to a buffer containing the samples
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 * @param nb_channels      the number of channels
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 * @param nb_samples       the number of samples in a single channel
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 * @param sample_fmt       the sample format
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 * @param align            buffer size alignment (0 = default, 1 = no alignment)
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 * @return                 0 on success or a negative error code on failure
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 */
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int av_samples_fill_arrays(uint8_t **audio_data, int *linesize,
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                           const uint8_t *buf,
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                           int nb_channels, int nb_samples,
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                           enum AVSampleFormat sample_fmt, int align);
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/**
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 * Allocate a samples buffer for nb_samples samples, and fill data pointers and
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 * linesize accordingly.
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 * The allocated samples buffer can be freed by using av_freep(&audio_data[0])
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 * Allocated data will be initialized to silence.
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 *
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 * @see enum AVSampleFormat
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 * The documentation for AVSampleFormat describes the data layout.
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 *
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 * @param[out] audio_data  array to be filled with the pointer for each channel
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 * @param[out] linesize    aligned size for audio buffer(s), may be NULL
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 * @param nb_channels      number of audio channels
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 * @param nb_samples       number of samples per channel
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 * @param align            buffer size alignment (0 = default, 1 = no alignment)
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 * @return                 0 on success or a negative error code on failure
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 * @see av_samples_fill_arrays()
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 */
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int av_samples_alloc(uint8_t **audio_data, int *linesize, int nb_channels,
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                     int nb_samples, enum AVSampleFormat sample_fmt, int align);
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/**
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 * Copy samples from src to dst.
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 *
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 * @param dst destination array of pointers to data planes
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 * @param src source array of pointers to data planes
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 * @param dst_offset offset in samples at which the data will be written to dst
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 * @param src_offset offset in samples at which the data will be read from src
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 * @param nb_samples number of samples to be copied
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 * @param nb_channels number of audio channels
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 * @param sample_fmt audio sample format
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 */
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int av_samples_copy(uint8_t **dst, uint8_t * const *src, int dst_offset,
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                    int src_offset, int nb_samples, int nb_channels,
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                    enum AVSampleFormat sample_fmt);
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/**
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 * Fill an audio buffer with silence.
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 *
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 * @param audio_data  array of pointers to data planes
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 * @param offset      offset in samples at which to start filling
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 * @param nb_samples  number of samples to fill
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 * @param nb_channels number of audio channels
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 * @param sample_fmt  audio sample format
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 */
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int av_samples_set_silence(uint8_t **audio_data, int offset, int nb_samples,
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                           int nb_channels, enum AVSampleFormat sample_fmt);
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#endif /* AVUTIL_SAMPLEFMT_H */
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