* Copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at>
* Copyright (c) 2008 Peter Ross
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * 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.
*
- * FFmpeg is distributed in the hope that it will be useful,
+ * 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 FFmpeg; if not, write to the Free Software
+ * License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
-#ifndef FFMPEG_AUDIOCONVERT_H
-#define FFMPEG_AUDIOCONVERT_H
+#ifndef AVCODEC_AUDIOCONVERT_H
+#define AVCODEC_AUDIOCONVERT_H
/**
- * @file audioconvert.h
+ * @file
* Audio format conversion routines
*/
+#include "libavutil/cpu.h"
#include "avcodec.h"
+#include "libavutil/audioconvert.h"
-
+#if FF_API_OLD_SAMPLE_FMT
/**
- * Generate string corresponding to the sample format with
- * number sample_fmt, or a header if sample_fmt is negative.
- *
- * @param[in] buf the buffer where to write the string
- * @param[in] buf_size the size of buf
- * @param[in] sample_fmt the number of the sample format to print the corresponding info string, or
- * a negative value to print the corresponding header.
- * Meaningful values for obtaining a sample format info vary from 0 to SAMPLE_FMT_NB -1.
+ * @deprecated Use av_get_sample_fmt_string() instead.
*/
+attribute_deprecated
void avcodec_sample_fmt_string(char *buf, int buf_size, int sample_fmt);
/**
- * @return NULL on error
+ * @deprecated Use av_get_sample_fmt_name() instead.
*/
+attribute_deprecated
const char *avcodec_get_sample_fmt_name(int sample_fmt);
/**
- * @return SAMPLE_FMT_NONE on error
+ * @deprecated Use av_get_sample_fmt() instead.
+ */
+attribute_deprecated
+enum AVSampleFormat avcodec_get_sample_fmt(const char* name);
+#endif
+
+#if FF_API_OLD_AUDIOCONVERT
+/**
+ * @deprecated Use av_get_channel_layout() instead.
+ */
+attribute_deprecated
+int64_t avcodec_get_channel_layout(const char *name);
+
+/**
+ * @deprecated Use av_get_channel_layout_string() instead.
+ */
+attribute_deprecated
+void avcodec_get_channel_layout_string(char *buf, int buf_size, int nb_channels, int64_t channel_layout);
+
+/**
+ * @deprecated Use av_get_channel_layout_nb_channels() instead.
+ */
+attribute_deprecated
+int avcodec_channel_layout_num_channels(int64_t channel_layout);
+#endif
+
+/**
+ * Guess the channel layout
+ * @param nb_channels
+ * @param codec_id Codec identifier, or CODEC_ID_NONE if unknown
+ * @param fmt_name Format name, or NULL if unknown
+ * @return Channel layout mask
+ */
+uint64_t avcodec_guess_channel_layout(int nb_channels, enum CodecID codec_id, const char *fmt_name);
+
+struct AVAudioConvert;
+typedef struct AVAudioConvert AVAudioConvert;
+
+/**
+ * Create an audio sample format converter context
+ * @param out_fmt Output sample format
+ * @param out_channels Number of output channels
+ * @param in_fmt Input sample format
+ * @param in_channels Number of input channels
+ * @param[in] matrix Channel mixing matrix (of dimension in_channel*out_channels). Set to NULL to ignore.
+ * @param flags See AV_CPU_FLAG_xx
+ * @return NULL on error
+ */
+AVAudioConvert *av_audio_convert_alloc(enum AVSampleFormat out_fmt, int out_channels,
+ enum AVSampleFormat in_fmt, int in_channels,
+ const float *matrix, int flags);
+
+/**
+ * Free audio sample format converter context
+ */
+void av_audio_convert_free(AVAudioConvert *ctx);
+
+/**
+ * Convert between audio sample formats
+ * @param[in] out array of output buffers for each channel. set to NULL to ignore processing of the given channel.
+ * @param[in] out_stride distance between consecutive output samples (measured in bytes)
+ * @param[in] in array of input buffers for each channel
+ * @param[in] in_stride distance between consecutive input samples (measured in bytes)
+ * @param len length of audio frame size (measured in samples)
*/
-enum SampleFormat avcodec_get_sample_fmt(const char* name);
+int av_audio_convert(AVAudioConvert *ctx,
+ void * const out[6], const int out_stride[6],
+ const void * const in[6], const int in_stride[6], int len);
-#endif /* FFMPEG_AUDIOCONVERT_H */
+#endif /* AVCODEC_AUDIOCONVERT_H */
projects / ffmpeg / blobdiff