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-rw-r--r--libavcodec/audioconvert.h34
1 files changed, 34 insertions, 0 deletions
diff --git a/libavcodec/audioconvert.h b/libavcodec/audioconvert.h
index 210cc87716..e10849bd3a 100644
--- a/libavcodec/audioconvert.h
+++ b/libavcodec/audioconvert.h
@@ -54,4 +54,38 @@ const char *avcodec_get_sample_fmt_name(int sample_fmt);
*/
enum SampleFormat avcodec_get_sample_fmt(const char* 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 FF_MM_xx
+ * @return NULL on error
+ */
+AVAudioConvert *av_audio_convert_alloc(enum SampleFormat out_fmt, int out_channels,
+ enum SampleFormat 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 input samples (measured in bytes)
+ * @param[in] in array of input buffers for each channel
+ * @param[in] in_stride distance between consecutive output samples (measured in bytes)
+ * @param len length of audio frame size (measured in samples)
+ */
+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 */