Merge remote branch 'qatar/master'

[ffmpeg] / libavcodec / audioconvert.h

diff --git a/libavcodec/audioconvert.h b/libavcodec/audioconvert.h
index c391759ebe718d3f15c738f5cce893dd4ecc6751..5f38cca5134b663ddb621cc49b1e8b1cf64756c3 100644 (file)
--- a/libavcodec/audioconvert.h
+++ b/libavcodec/audioconvert.h
@@ -24,35 +24,63 @@
 #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
  */
-enum SampleFormat avcodec_get_sample_fmt(const char* name);
+int64_t avcodec_guess_channel_layout(int nb_channels, enum CodecID codec_id, const char *fmt_name);
 
 struct AVAudioConvert;
 typedef struct AVAudioConvert AVAudioConvert;
@@ -64,11 +92,11 @@ typedef struct AVAudioConvert AVAudioConvert;
  * @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
+ * @param flags See AV_CPU_FLAG_xx
  * @return NULL on error
  */
-AVAudioConvert *av_audio_convert_alloc(enum SampleFormat out_fmt, int out_channels,
-                                       enum SampleFormat in_fmt, int in_channels,
+AVAudioConvert *av_audio_convert_alloc(enum AVSampleFormat out_fmt, int out_channels,
+                                       enum AVSampleFormat in_fmt, int in_channels,
                                        const float *matrix, int flags);
 
 /**
@@ -79,9 +107,9 @@ 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] 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 output samples (measured in bytes)
+ * @param[in] in_stride distance between consecutive input samples (measured in bytes)
  * @param len length of audio frame size (measured in samples)
  */
 int av_audio_convert(AVAudioConvert *ctx,