Revert "Merge commit '0517c9e0'" bring the old...

Revert "Merge commit '0517c9e0'" bring the old audio resampling API back

This reverts commit d3edc65d, reversing
changes made to 150de78d.

Conflicts:

	libavcodec/version.h

It seems there are several applications still using it
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

libavcodec/Makefile

@@ -27,6 +27,8 @@ OBJS = allcodecs.o                                                      \
        parser.o                                                         \
        raw.o                                                            \
        rawdec.o                                                         \
+       resample.o                                                       \
+       resample2.o                                                      \
        utils.o                                                          \
 
 # parts needed for many different codecs

libavcodec/avcodec.h

@@ -4096,6 +4096,103 @@ int avcodec_encode_subtitle(AVCodecContext *avctx, uint8_t *buf, int buf_size,
  * @}
  */
 
+#if FF_API_AVCODEC_RESAMPLE
+/**
+ * @defgroup lavc_resample Audio resampling
+ * @ingroup libavc
+ * @deprecated use libswresample instead
+ *
+ * @{
+ */
+struct ReSampleContext;
+struct AVResampleContext;
+
+typedef struct ReSampleContext ReSampleContext;
+
+/**
+ *  Initialize audio resampling context.
+ *
+ * @param output_channels  number of output channels
+ * @param input_channels   number of input channels
+ * @param output_rate      output sample rate
+ * @param input_rate       input sample rate
+ * @param sample_fmt_out   requested output sample format
+ * @param sample_fmt_in    input sample format
+ * @param filter_length    length of each FIR filter in the filterbank relative to the cutoff frequency
+ * @param log2_phase_count log2 of the number of entries in the polyphase filterbank
+ * @param linear           if 1 then the used FIR filter will be linearly interpolated
+                           between the 2 closest, if 0 the closest will be used
+ * @param cutoff           cutoff frequency, 1.0 corresponds to half the output sampling rate
+ * @return allocated ReSampleContext, NULL if error occurred
+ */
+attribute_deprecated
+ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
+                                        int output_rate, int input_rate,
+                                        enum AVSampleFormat sample_fmt_out,
+                                        enum AVSampleFormat sample_fmt_in,
+                                        int filter_length, int log2_phase_count,
+                                        int linear, double cutoff);
+
+attribute_deprecated
+int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples);
+
+/**
+ * Free resample context.
+ *
+ * @param s a non-NULL pointer to a resample context previously
+ *          created with av_audio_resample_init()
+ */
+attribute_deprecated
+void audio_resample_close(ReSampleContext *s);
+
+
+/**
+ * Initialize an audio resampler.
+ * Note, if either rate is not an integer then simply scale both rates up so they are.
+ * @param filter_length length of each FIR filter in the filterbank relative to the cutoff freq
+ * @param log2_phase_count log2 of the number of entries in the polyphase filterbank
+ * @param linear If 1 then the used FIR filter will be linearly interpolated
+                 between the 2 closest, if 0 the closest will be used
+ * @param cutoff cutoff frequency, 1.0 corresponds to half the output sampling rate
+ */
+attribute_deprecated
+struct AVResampleContext *av_resample_init(int out_rate, int in_rate, int filter_length, int log2_phase_count, int linear, double cutoff);
+
+/**
+ * Resample an array of samples using a previously configured context.
+ * @param src an array of unconsumed samples
+ * @param consumed the number of samples of src which have been consumed are returned here
+ * @param src_size the number of unconsumed samples available
+ * @param dst_size the amount of space in samples available in dst
+ * @param update_ctx If this is 0 then the context will not be modified, that way several channels can be resampled with the same context.
+ * @return the number of samples written in dst or -1 if an error occurred
+ */
+attribute_deprecated
+int av_resample(struct AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx);
+
+
+/**
+ * Compensate samplerate/timestamp drift. The compensation is done by changing
+ * the resampler parameters, so no audible clicks or similar distortions occur
+ * @param compensation_distance distance in output samples over which the compensation should be performed
+ * @param sample_delta number of output samples which should be output less
+ *
+ * example: av_resample_compensate(c, 10, 500)
+ * here instead of 510 samples only 500 samples would be output
+ *
+ * note, due to rounding the actual compensation might be slightly different,
+ * especially if the compensation_distance is large and the in_rate used during init is small
+ */
+attribute_deprecated
+void av_resample_compensate(struct AVResampleContext *c, int sample_delta, int compensation_distance);
+attribute_deprecated
+void av_resample_close(struct AVResampleContext *c);
+
+/**
+ * @}
+ */
+#endif
+
 /**
  * @addtogroup lavc_picture
  * @{

libavcodec/version.h

@@ -73,6 +73,9 @@
 #ifndef FF_API_CODEC_ID
 #define FF_API_CODEC_ID          (LIBAVCODEC_VERSION_MAJOR < 56)
 #endif
+#ifndef FF_API_AVCODEC_RESAMPLE
+#define FF_API_AVCODEC_RESAMPLE  (LIBAVCODEC_VERSION_MAJOR < 56)
+#endif
 #ifndef FF_API_MMI
 #define FF_API_MMI               (LIBAVCODEC_VERSION_MAJOR < 55)
 #endif