avcodec: Drop deprecated audio resample API

Deprecated in 03/2013.

libavcodec/Makefile

@@ -45,8 +45,6 @@ OBJS = allcodecs.o                                                      \
        profiles.o                                                       \
        qsv_api.o                                                        \
        raw.o                                                            \
-       resample.o                                                       \
-       resample2.o                                                      \
        utils.o                                                          \
        vorbis_parser.o                                                  \
        xiph.o                                                           \

libavcodec/avcodec.h

@@ -5516,103 +5516,6 @@ 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
-
 #if FF_API_AVPICTURE
 /**
  * @addtogroup lavc_picture

libavcodec/version.h

@@ -57,9 +57,6 @@
 #ifndef FF_API_AUDIO_CONVERT
 #define FF_API_AUDIO_CONVERT     (LIBAVCODEC_VERSION_MAJOR < 58)
 #endif
-#ifndef FF_API_AVCODEC_RESAMPLE
-#define FF_API_AVCODEC_RESAMPLE  FF_API_AUDIO_CONVERT
-#endif
 #ifndef FF_API_LOWRES
 #define FF_API_LOWRES            (LIBAVCODEC_VERSION_MAJOR < 59)
 #endif