comparison avcodec.h @ 8806:cbeaa8c0fe4f libavcodec

extend resampling API, add S16 internal conversion

8805:eda229beb608

#include <errno.h>
#include "libavutil/avutil.h"

#define LIBAVCODEC_VERSION_MAJOR 52
#define LIBAVCODEC_VERSION_MINOR 14
#define LIBAVCODEC_VERSION_MICRO  0

#define LIBAVCODEC_VERSION_INT  AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
                                               LIBAVCODEC_VERSION_MINOR, \
                                               LIBAVCODEC_VERSION_MICRO)

struct ReSampleContext;
struct AVResampleContext;

typedef struct ReSampleContext ReSampleContext;

ReSampleContext *audio_resample_init(int output_channels, int input_channels,
                                     int output_rate, int input_rate);
int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples);
void audio_resample_close(ReSampleContext *s);


/**

8806:cbeaa8c0fe4f

#include <errno.h>
#include "libavutil/avutil.h"

#define LIBAVCODEC_VERSION_MAJOR 52
#define LIBAVCODEC_VERSION_MINOR 15
#define LIBAVCODEC_VERSION_MICRO  0

#define LIBAVCODEC_VERSION_INT  AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
                                               LIBAVCODEC_VERSION_MINOR, \
                                               LIBAVCODEC_VERSION_MICRO)

struct ReSampleContext;
struct AVResampleContext;

typedef struct ReSampleContext ReSampleContext;

#if LIBAVCODEC_VERSION_MAJOR < 53
/**
 * @deprecated Use av_audio_resample_init() instead.
 */
attribute_deprecated ReSampleContext *audio_resample_init(int output_channels, int input_channels,
                                                          int output_rate, int input_rate);
#endif
/**
 *  Initializes 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 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
 * @return allocated ReSampleContext, NULL if error occured
 */
ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
                                        int output_rate, int input_rate,
                                        enum SampleFormat sample_fmt_out,
                                        enum SampleFormat sample_fmt_in,
                                        int filter_length, int log2_phase_count,
                                        int linear, double cutoff);

int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples);
void audio_resample_close(ReSampleContext *s);


/**