ffmpeg
之音频重采样
头文件
#include <libswresample/swresample.h>
定义
#define SWR_FLAG_RESAMPLE 1 ///< Force resampling even if equal sample rate
//TODO use int resample ?
//long term TODO can we enable this dynamically?
/** Dithering algorithms */
enum SwrDitherType {
SWR_DITHER_NONE = 0,
SWR_DITHER_RECTANGULAR,
SWR_DITHER_TRIANGULAR,
SWR_DITHER_TRIANGULAR_HIGHPASS,
SWR_DITHER_NS = 64, ///< not part of API/ABI
SWR_DITHER_NS_LIPSHITZ,
SWR_DITHER_NS_F_WEIGHTED,
SWR_DITHER_NS_MODIFIED_E_WEIGHTED,
SWR_DITHER_NS_IMPROVED_E_WEIGHTED,
SWR_DITHER_NS_SHIBATA,
SWR_DITHER_NS_LOW_SHIBATA,
SWR_DITHER_NS_HIGH_SHIBATA,
SWR_DITHER_NB, ///< not part of API/ABI
};
/** Resampling Engines */
enum SwrEngine {
SWR_ENGINE_SWR, /**< SW Resampler */
SWR_ENGINE_SOXR, /**< SoX Resampler */
SWR_ENGINE_NB, ///< not part of API/ABI
};
/** Resampling Filter Types */
enum SwrFilterType {
SWR_FILTER_TYPE_CUBIC, /**< Cubic */
SWR_FILTER_TYPE_BLACKMAN_NUTTALL, /**< Blackman Nuttall windowed sinc */
SWR_FILTER_TYPE_KAISER, /**< Kaiser windowed sinc */
};
/**
* @}
*/
/**
* The libswresample context. Unlike libavcodec and libavformat, this structure
* is opaque. This means that if you would like to set options, you must use
* the @ref avoptions API and cannot directly set values to members of the
* structure.
*/
typedef struct SwrContext SwrContext;
API
construct
构造
获取AVClass
/**
* 获取SWRContext的AVCLASS。 它可以组合使用 av_opt_search_fake_obj 用于检查选项。
*
* @see av_opt_find().
* @return the AVClass of SwrContext
*/
const AVClass *swr_get_class(void);
申请上下文
/**
* Allocate SwrContext.
*
* 如果你使用这个函数,你将需要设置参数(手动或用swr_alloc_set_opts()),然后再调用swr_init()。
*
* @see swr_alloc_set_opts(), swr_init(), swr_free()
* @return NULL on error, allocated context otherwise
*/
struct SwrContext *swr_alloc(void);
初始化上下文
/**
* 在用户参数设置完毕后初始化上下文。
* @note The context must be configured using the AVOption API.
*
* @see av_opt_set_int()
* @see av_opt_set_dict()
*
* @param[in,out] s Swr context to initialize
* @return AVERROR error code in case of failure.
*/
int swr_init(struct SwrContext *s);
检查上下文是否可初始化
/**
* Check whether an swr context has been initialized or not.
*
* @param[in] s Swr context to check
* @see swr_init()
* @return positive if it has been initialized, 0 if not initialized
*/
int swr_is_initialized(struct SwrContext *s);
申请上下文,并设置常用参数
/**
* Allocate SwrContext if needed and set/reset common parameters.
*
* This function does not require s to be allocated with swr_alloc(). On the
* other hand, swr_alloc() can use swr_alloc_set_opts() to set the parameters
* on the allocated context.
*
* @param s existing Swr context if available, or NULL if not
* @param out_ch_layout output channel layout (AV_CH_LAYOUT_*)
* @param out_sample_fmt output sample format (AV_SAMPLE_FMT_*).
* @param out_sample_rate output sample rate (frequency in Hz)
* @param in_ch_layout input channel layout (AV_CH_LAYOUT_*)
* @param in_sample_fmt input sample format (AV_SAMPLE_FMT_*).
* @param in_sample_rate input sample rate (frequency in Hz)
* @param log_offset logging level offset
* @param log_ctx parent logging context, can be NULL
*
* @see swr_init(), swr_free()
* @return NULL on error, allocated context otherwise
*/
struct SwrContext *swr_alloc_set_opts(struct SwrContext *s,
int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
int64_t in_ch_layout, enum AVSampleFormat in_sample_fmt, int in_sample_rate,
int log_offset, void *log_ctx);
destruct
销毁
释放上下文
/**
* Free the given SwrContext and set the pointer to NULL.
*
* @param[in] s a pointer to a pointer to Swr context
*/
void swr_free(struct SwrContext **s);
关闭上下文
/**
* Closes the context so that swr_is_initialized() returns 0.
*
* The context can be brought back to life by running swr_init(),
* swr_init() can also be used without swr_close().
* This function is mainly provided for simplifying the usecase
* where one tries to support libavresample and libswresample.
*
* @param[in,out] s Swr context to be closed
*/
void swr_close(struct SwrContext *s);
Core conversion functions
转换函数
转换函数(缓冲区)
/** Convert audio.
*
* in and in_count can be set to 0 to flush the last few samples out at the
* end.
*
* If more input is provided than output space, then the input will be buffered.
* You can avoid this buffering by using swr_get_out_samples() to retrieve an
* upper bound on the required number of output samples for the given number of
* input samples. Conversion will run directly without copying whenever possible.
*
* @param s allocated Swr context, with parameters set
* @param out output buffers, only the first one need be set in case of packed audio
* @param out_count amount of space available for output in samples per channel
* @param in input buffers, only the first one need to be set in case of packed audio
* @param in_count number of input samples available in one channel
*
* @return number of samples output per channel, negative value on error
*/
int swr_convert(struct SwrContext *s, uint8_t **out, int out_count,
const uint8_t **in , int in_count);
将时间戳从输入转换到输出
/**
* Convert the next timestamp from input to output
* timestamps are in 1/(in_sample_rate * out_sample_rate) units.
*
* @note There are 2 slightly differently behaving modes.
* @li When automatic timestamp compensation is not used, (min_compensation >= FLT_MAX)
* in this case timestamps will be passed through with delays compensated
* @li When automatic timestamp compensation is used, (min_compensation < FLT_MAX)
* in this case the output timestamps will match output sample numbers.
* See ffmpeg-resampler(1) for the two modes of compensation.
*
* @param s[in] initialized Swr context
* @param pts[in] timestamp for the next input sample, INT64_MIN if unknown
* @see swr_set_compensation(), swr_drop_output(), and swr_inject_silence() are
* function used internally for timestamp compensation.
* @return the output timestamp for the next output sample
*/
int64_t swr_next_pts(struct SwrContext *s, int64_t pts);
Low-level option setting functions
设置函数
激活重采样软补偿
/**
* Activate resampling compensation ("soft" compensation). This function is
* internally called when needed in swr_next_pts().
*
* @param[in,out] s allocated Swr context. If it is not initialized,
* or SWR_FLAG_RESAMPLE is not set, swr_init() is
* called with the flag set.
* @param[in] sample_delta delta in PTS per sample
* @param[in] compensation_distance number of samples to compensate for
* @return >= 0 on success, AVERROR error codes if:
* @li @c s is NULL,
* @li @c compensation_distance is less than 0,
* @li @c compensation_distance is 0 but sample_delta is not,
* @li compensation unsupported by resampler, or
* @li swr_init() fails when called.
*/
int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance);
自定义输入通道
/**
* Set a customized input channel mapping.
*
* @param[in,out] s allocated Swr context, not yet initialized
* @param[in] channel_map customized input channel mapping (array of channel
* indexes, -1 for a muted channel)
* @return >= 0 on success, or AVERROR error code in case of failure.
*/
int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map);
构建自定义矩阵
/**
* Generate a channel mixing matrix.
*
* This function is the one used internally by libswresample for building the
* default mixing matrix. It is made public just as a utility function for
* building custom matrices.
*
* @param in_layout input channel layout
* @param out_layout output channel layout
* @param center_mix_level mix level for the center channel
* @param surround_mix_level mix level for the surround channel(s)
* @param lfe_mix_level mix level for the low-frequency effects channel
* @param rematrix_maxval if 1.0, coefficients will be normalized to prevent
* overflow. if INT_MAX, coefficients will not be
* normalized.
* @param[out] matrix mixing coefficients; matrix[i + stride * o] is
* the weight of input channel i in output channel o.
* @param stride distance between adjacent input channels in the
* matrix array
* @param matrix_encoding matrixed stereo downmix mode (e.g. dplii)
* @param log_ctx parent logging context, can be NULL
* @return 0 on success, negative AVERROR code on failure
*/
int swr_build_matrix(uint64_t in_layout, uint64_t out_layout,
double center_mix_level, double surround_mix_level,
double lfe_mix_level, double rematrix_maxval,
double rematrix_volume, double *matrix,
int stride, enum AVMatrixEncoding matrix_encoding,
void *log_ctx);
自定义混音矩阵
/**
* Set a customized remix matrix.
*
* @param s allocated Swr context, not yet initialized
* @param matrix remix coefficients; matrix[i + stride * o] is
* the weight of input channel i in output channel o
* @param stride offset between lines of the matrix
* @return >= 0 on success, or AVERROR error code in case of failure.
*/
int swr_set_matrix(struct SwrContext *s, const double *matrix, int stride);
Sample handling functions处理函数
丢弃指定数量的输出样本
/**
* Drops the specified number of output samples.
*
* This function, along with swr_inject_silence(), is called by swr_next_pts()
* if needed for "hard" compensation.
*
* @param s allocated Swr context
* @param count number of samples to be dropped
*
* @return >= 0 on success, or a negative AVERROR code on failure
*/
int swr_drop_output(struct SwrContext *s, int count);
注入指定数量的静音样本
/**
* Injects the specified number of silence samples.
*
* This function, along with swr_drop_output(), is called by swr_next_pts()
* if needed for "hard" compensation.
*
* @param s allocated Swr context
* @param count number of samples to be dropped
*
* @return >= 0 on success, or a negative AVERROR code on failure
*/
int swr_inject_silence(struct SwrContext *s, int count);
获取下一个输入到输出的延时
/**
* Gets the delay the next input sample will experience relative to the next output sample.
*
* Swresample can buffer data if more input has been provided than available
* output space, also converting between sample rates needs a delay.
* This function returns the sum of all such delays.
* The exact delay is not necessarily an integer value in either input or
* output sample rate. Especially when downsampling by a large value, the
* output sample rate may be a poor choice to represent the delay, similarly
* for upsampling and the input sample rate.
*
* @param s swr context
* @param base timebase in which the returned delay will be:
* @li if it's set to 1 the returned delay is in seconds
* @li if it's set to 1000 the returned delay is in milliseconds
* @li if it's set to the input sample rate then the returned
* delay is in input samples
* @li if it's set to the output sample rate then the returned
* delay is in output samples
* @li if it's the least common multiple of in_sample_rate and
* out_sample_rate then an exact rounding-free delay will be
* returned
* @returns the delay in 1 / @c base units.
*/
int64_t swr_get_delay(struct SwrContext *s, int64_t base);
得到较于输入样本数的输入样本数
/**
* Find an upper bound on the number of samples that the next swr_convert
* call will output, if called with in_samples of input samples. This
* depends on the internal state, and anything changing the internal state
* (like further swr_convert() calls) will may change the number of samples
* swr_get_out_samples() returns for the same number of input samples.
*
* @param in_samples number of input samples.
* @note any call to swr_inject_silence(), swr_convert(), swr_next_pts()
* or swr_set_compensation() invalidates this limit
* @note it is recommended to pass the correct available buffer size
* to all functions like swr_convert() even if swr_get_out_samples()
* indicates that less would be used.
* @returns an upper bound on the number of samples that the next swr_convert
* will output or a negative value to indicate an error
*/
int swr_get_out_samples(struct SwrContext *s, int in_samples);
configure function
重采样版本号
/**
* Return the @ref LIBSWRESAMPLE_VERSION_INT constant.
*
* This is useful to check if the build-time libswresample has the same version
* as the run-time one.
*
* @returns the unsigned int-typed version
*/
unsigned swresample_version(void);
得到编译时的配置项
/**
* Return the swr build-time configuration.
*
* @returns the build-time @c ./configure flags
*/
const char *swresample_configuration(void);
获取许可证
/**
* Return the swr license.
*
* @returns the license of libswresample, determined at build-time
*/
const char *swresample_license(void);
AVFrame based API
转换输入帧,并写入输出帧
/**
* Convert the samples in the input AVFrame and write them to the output AVFrame.
*
* Input and output AVFrames must have channel_layout, sample_rate and format set.
*
* If the output AVFrame does not have the data pointers allocated the nb_samples
* field will be set using av_frame_get_buffer()
* is called to allocate the frame.
*
* The output AVFrame can be NULL or have fewer allocated samples than required.
* In this case, any remaining samples not written to the output will be added
* to an internal FIFO buffer, to be returned at the next call to this function
* or to swr_convert().
*
* If converting sample rate, there may be data remaining in the internal
* resampling delay buffer. swr_get_delay() tells the number of
* remaining samples. To get this data as output, call this function or
* swr_convert() with NULL input.
*
* If the SwrContext configuration does not match the output and
* input AVFrame settings the conversion does not take place and depending on
* which AVFrame is not matching AVERROR_OUTPUT_CHANGED, AVERROR_INPUT_CHANGED
* or the result of a bitwise-OR of them is returned.
*
* @see swr_delay()
* @see swr_convert()
* @see swr_get_delay()
*
* @param swr audio resample context
* @param output output AVFrame
* @param input input AVFrame
* @return 0 on success, AVERROR on failure or nonmatching
* configuration.
*/
int swr_convert_frame(SwrContext *swr,
AVFrame *output, const AVFrame *input);
使用输入帧和输出帧中的信息重新配置上下文
/**
* Configure or reconfigure the SwrContext using the information
* provided by the AVFrames.
*
* The original resampling context is reset even on failure.
* The function calls swr_close() internally if the context is open.
*
* @see swr_close();
*
* @param swr audio resample context
* @param output output AVFrame
* @param input input AVFrame
* @return 0 on success, AVERROR on failure.
*/
int swr_config_frame(SwrContext *swr, const AVFrame *out, const AVFrame *in);