关于使用音频重采样库lib样本速率
有关resample的库，有多个，例如ffmpeg的libswresample。 alsa使用一个名为libsamplerate的库，github链接目标为： https://github.com/erikd/lib sample rate在alsaloop和aplay中都广泛使用。 今天写DEMO来解决实际问题。 实际问题是如何将16bit 2通道的48000采样率的PCM转换成16bit 2通道的44100采样率
再废话一点，为什么要转换不同的采样率呢？ 因为如果同时播放不同采样率的音频文件，声卡实际上只是采样率设置，需要将多个不同的采样率转换为相同的采样率，混合后传递给声卡播放。 这就是这个问题的来源。
1 .在1. Ubuntu 14.04上安装libsampleratesudoapt-getinstalllibsamplerate-dev后，您可以参考以下信息：
大胆的小海豚@大胆的小海豚- e 431:~$ find/usr/- name sample rate.h/usr/include/sample rate.h大胆的小海豚@大胆的小海豚- e 4313360 ~
(1) src_new初始化；2 ) src_process转换开始；3 ) src_delete反向初始化完整源
/* *版权所有(c ) 2002-2011 erikdecastrolopoerikd @ mega-nerd.com * * * thisprogramisfreesoftware； youcanredistributeitand/or modify * * itunderthetermsofthegnugeneralpubliclicenseaspublishedby * * thefreesoftwarefoundation either版本2 of the license， or** (at your option ) anylaterversion.* * * thisprogramisdistributedinthehopethatitwithouteventheimpliedwarrantyyuser merchantabilityorfitnessforaparticularpurpose.see the * * gnugeneralpubliclicenseformorededition * youshouldhavereceivedacopytion if not，writetothefreesoftware * * foundation，Inc .59模板放置，Suite 330，Boston，MA 02111-1307， USA.*//* * thiscodeispartofsecretrabbitcodeakalibsamplerate.a commercial * * uselicenseforthiscodeisavailable， pleasesee : * * http://www.mega-nerd.com/src/procedure.html */* * * apidocumentationisavailablehere 3360 * * * * * ifdef _ _ cplusplus extern ' c ' { # endif/* _ cplusplus */* opaque }/* src _ dataisusedtopassdatatosrc _ simple (atosrc ) long input_frames_used，output_frames_gen； int end_of_input； 双精度src _ ratio； } SRC_DATA；/* src _ CB _ dataisusedwithcallbackbasedapi.*/typedef struct { long frames； float *data_in； } SRC_CB_DATA； /**
User supplied callback function type for use with src_callback_new()** and src_callback_read(). First parameter is the same pointer that was** passed into src_callback_new(). Second parameter is pointer to a** pointer. The user supplied callback function must modify *data to** point to the start of the user supplied float array. The user supplied** function must return the number of frames that **data points to.*/typedef long (*src_callback_t) (void *cb_data, float **data) ;/*** Standard initialisation function : return an anonymous pointer to the** internal state of the converter. Choose a converter from the enums below.** Error returned in *error.*/SRC_STATE* src_new (int converter_type, int channels, int *error) ;/*** Initilisation for callback based API : return an anonymous pointer to the** internal state of the converter. Choose a converter from the enums below.** The cb_data pointer can point to any data or be set to NULL. Whatever the** value, when processing, user supplied function "func" gets called with** cb_data as first parameter.*/SRC_STATE* src_callback_new (src_callback_t func, int converter_type, int channels, int *error, void* cb_data) ;/*** Cleanup all internal allocations.** Always returns NULL.*/SRC_STATE* src_delete (SRC_STATE *state) ;/*** Standard processing function.** Returns non zero on error.*/int src_process (SRC_STATE *state, SRC_DATA *data) ;/*** Callback based processing function. Read up to frames worth of data from** the converter int *data and return frames read or -1 on error.*/long src_callback_read (SRC_STATE *state, double src_ratio, long frames, float *data) ;/*** Simple interface for performing a single conversion from input buffer to** output buffer at a fixed conversion ratio.** Simple interface does not require initialisation as it can only operate on** a single buffer worth of audio.*/int src_simple (SRC_DATA *data, int converter_type, int channels) ;/*** This library contains a number of different sample rate converters,** numbered 0 through N.**** Return a string giving either a name or a more full description of each** sample rate converter or NULL if no sample rate converter exists for** the given value. The converters are sequentially numbered from 0 to N.*/const char *src_get_name (int converter_type) ;const char *src_get_description (int converter_type) ;const char *src_get_version (void) ;/*** Set a new SRC ratio. This allows step responses** in the conversion ratio.** Returns non zero on error.*/int src_set_ratio (SRC_STATE *state, double new_ratio) ;/*** Reset the internal SRC state.** Does not modify the quality settings.** Does not free any memory allocations.** Returns non zero on error.*/int src_reset (SRC_STATE *state) ;/*** Return TRUE if ratio is a valid conversion ratio, FALSE** otherwise.*/int src_is_valid_ratio (double ratio) ;/*** Return an error number.*/int src_error (SRC_STATE *state) ;/*** Convert the error number into a string.*/const char* src_strerror (int error) ;/*** The following enums can be used to set the interpolator type** using the function src_set_converter().*/enum{ SRC_SINC_BEST_QUALITY = 0, SRC_SINC_MEDIUM_QUALITY = 1, SRC_SINC_FASTEST = 2, SRC_ZERO_ORDER_HOLD = 3, SRC_LINEAR = 4,} ;/*** Extra helper functions for converting from short to float and** back again.*/void src_short_to_float_array (const short *in, float *out, int len) ;void src_float_to_short_array (const float *in, short *out, int len) ;void src_int_to_float_array (const int *in, float *out, int len) ;void src_float_to_int_array (const float *in, int *out, int len) ;#ifdef __cplusplus} /* extern "C" */#endif /* __cplusplus */#endif /* SAMPLERATE_H */ 3. 转换音频文件例子 sudo apt-get install samplerate-programs