FFmpeg音频重采样基本流程

流程概述

音频重采样的基本流程为：

1. 申请重采样器上下文
2. 设置重采样去上下文的参数
3. 初始化重采样器
4. 申请数据存放的缓冲区空间
5. 进行重采样

注意，要先设置参数再对重采样器初始化

用到的API

1. SwrContext重采样器上下文的结构体。此结构是不透明的，这意味着，如果要设置选项，诸如av_opt_set等函数来设置。

2. struct SwrContext *swr_alloc();，申请重采样器上下文。

3. int av_opt_set(void *obj, const char *name, const char *val, int search_flags);
int av_opt_set_int(void *obj, const char *name, int64_t val, int search_flags);
int av_opt_set_chlayout(void *obj, const char *name, const AVChannelLayout *layout, int search_flags);
   av_opt_set* 函数簇，这里仅列举几个。以av_opt_set为例，用于将给定name的obj字段设置为指定的val。第一个void* 的obj参数表示要设置的对象，第二个name参数表示要设置的字段名称，以字符串形式传入。例如obj为SwrContext* 对象，name为"in_sample_rate"就对应着SwrContext中的同名字段。中间的部分就为要设置的参数，最后的search_flags表示搜索搜索标志，一般设为0即可。

4. int swr_alloc_set_opts2(struct SwrContext **ps, const AVChannelLayout *out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate, const AVChannelLayout *in_ch_layout, enum AVSampleFormat in_sample_fmt, int in_sample_rate, int log_offset, void *log_ctx);如果还未分配则分配SwrContext，并设置/重置公共参数。就相当于alloc + set。

5. int swr_init(struct SwrContext *s);重采样去初始化。必须在设置过SwrContext 参数之后初始化。

6. int64_t av_rescale_rnd(int64_t a, int64_t b, int64_t c, enum AVRounding rnd)和int64_t av_rescale(int64_t a, int64_t b, int64_t c)都是用于计算的(a*b/c)，唯一的区别在于rnd可以设置向上取整向下取整等。

7. int av_samples_alloc_array_and_samples(uint8_t ***audio_data, int *linesize, int nb_channels, int nb_samples, enum AVSampleFormat sample_fmt, int align);
   申请一个 data[nb_channels][ch_data] 的二维数组，所以audio_data要作为一个三级指针传进去。

8. void av_freep(void *ptr);释放av_samples_alloc_array_and_samples申请的data。av_freep即使传入null也是安全的。用法示例：

   uint8_t *buf = av_malloc(16);
   av_freep(&buf);
   

9. int64_t swr_get_delay(struct SwrContext *s, int64_t base);获取下一个输入样本相对于下一个输出样本所经历的延迟帧数。

10. int swr_convert(struct SwrContext *s, uint8_t * const *out, int out_count, const uint8_t * const *in , int in_count);
    音频重采样，in和out是由av_samples_alloc_array_and_samples生成的data缓冲区。in_count和out_count则是对应的缓冲区大小的样本数。

11. int av_samples_get_buffer_size(int *linesize, int nb_channels, int nb_samples, enum AVSampleFormat sample_fmt, int align);
    获取给定音频参数所需的缓冲区大小。
    
    

tips

1. swr是software resample的缩写
2. nb_samples样本数，表示每帧的每个通道中的采样点数。
3. 重采样的三个关键参数：采样率、采样格式、声道布局。
4. 音频的planner格式的数据是分在多个数组中的，例如左右声道的data[0]中存放L声道的数据，data[1]中存放R声道的数据。而交错模式的数据则是按照LRLR…的顺序统一放到data[0]中的。
5. av_freep要取地址的原因，是因为要将指针置空，仅此而已。
6. 老版本的FFmpeg，例如在ffmpeg-4.2下，音频声道数只是一个单一的int型字段。而新版本的FFmpeg，以ffmpeg-7.0为例，则是将音频数据封装为一个AVChannelLayout结构体了。所以在设置 ‘layout’ 字段时，不能再用av_opt_set_int接口，而是要用av_opt_set_chlayout，name参数也要使用"in_chlayout"才行。

demo样例

重采样样例，参考：Examples - resample_audio.c

#include <iostream>
#include <fstream>
#include <string>
#include <cmath>
using namespace std;extern "C"
{
#include <libavutil/opt.h>
#include <libavutil/channel_layout.h>
#include <libavutil/samplefmt.h>
#include <libswresample/swresample.h>
}/* format转字符串 */
string string_sample_fmt(enum AVSampleFormat sample_fmt)
{// 定义sample_fmt_entry结构体，同时定义了一个数组struct sample_fmt_entry{enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;} sample_fmt_entries[] = {{ AV_SAMPLE_FMT_U8,  "u8",    "u8"    },{ AV_SAMPLE_FMT_S16, "s16be", "s16le" },{ AV_SAMPLE_FMT_S32, "s32be", "s32le" },{ AV_SAMPLE_FMT_FLT, "f32be", "f32le" },{ AV_SAMPLE_FMT_DBL, "f64be", "f64le" },};// 返回字符串const char* str_fmt = nullptr;int arr_len = FF_ARRAY_ELEMS(sample_fmt_entries);for (int i = 0; i < arr_len; i++){auto entry = sample_fmt_entries[i];if (sample_fmt == entry.sample_fmt){return AV_NE(entry.fmt_be, entry.fmt_le);}}
}/*** Fill dst buffer with nb_samples, generated starting from t.* 交错模式，函数摘自：https://ffmpeg.org/doxygen/7.0/resample_audio_8c-example.html* sin曲线，t表示当前所在的相位，周期为一帧所持续的时间*/
void fill_samples(double *dst, int nb_samples, int nb_channels, int sample_rate, double *t)
{int i, j;double tincr = 1.0 / sample_rate, *dstp = dst;const double c = 2 * M_PI * 440.0;/* generate sin tone with 440Hz frequency and duplicated channels */for (i = 0; i < nb_samples; i++) {*dstp = sin(c * *t);for (j = 1; j < nb_channels; j++)dstp[j] = dstp[0];dstp += nb_channels;*t += tincr;}
}int main()
{/* 采样参数定义 */// 输入参数int src_sample_rate = 48000;enum AVSampleFormat src_sample_fmt = AV_SAMPLE_FMT_DBL;AVChannelLayout src_ch_layout = AV_CHANNEL_LAYOUT_STEREO; // 立体声// 输出参数int dst_sample_rate = 44100;enum AVSampleFormat dst_sample_fmt = AV_SAMPLE_FMT_S16;AVChannelLayout dst_ch_layout = AV_CHANNEL_LAYOUT_STEREO; // 立体声// 创建重采样器上下文（暂且认为不会失败）SwrContext *swr_ctx = swr_alloc();/* 参数设置（SwrContext字段设置） */// 输入参数check_optset(av_opt_set_int(swr_ctx, "in_sample_rate", src_sample_rate, 0), __LINE__);check_optset(av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", src_sample_fmt, 0), __LINE__);check_optset(av_opt_set_chlayout(swr_ctx, "in_chlayout", &src_ch_layout, 0), __LINE__);// 输出参数check_optset(av_opt_set_int(swr_ctx, "out_sample_rate", dst_sample_rate, 0), __LINE__);check_optset(av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", dst_sample_fmt, 0), __LINE__);check_optset(av_opt_set_chlayout(swr_ctx, "out_chlayout", &dst_ch_layout, 0), __LINE__);// 参数设置完成后，初始化上下文swr_init(swr_ctx);// 给输入源分配内存空间uint8_t **src_data = nullptr;int src_linesize;int src_nb_samples = 1024; // 每个通道的样本数av_samples_alloc_array_and_samples(&src_data, &src_linesize, src_ch_layout.nb_channels,src_nb_samples, src_sample_fmt, 0);// 给输出源分配内存空间uint8_t **dst_data;int dst_linesize;// 计算输出的信道样本数：a * b / c，AV_ROUND_UP表示向上取整int dst_nb_samples = av_rescale_rnd(src_nb_samples, dst_sample_rate, src_sample_rate, AV_ROUND_UP);// 分配空间av_samples_alloc_array_and_samples(&dst_data, &dst_linesize, dst_ch_layout.nb_channels,dst_nb_samples, dst_sample_fmt, 0);// 采样转换double t = 0; // 时间，以输入源的时间为基准int max_nb_samples = dst_nb_samples;string dst_file_name = "out.pcm";ofstream dst_file(dst_file_name, ios_base::out | ios_base::binary);while(t < 10){// 生成输入源（模拟）fill_samples((double*)src_data[0], src_nb_samples, src_ch_layout.nb_channels, src_sample_rate, &t);// 获取延迟（dst音频相对src音频延迟的帧数）int64_t delay = swr_get_delay(swr_ctx, src_sample_rate);// 输出的信道样本数，a * b / cdst_nb_samples = av_rescale(delay + src_nb_samples, dst_sample_rate, src_sample_rate);// 如果输出缓冲区大小不够，重新申请空间if(dst_nb_samples > max_nb_samples){// 重新申请空间av_freep(&dst_data[0]);av_samples_alloc(dst_data, &dst_linesize, dst_ch_layout.nb_channels,dst_nb_samples, dst_sample_fmt, 1);max_nb_samples = dst_nb_samples;}// 音频重采样int ret = swr_convert(swr_ctx, dst_data, dst_nb_samples,(const uint8_t **)src_data, src_nb_samples);// 获取给定音频参数所需的缓冲区大小。int dst_buf_size = av_samples_get_buffer_size(&dst_linesize, dst_ch_layout.nb_channels,ret, dst_sample_fmt, 1);// writedst_file.write((char*)dst_data[0], dst_buf_size);}// clear and exit// TODO
}

附录 - SwrContext结构体字段

版本：ffmpeg-7.0

struct SwrContext {const AVClass *av_class;                        ///< AVClass used for AVOption and av_log()int log_level_offset;                           ///< logging level offsetvoid *log_ctx;                                  ///< parent logging contextenum AVSampleFormat  in_sample_fmt;             ///< input sample formatenum AVSampleFormat int_sample_fmt;             ///< internal sample format (AV_SAMPLE_FMT_FLTP or AV_SAMPLE_FMT_S16P)enum AVSampleFormat out_sample_fmt;             ///< output sample formatAVChannelLayout used_ch_layout;                 ///< number of used input channels (mapped channel count if channel_map, otherwise in.ch_count)AVChannelLayout  in_ch_layout;                  ///< input channel layoutAVChannelLayout out_ch_layout;                  ///< output channel layoutint      in_sample_rate;                        ///< input sample rateint     out_sample_rate;                        ///< output sample rateint flags;                                      ///< miscellaneous flags such as SWR_FLAG_RESAMPLEfloat slev;                                     ///< surround mixing levelfloat clev;                                     ///< center mixing levelfloat lfe_mix_level;                            ///< LFE mixing levelfloat rematrix_volume;                          ///< rematrixing volume coefficientfloat rematrix_maxval;                          ///< maximum value for rematrixing outputint matrix_encoding;                            /**< matrixed stereo encoding */const int *channel_map;                         ///< channel index (or -1 if muted channel) mapint engine;AVChannelLayout user_used_chlayout;             ///< User set used channel layoutAVChannelLayout user_in_chlayout;               ///< User set input channel layoutAVChannelLayout user_out_chlayout;              ///< User set output channel layoutenum AVSampleFormat user_int_sample_fmt;        ///< User set internal sample formatint user_dither_method;                         ///< User set dither methodstruct DitherContext dither;int filter_size;                                /**< length of each FIR filter in the resampling filterbank relative to the cutoff frequency */int phase_shift;                                /**< log2 of the number of entries in the resampling polyphase filterbank */int linear_interp;                              /**< if 1 then the resampling FIR filter will be linearly interpolated */int exact_rational;                             /**< if 1 then enable non power of 2 phase_count */double cutoff;                                  /**< resampling cutoff frequency (swr: 6dB point; soxr: 0dB point). 1.0 corresponds to half the output sample rate */int filter_type;                                /**< swr resampling filter type */double kaiser_beta;                                /**< swr beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) */double precision;                               /**< soxr resampling precision (in bits) */int cheby;                                      /**< soxr: if 1 then passband rolloff will be none (Chebyshev) & irrational ratio approximation precision will be higher */float min_compensation;                         ///< swr minimum below which no compensation will happenfloat min_hard_compensation;                    ///< swr minimum below which no silence inject / sample drop will happenfloat soft_compensation_duration;               ///< swr duration over which soft compensation is appliedfloat max_soft_compensation;                    ///< swr maximum soft compensation in seconds over soft_compensation_durationfloat async;                                    ///< swr simple 1 parameter async, similar to ffmpegs -asyncint64_t firstpts_in_samples;                    ///< swr first pts in samplesint resample_first;                             ///< 1 if resampling must come first, 0 if rematrixingint rematrix;                                   ///< flag to indicate if rematrixing is needed (basically if input and output layouts mismatch)int rematrix_custom;                            ///< flag to indicate that a custom matrix has been definedAudioData in;                                   ///< input audio dataAudioData postin;                               ///< post-input audio data: used for rematrix/resampleAudioData midbuf;                               ///< intermediate audio data (postin/preout)AudioData preout;                               ///< pre-output audio data: used for rematrix/resampleAudioData out;                                  ///< converted output audio dataAudioData in_buffer;                            ///< cached audio data (convert and resample purpose)AudioData silence;                              ///< temporary with silenceAudioData drop_temp;                            ///< temporary used to discard outputint in_buffer_index;                            ///< cached buffer positionint in_buffer_count;                            ///< cached buffer lengthint resample_in_constraint;                     ///< 1 if the input end was reach before the output end, 0 otherwiseint flushed;                                    ///< 1 if data is to be flushed and no further input is expectedint64_t outpts;                                 ///< output PTSint64_t firstpts;                               ///< first PTSint drop_output;                                ///< number of output samples to dropdouble delayed_samples_fixup;                   ///< soxr 0.1.1: needed to fixup delayed_samples after flush has been called.struct AudioConvert *in_convert;                ///< input conversion contextstruct AudioConvert *out_convert;               ///< output conversion contextstruct AudioConvert *full_convert;              ///< full conversion context (single conversion for input and output)struct ResampleContext *resample;               ///< resampling contextstruct Resampler const *resampler;              ///< resampler virtual function tabledouble matrix[SWR_CH_MAX][SWR_CH_MAX];          ///< floating point rematrixing coefficientsfloat matrix_flt[SWR_CH_MAX][SWR_CH_MAX];       ///< single precision floating point rematrixing coefficientsuint8_t *native_matrix;uint8_t *native_one;uint8_t *native_simd_one;uint8_t *native_simd_matrix;int32_t matrix32[SWR_CH_MAX][SWR_CH_MAX];       ///< 17.15 fixed point rematrixing coefficientsuint8_t matrix_ch[SWR_CH_MAX][SWR_CH_MAX+1];    ///< Lists of input channels per output channel that have non zero rematrixing coefficientsmix_1_1_func_type *mix_1_1_f;mix_1_1_func_type *mix_1_1_simd;mix_2_1_func_type *mix_2_1_f;mix_2_1_func_type *mix_2_1_simd;mix_any_func_type *mix_any_f;/* TODO: callbacks for ASM optimizations */
};