目录
流程概述
音频重采样的基本流程为：
- 申请重采样器上下文
- 设置重采样去上下文的参数
- 初始化重采样器
- 申请数据存放的缓冲区空间
- 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
-
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。 -
struct SwrContext *swr_alloc();，申请重采样器上下文。 -
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。
-
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。 -
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。 -
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。 -
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。
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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：cuint8_t *buf = av_malloc(16); av_freep(&buf); -
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。 -
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。
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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
- swr是software resample的缩写
- nb_samples样本数，表示每帧的每个通道中的采样点数。
- 重采样的三个关键参数：采样率、采样格式、声道布局。
- 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。
- av_freep要取地址的原因，是因为要将指针置空，仅此而已。
- 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
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 / c
dst_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);
// write
dst_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
c
struct SwrContext {
const AVClass *av_class; ///< AVClass used for AVOption and av_log()
int log_level_offset; ///< logging level offset
void *log_ctx; ///< parent logging context
enum AVSampleFormat in_sample_fmt; ///< input sample format
enum AVSampleFormat int_sample_fmt; ///< internal sample format (AV_SAMPLE_FMT_FLTP or AV_SAMPLE_FMT_S16P)
enum AVSampleFormat out_sample_fmt; ///< output sample format
AVChannelLayout used_ch_layout; ///< number of used input channels (mapped channel count if channel_map, otherwise in.ch_count)
AVChannelLayout in_ch_layout; ///< input channel layout
AVChannelLayout out_ch_layout; ///< output channel layout
int in_sample_rate; ///< input sample rate
int out_sample_rate; ///< output sample rate
int flags; ///< miscellaneous flags such as SWR_FLAG_RESAMPLE
float slev; ///< surround mixing level
float clev; ///< center mixing level
float lfe_mix_level; ///< LFE mixing level
float rematrix_volume; ///< rematrixing volume coefficient
float rematrix_maxval; ///< maximum value for rematrixing output
int matrix_encoding; /**< matrixed stereo encoding */
const int *channel_map; ///< channel index (or -1 if muted channel) map
int engine;
AVChannelLayout user_used_chlayout; ///< User set used channel layout
AVChannelLayout user_in_chlayout; ///< User set input channel layout
AVChannelLayout user_out_chlayout; ///< User set output channel layout
enum AVSampleFormat user_int_sample_fmt; ///< User set internal sample format
int user_dither_method; ///< User set dither method
struct 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 happen
float min_hard_compensation; ///< swr minimum below which no silence inject / sample drop will happen
float soft_compensation_duration; ///< swr duration over which soft compensation is applied
float max_soft_compensation; ///< swr maximum soft compensation in seconds over soft_compensation_duration
float async; ///< swr simple 1 parameter async, similar to ffmpegs -async
int64_t firstpts_in_samples; ///< swr first pts in samples
int resample_first; ///< 1 if resampling must come first, 0 if rematrixing
int 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 defined
AudioData in; ///< input audio data
AudioData postin; ///< post-input audio data: used for rematrix/resample
AudioData midbuf; ///< intermediate audio data (postin/preout)
AudioData preout; ///< pre-output audio data: used for rematrix/resample
AudioData out; ///< converted output audio data
AudioData in_buffer; ///< cached audio data (convert and resample purpose)
AudioData silence; ///< temporary with silence
AudioData drop_temp; ///< temporary used to discard output
int in_buffer_index; ///< cached buffer position
int in_buffer_count; ///< cached buffer length
int resample_in_constraint; ///< 1 if the input end was reach before the output end, 0 otherwise
int flushed; ///< 1 if data is to be flushed and no further input is expected
int64_t outpts; ///< output PTS
int64_t firstpts; ///< first PTS
int drop_output; ///< number of output samples to drop
double delayed_samples_fixup; ///< soxr 0.1.1: needed to fixup delayed_samples after flush has been called.
struct AudioConvert *in_convert; ///< input conversion context
struct AudioConvert *out_convert; ///< output conversion context
struct AudioConvert *full_convert; ///< full conversion context (single conversion for input and output)
struct ResampleContext *resample; ///< resampling context
struct Resampler const *resampler; ///< resampler virtual function table
double matrix[SWR_CH_MAX][SWR_CH_MAX]; ///< floating point rematrixing coefficients
float matrix_flt[SWR_CH_MAX][SWR_CH_MAX]; ///< single precision floating point rematrixing coefficients
uint8_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 coefficients
uint8_t matrix_ch[SWR_CH_MAX][SWR_CH_MAX+1]; ///< Lists of input channels per output channel that have non zero rematrixing coefficients
mix_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 */
};