FFmpeg源码分析：swr_convert()音频格式转换

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FFmpeg在libswresample模块提供提供音频转换函数，以前使用的libavresample模块已经过时。根据官方文档说明：libswresample提供深度优化的音频重采样、声道布局转换与格式转换。音频重采样过程是先建立原始音频信号，然后重新采样。重采样分为上采样和下采样，其中上采样需要插值，下采样需要抽取。从高采样率到低采样率转换是一种有损过程，FFmpeg提供若干选项和算法进行重采样。

1、libswresample模块介绍

FFmpeg关于libswresample模块介绍，包括重采样、格式转换与声道布局转换，具体文档可查看：https://ffmpeg/libswresample.html。具体描述如下：

The libswresample library performs highly optimized audio resampling, rematrixing and sample format conversion operations.

Specifically, this library performs the following conversions:

Resampling: is the process of changing the audio rate, for example from a high sample rate of 44100Hz to 8000Hz. 
Audio conversion from high to low sample rate is a lossy process. Several resampling options and algorithms are available.
Format conversion: is the process of converting the type of samples, for example from 16-bit signed samples to unsigned 8-bit or float samples. 
It also handles packing conversion, when passing from packed layout.
Rematrixing: is the process of changing the channel layout, for example from stereo to mono. 
When the input channels cannot be mapped to the output streams, the process is lossy, since it involves different gain factors and mixing.

2、SwrContext结构体

SwrContext是音频转换的结构体，位于swresample_internal.h头文件中：

struct SwrContext {                           
    enum AVSampleFormat  in_sample_fmt;      // input sample format
    enum AVSampleFormat int_sample_fmt;      // internal sample format
    enum AVSampleFormat out_sample_fmt;      // output sample format
    int64_t  in_ch_layout;                   // input channel layout
    int64_t 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 used_ch_count;                       // number of used input channels
    int engine;

    int user_in_ch_count;                    // User set input channel count
    int user_out_ch_count;                   // User set output channel count
    int user_used_ch_count;                  // User set used channel count
    int64_t user_in_ch_layout;               // User set input channel layout
    int64_t user_out_ch_layout;              // 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 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
    int filter_type;                         // swr resampling filter type
    double kaiser_beta;                      // swr beta value for Kaiser window                   

    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
    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
    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
    AudioData preout;                        // pre-output audio data: used for rematrix/resample
    AudioData out;                           // converted output audio data
    AudioData in_buffer;                     
    AudioData silence;                                                     

    struct AudioConvert *in_convert;             // input conversion context
    struct AudioConvert *out_convert;            // output conversion context
    struct AudioConvert *full_convert;           // full conversion context
    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];    // rematrixing coefficients
    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
};

3、swr_alloc与swr_alloc_set_opts

swr_alloc()函数用于分配SwrContext结构体，需要在swr_init()之前调用。而swr_alloc_set_opts()在swr_alloc()基础上，配置相关options参数，包括输入输出的采样格式、采样率、声道布局。官方使用描述如下：

SwrContext *swr = swr_alloc();
av_opt_set_channel_layout(swr, "in_channel_layout",  AV_CH_LAYOUT_5POINT1, 0);
av_opt_set_channel_layout(swr, "out_channel_layout", AV_CH_LAYOUT_STEREO,  0);
av_opt_set_int(swr, "in_sample_rate",     48000,                0);
av_opt_set_int(swr, "out_sample_rate",    44100,                0);
av_opt_set_sample_fmt(swr, "in_sample_fmt",  AV_SAMPLE_FMT_FLTP, 0);
av_opt_set_sample_fmt(swr, "out_sample_fmt", AV_SAMPLE_FMT_S16,  0);

// The same job can be done using swr_alloc_set_opts() as well:
SwrContext *swr = swr_alloc_set_opts(NULL,  // we're allocating a new context                       
                      AV_CH_LAYOUT_STEREO,  // out_ch_layout
                      AV_SAMPLE_FMT_S16,    // out_sample_fmt
                      44100,                // out_sample_rate
                      AV_CH_LAYOUT_5POINT1, // in_ch_layout
                      AV_SAMPLE_FMT_FLTP,   // in_sample_fmt
                      48000,                // in_sample_rate
                      0,                    // log_offset
                      NULL);                // log_ctx

4、swr_init

swr_init()用于初始化SwrContext上下文，内容包括传入参数进行校验、参数赋值、选择重采样器、分配输入输出转换器：

int swr_init(struct SwrContext *s){
    int ret;
    char l1[1024], l2[1024];

    clear_context(s);
    // 检查参数
    if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
        return AVERROR(EINVAL);
    }
    if(s->out_sample_fmt >= AV_SAMPLE_FMT_NB){
        return AVERROR(EINVAL);
    }
    if(s-> in_sample_rate <= 0){
        return AVERROR(EINVAL);
    }
    if(s->out_sample_rate <= 0){
        return AVERROR(EINVAL);
    }
	// 参数赋值
    s->out.ch_count  = s-> user_out_ch_count;
    s-> in.ch_count  = s->  user_in_ch_count;
    s->used_ch_count = s->user_used_ch_count;
    s-> in_ch_layout = s-> user_in_ch_layout;
    s->out_ch_layout = s->user_out_ch_layout;
    s->int_sample_fmt= s->user_int_sample_fmt;
    s->dither.method = s->user_dither_method;
    ......
    // 选择重采样器
    switch(s->engine){
#if CONFIG_LIBSOXR
        case SWR_ENGINE_SOXR: s->resampler = &swri_soxr_resampler; break;
#endif
        case SWR_ENGINE_SWR : s->resampler = &swri_resampler; break;
        default:
            av_log(s, AV_LOG_ERROR, "resampling engine is unavailable\n");
            return AVERROR(EINVAL);
    }
    ......
    // 分配输入输出的转换器
    s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt,
                                             s-> in_sample_fmt, 
											 s->used_ch_count, 
											 s->channel_map, 0);
    s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt,
                                             s->int_sample_fmt, 
											 s->out.ch_count, NULL, 0);
    // 如果需要声道转换，初始化声道转换函数
    if(s->rematrix || s->dither.method) {
        ret = swri_rematrix_init(s);
        if (ret < 0)
            goto fail;
    }
    ......
    return 0;
fail:
    swr_close(s);
    return ret;

}

5、swr_convert

swr_convert()主要是调用内部方法swr_convert_internal()进行音频转换。在音频流末尾，可以把in_arg和in_count两个参数设为0，从缓冲区刷新最后的音频数据。如果输入有多个采样数，将会在缓冲区进行缓存：

int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], 
				int out_count, const uint8_t *in_arg [SWR_CH_MAX], 
				int  in_count){
    AudioData * in= &s->in;
    AudioData *out= &s->out;
    int av_unused max_output;
    // 判断是否已经初始化
    if (!swr_is_initialized(s)) {
        return AVERROR(EINVAL);
    }
    ......
    if(s->resample){
		// 调用内部方法进行音频转换
        int ret = swr_convert_internal(s, out, out_count, in, in_count);
        if(ret>0 && !s->drop_output)
            s->outpts += ret * (int64_t)s->in_sample_rate;

        av_assert2(max_output < 0 || ret < 0 || ret <= max_output);

        return ret;
    }else{
        AudioData tmp= *in;
        int ret2=0;
        int ret, size;
        size = FFMIN(out_count, s->in_buffer_count);
        if(size){
            buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
            ret= swr_convert_internal(s, out, size, &tmp, size);
            if(ret<0)
                return ret;
            ret2= ret;
            s->in_buffer_count -= ret;
            s->in_buffer_index += ret;
            buf_set(out, out, ret);
            out_count -= ret;
            if(!s->in_buffer_count)
                s->in_buffer_index = 0;
        }
        ......
        return ret2;
    }
}

swr_convert_internal()的代码如下：

static int swr_convert_internal(struct SwrContext *s, 
                                AudioData *out, int out_count,
                                AudioData *in , int  in_count){

    // 如果是全量转换，直接转换，然后返回结果
    if(s->full_convert){
        swri_audio_convert(s->full_convert, out, in, in_count);
        return out_count;
    }
    // 重新分配缓冲区
    if((ret=swri_realloc_audio(&s->postin, in_count))<0)
        return ret;
    if(s->resample_first){
        av_assert0(s->midbuf.ch_count == s->used_ch_count);
        if((ret=swri_realloc_audio(&s->midbuf, out_count))<0)
            return ret;
    }else{
        av_assert0(s->midbuf.ch_count ==  s->out.ch_count);
        if((ret=swri_realloc_audio(&s->midbuf,  in_count))<0)
            return ret;
    }
    if((ret=swri_realloc_audio(&s->preout, out_count))<0)
        return ret;
    // 没有转换部分，执行音频转换
    if(in != postin){
        swri_audio_convert(s->in_convert, postin, in, in_count);
    }

    if(s->resample_first){
        if(postin != midbuf)
            out_count= resample(s, midbuf, out_count, postin, in_count);
        if(midbuf != preout)
            swri_rematrix(s, preout, midbuf, out_count, preout==out);
    }else{
        if(postin != midbuf)
            swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
        if(midbuf != preout)
            out_count= resample(s, preout, out_count, midbuf, in_count);
    }
    ......
    return out_count;
}

官方给出的音频转换处理demo如下：

uint8_t **input;
int in_samples;
while (get_input(&input, &in_samples)) {
    uint8_t *output;
    int out_samples = av_rescale_rnd(swr_get_delay(swr, 48000) 
	                                 + in_samples, 44100, 48000, 
									 AV_ROUND_UP);
    av_samples_alloc(&output, NULL, 2, out_samples,
                     AV_SAMPLE_FMT_S16, 0);
    out_samples = swr_convert(swr, &output, out_samples,
                                     input, in_samples);
    handle_output(output, out_samples);
    av_freep(&output);
}

6、swr_close与swr_free

swr_close()用于关闭SwrContext上下文，而swr_free()除了关闭上下文还释放指针。代码如下：

void swr_free(SwrContext **ss){
    SwrContext *s= *ss;
    if(s){
        clear_context(s);
        if (s->resampler)
            s->resampler->free(&s->resample);
    }

    av_freep(ss);
}

void swr_close(SwrContext *s){
    clear_context(s);
}

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