在进行音频播放时，有时视频流不能知足播放要求，须要对声音的相关属性如：通道数，采样率，样本存储方式进行变动播放，也就是音频重采样。ffmpeg提供了SwrContext进行转换。ide
typedef struct SwrContext SwrContext;
声音在录制时在不一样空间位置用不一样录音设备采样的声音信号，声音在播放时采用相应个数的扬声器播放。采用多通道的方式是为了丰富声音的现场感。经常使用的立体声有2个通道，环绕立体声3个通道。数字音频就是有一连串的样本流组成，立体声每次采用要采两次。有点相似视频中的YUV各个份量。函数
把模拟信号转换成数字信号在计算机中处理，须要按照必定的采样率采样，样本值就是声音波形中的一个值。音频在播放时按照采样率进行，采样率越高声音的连续性就越好，因为人的听觉器官分辨能力的局限，每每这些数值达到某种程度就能够知足人对“连续”性的需求了。例如22050和44100的采样率就是电台和CD 经常使用的采样率。相似视频中的帧率。布局
单位时间所需的空间存储。比特率反应的是视频或者音频一个样本全部的信息量，越大含有的信息量就大。视频中，图像分辨率越大，一帧就越大，实时解码就容易饥渴，传输带宽须要越大，存储空间就越大。音频中描述一个样本就越准确。学习
视频中帧就是一个图片采样。音频中一帧通常包含多个样本，如AAC格式会包含1024个样本。ui
音频中一个样本存储方式。列举ffmpeg中的样本格式spa
enum AVSampleFormat {
AV_SAMPLE_FMT_NONE = -1,
AV_SAMPLE_FMT_U8, ///< unsigned 8 bits
AV_SAMPLE_FMT_S16, ///< signed 16 bits
AV_SAMPLE_FMT_S32, ///< signed 32 bits
AV_SAMPLE_FMT_FLT, ///< float
AV_SAMPLE_FMT_DBL, ///< double
AV_SAMPLE_FMT_U8P, ///< unsigned 8 bits, planar
AV_SAMPLE_FMT_S16P, ///< signed 16 bits, planar
AV_SAMPLE_FMT_S32P, ///< signed 32 bits, planar
AV_SAMPLE_FMT_FLTP, ///< float, planar
AV_SAMPLE_FMT_DBLP, ///< double, planar
AV_SAMPLE_FMT_NB ///< Number of sample formats. DO NOT USE if linking dynamically
};
讲一下AV_SAMPLE_FMT_S16和AV_SAMPLE_FMT_S16P格式，AV_SAMPLE_FMT_S16保存一个样本采用有符号16bit交叉存储的方式，AV_SAMPLE_FMT_S16P保存一个样本采用有符号16bit平面存储的方式。举例有两个通道，通道1数据流 c1 c1 c1c1... , 通道2数据流 c2 c2 c2 c2....net
平面存储方式：c1 c1 c1c1... c2 c2 c2 c2...code
交叉存储方式：c1, c2,c1, c2, c1, c2, ...orm
AVFrame中平面方式planar每一个通道数据存储在data[0], data[1]中，长度为linesize[0],linesize[1],交叉方式则全部的数据都存储在data[0],长度为linesize[0]。视频
例程是ffmpeg2.4源代码目录下的doc/examples/resampling_audio.c文件，为便于学习作部分修改。
static int get_format_from_sample_fmt(const char **fmt,
enum AVSampleFormat sample_fmt)
{
int i;
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" },
};
*fmt = NULL;
for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++)
{
struct sample_fmt_entry *entry = &sample_fmt_entries[i];
if (sample_fmt == entry->sample_fmt)
{
*fmt = AV_NE(entry->fmt_be, entry->fmt_le);
return 0;
}
}
fprintf(stderr,
"Sample format %s not supported as output format\n",
av_get_sample_fmt_name(sample_fmt));
return AVERROR(EINVAL);
}
/**
* Fill dst buffer with nb_samples, generated starting from t.
*至关因而声源 产生一个正弦波形的声波
* dst 保存声音数据返回个调用者 nb_samples 采用的样本数 nb_channels 声音通道数，代表单次采样的样本数 t采用开始时间
*正弦波形就是一个生源，实际中复杂的声音都是经过波形叠加成的。
*以 sample_rate采样率，从时间t开始采样，采样通道为2，每一个通道的数据相同，从频率为440HZ的波形上采样，造成声源
*/
static void fill_samples(double *dst, int nb_samples, int nb_channels, int sample_rate, double *t)
{
int i, j;
//采样时间间隔 tincr
double tincr = 1.0 / sample_rate, *dstp = dst;
//正弦波y=Asin（ωx+φ）+h 最小正周期T=2π/|ω| 因此440HZ是正弦波的频率
const double c = 2 * M_PI * 440.0;
/* generate sin tone with 440Hz frequency and duplicated channels */
//填充每一个通道数据 采用交叉存储
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(int argc, char **argv)
{
// AV_CH_LAYOUT_STEREO 声音布局立体声 AV_CH_LAYOUT_SURROUND 声音布局环绕立体声
int64_t src_ch_layout = AV_CH_LAYOUT_STEREO, dst_ch_layout = AV_CH_LAYOUT_SURROUND;
//声音采样率
int src_rate = 48000, dst_rate = 44100;
uint8_t **src_data = NULL, **dst_data = NULL;
int src_nb_channels = 0, dst_nb_channels = 0;
int src_linesize, dst_linesize;
//每次采用样本数
int src_nb_samples = 1024, dst_nb_samples, max_dst_nb_samples;
//样本存储格式
enum AVSampleFormat src_sample_fmt = AV_SAMPLE_FMT_DBL, dst_sample_fmt = AV_SAMPLE_FMT_S16;
const char *dst_filename = NULL;
FILE *dst_file;
int dst_bufsize;
const char *fmt;
//重采样上下文
struct SwrContext *swr_ctx;
double t;
int ret;
if (argc != 2)
{
fprintf(stderr, "Usage: %s output_file\n"
"API example program to show how to resample an audio stream with libswresample.\n"
"This program generates a series of audio frames, resamples them to a specified "
"output format and rate and saves them to an output file named output_file.\n",
argv[0]);
exit(1);
}
dst_filename = argv[1];
dst_file = fopen(dst_filename, "wb");
if (!dst_file)
{
fprintf(stderr, "Could not open destination file %s\n", dst_filename);
exit(1);
}
/* create resampler context */
//初始化常采样上下文
swr_ctx = swr_alloc();
if (!swr_ctx)
{
fprintf(stderr, "Could not allocate resampler context\n");
ret = AVERROR(ENOMEM);
goto end;
}
/* set options */
//设置源通道布局
av_opt_set_int(swr_ctx, "in_channel_layout", src_ch_layout, 0);
//设置源通道采样率
av_opt_set_int(swr_ctx, "in_sample_rate", src_rate, 0);
//设置源通道样本格式
av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", src_sample_fmt, 0);
//目标通道布局
av_opt_set_int(swr_ctx, "out_channel_layout", dst_ch_layout, 0);
//目标采用率
av_opt_set_int(swr_ctx, "out_sample_rate", dst_rate, 0);
//目标样本格式
av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", dst_sample_fmt, 0);
/* initialize the resampling context */
if ((ret = swr_init(swr_ctx)) < 0)
{
fprintf(stderr, "Failed to initialize the resampling context\n");
goto end;
}
/* allocate source and destination samples buffers */
//获取源通道数
src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout);
//分配源声音所须要空间 src_linesize= src_nb_channels× src_nb_samples×sizeof(double)
ret = av_samples_alloc_array_and_samples(&src_data, &src_linesize, src_nb_channels,
src_nb_samples, src_sample_fmt, 0);
if (ret < 0)
{
fprintf(stderr, "Could not allocate source samples\n");
goto end;
}
/* compute the number of converted samples: buffering is avoided
* ensuring that the output buffer will contain at least all the
* converted input samples */
//计算目标样本数 转换先后的样本数不同 抓住一点 采样时间相等
//src_nb_samples/src_rate=dst_nb_samples/dst_rate
max_dst_nb_samples = dst_nb_samples = av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
/* buffer is going to be directly written to a rawaudio file, no alignment */
dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout);
ret = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize, dst_nb_channels,
dst_nb_samples, dst_sample_fmt, 0);
if (ret < 0)
{
fprintf(stderr, "Could not allocate destination samples\n");
goto end;
}
t = 0;
do {
/* generate synthetic audio */
fill_samples((double *)src_data[0], src_nb_samples, src_nb_channels, src_rate, &t);
/* compute destination number of samples */
//swr_get_delay(swr_ctx, src_rate)延迟时间 源采样率为单位的样本数
dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, src_rate) +
src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
if (dst_nb_samples > max_dst_nb_samples)
{
av_freep(&dst_data[0]);
ret = av_samples_alloc(dst_data, &dst_linesize, dst_nb_channels,
dst_nb_samples, dst_sample_fmt, 1);
if (ret < 0)
break;
max_dst_nb_samples = dst_nb_samples;
}
/* convert to destination format */
//ret 实际转换获得的样本数
ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, (const uint8_t **)src_data, src_nb_samples);
if (ret < 0)
{
fprintf(stderr, "Error while converting\n");
goto end;
}
dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels,
ret, dst_sample_fmt, 1);
if (dst_bufsize < 0)
{
fprintf(stderr, "Could not get sample buffer size\n");
goto end;
}
printf("t:%f in:%d out:%d\n", t, src_nb_samples, ret);
fwrite(dst_data[0], 1, dst_bufsize, dst_file);
} while (t < 10);
if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt)) < 0)
goto end;
fprintf(stderr, "Resampling succeeded. Play the output file with the command:\n"
"ffplay -f %s -channel_layout %lld -channels %d -ar %d %s\n",
fmt, dst_ch_layout, dst_nb_channels, dst_rate, dst_filename);
while(1)
{
Sleep(50);
}
end:
fclose(dst_file);
if (src_data)
av_freep(&src_data[0]);
av_freep(&src_data);
if (dst_data)
av_freep(&dst_data[0]);
av_freep(&dst_data);
swr_free(&swr_ctx);
return ret < 0;
}
编译环境：Win7_32bit+VS2010
FFMPEG版本：ffmpeg-2.4
源代码下载地址：http://download.csdn.net/detail/hiwubihe/9593005
技术在于交流、沟通，转载请注明出处并保持做品的完整性。