ffmpeg解码音频数据时，进行重采样（即改变文件原有的采样率）

我们使用ffmpeg解码音频的时候，往往需要改变原音频的采样率，即需要重采样。

比如一音乐文件的采样率22050，而播放端往往是固定的采样率，比如44100。在这种情况下，如果把解码出来的数据直接播放，会产生快进的效果。这个时候就需要对解码出来的数据作一次重采样，将数据转化为44100采样率下的数据，才能正确播放。

ffmpeg提供了一组用来重采样的API，主要如下：

/** *  Initialize audio resampling context. * * @param output_channels  number of output channels * @param input_channels   number of input channels * @param output_rate      output sample rate * @param input_rate       input sample rate * @param sample_fmt_out   requested output sample format * @param sample_fmt_in    input sample format * @param filter_length    length of each FIR filter in the filterbank relative to the cutoff frequency * @param log2_phase_count log2 of the number of entries in the polyphase filterbank * @param linear           if 1 then the used FIR filter will be linearly interpolated                           between the 2 closest, if 0 the closest will be used * @param cutoff           cutoff frequency, 1.0 corresponds to half the output sampling rate * @return allocated ReSampleContext, NULL if error occured */ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,                                        int output_rate, int input_rate,                                        enum AVSampleFormat sample_fmt_out,                                        enum AVSampleFormat sample_fmt_in,                                        int filter_length, int log2_phase_count,                                        int linear, double cutoff);int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples);/** * Free resample context. * * @param s a non-NULL pointer to a resample context previously *          created with av_audio_resample_init() */void audio_resample_close(ReSampleContext *s);

函数av_audio_resample_init()用来初始化重采样的参数，前4个参数很好理解；后6个参数基本上是使用缺省参数，分别为：

AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16,16, 10, 0, 1
函数audio_resample()用来重采样，前3个参数都好理解，最后一个参数是指“原数据的采样个数”，而不是input的bytes数。该函数的返回值也是采样个数，不过是重采样之后的。

函数audio_resample_close()用来清理重采样时分配的资源。

相关代码如下：

初始化

// need to do re-sampleif (m_codec_ctx->sample_rate != m_out_samplerate){LOGW("%s, need re-sample, initialize re-sample engine! out channels:%d, out sample rate:%d hz, in channels:%d, in sample rate:%d",__FUNCTION__, 2, m_out_samplerate, m_codec_ctx->channels, m_codec_ctx->sample_rate);m_resample_engine = av_audio_resample_init(2, m_codec_ctx->channels, m_out_samplerate, m_codec_ctx->sample_rate, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16, 16, 10, 0, 1);}

重采样代码

frame_size = AVCODEC_MAX_AUDIO_FRAME_SIZE;memset(m_audio_buff, 0, AVCODEC_MAX_AUDIO_FRAME_SIZE);decoded_len = avcodec_decode_audio3(m_codec_ctx, (short *)m_audio_buff, &frame_size, &m_avpkt);LOGI("%s, current decoded size:%d", __FUNCTION__, decoded_len);if (decoded_len > 0){m_avpkt.size -= decoded_len;m_avpkt.data += decoded_len;decoded_audio_len = frame_size;valid_data_pointer = m_audio_buff;// need to re-sampleif (m_resample_engine){// convert byte to shortint after_resampled_len = 0;int before_resampled_len = frame_size/(2 * m_codec_ctx->channels);memset(m_audio_resampled_buff, 0, AVCODEC_MAX_AUDIO_FRAME_SIZE);after_resampled_len = audio_resample(m_resample_engine, (short *)m_audio_resampled_buff, (short *)m_audio_buff, before_resampled_len);LOGI("%s, re-sampled! length in:%d, length out:%d", __FUNCTION__, before_resampled_len, after_resampled_len);decoded_audio_len = after_resampled_len * 2 * 2;//convert short to byte, and 2 channelsvalid_data_pointer = m_audio_resampled_buff;}      memcpy(buff+copied_len, valid_data_pointer, decoded_audio_len);      copied_len += decoded_audio_len;      LOGI("%s, copy1, %d bytes has copied to output buff, total:%d!", __FUNCTION__, decoded_audio_len, copied_len);                                                                                                                        }

释放代码

if (m_resample_engine){audio_resample_close(m_resample_engine);m_resample_engine = 0;}