我试图将用户的声音与音乐混合并将其保存到文件中。
我创建了2个Decoders - 1个用于语音,1个用于音乐,并将它们放入Mixer的输入中。我解码每个帧并使用FILE / createWAV / fwrite将其保存到文件。
当我的歌曲是.wav并且具有相同的sampleRate和samplesPerFrame作为录制的声音(48000/1024)时,一切都很完美。
但是,当我想使用带有不同参数的.mp3文件(44100/1152)时,最终文件不正确 - 它被拉伸或有一些噼啪声。我认为这是因为我们为每个解码器得到不同的采样解码,当它被放入混音器或保存到文件时 - 这些样本之间的差异丢失了。
据我所知,当我们做voiceDecoder->decode(buffer, &samplesDecoded)
时,它会被samplePosition
移动samplesDecoded
。
我试图做的是使用两个解码器的最小值。然而根据上面的句子,每个循环迭代歌曲将松散(1152 - 1024 = 128)128个样本,所以我也试图寻找songDecoder与voiceDecoder相同:songDecoder->seek(voiceDecoder->samplePosition, true)
但它导致完全不正确的文件。
总结一下:当每个解码器具有不同的采样率和samplesPerFrame时,如何使用2个解码器处理混频器/离线处理?
码:
void AudioProcessor::startProcessing() {
SuperpoweredStereoMixer *mixer = new SuperpoweredStereoMixer();
float *mixerInputs_[] = {0,0,0,0};
float *mixerOutputs_[] = {0,0};
float inputLevels_[]= {0.5f, 0.5f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f};
float outputLevels_[] = { 1.0f, 1.0f };
SuperpoweredDecoder *voiceDecoder = new SuperpoweredDecoder();
SuperpoweredDecoder *songDecoder = new SuperpoweredDecoder();
if (voiceDecoder->open(voiceInputPath, false) || songDecoder->open(songInputPath, false, songOffset, songLength)) {
delete voiceDecoder;
delete songDecoder;
delete mixer;
callJavaVoidMethodWithBoolParam(jvm, jObject, processingFinishedMethodId, false);
return;
};
FILE *fd = createWAV(outputPath, songDecoder->samplerate, 2);
if (!fd) {
delete voiceDecoder;
delete songDecoder;
delete mixer;
callJavaVoidMethodWithBoolParam(jvm, jObject, processingFinishedMethodId, false);
return;
};
// Create a buffer for the 16-bit integer samples coming from the decoder.
short int *voiceIntBuffer = (short int *)malloc(voiceDecoder->samplesPerFrame * 4 * sizeof(short int) + 32768);
short int *songIntBuffer = (short int *)malloc(songDecoder->samplesPerFrame * 4 * sizeof(short int) + 32768);
short int *outputIntBuffer = (short int *)malloc(voiceDecoder->samplesPerFrame * 4 * sizeof(short int) + 32768);
// Create a buffer for the 32-bit floating point samples required by the effect.
float *voiceFloatBuffer = (float *)malloc(voiceDecoder->samplesPerFrame * 4 * sizeof(float) + 32768);
float *songFloatBuffer = (float *)malloc(songDecoder->samplesPerFrame * 4 * sizeof(float) + 32768);
float *outputFloatBuffer = (float *)malloc(voiceDecoder->samplesPerFrame * 4 * sizeof(float) + 32768);
bool isError = false;
// Processing.
while (true) {
if (isCanceled) {
isError = true;
break;
}
// Decode one frame. samplesDecoded will be overwritten with the actual decoded number of samples.
unsigned int voiceSamplesDecoded = voiceDecoder->samplesPerFrame;
if (voiceDecoder->decode(voiceIntBuffer, &voiceSamplesDecoded) == SUPERPOWEREDDECODER_ERROR) {
break;
}
if (voiceSamplesDecoded < 1) {
break;
}
//
// Decode one frame. samplesDecoded will be overwritten with the actual decoded number of samples.
unsigned int songSamplesDecoded = songDecoder->samplesPerFrame;
if (songDecoder->decode(songIntBuffer, &songSamplesDecoded) == SUPERPOWEREDDECODER_ERROR) {
break;
}
if (songSamplesDecoded < 1) {
break;
}
unsigned int samplesDecoded = static_cast<unsigned int>(fmin(voiceSamplesDecoded, songSamplesDecoded));
// Convert the decoded PCM samples from 16-bit integer to 32-bit floating point.
SuperpoweredShortIntToFloat(voiceIntBuffer, voiceFloatBuffer, samplesDecoded);
SuperpoweredShortIntToFloat(songIntBuffer, songFloatBuffer, samplesDecoded);
//setup mixer inputs
mixerInputs_[0] = voiceFloatBuffer;
mixerInputs_[1] = songFloatBuffer;
mixerInputs_[2] = NULL;
mixerInputs_[3] = NULL;
// setup mixer outputs, might have two separate outputs (L/R) if second not null
mixerOutputs_[0] = outputFloatBuffer;
mixerOutputs_[1] = NULL;
mixer->process(mixerInputs_, mixerOutputs_, inputLevels_, outputLevels_, NULL, NULL, samplesDecoded);
// Convert the PCM samples from 32-bit floating point to 16-bit integer.
SuperpoweredFloatToShortInt(outputFloatBuffer, outputIntBuffer, samplesDecoded);
// Write the audio to disk.
fwrite(outputIntBuffer, 1, samplesDecoded * 4, fd);
// songDecoder->seek(voiceDecoder->samplePosition, true);
}
// Cleanup.
closeWAV(fd);
delete voiceDecoder;
delete songDecoder;
delete mixer;
free(voiceIntBuffer);
free(voiceFloatBuffer);
free(songIntBuffer);
free(songFloatBuffer);
free(outputFloatBuffer);
free(outputIntBuffer);
}
提前致谢!
您需要使用SuperpoweredResampler类匹配采样率。对于两个输入,您还需要一些循环缓冲区,因为在许多情况下可用的样本数量不匹配。
好的,所以我设法让它工作。我做了@Gabor提出的但是没有完全正常工作。我缺少的是频道 - 我必须将它包含在我的缓冲区/移位操作中,现在它很好!