测试一个放美团的demo,拉到底部会跳动

[15202284857]

您好,我测试了一个美团的demo,并且也用到了项目中,只是没有上下拉刷新,泥浆上下拉刷新去掉之后,将每一个cell的高度随机设置,我随便写到 arc4random()%300 + 100;这样你拉到底部,触发弹性效果的时候,松手会突然的回弹跳动,体验十分的不好,(如果没有重现,多拉几次,或者拉的距离大点就会出现)没有系统默认的那种回弹舒服,麻烦是否可以修复下上面的问题呢

[adamstark]

Do you mean the size in bytes? Or the length in the number of audio samples? :)

[rfdnl]

Hello there again, Mr. Adam Stark

Sorry for the late reply. I have a Wave Sound file of 21,846 bytes (21.3 KB), length of some microseconds and BitRate of 1411 kbps, based on Properties window. After I load the file, I try to check the properties of the audio file based on your library, BitDepth(16), SampleRate(44100), Channel(2), Samples(5408), I try to get the buffer size by multiplying Channel and Samples = 10816. Following is how I load the audio file and create the audio buffer to be used with OpenAL Soft. Am I doing it correctly?

// load AudioFile<double> ALuint buffer; alGenBuffers((ALuint)1, &buffer); int bufferSize = numChannel * numSamples; double* bufferData = (double*)malloc(bufferSize); alBufferData(buffer, AL_FORMAT_STEREO16, bufferData, bufferSize, sampleRate);

Edit: The problem with the current code is the sound is played but I think only 1/4 part of the file was played. I think something is wrong with the way I count the bufferSize.

[adamstark]

I'm afraid I'm not familiar with OpenAL Soft - is the AudioFile library being used here? :)

[ScrappyCocco]

Did he managed to load a file into openal? I'm having the same difficulty opening the file with AudioFile and loading it to openal

[matts-dev]

For those in the future looking to use this to pass PCM data to OpenAL.

It appears this library doesn't provide a function directly to get the PCM data, but it does do this work -- just for a different purpose. In the process of writing to a WAV file it writes the data in PCM in correct format. see "bool AudioFile<T>::saveToWaveFile" function. the DATA chunk portion.

I extracted this out into its own function

/** @note extracted this from writing to WAV file -- this is not part of the original library. */
template <class T>
bool AudioFile<T>::savePCMToBuffer(std::vector<uint8_t>& fileData)
{
	fileData.clear();

	int16_t audioFormat = bitDepth == 32 ? WavAudioFormat::IEEEFloat : WavAudioFormat::PCM;

	for (int i = 0; i < getNumSamplesPerChannel(); i++)
	{
		for (int channel = 0; channel < getNumChannels(); channel++)
		{
			if (bitDepth == 8)
			{
				uint8_t byte = sampleToSingleByte(samples[channel][i]);
				fileData.push_back(byte);
			}
			else if (bitDepth == 16)
			{
				int16_t sampleAsInt = sampleToSixteenBitInt(samples[channel][i]);
				addInt16ToFileData(fileData, sampleAsInt);
			}
			else if (bitDepth == 24)
			{
				int32_t sampleAsIntAgain = (int32_t)(samples[channel][i] * (T)8388608.);

				uint8_t bytes[3];
				bytes[2] = (uint8_t)(sampleAsIntAgain >> 16) & 0xFF;
				bytes[1] = (uint8_t)(sampleAsIntAgain >> 8) & 0xFF;
				bytes[0] = (uint8_t)sampleAsIntAgain & 0xFF;

				fileData.push_back(bytes[0]);
				fileData.push_back(bytes[1]);
				fileData.push_back(bytes[2]);
			}
			else if (bitDepth == 32)
			{
				int32_t sampleAsInt;

				if (audioFormat == WavAudioFormat::IEEEFloat)
					sampleAsInt = (int32_t) reinterpret_cast<int32_t&> (samples[channel][i]);
				else // assume PCM
					sampleAsInt = (int32_t)(samples[channel][i] * std::numeric_limits<int32_t>::max());

				addInt32ToFileData(fileData, sampleAsInt, Endianness::LittleEndian);
			}
			else
			{
				assert(false && "Trying to write a file with unsupported bit depth");
				return false;
			}
		}
	}

	return true;
}

basically converted this example to use AudioFIile, it is roughly below but may be missing some peices https://github.com/radiilab/3D-sound/blob/master/openal-example.c

It looks something like this

AudioFile<float> audioFile;
audioFile.load("sounds/TestSound.wav"))

ALuint buffer;
alGenBuffers(1, &buffer);

std::vector<uint8_t> pcmData_uint8;
audioFile.savePCMToBuffer(pcmData_uint8);

alBufferData(buffer, getALFormat(audioFile.getNumChannels(),  audioFile.getBitDepth()),
		/*bufferData*/ reinterpret_cast<ALvoid*>(pcmData_uint8.data()), pcmData_uint8.size(), audioFile.getSampleRate());

OpenAL expects the size in bytes.

This correctly played the sounds for me.

	alListener3f(AL_POSITION, 0, 0, 1.0f);
	alListener3f(AL_VELOCITY, 0, 0, 0);
	alListenerfv(AL_ORIENTATION, listenerOri);

	alGenSources((ALuint)1, &source);
	alSourcef(source, AL_PITCH, 1);
	alSourcef(source, AL_GAIN, 1);
	alSource3f(source, AL_POSITION, 0, 0, 0);
	alSource3f(source, AL_VELOCITY, 0, 0, 0);
	alSourcei(source, AL_LOOPING, AL_FALSE);
	alSourcei(source, AL_BUFFER, buffer);

	alSourcePlay(source);
	alGetSourcei(source, AL_SOURCE_STATE, &source_state);
	while (source_state == AL_PLAYING) {
		alGetSourcei(source, AL_SOURCE_STATE, &source_state);
	}

You can see how to use AudilFile to get the data into a format for OpenAL.