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C# IAudioStream.ReadSamples方法代码示例

本文整理汇总了C#中IAudioStream.ReadSamples方法的典型用法代码示例。如果您正苦于以下问题:C# IAudioStream.ReadSamples方法的具体用法?C# IAudioStream.ReadSamples怎么用?C# IAudioStream.ReadSamples使用的例子?那么恭喜您, 这里精选的方法代码示例或许可以为您提供帮助。您也可以进一步了解该方法所在IAudioStream的用法示例。


在下文中一共展示了IAudioStream.ReadSamples方法的4个代码示例,这些例子默认根据受欢迎程度排序。您可以为喜欢或者感觉有用的代码点赞,您的评价将有助于系统推荐出更棒的C#代码示例。

示例1: Encode


//.........这里部分代码省略.........
            part1->_lastBlockSamples = lbSamples;
            part1->_lastBlockTotal = lbTotal;
            part1->_dataInterval = 0x3800;
            part1->_bitsPerSample = 4;

            //Create one ADPCMInfo for each channel
            int* adpcData = stackalloc int[channels];
            ADPCMInfo** pAdpcm = (ADPCMInfo**)adpcData;
            for (int i = 0; i < channels; i++)
                *(pAdpcm[i] = head->GetChannelInfo(i)) = new ADPCMInfo() { _pad = 0 };

            //Create buffer for each channel
            int* bufferData = stackalloc int[channels];
            short** channelBuffers = (short**)bufferData;
            int bufferSamples = totalSamples + 2; //Add two samples for initial yn values
            for (int i = 0; i < channels; i++)
            {
                channelBuffers[i] = tPtr = (short*)Marshal.AllocHGlobal(bufferSamples * 2); //Two bytes per sample

                //Zero padding samples and initial yn values
                for (int x = 0; x < (loopPadding + 2); x++)
                    *tPtr++ = 0;
            }

            //Fill buffers
            stream.SamplePosition = 0;
            short* sampleBuffer = stackalloc short[channels];

            for (int i = 2; i < bufferSamples; i++)
            {
                if (stream.SamplePosition == stream.LoopEndSample && looped)
                    stream.SamplePosition = stream.LoopStartSample;

                stream.ReadSamples(sampleBuffer, 1);
                for (int x = 0; x < channels; x++)
                    channelBuffers[x][i] = sampleBuffer[x];
            }

            //Calculate coefs
            for (int i = 0; i < channels; i++)
                AudioConverter.CalcCoefs(channelBuffers[i] + 2, totalSamples, (short*)pAdpcm[i], progress);

            //Encode blocks
            byte* dPtr = (byte*)data->Data;
            bshort* pyn = (bshort*)adpc->Data;
            for (int sIndex = 0, bIndex = 1; sIndex < totalSamples; sIndex += 0x3800, bIndex++)
            {
                int blockSamples = Math.Min(totalSamples - sIndex, 0x3800);
                for (int x = 0; x < channels; x++)
                {
                    short* sPtr = channelBuffers[x] + sIndex;

                    //Set block yn values
                    if (bIndex != blocks)
                    {
                        *pyn++ = sPtr[0x3801];
                        *pyn++ = sPtr[0x3800];
                    }

                    //Encode block (include yn in sPtr)
                    AudioConverter.EncodeBlock(sPtr, blockSamples, dPtr, (short*)pAdpcm[x]);

                    //Set initial ps
                    if (bIndex == 1)
                        pAdpcm[x]->_ps = *dPtr;
开发者ID:blahblahblahblah831,项目名称:brawltools2,代码行数:66,代码来源:RSTMConverter.cs

示例2: Encode


//.........这里部分代码省略.........
                    _volFrontRight = 1,
                    _adpcmInfoOffset = waveSize + tableSize + channelSize + i * 0x30
                };
            }

            //Create one ADPCMInfo for each channel
            int* adpcData = stackalloc int[channels];
            ADPCMInfo** pAdpcm = (ADPCMInfo**)adpcData;
            for (int i = 0; i < channels; i++)
                *(pAdpcm[i] = wave->GetADPCMInfo(i)) = new ADPCMInfo();

            //Create buffer for each channel
            int* bufferData = stackalloc int[channels];
            short** channelBuffers = (short**)bufferData;
            int bufferSamples = totalSamples + 2; //Add two samples for initial yn values
            for (int i = 0; i < channels; i++)
            {
                channelBuffers[i] = tPtr = (short*)Marshal.AllocHGlobal(bufferSamples * 2); //Two bytes per sample

                //Zero padding samples and initial yn values
                //for (int x = 0; x < (loopPadding + 2); x++)
                //    *tPtr++ = 0;
            }

            //Fill buffers
            stream.SamplePosition = 0;
            short* sampleBuffer = stackalloc short[channels];

            for (int i = 2; i < bufferSamples; i++)
            {
                //if (stream.SamplePosition == stream.LoopEndSample && looped)
                //    stream.SamplePosition = stream.LoopStartSample;

                stream.ReadSamples(sampleBuffer, 1);
                for (int x = 0; x < channels; x++)
                    channelBuffers[x][i] = sampleBuffer[x];
            }

            //Calculate coefs
            for (int i = 0; i < channels; i++)
                AudioConverter.CalcCoefs(channelBuffers[i] + 2, totalSamples, (short*)pAdpcm[i], progress);

            //Encode blocks
            byte* dPtr = (byte*)wave + entrySize;
            for (int sIndex = 0, bIndex = 1; sIndex < totalSamples; sIndex += samplesPerBlock, bIndex++)
            {
                int blockSamples = Math.Min(totalSamples - sIndex, samplesPerBlock);
                for (int x = 0; x < channels; x++)
                {
                    channelInfo[x]._channelDataOffset = (int)(dPtr - ((byte*)wave + entrySize));
                    short* sPtr = channelBuffers[x] + sIndex;

                    //Set block yn values
                    if (bIndex != blocks)
                    {
                        pAdpcm[x]->_yn1 = sPtr[samplesPerBlock + 1];
                        pAdpcm[x]->_yn2 = sPtr[samplesPerBlock];
                    }

                    //Encode block (include yn in sPtr)
                    AudioConverter.EncodeBlock(sPtr, blockSamples, dPtr, (short*)pAdpcm[x]);

                    //Set initial ps
                    if (bIndex == 1)
                        pAdpcm[x]->_ps = *dPtr;
开发者ID:blahblahblahblah831,项目名称:brawltools2,代码行数:66,代码来源:RSARSoundConverter.cs

示例3: ToFile

        public static void ToFile(IAudioStream source, string path)
        {
            using (FileStream stream = new FileStream(path, FileMode.OpenOrCreate, FileAccess.ReadWrite, FileShare.None, 8, FileOptions.SequentialScan))
            {
                //Estimate size
                int outLen = 44 + (source.Samples * source.Channels * 2);

                //Create file map
                stream.SetLength(outLen);
                using (FileMap map = FileMap.FromStreamInternal(stream, FileMapProtect.ReadWrite, 0, outLen))
                {
                    RIFFHeader* riff = (RIFFHeader*)map.Address;
                    *riff = new RIFFHeader(1, source.Channels, 16, source.Frequency, source.Samples);

                    source.SamplePosition = 0;
                    source.ReadSamples(map.Address + 44, source.Samples);
                }
            }
        }
开发者ID:blahblahblahblah831,项目名称:brawltools2,代码行数:19,代码来源:WAV.cs

示例4: Fill

        public void Fill(IAudioStream stream, bool loop)
        {
            int blockAlign = stream.BitsPerSample * stream.Channels / 8;
            int samplePos = stream.SamplePosition;
            int sampleCount = _sampleLength;
            int samplesRead;
            bool end = false;

            loop = loop && stream.IsLooping;
            int lastSample = loop ? stream.LoopEndSample : stream.Samples;

            VoidPtr blockAddr = _part1Address;
            int blockRemaining = _part1Samples;

            while (sampleCount > 0)
            {
                //Get current block sample count
                int blockSamples = Math.Min(blockRemaining, sampleCount);

                //Fill zeros
                if (end)
                    Memory.Fill(blockAddr, (uint)(blockSamples * blockAlign), 0);
                else
                {
                    //Do we extend within last sample range?
                    if ((samplePos <= lastSample) && (lastSample < (samplePos + blockSamples)))
                    {
                        blockSamples = lastSample - samplePos;
                        end = true;
                    }

                    samplesRead = stream.ReadSamples(blockAddr, blockSamples);
                    samplePos += samplesRead;

                    if (samplesRead < blockSamples)
                    {
                        blockSamples = samplesRead;
                        end = true;
                    }
                    else if (loop && end)
                    {
                        stream.Wrap();
                        if (samplePos == stream.SamplePosition)
                        {
                            samplePos = -1;
                            break;
                        }
                        samplePos = stream.SamplePosition;
                        end = false;
                    }
                }

                blockAddr += blockSamples * blockAlign;
                blockRemaining -= blockSamples;

                //Wrap to second buffer
                if (blockRemaining <= 0)
                {
                    blockAddr = _part2Address;
                    blockRemaining = _part2Samples;
                }

                sampleCount -= blockSamples;
            }
        }
开发者ID:blahblahblahblah831,项目名称:brawltools2,代码行数:65,代码来源:BufferData.cs


注:本文中的IAudioStream.ReadSamples方法示例由纯净天空整理自Github/MSDocs等开源代码及文档管理平台,相关代码片段筛选自各路编程大神贡献的开源项目,源码版权归原作者所有,传播和使用请参考对应项目的License;未经允许,请勿转载。