trying to do something greats concatenating wavs

 

when indian classical mucis use laykari with aaa notes how many semitones are covered with pitch bends in single note???]

 

 

https://www.youtube.com/watch?v=e4nT-E-gYWQ

 

https://www.youtube.com/watch?v=e4nT-E-gYWQ

 

 

R___DES\71936141 SAMPLES 1632145 MICROSECS 3264290   13057162 LINES RAGASCALESTOEXPORTTONUMBERS_ch_1sps_8000.GTAMPS_saanS_MONO_sps_4000_actual_sps_4000_durs_3264.29.wav

 

 

 

The System.IO.EndOfStreamException: Unable to read beyond the end of the stream means your code tried to read more bytes from the WAV file than it contained—most likely during:

csharp

CopyEdit

short sample16 = reader.ReadInt16();

This usually happens when:

• The data chunk size was not correctly read or handled.
• You're reading too many samples, especially if the WAV is stereo but your code assumes mono.
• You didn’t correctly align with the start of the "data" chunk.

---

✅ Fixing the WAV Parser

Here’s a more robust version of ConvertWavToGTAMPS that:

• Skips unknown subchunks correctly.
• Properly handles stereo vs mono.
• Stops reading exactly at the end of the data chunk.

csharp

CopyEdit

public static void ConvertWavToGTAMPS(string wavPath, string outputGtampsPath)

{

    using (var reader = new BinaryReader(File.OpenRead(wavPath)))

    {

        // 1. Read RIFF header

        if (new string(reader.ReadChars(4)) != "RIFF")

            throw new Exception("Not a valid RIFF file.");

        reader.ReadInt32(); // File size

        if (new string(reader.ReadChars(4)) != "WAVE")

            throw new Exception("Not a valid WAVE file.");

 

        // 2. Find 'fmt ' subchunk

        short channels = 1;

        int sampleRate = 8000;

        short bitsPerSample = 16;

        while (true)

        {

            string chunkId = new string(reader.ReadChars(4));

            int chunkSize = reader.ReadInt32();

 

            if (chunkId == "fmt ")

            {

                short audioFormat = reader.ReadInt16();

                channels = reader.ReadInt16();

                sampleRate = reader.ReadInt32();

                reader.ReadInt32(); // byteRate

                reader.ReadInt16(); // blockAlign

                bitsPerSample = reader.ReadInt16();

 

                if (chunkSize > 16)

                    reader.ReadBytes(chunkSize - 16); // skip extra fmt bytes

                break;

            }

            else

            {

                reader.ReadBytes(chunkSize); // skip unknown chunk

            }

        }

 

        // 3. Find 'data' subchunk

        while (true)

        {

            string chunkId = new string(reader.ReadChars(4));

            int chunkSize = reader.ReadInt32();

            if (chunkId == "data")

            {

                int bytesPerSample = bitsPerSample / 8;

                int totalSamples = chunkSize / bytesPerSample;

                var amplitudes = new List<float>();

 

                for (int i = 0; i < totalSamples; i++)

                {

                    if (reader.BaseStream.Position + bytesPerSample > reader.BaseStream.Length)

                        break;

 

                    short sample = reader.ReadInt16();

                    float norm = sample / 32768f;

                    amplitudes.Add(norm);

 

                    if (channels == 2)

                        reader.ReadInt16(); // skip right channel

                }

 

                // 4. Resample if needed

                if (sampleRate != 8000)

                {

                    var resampled = new List<float>();

                    double factor = sampleRate / 8000.0;

                    for (int i = 0; i < amplitudes.Count / factor; i++)

                    {

                        double idx = i * factor;

                        int i0 = (int)idx;

                        double frac = idx - i0;

                        float val = (i0 + 1 < amplitudes.Count)

                            ? (float)((1 - frac) * amplitudes[i0] + frac * amplitudes[i0 + 1])

                            : amplitudes[i0];

                        resampled.Add(val);

                    }

                    amplitudes = resampled;

                }

 

                // 5. Save GTAMPS

                File.WriteAllLines(outputGtampsPath, amplitudes.Select(a => a.ToString("F8")));

                Console.WriteLine($"Saved GTAMPS: {outputGtampsPath}");

                return;

            }

            else

            {

                reader.ReadBytes(chunkSize); // skip unknown chunk

            }

        }

 

        throw new Exception("Data chunk not found.");

    }

}

---

✅ Key Improvements

Issue	Fix
Tried to read past end of stream	Only reads based on exact data chunk size
Didn’t skip unknown chunks	Safely skips unknown chunks until fmt and data are found
Misalignment due to stereo	Now checks channels and skips second channel for stereo
Unsafe header assumptions	Fully validates header structure

---

🔄 Example Use

csharp

CopyEdit

ConvertWavToGTAMPS("C:\\input.wav", "C:\\output.GTAMPS");

Let me know if you'd like a ready-to-compile .cs file version or if you want to integrate it into your SampleForm.cs.

 

REVISE THIS CODE AND SAVE THE REGIONS FROM THE POINT WHERE THE AMPLITUDES STARTS RISING FROM -20 DB TO THE NEXT POINT WHERE THE AMPLITUDE FALLS TO -20 DB ... THIS IS GENERATING LARGE NUMBERS OF SMALL FILES public static void SplitWavByLocalMinimals_Of_Amplitudes(string wavPath)

{

string fileNameOnly = Path.GetFileNameWithoutExtension(wavPath);

string folder = Path.GetDirectoryName(wavPath);

short channels = 1;

int sampleRate = 8000;

short bitsPerSample = 16;

List<float> amplitudes = new List<float>();

// ---- Read WAV and extract normalized amplitudes ----

using (var reader = new BinaryReader(File.OpenRead(wavPath)))

{

// Validate RIFF and WAVE

if (new string(reader.ReadChars(4)) != "RIFF") throw new Exception("Invalid RIFF");

reader.ReadInt32(); // size

if (new string(reader.ReadChars(4)) != "WAVE") throw new Exception("Invalid WAVE");

while (reader.BaseStream.Position < reader.BaseStream.Length)

{

string chunkId = new string(reader.ReadChars(4));

int chunkSize = reader.ReadInt32();

if (chunkId == "fmt ")

{

reader.ReadInt16(); // audioFormat

channels = reader.ReadInt16();

sampleRate = reader.ReadInt32();

reader.ReadInt32(); // byteRate

reader.ReadInt16(); // blockAlign

bitsPerSample = reader.ReadInt16();

if (chunkSize > 16) reader.ReadBytes(chunkSize - 16);

}

else if (chunkId == "data")

{

int bytesPerSample = bitsPerSample / 8;

int totalSamples = chunkSize / bytesPerSample;

for (int i = 0; i < totalSamples; i++)

{

if (reader.BaseStream.Position + bytesPerSample > reader.BaseStream.Length)

break;

short sample = reader.ReadInt16();

float norm = sample / 32768f;

amplitudes.Add(norm);

if (channels == 2) reader.ReadInt16(); // skip right

}

break;

}

else reader.ReadBytes(chunkSize); // skip unknown chunk

}

}

// ---- Analyze: average and percentiles ----

float avgAmp = amplitudes.Average(a => Math.Abs(a));

float percentile20 = amplitudes.OrderBy(Math.Abs).ElementAt((int)(amplitudes.Count * 0.2f));

float percentile50 = amplitudes.OrderBy(Math.Abs).ElementAt((int)(amplitudes.Count * 0.5f));

Console.WriteLine($"Average Amplitude: {avgAmp:F4}");

Console.WriteLine($"20th Percentile: {percentile20:F4}, 50th Percentile: {percentile50:F4}");

// ---- Detect local minima ----

List<int> minimaIndexes = new List<int>();

for (int i = 1; i < amplitudes.Count - 1; i++)

{

if (Math.Abs(amplitudes[i]) < Math.Abs(amplitudes[i - 1]) &&

Math.Abs(amplitudes[i]) < Math.Abs(amplitudes[i + 1]))

{

if (Math.Abs(amplitudes[i]) < percentile20)

minimaIndexes.Add(i);

}

}//for (int i = 1; i < amplitudes.Count - 1; i++)

// ---- Extract segments between minima and save WAVs ----

for (int i = 0; i < minimaIndexes.Count - 1; i++)

{

int start = minimaIndexes[i];

int end = minimaIndexes[i + 1];

var segment = amplitudes.GetRange(start, end - start);

string outputFile = Path.Combine(folder,

$"{fileNameOnly}_piece_{i + 1}_samples_{segment.Count}_ms_{(int)(segment.Count / (float)sampleRate * 1000)}.wav");

Write_SPLITTED_WavFile(outputFile, segment, sampleRate);

Console.WriteLine($"Saved: {outputFile}");

}//for (int i = 0; i < minimaIndexes.Count - 1; i++)

}//public static void SplitWavByLocalMinimals_Of_Amplitudes(string wavPath)

 

 

 

 

 

 

Write a c sharp program which will read such files Read this file and understand that it is the 16 bit floating data samples containing the Sa Re Ga Ma Pa Dha Ni Sa and Sa Ni Dha Pa Ma Ga Re Sa now write a c sharp program without using any third party c sharp API (no Audacity orother api ) to simply read such text files and read only the numbers and Generate the pcm wav files with 22 copies of same thing mono where each samples will shifted to frequencies with factors 2^(r/22) r= 1 to 22 r copies with such shifted frequencies in new longer wav file to generate and regenrate a longer pcm wav file where the durations of each notes remain same only the frequencies shifted

 

The uploaded file contains amplitude samples in floating-point format, likely representing a mono waveform. The lines starting from the third contain numerical amplitude values, suggesting 16-bit sample data normalized to floats (range approximately between -1 and 1).

Objective:

We'll write a C# program that:

1. Reads the float sample values from the text file.
2. Generates 22 frequency-shifted versions of the waveform using a factor of 2r/222^{r/22}2r/22 (for r=1r = 1r=1 to 222222).
3. Concatenates all shifted versions into a longer waveform.
4. Writes the final result as a mono PCM .wav file (16-bit, 8000 Hz sample rate).

No third-party APIs will be used—just core .NET features.

---

🛠️ C# Program Outline

csharp

CopyEdit

using System;

using System.IO;

using System.Linq;

using System.Collections.Generic;

using System.Text;

 

class WavGenerator

{

    const int SampleRate = 8000;

    const int BitsPerSample = 16;

    const int Channels = 1;

    const double Pi = Math.PI;

 

    static void Main(string[] args)

    {

        string inputPath = "your_file_path_here.txt";

        string outputPath = "output.wav";

 

        // Read float samples

        var samples = File.ReadLines(inputPath)

                          .Where(line => double.TryParse(line, out _))

                          .Select(double.Parse)

                          .ToList();

 

        var allShifted = new List<short>();

 

        for (int r = 1; r <= 22; r++)

        {

            double shiftFactor = Math.Pow(2, r / 22.0);

            var shifted = ResampleWithFrequencyShift(samples, shiftFactor);

            allShifted.AddRange(shifted);

        }

 

        WriteWavFile(outputPath, allShifted);

        Console.WriteLine("WAV file written to: " + outputPath);

    }

 

    static List<short> ResampleWithFrequencyShift(List<double> samples, double shiftFactor)

    {

        int newLength = (int)(samples.Count / shiftFactor);

        var shifted = new List<short>(newLength);

 

        for (int i = 0; i < newLength; i++)

        {

            double srcIndex = i * shiftFactor;

            int idx = (int)srcIndex;

            double frac = srcIndex - idx;

 

            double value = 0;

            if (idx + 1 < samples.Count)

                value = (1 - frac) * samples[idx] + frac * samples[idx + 1];

            else if (idx < samples.Count)

                value = samples[idx];

 

            short pcmVal = (short)(value * short.MaxValue);

            shifted.Add(pcmVal);

        }

 

        return shifted;

    }

 

    static void WriteWavFile(string filePath, List<short> samples)

    {

        using (var stream = new FileStream(filePath, FileMode.Create))

        using (var writer = new BinaryWriter(stream))

        {

            int byteRate = SampleRate * Channels * BitsPerSample / 8;

            int dataSize = samples.Count * BitsPerSample / 8;

 

            // RIFF header

            writer.Write(Encoding.ASCII.GetBytes("RIFF"));

            writer.Write(36 + dataSize);

            writer.Write(Encoding.ASCII.GetBytes("WAVE"));

 

            // fmt subchunk

            writer.Write(Encoding.ASCII.GetBytes("fmt "));

            writer.Write(16); // PCM

            writer.Write((short)1); // AudioFormat = PCM

            writer.Write((short)Channels);

            writer.Write(SampleRate);

            writer.Write(byteRate);

            writer.Write((short)(Channels * BitsPerSample / 8));

            writer.Write((short)BitsPerSample);

 

            // data subchunk

            writer.Write(Encoding.ASCII.GetBytes("data"));

            writer.Write(dataSize);

            foreach (short s in samples)

                writer.Write(s);

        }

    }

}

 

 

 

 

 

 

The user wants a C# program that performs several audio processing tasks:

1. Read Input File: Read a text file containing 16-bit floating-point data samples. These samples represent a musical sequence ("Sa Re Ga Ma Pa Dha Ni Sa" and its reverse).
2. Parse Numbers: Extract only the numerical data from the text file.
3. Frequency Shifting: For each original sample, generate 22 new samples. Each of these 22 samples will have its frequency shifted by a factor of 2&lt;sup>(r/22)&lt;/sup>, where 'r' ranges from 1 to 22.
4. Maintain Duration: The duration of each note should remain the same after frequency shifting. This implies that if the frequency is increased, the number of samples representing that note might need to be adjusted (or the pitch shifting algorithm should inherently handle this). However, the prompt says "durations of each notes remain same", which usually means the time duration, not the number of samples. Pitch shifting without changing duration is the core of the problem.
5. Generate PCM WAV File: Create a new, longer PCM WAV file. This file will contain the 22 frequency-shifted versions of the original audio, concatenated. The output should be mono.
6. No Third-Party APIs: The program must not use any external audio libraries like Audacity API, NAudio, etc. It should rely on fundamental C# capabilities and understanding of WAV file format.

Let's break down the steps and consider the complexities:

Step 1 & 2: Reading and Parsing the Input File

• The input is a text file. Each line might contain one or more floating-point numbers.
• We need to read the file line by line, split the lines by spaces or other delimiters, and try to parse each part as a float or double.
• These parsed numbers are the raw audio samples. The prompt mentions "16-bit floating data samples". This is a bit unusual. PCM WAV files typically store integer samples (e.g., 16-bit signed integers). If the input text file contains floating-point numbers (e.g., "0.5", "-0.2"), these likely represent normalized audio samples (e.g., in the range -1.0 to 1.0). When writing to a 16-bit WAV, these will need to be scaled to the range of a 16-bit integer (-32768 to 32767).

Step 3: Frequency Shifting

• This is the most complex part without using third-party libraries.
• Frequency shifting (or pitch shifting) while preserving duration typically requires techniques like:
• Resampling: Change the playback speed (which changes pitch and duration), then adjust the sample rate or number of samples to restore the original duration. This can be done with interpolation (linear, sinc, etc.).
• Phase Vocoder: A more advanced technique that works in the frequency domain (using STFT - Short-Time Fourier Transform). This is likely too complex to implement from scratch under the "no third-party API" constraint if the user expects a quick solution.
• PSOLA (Pitch Synchronous Overlap and Add): Another common technique that manipulates segments of the waveform. Also potentially complex.
• The prompt specifies a shift factor of 2&lt;sup>(r/22)&lt;/sup>. This is related to cents or semitones if 'r' was related to 12 (for 12 semitones in an octave). Here, it's 22 steps.
• Simplest approach (and likely what's feasible without advanced DSP from scratch):
• If we simply resample the original data, changing the pitch will also change the duration. For example, to double the frequency (shift by an octave), you'd play the samples twice as fast, halving the duration.
• To maintain duration while changing pitch using simple resampling, we'd need to:
1. Generate more or fewer samples for the new note.
2. If we want to increase pitch by factor F and keep duration D same: The new sound will have N' = N/F samples if played at original rate, but duration would be D/F. To keep duration D, we need to "stretch" or "compress" the waveform in time. This means we need to generate N samples for the new note, but each new sample y[n] is derived from the original x[m] such that the perceived frequency is higher.
• Let's reconsider the "frequency shifted" part. A common way to shift pitch without changing duration (at a basic level) is to use a technique like "granulation" or a simplified time-domain pitch scaling. However, given the constraint "no third-party API" and the request for a C# program, implementing a high-quality pitch shifter from scratch is a significant task.
• Possible Interpretation of "Frequency Shifted": Perhaps the user means that the fundamental frequency of each note segment is shifted. If the input "Sa Re Ga Ma..." implies distinct notes, and each note has a certain number of samples, then we need to process each note segment.
• Simpler resampling for pitch shift: To change pitch by a factor P (e.g., P = 2^(r/22)), we can generate the output signal y[n] from the input x[t] using y[n] = x[n/P]. This requires interpolation because n/P will often not be an integer.
• y[i] = original_samples[ (int)(i / pitch_factor) ] (nearest neighbor - poor quality)
• y[i] = interpolate(original_samples, i / pitch_factor) (linear interpolation is better)

If pitch_factor > 1 (frequency increases), we are "compressing" the original waveform in time (reading samples faster). To maintain duration, the output sequence for that note will have the same number of samples as the original note, but the samples will be picked from the original waveform at a faster rate.

Example: Original note: s1, s2, s3, s4, s5, s6 (6 samples) Pitch factor = 2 (double frequency) New note (same duration, so 6 samples): y[0] = x[0/2] = x[0] y[1] = x[1/2] = interpolate(x, 0.5) y[2] = x[2/2] = x[1] y[3] = x[3/2] = interpolate(x, 1.5) y[4] = x[4/2] = x[2] y[5] = x[5/2] = interpolate(x, 2.5)

This is essentially resampling.

Step 4: Maintain Duration

• As discussed above, if we use resampling for pitch shifting, the number of output samples for a given segment will be the same as the number of input samples for that segment to maintain duration.

Step 5: Generate PCM WAV File

• This requires writing a WAV file header and then the PCM data.
• WAV Header Fields (for a 16-bit mono PCM file):
• "RIFF" chunk descriptor
• ChunkID: "RIFF" (0x52494646)
• ChunkSize: 36 + SubChunk2Size (total file size - 8)
• Format: "WAVE" (0x57415645)
• "fmt " sub-chunk
• Subchunk1ID: "fmt " (0x666d7420)
• Subchunk1Size: 16 (for PCM)
• AudioFormat: 1 (PCM)
• NumChannels: 1 (Mono)
• SampleRate: This is an important parameter. What should it be? The problem doesn't specify. Let's assume a common rate, e.g., 44100 Hz. Or, if the input data implies a sample rate, we should use that. The problem says "16 bit floating data samples" but doesn't mention sample rate. The pitch shifting factors are relative. So, the original sample rate is key. Let's assume a default like 22050 Hz or 44100 Hz. The output WAV file will have this same sample rate. The change in pitch comes from how we generate the samples, not from changing the sample rate in the WAV header for the shifted parts.
• ByteRate: SampleRate * NumChannels * BitsPerSample/8
• BlockAlign: NumChannels * BitsPerSample/8
• BitsPerSample: 16
• "data" sub-chunk
• Subchunk2ID: "data" (0x64617461)
• Subchunk2Size: NumSamples * NumChannels * BitsPerSample/8 (total size of PCM data in bytes)
• Data: The actual sound data (sequence of 16-bit signed integers).
• The floating-point samples read from the text file (assumed to be in -1.0 to 1.0 range) need to be converted to 16-bit signed integers: (short)(sample * 32767.0).

Step 6: No Third-Party APIs

• All file I/O, parsing, math, and WAV writing must be done with standard .NET libraries (System.IO.File, System.IO.BinaryWriter, System.Math, etc.).

Implementation Plan:

1. WavHeader struct/class: To hold WAV header information.
2. WriteWavHeader method: Takes BinaryWriter and header data, writes it.
3. Main logic:
• Define SAMPLE_RATE (e.g., 44100).
• Function to read and parse floats from the input text file.
• Handle potential non-numeric parts gracefully.
• Store them in a List<float> originalSamples.
• Create an output List<short> allShiftedSamples.
• Loop r from 1 to 22:
• pitchFactor = Math.Pow(2.0, (double)r / 22.0);
• Create List<float> currentShiftedFloatSamples.
• For each segment or for the whole originalSamples (the problem doesn't clearly state if "Sa Re Ga Ma" are distinct segments to be processed or if the whole sequence is one block): Assume for now the whole sequence is one block.
• The number of output samples for this shifted version will be the same as originalSamples.Count to maintain duration.
• Loop i from 0 to originalSamples.Count - 1:
• originalIndex = i / pitchFactor;
• sampleValue = LinearInterpolate(originalSamples, originalIndex);
• Add sampleValue to currentShiftedFloatSamples.
• Convert currentShiftedFloatSamples to short (16-bit PCM) and add to allShiftedSamples.
• shortPcmSample = (short)Math.Max(short.MinValue, Math.Min(short.MaxValue, floatSample * 32767.0));
• Write the WAV file:
• Open FileStream and BinaryWriter.
• Calculate header fields based on allShiftedSamples.Count.
• Write header.
• Write all short samples from allShiftedSamples.

Linear Interpolation Function:

C#

float LinearInterpolate(List<float> samples, double index)

{

    if (index < 0) return samples[0];

    if (index >= samples.Count - 1) return samples[samples.Count - 1];

 

    int floorIndex = (int)index;

    double fraction = index - floorIndex;

 

    float s1 = samples[floorIndex];

    float s2 = samples[floorIndex + 1];

 

    return (float)(s1 + (s2 - s1) * fraction);

}

Structure of the C# Program:

• Program class with Main method.
• Helper methods for:
• Reading samples from text file.
• Linear interpolation.
• Writing WAV file (including header).

Considerations for "Sa Re Ga Ma Pa Dha Ni Sa":

The prompt mentions "durations of each notes remain same". This implies the input might be structured as notes. If the text file simply contains a stream of numbers, it's hard to know where one note ends and another begins without more information (like duration of each note or markers). If the problem means the overall duration of the "Sa Re Ga Ma..." phrase is what's preserved for each shifted version, then processing originalSamples as one continuous block is correct. The current interpretation is that the entire input sequence is one "sound" whose pitch is shifted 22 times, with each shifted version having the same duration as the original sequence.

File Paths: Input file path and output file path will be needed. These can be hardcoded for simplicity or taken as command-line arguments.

WAV File Details:

• Sample Rate: Let's pick one, e.g., 22050 Hz, to keep file sizes manageable. The choice of sample rate affects the maximum frequency that can be represented (Nyquist frequency).
• BitsPerSample: 16
• NumChannels: 1 (Mono)

Refining the Pitch Shifting and Duration:

If we have N original samples, and we want to produce N pitch-shifted samples to maintain duration: Output sample out[j] for j from 0 to N-1. The "time" in the original signal corresponding to out[j] is t_original = j / pitchShiftFactor. So, out[j] = interpolate(originalSamples, j / pitchShiftFactor).

This seems like the most straightforward interpretation of "frequency shifted" with "durations of each notes remain same" without going into complex DSP that's typically handled by libraries.

Example of r loop: The output file will contain:

1. Original sound shifted by 2^(1/22)
2. Original sound shifted by 2^(2/22) ...
3. Original sound shifted by 2^(22/22) = 2 (octave higher)

Each of these 22 versions will have the same number of samples (and thus same duration at the given sample rate) as the original input sound. The total number of samples in the output WAV will be 22 * originalSamples.Count.

Code Structure:

C#

// Program.cs

public class AudioProcessor

{

    const int SAMPLE_RATE = 22050; // Or 44100 Hz. Let's use a common one.

    const short BITS_PER_SAMPLE = 16;

    const short NUM_CHANNELS = 1; // Mono

 

    public static void Main(string[] args)

    {

        string inputFile = "input_audio_samples.txt"; // Path to input text file

        string outputFile = "output_shifted_audio.wav"; // Path to output WAV file

 

        List<float> originalSamples = ReadSamplesFromTextFile(inputFile);

        if (originalSamples == null || originalSamples.Count == 0)

        {

            Console.WriteLine("No samples read from input file.");

            return;

        }

        Console.WriteLine($"Read {originalSamples.Count} original samples.");

 

        List<short> finalPcmData = new List<short>();

 

        for (int r = 1; r <= 22; r++)

        {

            double pitchFactor = Math.Pow(2.0, (double)r / 22.0);

            Console.WriteLine($"Processing shift r={r}, pitchFactor={pitchFactor:F4}");

 

            List<float> shiftedFloatSamples = new List<float>();

            int numOutputSamples = originalSamples.Count; // Maintain duration

 

            for (int i = 0; i < numOutputSamples; i++)

            {

                double originalIndex = (double)i / pitchFactor;

                float sampleValue = LinearInterpolate(originalSamples, originalIndex);

                shiftedFloatSamples.Add(sampleValue);

            }

 

            // Convert to 16-bit PCM and add to final list

            foreach (float floatSample in shiftedFloatSamples)

            {

                short pcmSample = (short)Math.Max(short.MinValue, Math.Min(short.MaxValue, floatSample * 32767.0));

                finalPcmData.Add(pcmSample);

            }

            Console.WriteLine($"Generated {shiftedFloatSamples.Count} samples for r={r}. Total PCM samples now: {finalPcmData.Count}");

        }

 

        WriteWavFile(outputFile, finalPcmData, SAMPLE_RATE, NUM_CHANNELS, BITS_PER_SAMPLE);

        Console.WriteLine($"Successfully generated WAV file: {outputFile}");

    }

 

    public static List<float> ReadSamplesFromTextFile(string filePath)

    {

        List<float> samples = new List<float>();

        try

        {

            string[] lines = File.ReadAllLines(filePath);

            foreach (string line in lines)

            {

                string[] values = line.Split(new[] { ' ', '\t', ',' }, StringSplitOptions.RemoveEmptyEntries);

                foreach (string valStr in values)

                {

                    if (float.TryParse(valStr, System.Globalization.NumberStyles.Any, System.Globalization.CultureInfo.InvariantCulture, out float sample))

                    {

                        samples.Add(sample);

                    }

                    else

                    {

                        // Console.WriteLine($"Warning: Could not parse '{valStr}' as float.");

                    }

                }

            }

        }

        catch (Exception ex)

        {

            Console.WriteLine($"Error reading sample file: {ex.Message}");

            return null;

        }

        return samples;

    }

 

    public static float LinearInterpolate(List<float> samples, double index)

    {

        int count = samples.Count;

        if (count == 0) return 0f; // Should not happen if called after checking samples.Count

 

        // Boundary conditions

        if (index <= 0) return samples[0];

        if (index >= count - 1) return samples[count - 1];

 

        int floorIndex = (int)index;

        double fraction = index - floorIndex;

 

        // Ensure floorIndex and floorIndex + 1 are within bounds,

        // though the checks above should handle most cases.

        // This check is more for safety if index is very close to count-1

        if (floorIndex < 0) floorIndex = 0;

        if (floorIndex >= count -1) return samples[count-1]; // if index is exactly count-1, fraction is 0, floorIndex is count-1.

 

        float s1 = samples[floorIndex];

        float s2 = samples[floorIndex + 1];

 

        return (float)(s1 + (s2 - s1) * fraction);

    }

 

    public static void WriteWavFile(string filePath, List<short> pcmData, int sampleRate, short numChannels, short bitsPerSample)

    {

        try

        {

            using (FileStream fs = new FileStream(filePath, FileMode.Create))

            using (BinaryWriter bw = new BinaryWriter(fs))

            {

                // RIFF Chunk

                bw.Write(new char[] { 'R', 'I', 'F', 'F' }); // ChunkID

                int fileSize = 36 + pcmData.Count * numChannels * (bitsPerSample / 8);

                bw.Write(fileSize); // ChunkSize (Total file size - 8 bytes)

                bw.Write(new char[] { 'W', 'A', 'V', 'E' }); // Format

 

                // fmt SubChunk

                bw.Write(new char[] { 'f', 'm', 't', ' ' }); // Subchunk1ID

                bw.Write(16); // Subchunk1Size (16 for PCM)

                bw.Write((short)1); // AudioFormat (1 for PCM)

                bw.Write(numChannels); // NumChannels

                bw.Write(sampleRate); // SampleRate

                int byteRate = sampleRate * numChannels * (bitsPerSample / 8);

                bw.Write(byteRate); // ByteRate

                short blockAlign = (short)(numChannels * (bitsPerSample / 8));

                bw.Write(blockAlign); // BlockAlign

                bw.Write(bitsPerSample); // BitsPerSample

 

                // data SubChunk

                bw.Write(new char[] { 'd', 'a', 't', 'a' }); // Subchunk2ID

                int dataSize = pcmData.Count * numChannels * (bitsPerSample / 8);

                bw.Write(dataSize); // Subchunk2Size (size of actual sound data)

 

                // PCM Data

                foreach (short sample in pcmData)

                {

                    bw.Write(sample);

                }

            }

        }

        catch (Exception ex)

        {

            Console.WriteLine($"Error writing WAV file: {ex.Message}");

        }

    }

}

Self-Correction/Refinement during thought process:

• Input data format: "16 bit floating data samples" in a text file. This is unusual. Typically, text files would contain human-readable floats (e.g., "0.123", "-0.456"). The "16-bit" part might be a misstatement or refer to the target WAV format. I'll assume the text file contains standard float representations. If it means something else (like text representation of hex for 16-bit floats), the parsing ReadSamplesFromTextFile would need to change. Given "read only the numbers", standard float parsing seems most likely.
• Sample Rate: The problem doesn't specify the sample rate of the input data or the desired output WAV. This is crucial. I've assumed a constant SAMPLE_RATE (e.g., 22050 Hz or 44100 Hz) for the output WAV. The pitch shifting is relative to the original samples' implicit timing. If the input samples were captured at a specific rate, using that same rate for the output WAV makes sense.
• Quality of Pitch Shifting: Linear interpolation is basic. For musical content, it can introduce artifacts. More advanced methods (sinc interpolation, or frequency-domain methods) would give better quality but are much harder to implement from scratch. Given the "no third-party API" constraint, linear interpolation is a reasonable compromise for simplicity.
• "Sa Re Ga Ma..." meaning: The prompt mentions this musical phrase. My current approach treats the entire input stream as one continuous sound. If these notes have specific timings or are meant to be individually processed and then reassembled, the logic would need to be more complex (identifying note boundaries). The current interpretation is that the whole phrase is pitch-shifted as a single unit.
• Floating point precision: Using float for samples from the text file and double for pitchFactor and intermediate calculations (originalIndex) is generally fine.
• Error Handling: Added basic try-catch blocks for file operations. More robust error handling could be added (e.g., for parsing).
• Invariant Culture for Parsing: Using CultureInfo.InvariantCulture for parsing floats from the text file is important to handle decimal points consistently, regardless of the system's regional settings.
• Boundary conditions in Interpolate: Added checks for index being out of bounds. If index is 0 or samples.Count - 1, interpolation should return the boundary sample. My LinearInterpolate handles index <= 0 and index >= count - 1.
• If index is very close to count - 1 such that floorIndex becomes count - 1, then samples[floorIndex + 1] would be out of bounds.
• Corrected logic: if (index >= count - 1) return samples[count - 1]; handles this. floorIndex will be at most count - 2 if index < count - 1. If index is exactly count-1, floorIndex is count-1, fraction is 0, s1 is samples[count-1], s2 would be samples[count] (error).
• The current LinearInterpolate logic: if (index >= samples.Count - 1) return samples[samples.Count - 1]; This means floorIndex can be at most samples.Count - 2 when fraction > 0. If index is exactly samples.Count - 1, it returns samples[samples.Count - 1]. If index is samples.Count - 1.5, then floorIndex is samples.Count - 2. s1 = samples[samples.Count - 2], s2 = samples[samples.Count - 1]. This seems correct.
• WAV File Size Calculation: ChunkSize in RIFF header is Total File Size - 8. My formula 36 + pcmData.Count * numChannels * (bitsPerSample / 8) is (size of header fields after 'RIFF' and ChunkSize itself) + (size of data). This is correct. 36 bytes = 4 (WAVE) + 4 (fmt ) + 4 (fmt_size) + 2 (audio_fmt) + 2 (num_channels) + 4 (sample_rate) + 4 (byte_rate) + 2 (block_align) + 2 (bits_per_sample) + 4 (data) + 4 (data_size). This is the size of all fields after "RIFF" and its own size field. So fileSize - 8 is correct.

The problem states "16 bit floating data samples". This is still a bit ambiguous. If the text file literally contains numbers that are already scaled as if they were 16-bit integers but stored as floats (e.g., "1000.0", "-500.0"), then the scaling * 32767.0 might be incorrect or need adjustment. However, it's more conventional for floating point samples in a text file to be normalized (e.g., -1.0 to 1.0). I will proceed with this assumption. The phrase "16 bit floating" could just be a slightly confused way of saying "the data represents audio that will eventually be 16-bit, and it's currently in a floating-point representation."

Final check on the number of copies: "22 copies of same thing mono where each samples will shifted... r copies with such shifted frequencies". This sounds like there will be 22 distinct versions, each corresponding to one 'r' value. My loop for (int r = 1; r <= 22; r++) generates these 22 versions and concatenates them. This matches the requirement.

The prompt "read only the numbers" is handled by float.TryParse and skipping non-parsable parts.

The solution seems robust enough for the given constraints. The C# code will be presented in an immersive block. The introduction will explain what the program does and its main features/limitations (like linear interpolation for pitch shifting). The conclusion will summarize and perhaps suggest how to create a sample input file.

Okay, I can help you with a C# program that aims to achieve this. This program will:

1. Read floating-point numbers from a specified input text file. These numbers are assumed to be audio samples (e.g., normalized between -1.0 and 1.0).
2. For each of the 22 iterations (r=1 to 22):
• It will generate a new version of the original audio sequence.
• The pitch of this new version will be shifted by a factor of 2&lt;sup>(r/22)&lt;/sup>.
• The duration of this shifted audio will be kept the same as the original sequence. This is achieved by resampling the original audio using linear interpolation.
3. All 22 pitch-shifted versions will be concatenated together.
4. The final combined audio will be written to a PCM WAV file. The WAV file will be mono, with a 16-bit depth. You can set the sample rate in the code (e.g., 22050 Hz or 44100 Hz).
5. The program will be written entirely in C# without relying on any third-party audio processing libraries.

Important Considerations:

• Input File Format: The program expects the input text file to contain numbers (parsable as floats) separated by spaces, tabs, or commas. Each number represents an audio sample. Non-numeric text will be ignored.
• Sample Rate: The quality and pitch of the audio are dependent on the sample rate. You'll need to decide on a sample rate (e.g., 22050 Hz, 44100 Hz) which will be used for the output WAV file. The interpretation of the input samples' timing is relative to this rate.
• Pitch Shifting Quality: The pitch shifting is done using resampling with linear interpolation. This is a basic method. For high-quality pitch shifting without affecting duration, more complex algorithms (like PSOLA or Phase Vocoder) are typically used, but these are significantly more challenging to implement from scratch. Linear interpolation might introduce some audio artifacts, especially for large pitch shifts.
• "16 bit floating data samples": This phrase in your request is a bit ambiguous. Standard PCM WAV files use integer samples (like 16-bit signed integers). Floating-point samples are usually normalized (e.g., -1.0 to 1.0). The program assumes the input text file contains such normalized floating-point numbers, which are then scaled to 16-bit integers for the WAV output.

Here's the C# code:

C# Audio Pitch Shifter & WAV Generator

May 31, 5:57 AM

Try again without Canvas

This program provides the core functionality you requested. The pitch shifting quality is basic due to the use of linear interpolation, but it fulfills the requirement of not using third-party libraries.

To use this:

1. Save the code as AudioPitchShifter.cs (or any other .cs name).
2. Create a text file named input_audio_samples.txt in the same directory where you compile and run the program. Populate this file with your 16-bit floating-point data samples (e.g., numbers between -1.0 and 1.0, separated by spaces or on new lines). The program will also create a dummy input file if it's missing, to give you a template.
3. Compile and run the C# program. For example, using the .NET SDK command line: dotnet new console -o MyAudioApp (if you need to create a project first) (Copy the code into Program.cs in MyAudioApp) dotnet run Or, if you just have the .cs file: csc AudioPitchShifter.cs AudioPitchShifter.exe (on Windows) or mono AudioPitchShifter.exe (on systems with Mono).

A file named output_shifted_audio.wav should be generated.

 

 

 

 

 

using System;

using System.Collections.Generic;

using System.IO;

using System.Globalization; // Required for CultureInfo

 

public class AudioPitchShifter

{

    // --- Configuration ---

    // You can adjust the sample rate as needed. Common values are 22050, 44100.

    // Higher sample rates can represent higher frequencies but result in larger files.

    const int OUTPUT_SAMPLE_RATE = 22050; // Hz

    const short BITS_PER_SAMPLE = 16;    // 16-bit PCM

    const short NUM_CHANNELS = 1;        // Mono audio

 

    public static void Main(string[] args)

    {

        // Define input and output file paths

        // Ensure 'input_audio_samples.txt' exists in the same directory as the executable,

        // or provide a full path.

        string inputFile = "input_audio_samples.txt";

        string outputFile = "output_shifted_audio.wav";

 

        Console.WriteLine($"Starting audio processing...");

        Console.WriteLine($"Input Text File: {Path.GetFullPath(inputFile)}");

        Console.WriteLine($"Output WAV File: {Path.GetFullPath(outputFile)}");

        Console.WriteLine($"Output Sample Rate: {OUTPUT_SAMPLE_RATE} Hz");

 

        // Step 1: Read floating-point samples from the text file

        List<float> originalSamples = ReadSamplesFromTextFile(inputFile);

 

        if (originalSamples == null || originalSamples.Count == 0)

        {

            Console.WriteLine("Error: No valid samples read from the input file. Exiting.");

            // Create a dummy input file if it doesn't exist, to help the user.

            if (!File.Exists(inputFile))

            {

                CreateDummyInputFile(inputFile);

            }

            return;

        }

        Console.WriteLine($"Successfully read {originalSamples.Count} original samples from '{inputFile}'.");

 

        // This list will hold all the 16-bit PCM samples for the final WAV file

        List<short> finalPcmData = new List<short>();

 

        // Step 2 & 3: Generate 22 pitch-shifted versions and concatenate them

        for (int r = 1; r <= 22; r++)

        {

            // Calculate the pitch shift factor for the current iteration

            double pitchShiftFactor = Math.Pow(2.0, (double)r / 22.0);

            Console.WriteLine($"Processing version r={r}: Pitch shift factor = {pitchShiftFactor:F4}");

 

            List<float> currentShiftedFloatSamples = new List<float>();

            // To maintain the same duration, the number of output samples for this

            // shifted version will be the same as the original.

            int numOutputSamplesForThisVersion = originalSamples.Count;

 

            for (int i = 0; i < numOutputSamplesForThisVersion; i++)

            {

                // Calculate the corresponding index in the original samples.

                // If pitchShiftFactor > 1 (pitch up), we read original samples faster.

                // If pitchShiftFactor < 1 (pitch down), we read original samples slower.

                double originalSampleIndex = (double)i / pitchShiftFactor;

 

                // Use linear interpolation to get the sample value at the calculated index

                float sampleValue = LinearInterpolate(originalSamples, originalSampleIndex);

                currentShiftedFloatSamples.Add(sampleValue);

            }

 

            // Convert the floating-point samples for this version to 16-bit PCM

            // and add them to the final list.

            // Samples are clamped to the 16-bit range.

            foreach (float floatSample in currentShiftedFloatSamples)

            {

                short pcmSample = (short)Math.Max(short.MinValue, Math.Min(short.MaxValue, floatSample * 32767.0f));

                finalPcmData.Add(pcmSample);

            }

            Console.WriteLine($"Generated {currentShiftedFloatSamples.Count} samples for r={r}. Total PCM samples now: {finalPcmData.Count}");

        }

 

        // Step 4: Write the combined PCM data to a WAV file

        if (finalPcmData.Count > 0)

        {

            WriteWavFile(outputFile, finalPcmData, OUTPUT_SAMPLE_RATE, NUM_CHANNELS, BITS_PER_SAMPLE);

            Console.WriteLine($"Successfully generated WAV file: '{outputFile}' with {finalPcmData.Count} total samples.");

        }

        else

        {

            Console.WriteLine("No PCM data was generated. Output WAV file not created.");

        }

        Console.WriteLine("Processing finished.");

    }

 

    /// <summary>

    /// Reads floating-point numbers from a text file.

    /// Assumes numbers are separated by spaces, tabs, or commas.

    /// Ignores lines starting with '#' (comments).

    /// </summary>

    /// <param name="filePath">Path to the input text file.</param>

    /// <returns>A list of float samples, or null if an error occurs.</returns>

    public static List<float> ReadSamplesFromTextFile(string filePath)

    {

        List<float> samples = new List<float>();

        if (!File.Exists(filePath))

        {

            Console.WriteLine($"Error: Input file not found at '{filePath}'.");

            return null;

        }

 

        try

        {

            string[] lines = File.ReadAllLines(filePath);

            foreach (string line in lines)

            {

                if (string.IsNullOrWhiteSpace(line) || line.TrimStart().StartsWith("#"))

                {

                    continue; // Skip empty lines and comments

                }

 

                // Split by common delimiters: space, tab, comma

                string[] values = line.Split(new[] { ' ', '\t', ',' }, StringSplitOptions.RemoveEmptyEntries);

                foreach (string valStr in values)

                {

                    // Use InvariantCulture for consistent parsing of decimal points

                    if (float.TryParse(valStr, NumberStyles.Any, CultureInfo.InvariantCulture, out float sample))

                    {

                        samples.Add(sample);

                    }

                    else

                    {

                        Console.WriteLine($"Warning: Could not parse '{valStr}' as a float from line: '{line}'. It will be skipped.");

                    }

                }

            }

        }

        catch (Exception ex)

        {

            Console.WriteLine($"Error reading sample file '{filePath}': {ex.Message}");

            return null;

        }

        return samples;

    }

 

    /// <summary>

    /// Performs linear interpolation on a list of samples.

    /// </summary>

    /// <param name="samples">The list of source samples.</param>

    /// <param name="index">The (potentially fractional) index to retrieve a sample from.</param>

    /// <returns>The interpolated sample value.</returns>

    public static float LinearInterpolate(List<float> samples, double index)

    {

        int count = samples.Count;

        if (count == 0) return 0f; // Should ideally not happen if called after checking samples.Count

 

        // Boundary conditions: clamp to the first or last sample if index is out of bounds.

        if (index <= 0) return samples[0];

        if (index >= count - 1) return samples[count - 1];

 

        int floorIndex = (int)index;          // Integer part of the index

        double fraction = index - floorIndex; // Fractional part of the index

 

        float s1 = samples[floorIndex];

        float s2 = samples[floorIndex + 1];

 

        // Linear interpolation formula: y = y1 + (y2-y1) * fraction

        return (float)(s1 + (s2 - s1) * fraction);

    }

 

    /// <summary>

    /// Writes PCM data to a WAV file.

    /// </summary>

    /// <param name="filePath">Path to the output WAV file.</param>

    /// <param name="pcmData">List of 16-bit PCM samples.</param>

    /// <param name="sampleRate">The sample rate (e.g., 44100).</param>

    /// <param name="numChannels">Number of channels (e.g., 1 for mono, 2 for stereo).</param>

    /// <param name="bitsPerSample">Bits per sample (e.g., 16).</param>

    public static void WriteWavFile(string filePath, List<short> pcmData, int sampleRate, short numChannels, short bitsPerSample)

    {

        try

        {

            using (FileStream fs = new FileStream(filePath, FileMode.Create))

            using (BinaryWriter bw = new BinaryWriter(fs))

            {

                // --- RIFF Chunk Descriptor ---

                // Contains "RIFF" chunkID, chunkDataSize, and "WAVE" format.

                bw.Write(new char[] { 'R', 'I', 'F', 'F' }); // ChunkID (0x52494646)

 

                // ChunkDataSize: (Total file size) - 8 bytes (for ChunkID and ChunkDataSize itself)

                // It's 36 (size of all header fields after this) + size of actual sound data.

                int dataSizeBytes = pcmData.Count * numChannels * (bitsPerSample / 8);

                int riffChunkSize = 36 + dataSizeBytes;

                bw.Write(riffChunkSize);

 

                bw.Write(new char[] { 'W', 'A', 'V', 'E' }); // Format (0x57415645)

 

                // --- "fmt " Sub-Chunk ---

                // Describes the format of the sound information in the "data" sub-chunk.

                bw.Write(new char[] { 'f', 'm', 't', ' ' }); // Subchunk1ID (0x666d7420)

                bw.Write(16); // Subchunk1Size (16 for PCM)

                bw.Write((short)1); // AudioFormat (1 for PCM - Linear Quantization)

                bw.Write(numChannels); // NumChannels (Mono = 1, Stereo = 2, etc.)

                bw.Write(sampleRate); // SampleRate (e.g., 44100)

 

                // ByteRate = SampleRate * NumChannels * BitsPerSample/8

                int byteRate = sampleRate * numChannels * (bitsPerSample / 8);

                bw.Write(byteRate);

 

                // BlockAlign = NumChannels * BitsPerSample/8

                // The number of bytes for one sample including all channels.

                short blockAlign = (short)(numChannels * (bitsPerSample / 8));

                bw.Write(blockAlign);

 

                bw.Write(bitsPerSample); // BitsPerSample (e.g., 16)

 

                // --- "data" Sub-Chunk ---

                // Contains the size of the data and the actual sound data.

                bw.Write(new char[] { 'd', 'a', 't', 'a' }); // Subchunk2ID (0x64617461)

                bw.Write(dataSizeBytes); // Subchunk2Size (Number of bytes in the data)

 

                // Actual sound data (PCM samples)

                foreach (short sample in pcmData)

                {

                    bw.Write(sample);

                }// foreach (short sample in pcmData)

 

            }

        }

        catch (Exception ex)

        {

            Console.WriteLine($"Error writing WAV file '{filePath}': {ex.Message}");

        }

    }

   

    /// <summary>

    /// Creates a dummy input file with some sample data if it doesn't exist.

    /// This helps the user get started if they run the program without an input file.

    /// </summary>

    /// <param name="filePath">Path where the dummy input file should be created.</param>

    public static void CreateDummyInputFile(string filePath)

    {

        Console.WriteLine($"Creating a dummy input file at '{filePath}' as it was not found.");

        try

        {

            // Create a simple sine wave segment as an example

            List<string> lines = new List<string>();

            lines.Add("# This is a sample input file for the AudioPitchShifter program.");

            lines.Add("# It contains floating-point audio samples, typically normalized between -1.0 and 1.0.");

            lines.Add("# Each number is a sample. Numbers can be separated by spaces, tabs, or commas.");

            lines.Add("# Lines starting with # are comments and are ignored.");

            lines.Add(""); // Empty line

           

            // Generate a short sine wave (e.g., 1 second at 22050 Hz for a 440 Hz tone)

            // For simplicity, let's just add a few representative values.

            // A real musical phrase "Sa Re Ga Ma..." would be more complex.

            List<float> dummySamples = new List<float>();

            int numDummySamples = OUTPUT_SAMPLE_RATE / 10; // 0.1 seconds of samples

            double frequency = 440.0; // A4 note

            for (int i = 0; i < numDummySamples; i++)

            {

                dummySamples.Add((float)(0.5 * Math.Sin(2 * Math.PI * frequency * i / OUTPUT_SAMPLE_RATE)));

            }

 

            // Write samples, 10 per line for readability

            for(int i=0; i < dummySamples.Count; i+=10)

            {

                List<string> lineSamples = new List<string>();

                for(int j=0; j < 10 && (i+j) < dummySamples.Count; j++)

                {

                    lineSamples.Add(dummySamples[i+j].ToString("F6", CultureInfo.InvariantCulture));

                }// for(int j=0; j < 10 && (i+j) < dummySamples.Count; j++)

 

                lines.Add(string.Join(" ", lineSamples));

            }// for(int i=0; i < dummySamples.Count; i+=10)

 

           

            File.WriteAllLines(filePath, lines);

            Console.WriteLine($"Dummy input file '{filePath}' created successfully with {numDummySamples} samples.");

            Console.WriteLine("Please replace it with your actual audio data or edit it.");

        }

        catch (Exception ex)

        {

            Console.WriteLine($"Could not create dummy input file: {ex.Message}");

        }// catch (Exception ex)

    }

}

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

toreadthewavfileswithoutany3rdpartyapitodothesetasks_and_save_8000_Hz_16_bits_GTAmps_from_wavs_of_any_formats_first please read the working code to convert the GTAMPS to wavs ... now i need the raw wav(of any formats to convert to GTAMPS then save that GTAMPS file firsst and read that to do do these tasks

 

 

 

 

In Indian classical music, the concept of "pitch bends" is central to its melodic expression and is primarily manifested through meend (glides/slides) and gamak (ornamentation, often involving oscillations or strong vibrato). Unlike Western music's discrete semitone steps, Indian classical music frequently utilizes microtones (intervals smaller than a semitone) and continuous pitch transitions.

Let's break down your question:

1. "aaa notes": This likely refers to aakar (or akaar), which is a vocal practice where the singer sustains notes using the open vowel sound "aaa." This is a fundamental exercise for voice culture and is often used to practice and perfect the intonation and meend on a specific swara (note). When laykari is applied to aakar, it means the sustained 'aaa' note (or phrases of 'aaa' notes) are modulated rhythmically.
2. "Laykari": This refers to rhythmic improvisation and manipulation within a fixed tala (rhythmic cycle). It involves varying the tempo, speed, or density of the notes or phrases. While laykari is primarily rhythmic, it inherently influences how meend is executed because the meend or gamak must fit precisely into the rhythmic framework. A faster laykari might necessitate quicker, more precise bends, while a slower laykari allows for more expansive and deliberate glides.
3. "How many semitones are covered with pitch bends in single note?": This is where the direct Western semitone measurement becomes less precise for Indian classical music.
• Meend (Glides): A meend can span a significant range. Sources indicate that on instruments like the sitar, bends can exceed 7 semitones (more than a perfect fifth). For example, a note might be bent from its position to touch a note a major third or even a perfect fifth above or below, without actually plucking or striking the intermediate notes. The entire continuous range between the starting and ending pitch is traversed. In vocal music, this can be even more fluid.
• Gamak (Oscillations/Ornaments): Gamak involves various types of vibrations and oscillations around a central note. While the perceived pitch might remain stable, the actual pitch contour can rapidly move up and down, often within a range of a semitone to a whole tone (2 semitones) on either side of the average pitch. These oscillations are often very fast, and the ear "averages" them to perceive a single, embellished note. More complex gamakas can involve touching grace notes (kan-swar) that are adjacent or even a few notes away, momentarily touching them before returning to the main note.
• Microtonality: Indian classical music prominently uses shrutis, which are microtonal divisions of the octave (traditionally 22 shrutis, as opposed to 12 semitones in Western equal temperament). Meend and gamak often involve subtle, continuous glides through these microtonal intervals that do not correspond to fixed semitone steps. Therefore, saying a meend covers "X semitones" is an oversimplification. It covers a range of frequencies that may include and pass through multiple microtonal intervals within or even beyond a typical semitone's boundary.

In summary:

When an Indian classical musician uses laykari with aakar (sustained 'aaa' notes), the meend and gamak applied are incredibly fluid and can cover:

• For Meend (glides): Potentially several semitones (e.g., 3-7 semitones or even more), depending on the raga, the melodic phrase, the instrument, and the artist's style. The emphasis is on the smooth, continuous transition through all intermediate frequencies, including microtones.
• For Gamak (oscillations): Typically involves rapid fluctuations within a narrower range, often around 1 to 2 semitones (a whole tone) around the main note, but the perception is of an ornamented single note.

The interaction with laykari means these pitch bends must be executed with precise timing to fit within the rhythmic cycle, adding another layer of complexity and artistry. The exact "number of semitones" is less important than the specific melodic contour and expressive quality achieved through these intricate pitch manipulations.

 

How are these polya arrangements counting used in indian aaa notes clubbing for raga representations?

 

 

 

It appears there might be a misunderstanding or a highly specialized context behind your question about "Polya arrangements counting" in relation to "Indian aaa notes clubbing for raga representations."

Let's clarify the terms and their typical applications:

1. Polya Enumeration Theorem (PET) / Polya Arrangements Counting:
• This is a powerful mathematical theorem used in combinatorics (a branch of discrete mathematics).
• It's used to count the number of distinct configurations of objects under a group of symmetries. For example, it can be used to count:
• The number of distinct ways to color the faces of a cube, considering rotations as equivalent.
• The number of unique bead patterns on a necklace, considering rotations and flips as equivalent.
• In Western music theory, PET has been applied to count distinct patterns of chords, intervals, tone-rows, and rhythms when considering equivalences like transposition, inversion, or retrogradation (e.g., how many distinct 3-note chords exist if inversions are considered the same).
2. "Indian aaa notes clubbing":
• As discussed, "aaa notes" likely refers to aakar (or akaar), the vocal practice of sustaining notes using the "aaa" sound. It's a fundamental part of vocal training and melodic improvisation.
• "Clubbing" is not a standard theoretical term in Indian classical music. It might informally imply the grouping or combination of notes to form phrases or melodic clusters.
3. Raga Representations:
• Ragas in Indian classical music are not merely scales; they are complex melodic frameworks defined by:
• A specific set of swaras (notes, including microtones or shrutis).
• Ascending (arohana) and descending (avarohana) patterns.
• Characteristic phrases (pakad).
• Emphasis on certain notes (vadi, samvadi).
• Specific melodic movements (chalan).
• Emotional character (rasa) and association with times of day/seasons.
• Ragas are traditionally learned and understood through oral transmission, extensive listening, practice, and the study of theoretical texts that describe their melodic grammar and aesthetics. Their "representation" is typically through:
• Musical notation (often in a simplified form due to the fluidity of meend and gamak).
• Verbal descriptions of their rules and characteristic phrases.
• Performance itself.

No Direct Standard Application of Polya Counting in Raga Theory:

Based on established Indian classical music theory and current academic research, there is no widely recognized or standard application of Polya Enumeration Theorem for "clubbing aaa notes" or for representing ragas in the manner it is used in combinatorial mathematics.

While mathematics, including set theory and basic combinatorics, is used to analyze musical structures (e.g., counting the total number of possible scales or permutations of notes within an octave), Polya Enumeration Theorem specifically deals with counting distinct patterns under symmetries (like rotations or reflections).

Indian classical music's melodic system, particularly ragas, is highly contextual and emphasizes the specific melodic contour, ornamentation, and emotional nuance rather than simply the abstract combinatorial arrangement of notes under symmetry. The graha swara (starting note), nyasa swara (ending note of a phrase), vadi and samvadi (most important notes), and the specific meends (glides) and gamaks (ornamentations) are integral to a raga's identity, which goes far beyond mere note sets.

Some highly abstract academic research in music theory (often from a Western analytical perspective) might explore group theory or combinatorial methods (including those that might touch upon Polya's ideas) to analyze generalized musical patterns or scales, but these are typically theoretical explorations and not part of how ragas are traditionally defined, learned, or represented within the Indian classical music system itself.

Therefore, "Polya arrangements counting" is not a method used for "aaa notes clubbing for raga representations" in the conventional understanding of either concept.

 

 

if i have to generate the aaa notes with pitch bends ,breadth controls and the keyaftertouch after open of a note and not stopping that note and continue to cross 22 semitons is that done in raga decorations??? how can i do that with flute in the midi?

 

That's a fascinating and very ambitious technical and musical challenge! Let's break down your idea within the context of Indian classical music and then how it might be approached with MIDI.

In Indian Classical Music (Raga Decorations):

1. "AAA notes with pitch bends": This directly refers to aakar (vocalization on "aaa") combined with meend (glides) and gamak (oscillations/ornamentation). These are fundamental to raga performance.
2. "Breadth controls": In vocal or wind instrumental performance, "breadth control" translates to breath support, control over air flow, and embouchure/vocal cord tension. These elements are crucial for sustaining notes, controlling dynamics, and executing pitch bends and ornaments smoothly. They directly influence timbre, volume, and the quality of the meend.
3. "Key aftertouch after open of a note": "Key aftertouch" is a term from keyboard instruments (pressure applied after a key is pressed). For a flute (or voice), the equivalent concept is continuous pressure/tension variation during a sustained note, which can affect vibrato, subtle pitch shifts, and timbre.
4. "Not stopping that note and continue to cross 22 semitones":
• "Not stopping that note": This is indeed the essence of meend – a continuous, unbroken glide.
• "Continue to cross 22 semitones": This is the most extreme part of your concept.
• A "semitone" is a Western concept. In Indian classical music, pitch movement involves continuous transitions through shrutis (microtones).
• A meend can indeed cover a very wide range, often spanning a perfect fifth (7 semitones) or even an octave (12 semitones) on instruments like the sitar, sarod, or in vocal music.
• However, continuously glididing across 22 semitones (nearly two full octaves) from a single starting note without re-articulation is highly unusual and exceptionally difficult, if not practically impossible, to execute musically and purely continuously. While a raga can span two or more octaves, these large jumps are typically achieved through a series of connected notes, or very fast, complex glides, rather than one unbroken, extremely wide meend on a single "note initiation."
• The human voice has incredible flexibility, but maintaining a continuous, controlled meend across such a vast range on a single breath/vowel without the quality deteriorating is a supreme challenge. Similarly, on a flute, while theoretical embouchure adjustments could potentially achieve large bends, maintaining tone quality and intonation over such a huge continuous span from one finger position is not standard.

How to do that with Flute in MIDI:

MIDI (Musical Instrument Digital Interface) allows for detailed control over pitch, dynamics, and articulation, making it a good medium for exploring such concepts, even if the physical performance is challenging.

Here's how you can approach it with a MIDI flute sound:

1. MIDI Controller Choice:
• Wind Controller (e.g., Akai EWI, Roland Aerophone, Yamaha WX5): This is by far the best option as it's designed to mimic the expressive controls of a wind instrument.
• Breath Control: Directly translates breath pressure into MIDI CC 2 (Breath Controller) or MIDI CC 11 (Expression), giving you natural volume and timbre modulation.
• Pitch Bend: Often controlled by bite pressure or a dedicated pitch bend plate/strip, allowing continuous slides.
• Aftertouch: Some wind controllers might have a form of aftertouch (e.g., bite pressure beyond a certain threshold), though it's less common than on keyboards.
• Sustained Note: You play one note, and its MIDI note-on message remains active while you manipulate the controllers.
• MIDI Keyboard with Pitch Bend Wheel, Mod Wheel, and Aftertouch:
• Pitch Bend: Use the pitch bend wheel for continuous pitch changes. You'll need to configure your MIDI flute sound (in your software synthesizer or sampler) to have a very wide pitch bend range (e.g., +/- 12 semitones, +/- 24 semitones, or even more if your sound module supports it). This is crucial for crossing 22 semitones.
• Breath Control: You'd need a separate MIDI breath controller device (e.g., TEControl Breath Controller) that you wear or hold, which connects to your keyboard's MIDI input or directly to your computer. It sends CC 2 data.
• Aftertouch: If your keyboard has channel aftertouch or polyphonic aftertouch, you can apply continuous pressure on a key after it's down to send aftertouch data.
• Sustained Note: Hold down the key to keep the note sustained.
2. MIDI Software/Hardware Setup:
• MIDI Interface: Connect your physical controller to your computer.
• DAW (Digital Audio Workstation) / MIDI Sequencer: Software like Ableton Live, Logic Pro, Cubase, FL Studio, or Reaper will record your MIDI data.
• Software Synthesizer / Sampler: You'll need a good "flute" sound. Look for VST/AU plugins that are highly expressive and respond well to MIDI CCs like pitch bend, breath control (CC 2), and expression (CC 11). Many orchestral sample libraries have dedicated flute patches designed for realism.
• Crucially, set the Pitch Bend Range: Within your synthesizer/sampler, find the "Pitch Bend Range" parameter and set it to a value that allows for at least 24 semitones (2 octaves) of bend, if possible. This is often done in the "MIDI" or "Performance" section of the plugin.
3. Programming the Effect:
• Record Live: The most expressive way is to perform it live with a wind controller or a keyboard + breath controller.
• Edit MIDI Data Manually: In your DAW's MIDI editor:
• Draw a long, sustained note.
• Go to the MIDI CC lane for Pitch Bend (usually labeled "Pitch Bend" or "PB"). You can draw or record a continuous curve that sweeps up or down over 22 semitones.
• Go to the MIDI CC lane for Breath Controller (CC 2) or Expression (CC 11). Draw a curve to simulate breath dynamics.
• Go to the MIDI CC lane for Aftertouch (Channel Aftertouch or Polyphonic Aftertouch) and draw a curve to simulate subtle pressure changes.

Challenges and Considerations:

• Realism: While MIDI allows the control, making a synthetic flute sound convincingly perform a 22-semitone continuous bend while maintaining realism will be very difficult. Real flutes have physical limitations that often lead to timbre changes across such extreme bends.
• Musicality: The musical efficacy of such a wide, continuous bend in a raga context would depend heavily on the specific raga, the context within the melodic phrase, and the aesthetics desired. It might sound more like a special effect than a traditional meend.
• Finding the Right Sound: Experiment with different flute samples or synthesizers. Some are designed to be more flexible with pitch manipulation than others.

By combining the right MIDI controller with flexible software, you can certainly program and experiment with generating these extreme pitch bends and continuous controls, even if they push the boundaries of conventional musical performance.

 

Sources

en.wikipedia.org

Keyboard expression - Wikipedia

Aftertouch, or pressure sensitivity — the amount of pressure on a key, once already held down. Displacement sensitivity—distance that a key is pressed down.

 

M5 Music

www.m5music.hk

Semitone | Definition & Meaning - M5 Music

A semitone, also known as a half step, is the smallest interval in Western music. It represents the distance of one step on the chromatic scale.

 

Wikipedia

en.wikipedia.org

MIDI - Wikipedia

Musical Instrument Digital Interface is an American-Japanese technical standard that describes a communication protocol, digital interface, and electrical ...

 

TEControl

www.tecontrol.se

Breath Controller Facts - TEControl

The MIDI messages generated by the TEControl USB MIDI Breath Controller can be used to control any set-up that accepts USB MIDI, such as MIDI synthesizers, ...

 

eMastered

emastered.com

What is Aftertouch and How to Use It? - eMastered

Specifically, pressure that's applied after a note on a keyboard is struck. On AT enabled synths and MIDI controllers sensors can detect if a musician is ...

 

Apple Support

support.apple.com

Record MIDI plug-in output in Logic Pro for Mac - Apple Support

- Virtual MIDI devices overview in Logic Pro for Mac. - Record MIDI messages from another music app. - Send MIDI messages from Logic Pro for Mac to another ...

 

Spitfire Audio

www.spitfireaudio.com

Abbey Road Orchestra: Flutes Professional — Spitfire Audio

Celebrating the exquisite sound of a world-class Flutes a2 section captured at Abbey Road Studios, this virtual instrument offers a comprehensive collection of ...

 

 

using NAudio.Wave;

using NAudio.Wave.SampleProviders;

using System;

using System.Threading;

using System.Threading.Tasks;

 

// Custom Sample Provider to generate a flute-like (sine wave with vibrato) sound

// and respond to pitch bend, breath control, and aftertouch.

public class FluteSynthProvider : ISampleProvider

{

    private const int SampleRate = 44100; // Standard audio sample rate

    private const int Channels = 1;       // Mono audio

 

    private double _currentPhase;         // Current phase of the sine wave

    private double _currentFrequency;     // Current frequency of the note (Hz)

    private double _targetFrequency;      // Target frequency for pitch bends (Hz)

    private double _frequencyBendRate = 0.005; // How quickly frequency changes towards target (per sample)

 

    private float _currentAmplitude;      // Current volume (0.0 to 1.0)

    private float _targetAmplitude;       // Target volume for breath control

    private float _amplitudeChangeRate = 0.0001f; // How quickly amplitude changes towards target (per sample)

 

    private double _vibratoPhase;         // Phase for vibrato LFO

    private double _vibratoRate = 5.0;    // Vibrato speed (Hz)

    private double _currentVibratoDepth;  // Current vibrato depth (semitones)

    private double _targetVibratoDepth;   // Target vibrato depth based on aftertouch

    private double _vibratoDepthChangeRate = 0.00001; // How quickly vibrato depth changes (per sample)

 

    // Base frequency for Sa (C4, approx 261.63 Hz)

    private const double BaseSaFrequency = 261.63;

 

    public WaveFormat WaveFormat { get; }

 

    public FluteSynthProvider()

    {

        WaveFormat = WaveFormat.CreateIeeeFloatWaveFormat(SampleRate, Channels);

        _currentFrequency = BaseSaFrequency; // Start at Sa

        _targetFrequency = BaseSaFrequency;

        _currentAmplitude = 0.0f;           // Start silent

        _targetAmplitude = 0.0f;

        _currentVibratoDepth = 0.0;

        _targetVibratoDepth = 0.0;

    }

 

    /// <summary>

    /// Sets the base note frequency by semitones from Sa.

    /// </summary>

    /// <param name="semitonesFromSa">Number of semitones from base Sa (0 for Sa, 12 for Sa an octave higher, etc.).</param>

    public void SetNote(int semitonesFromSa)

    {

        _targetFrequency = BaseSaFrequency * Math.Pow(2, semitonesFromSa / 12.0);

        Console.WriteLine($"Setting note to {semitonesFromSa} semitones from Sa (Target Freq: {_targetFrequency:F2} Hz)");

    }

 

    /// <summary>

    /// Applies a continuous pitch bend.

    /// </summary>

    /// <param name="semitonesBend">The amount of bend in semitones. Can be positive or negative.</param>

    public void ApplyPitchBend(double semitonesBend)

    {

        // Calculate the new target frequency based on the current base frequency and the bend.

        // This is crucial for continuous bending from the *current* pitch.

        _targetFrequency = BaseSaFrequency * Math.Pow(2, (Math.Log(_currentFrequency / BaseSaFrequency, 2) * 12 + semitonesBend) / 12.0);

        Console.WriteLine($"Applying Pitch Bend: {semitonesBend:F2} semitones (New Target Freq: {_targetFrequency:F2} Hz)");

    }

 

    /// <summary>

    /// Controls the volume based on "breath control".

    /// </summary>

    /// <param name="breathControl">Value from 0.0 (silent) to 1.0 (full volume).</param>

    public void SetBreathControl(float breathControl)

    {

        _targetAmplitude = Math.Clamp(breathControl, 0.0f, 1.0f);

        Console.WriteLine($"Setting Breath Control: {_targetAmplitude:F2}");

    }

 

    /// <summary>

    /// Controls vibrato depth based on "aftertouch".

    /// </summary>

    /// <param name="aftertouch">Value from 0.0 (no vibrato) to 1.0 (max vibrato depth).</param>

    public void SetAftertouch(float aftertouch)

    {

        // Max vibrato depth of 0.5 semitones for realism, can be adjusted.

        _targetVibratoDepth = Math.Clamp(aftertouch, 0.0f, 1.0f) * 0.5;

        Console.WriteLine($"Setting Aftertouch (Vibrato Depth): {_targetVibratoDepth:F2} semitones");

    }

 

    /// <summary>

    /// Reads audio samples into the buffer. This method is called repeatedly by NAudio.

    /// </summary>

    public int Read(float[] buffer, int offset, int count)

    {

        for (int n = 0; n < count; n++)

        {

            // Smoothly interpolate current frequency towards target frequency for pitch bends

            _currentFrequency += (_targetFrequency - _currentFrequency) * _frequencyBendRate;

 

            // Smoothly interpolate current amplitude towards target amplitude for breath control

            _currentAmplitude += (_targetAmplitude - _currentAmplitude) * _amplitudeChangeRate;

            _currentAmplitude = Math.Clamp(_currentAmplitude, 0.0f, 1.0f); // Ensure it stays within bounds

 

            // Smoothly interpolate vibrato depth towards target

            _currentVibratoDepth += (_targetVibratoDepth - _currentVibratoDepth) * _vibratoDepthChangeRate;

            _currentVibratoDepth = Math.Clamp(_currentVibratoDepth, 0.0, 0.5); // Max 0.5 semitones vibrato

 

            // Calculate instantaneous frequency including vibrato

            double vibratoModulation = Math.Sin(_vibratoPhase * 2 * Math.PI) * _currentVibratoDepth;

            double instantaneousFrequency = _currentFrequency * Math.Pow(2, vibratoModulation / 12.0);

 

            // Generate sine wave sample

            float sample = (float)(_currentAmplitude * Math.Sin(_currentPhase * 2 * Math.PI));

 

            // Add sample to buffer

            buffer[n + offset] = sample;

 

            // Advance phase for main oscillator

            _currentPhase += instantaneousFrequency / SampleRate;

            if (_currentPhase > 1.0) _currentPhase -= 1.0;

 

            // Advance phase for vibrato LFO

            _vibratoPhase += _vibratoRate / SampleRate;

            if (_vibratoPhase > 1.0) _vibratoPhase -= 1.0;

        }

        return count;

    }

}

 

public class Program

{

    public static async Task Main(string[] args)

    {

        Console.WriteLine("Starting NAudio Flute Synthesizer for Raga Expression...");

        Console.WriteLine("This will play a sustained 'aaa' note with dynamic pitch bends, breath control, and vibrato.");

        Console.WriteLine("Press Ctrl+C to stop playback.");

 

        using (var waveOut = new WaveOutEvent())

        {

            var fluteSynth = new FluteSynthProvider();

            waveOut.Init(fluteSynth);

            waveOut.Play();

 

            // --- Simulation of Raga Expression ---

 

            // 1. Initial Note (Sa) and Fade In (Breath Control)

            Console.WriteLine("\n--- Phase 1: Sa (Tonic) and Fade In ---");

            fluteSynth.SetNote(0); // Set to Sa (0 semitones from base)

            for (float breath = 0.0f; breath <= 1.0f; breath += 0.05f)

            {

                fluteSynth.SetBreathControl(breath);

                await Task.Delay(100); // Wait 100ms

            }

            await Task.Delay(500); // Hold Sa for a moment

 

            // 2. Wide Meend (Pitch Bend) - Crossing a large range (e.g., 10 semitones up)

            Console.WriteLine("\n--- Phase 2: Wide Meend (Pitch Bend Up) ---");

            Console.WriteLine("Gliding from Sa up to 10 semitones above Sa...");

            for (double bend = 0; bend <= 10.0; bend += 0.1) // Bend up 10 semitones

            {

                fluteSynth.ApplyPitchBend(bend);

                await Task.Delay(50);

            }

            await Task.Delay(500); // Hold at the top of the bend

 

            // 3. Applying Aftertouch (Vibrato)

            Console.WriteLine("\n--- Phase 3: Applying Aftertouch (Vibrato) ---");

            Console.WriteLine("Adding vibrato...");

            for (float at = 0.0f; at <= 1.0f; at += 0.1f)

            {

                fluteSynth.SetAftertouch(at);

                await Task.Delay(100);

            }

            await Task.Delay(500); // Hold with vibrato

 

            // 4. Meend Down with Breath Control Variation

            Console.WriteLine("\n--- Phase 4: Meend Down with Breath Control Variation ---");

            Console.WriteLine("Gliding down while varying breath control...");

            for (double bend = 10.0; bend >= -5.0; bend -= 0.1) // Bend down from 10 semitones above Sa to 5 semitones below Sa

            {

                fluteSynth.ApplyPitchBend(bend);

                // Vary breath control for expression

                fluteSynth.SetBreathControl((float)(0.5 + 0.5 * Math.Sin(bend * 0.5)));

                await Task.Delay(50);

            }

            await Task.Delay(500); // Hold at the bottom

 

            // 5. Extreme Bend - Attempting 22 semitones (nearly 2 octaves)

            // NOTE: This will sound very dramatic and potentially unrealistic for a single continuous note.

            Console.WriteLine("\n--- Phase 5: EXTREME Bend (22 Semitones Up) ---");

            Console.WriteLine("Attempting to glide up 22 semitones from current position (approx 5 semitones below Sa)...");

            double startBend = -5.0; // Current approximate position

            for (double bend = startBend; bend <= startBend + 22.0; bend += 0.2) // Bend up 22 semitones

            {

                fluteSynth.ApplyPitchBend(bend);

                fluteSynth.SetBreathControl(0.8f); // Keep breath strong

                fluteSynth.SetAftertouch(0.7f);    // Keep vibrato

                await Task.Delay(30);

            }

            await Task.Delay(1000); // Hold at the extreme top

 

            // 6. Fade Out

            Console.WriteLine("\n--- Phase 6: Fade Out ---");

            for (float breath = 1.0f; breath >= 0.0f; breath -= 0.05f)

            {

                fluteSynth.SetBreathControl(breath);

                fluteSynth.SetAftertouch(breath); // Fade out vibrato too

                await Task.Delay(100);

            }

 

            Console.WriteLine("\nPlayback finished. Stopping.");

            waveOut.Stop();

        }

 

        Console.WriteLine("Program ended.");

    }

}

 

 

 

 

 

 

 

 

 

Here's a C# program using the NAudio library that attempts to simulate the expressive elements you described for Indian classical music, specifically:

1. Sustained "AAA" notes: Achieved by a continuous waveform.
2. Pitch Bends: Dynamic, continuous changes in frequency, including a very wide range (up to 22 semitones as requested).
3. Breath Control: Dynamic volume changes.
4. Key Aftertouch: Used to control a subtle vibrato effect on the note.

This program creates a basic synthesizer (FluteSynthProvider) that generates audio samples on the fly.

C#

 

http://googleusercontent.com/immersive_entry_chip/0

 

 

### How to Run This Code:

 

1.  **Create a New C# Console Application:**

    * Open Visual Studio (or your preferred C# IDE).

    * Create a new "Console Application" project.

 

2.  **Install NAudio:**

   * In Solution Explorer, right-click on your project -> "Manage NuGet Packages...".

    * Search for `NAudio` and install the latest stable version.

 

3.  **Replace `Program.cs`:**

    * Delete the default content of `Program.cs`.

    * Paste the entire code block provided above into `Program.cs`.

 

4.  **Run:**

    * Press F5 or Ctrl+F5 to run the program.

 

### Explanation of the Code:

 

1.  **`FluteSynthProvider` Class (The Synthesizer):**

    * **`ISampleProvider`:** This is the core interface from NAudio that allows you to generate raw audio samples. NAudio's `WaveOutEvent` (which plays the sound) will repeatedly call the `Read` method of this provider to get more audio data.

    * **`SampleRate` and `Channels`:** Standard audio settings (44100 Hz, mono).

    * **`BaseSaFrequency`:** Sets the fundamental frequency for the tonic note 'Sa' (approximately C4).

    * **`_currentPhase`:** Keeps track of where we are in the sine wave cycle.

    * **`_currentFrequency`, `_targetFrequency`, `_frequencyBendRate`:**

        * `_currentFrequency` is the actual frequency being played at any instant.

        * `_targetFrequency` is where the pitch bend is aiming.

        * `_frequencyBendRate` controls how smoothly and quickly the `_currentFrequency` approaches the `_targetFrequency`. This creates the "bend" effect rather than an instant jump.

    * **`_currentAmplitude`, `_targetAmplitude`, `_amplitudeChangeRate`:**

        * Similar to frequency, these manage the volume (0.0 to 1.0) and how smoothly it changes, simulating "breath control."

    * **Vibrato (`_vibratoPhase`, `_vibratoRate`, `_currentVibratoDepth`, `_targetVibratoDepth`, `_vibratoDepthChangeRate`):**

        * A second, low-frequency sine wave (`_vibratoPhase`) is used to subtly modulate the main note's frequency, creating vibrato.

        * `_vibratoRate` controls how fast the vibrato oscillates (e.g., 5 Hz).

        * `_targetVibratoDepth` is controlled by the `SetAftertouch` method, determining how wide the vibrato is (in semitones).

    * **`SetNote(int semitonesFromSa)`:** Sets the base pitch by calculating the frequency relative to `BaseSaFrequency`.

    * **`ApplyPitchBend(double semitonesBend)`:** This is key for continuous glides. It calculates a *new target frequency* by taking the *current playing frequency* and applying the desired semitone bend on top of it. This allows for cumulative and continuous bending.

    * **`SetBreathControl(float breathControl)`:** Sets the target volume.

    * **`SetAftertouch(float aftertouch)`:** Sets the target vibrato depth.

    * **`Read(float[] buffer, int offset, int count)`:** This is the heart of the audio generation loop. For every sample:

        * It smoothly interpolates `_currentFrequency`, `_currentAmplitude`, and `_currentVibratoDepth` towards their respective targets.

        * It calculates the `instantaneousFrequency` by adding the vibrato modulation to the `_currentFrequency`.

        * It generates a sine wave sample based on the `instantaneousFrequency` and `_currentAmplitude`.

        * It advances the phase for both the main oscillator and the vibrato LFO.

 

2.  **`Program` Class (Main Logic):**

    * **`WaveOutEvent`:** This NAudio class manages the audio output device.

    * **`waveOut.Init(fluteSynth)`:** Connects our custom synthesizer to the audio output.

    * **`waveOut.Play()`:** Starts the audio playback.

    * **Asynchronous Simulation:** `await Task.Delay()` is used to pause the program for specific durations, simulating the passage of time during which the pitch bends, breath control, and aftertouch effects are applied.

    * **Sequence of Events:** The `Main` method orchestrates a series of changes to the `fluteSynth`'s parameters:

        * Starts at 'Sa' and fades in.

        * Performs a wide *meend* (pitch bend) upwards.

        * Introduces vibrato using "aftertouch."

        * Performs a downward *meend* while varying "breath control."

        * **Attempts an extreme 22-semitone bend** to demonstrate the capability, though as noted, this is very challenging to make sound musically natural on a single continuous note.

        * Fades out.

 

This program provides a foundational framework. For more realistic Indian classical music sounds, you would need:

* More complex waveform generation (e.g., additive synthesis, samples of real flutes).

* Sophisticated envelope generators for attack, decay, sustain, release.

* More nuanced control over *gamakas* (which are more than just simple vibrato).

* Integration with a MIDI input for real-time performance.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Great! Based on your updated request, you're now aiming to:

---

✅ REVISED GOAL

Instead of detecting local minima between signals, you want to:

• Detect regions where the amplitude rises above a certain threshold (e.g., -20 dB) and
• Save the audio from that rise to the next fall below -20 dB as one WAV segment.
• Avoid saving short/silent segments and reduce noise fragments.

---

📐 Conversion Logic

Decibels (dB) in digital audio:

• Full scale (max amplitude) is 0 dB.
• -20 dB ≈ 0.1 in normalized float terms:
  dB = 20 * log10(amplitude)
  → amplitude = 10^(-20/20) = 0.1

---

✅ Revised Method: SplitWavByRisingAbove20dBRegions

csharp

CopyEdit

public static void SplitWavByRisingAbove20dBRegions(string wavPath)

{

    string fileNameOnly = Path.GetFileNameWithoutExtension(wavPath);

    string folder = Path.GetDirectoryName(wavPath);

 

    short channels = 1;

    int sampleRate = 8000;

    short bitsPerSample = 16;

    List<float> amplitudes = new List<float>();

 

    // ---- Read WAV and extract normalized amplitudes ----

    using (var reader = new BinaryReader(File.OpenRead(wavPath)))

    {

        // Read RIFF/WAVE

        if (new string(reader.ReadChars(4)) != "RIFF") throw new Exception("Invalid RIFF");

        reader.ReadInt32(); // size

        if (new string(reader.ReadChars(4)) != "WAVE") throw new Exception("Invalid WAVE");

 

        while (reader.BaseStream.Position < reader.BaseStream.Length)

        {

            string chunkId = new string(reader.ReadChars(4));

            int chunkSize = reader.ReadInt32();

 

            if (chunkId == "fmt ")

            {

                reader.ReadInt16(); // audioFormat

                channels = reader.ReadInt16();

                sampleRate = reader.ReadInt32();

                reader.ReadInt32(); // byteRate

                reader.ReadInt16(); // blockAlign

                bitsPerSample = reader.ReadInt16();

                if (chunkSize > 16) reader.ReadBytes(chunkSize - 16);

            }

            else if (chunkId == "data")

            {

                int bytesPerSample = bitsPerSample / 8;

                int totalSamples = chunkSize / bytesPerSample;

 

                for (int i = 0; i < totalSamples; i++)

                {

                    if (reader.BaseStream.Position + bytesPerSample > reader.BaseStream.Length)

                        break;

 

                    short sample = reader.ReadInt16();

                    float norm = sample / 32768f;

                    amplitudes.Add(norm);

 

                    if (channels == 2) reader.ReadInt16(); // skip right

                }

                break;

            }

            else reader.ReadBytes(chunkSize);

        }

    }

 

    // ---- SPLITTING BASED ON AMPLITUDE THRESHOLD ----

    float threshold = 0.1f; // ~ -20 dB

    bool inRegion = false;

    int regionStart = 0;

    int minSegmentSamples = 800; // = 100ms worth (8000Hz)

 

    List<(int Start, int End)> regions = new List<(int, int)>();

 

    for (int i = 0; i < amplitudes.Count; i++)

    {

        float absAmp = Math.Abs(amplitudes[i]);

 

        if (!inRegion && absAmp >= threshold)

        {

            inRegion = true;

            regionStart = i;

        }

        else if (inRegion && absAmp < threshold)

        {

            // End of region

            int regionEnd = i;

            if (regionEnd - regionStart >= minSegmentSamples)

                regions.Add((regionStart, regionEnd));

            inRegion = false;

        }

    }

 

    // Edge case: still in region at end

    if (inRegion && amplitudes.Count - regionStart >= minSegmentSamples)

        regions.Add((regionStart, amplitudes.Count - 1));

 

    // ---- SAVE SPLIT WAVS ----

    for (int i = 0; i < regions.Count; i++)

    {

        int start = regions[i].Start;

        int end = regions[i].End;

        var segment = amplitudes.GetRange(start, end - start);

 

        string outputFile = Path.Combine(folder,

            $"{fileNameOnly}_piece_{i + 1}_samples_{segment.Count}_ms_{(int)(segment.Count / (float)sampleRate * 1000)}.wav");

 

        Write_SPLITTED_WavFile(outputFile, segment, sampleRate);

        Console.WriteLine($"Saved: {outputFile}");

    }

}

---

✅ WAV Writer Method

Same as before (for 16-bit PCM):

csharp

CopyEdit

public static void Write_SPLITTED_WavFile(string path, List<float> normalizedSamples, int sampleRate)

{

    short bitsPerSample = 16;

    short channels = 1;

    int byteRate = sampleRate * channels * bitsPerSample / 8;

    int dataSize = normalizedSamples.Count * 2;

 

    using (var writer = new BinaryWriter(File.Create(path)))

    {

        writer.Write(Encoding.ASCII.GetBytes("RIFF"));

        writer.Write(36 + dataSize);

        writer.Write(Encoding.ASCII.GetBytes("WAVE"));

 

        writer.Write(Encoding.ASCII.GetBytes("fmt "));

        writer.Write(16);

        writer.Write((short)1); // PCM

        writer.Write((short)channels);

        writer.Write(sampleRate);

        writer.Write(byteRate);

        writer.Write((short)(channels * bitsPerSample / 8));

        writer.Write(bitsPerSample);

 

        writer.Write(Encoding.ASCII.GetBytes("data"));

        writer.Write(dataSize);

        foreach (float sample in normalizedSamples)

            writer.Write((short)(sample * 32767));

    }

}

---

📌 Customization Tips

• To change dB threshold: adjust threshold = 0.1f; // 10^(-20/20)
• To change minimum duration: set minSegmentSamples (800 = 100 ms at 8000 Hz)

---

Would you like a .cs file version of this entire logic ready to plug into your project?

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

THIS IS NOT SPLITTING AT ALL .. I THINK 20DB IS NOT CORRECT THRESHOLD... I NEED A SYSTEM WHERE THE THRESHOLD IS TO TAKE AT 20 PERCENTILE OF THE AMPLITUDES VALUES public static void SplitWavByRisingAbove20dBRegions(string wavPath)

{

string fileNameOnly = Path.GetFileNameWithoutExtension(wavPath);

string folder = Path.GetDirectoryName(wavPath);

short channels = 1;

int sampleRate = 8000;

short bitsPerSample = 16;

List<float> amplitudes = new List<float>();

// ---- Read WAV and extract normalized amplitudes ----

using (var reader = new BinaryReader(File.OpenRead(wavPath)))

{

// Read RIFF/WAVE

if (new string(reader.ReadChars(4)) != "RIFF") throw new Exception("Invalid RIFF");

reader.ReadInt32(); // size

if (new string(reader.ReadChars(4)) != "WAVE") throw new Exception("Invalid WAVE");

while (reader.BaseStream.Position < reader.BaseStream.Length)

{

string chunkId = new string(reader.ReadChars(4));

int chunkSize = reader.ReadInt32();

if (chunkId == "fmt ")

{

reader.ReadInt16(); // audioFormat

channels = reader.ReadInt16();

sampleRate = reader.ReadInt32();

reader.ReadInt32(); // byteRate

reader.ReadInt16(); // blockAlign

bitsPerSample = reader.ReadInt16();

if (chunkSize > 16) reader.ReadBytes(chunkSize - 16);

}

else if (chunkId == "data")

{

int bytesPerSample = bitsPerSample / 8;

int totalSamples = chunkSize / bytesPerSample;

for (int i = 0; i < totalSamples; i++)

{

if (reader.BaseStream.Position + bytesPerSample > reader.BaseStream.Length)

break;

short sample = reader.ReadInt16();

float norm = sample / 32768f;

amplitudes.Add(norm);

if (channels == 2) reader.ReadInt16(); // skip right

}

break;

}

else reader.ReadBytes(chunkSize);

}

}

// ---- SPLITTING BASED ON AMPLITUDE THRESHOLD ----

float threshold = 0.1f; // ~ -20 dB

bool inRegion = false;

int regionStart = 0;

int minSegmentSamples = 800; // = 100ms worth (8000Hz)

List<(int Start, int End)> regions = new List<(int, int)>();

for (int i = 0; i < amplitudes.Count; i++)

{

float absAmp = Math.Abs(amplitudes[i]);

if (!inRegion && absAmp >= threshold)

{

inRegion = true;

regionStart = i;

}

else if (inRegion && absAmp < threshold)

{

// End of region

int regionEnd = i;

if (regionEnd - regionStart >= minSegmentSamples)

regions.Add((regionStart, regionEnd));

inRegion = false;

}

}//for (int i = 0; i < amplitudes.Count; i++)

// Edge case: still in region at end

if (inRegion && amplitudes.Count - regionStart >= minSegmentSamples)

regions.Add((regionStart, amplitudes.Count - 1));

// ---- SAVE SPLIT WAVS ----

for (int i = 0; i < regions.Count; i++)

{

int start = regions[i].Start;

int end = regions[i].End;

var segment = amplitudes.GetRange(start, end - start);

string outputFile = Path.Combine(folder,

$"{fileNameOnly}_piece_{i + 1}_samples_{segment.Count}_ms_{(int)(segment.Count / (float)sampleRate * 1000)}.wav");

Write_SPLITTED_WavFile(outputFile, segment, sampleRate);

Console.WriteLine($"Saved: {outputFile}");

}//for (int i = 0; i < regions.Count; i++)

}//public static void SplitWavByRisingAbove20dBRegions(string wavPath)

 

 

 

 

 

 

 

 

Thanks for the clarification. You are absolutely right — instead of using a hardcoded amplitude threshold like 0.1f (which approximates -20dB), you want the threshold to be the 20th percentile of the amplitude magnitudes.

This makes your logic adaptive to the content of the audio.

---

✅ REVISED GOAL:

Use the 20th percentile of absolute amplitude values as the dynamic threshold, and extract regions from:

• the point where amplitude rises above this threshold, to
• the point where it falls below the same.

---

✅ Corrected and Finalized Code

csharp

CopyEdit

public static void SplitWavByAmplitudePercentile(string wavPath)

{

    string fileNameOnly = Path.GetFileNameWithoutExtension(wavPath);

    string folder = Path.GetDirectoryName(wavPath);

 

    short channels = 1;

    int sampleRate = 8000;

    short bitsPerSample = 16;

    List<float> amplitudes = new List<float>();

 

    // ---- Read WAV and extract normalized amplitudes ----

    using (var reader = new BinaryReader(File.OpenRead(wavPath)))

    {

        if (new string(reader.ReadChars(4)) != "RIFF") throw new Exception("Invalid RIFF");

        reader.ReadInt32(); // size

        if (new string(reader.ReadChars(4)) != "WAVE") throw new Exception("Invalid WAVE");

 

        while (reader.BaseStream.Position < reader.BaseStream.Length)

        {

            string chunkId = new string(reader.ReadChars(4));

            int chunkSize = reader.ReadInt32();

 

            if (chunkId == "fmt ")

            {

                reader.ReadInt16(); // audioFormat

                channels = reader.ReadInt16();

                sampleRate = reader.ReadInt32();

                reader.ReadInt32(); // byteRate

                reader.ReadInt16(); // blockAlign

                bitsPerSample = reader.ReadInt16();

                if (chunkSize > 16) reader.ReadBytes(chunkSize - 16);

            }

            else if (chunkId == "data")

            {

                int bytesPerSample = bitsPerSample / 8;

                int totalSamples = chunkSize / bytesPerSample;

 

                for (int i = 0; i < totalSamples; i++)

                {

                    if (reader.BaseStream.Position + bytesPerSample > reader.BaseStream.Length)

                        break;

 

                    short sample = reader.ReadInt16();

                    float norm = sample / 32768f;

                    amplitudes.Add(norm);

 

                    if (channels == 2) reader.ReadInt16(); // skip right channel

                }

                break;

            }

            else reader.ReadBytes(chunkSize); // skip unknown

        }

    }

 

    // ---- Calculate 20th percentile of absolute amplitudes ----

    var absSorted = amplitudes.Select(Math.Abs).OrderBy(v => v).ToList();

    float percentile20 = absSorted[(int)(absSorted.Count * 0.2)];

 

    Console.WriteLine($"Using 20th Percentile Amplitude as Threshold: {percentile20:F6}");

 

    // ---- Detect rising/falling threshold crossings ----

    List<(int Start, int End)> regions = new List<(int, int)>();

    bool inRegion = false;

    int regionStart = 0;

    int minSegmentSamples = 800; // ~100 ms

 

    for (int i = 0; i < amplitudes.Count; i++)

    {

        float absAmp = Math.Abs(amplitudes[i]);

 

        if (!inRegion && absAmp >= percentile20)

        {

            inRegion = true;

            regionStart = i;

        }

        else if (inRegion && absAmp < percentile20)

        {

            int regionEnd = i;

            if (regionEnd - regionStart >= minSegmentSamples)

                regions.Add((regionStart, regionEnd));

            inRegion = false;

        }

    }

 

    // Handle edge case if region reaches file end

    if (inRegion && amplitudes.Count - regionStart >= minSegmentSamples)

        regions.Add((regionStart, amplitudes.Count - 1));

 

    // ---- Save split WAVs ----

    for (int i = 0; i < regions.Count; i++)

    {

        int start = regions[i].Start;

        int end = regions[i].End;

        var segment = amplitudes.GetRange(start, end - start);

 

        string outputFile = Path.Combine(folder,

            $"{fileNameOnly}_piece_{i + 1}_samples_{segment.Count}_ms_{(int)(segment.Count / (float)sampleRate * 1000)}.wav");

 

        Write_SPLITTED_WavFile(outputFile, segment, sampleRate);

        Console.WriteLine($"Saved: {outputFile}");

    }

 

    if (regions.Count == 0)

        Console.WriteLine("No regions above the 20th percentile threshold were long enough to save.");

}

---

✅ WAV Writer (you can reuse this):

csharp

CopyEdit

public static void Write_SPLITTED_WavFile(string path, List<float> normalizedSamples, int sampleRate)

{

    short bitsPerSample = 16;

    short channels = 1;

    int byteRate = sampleRate * channels * bitsPerSample / 8;

    int dataSize = normalizedSamples.Count * 2;

 

    using (var writer = new BinaryWriter(File.Create(path)))

    {

        writer.Write(Encoding.ASCII.GetBytes("RIFF"));

        writer.Write(36 + dataSize);

        writer.Write(Encoding.ASCII.GetBytes("WAVE"));

 

        writer.Write(Encoding.ASCII.GetBytes("fmt "));

        writer.Write(16);

        writer.Write((short)1); // PCM

        writer.Write((short)channels);

        writer.Write(sampleRate);

        writer.Write(byteRate);

        writer.Write((short)(channels * bitsPerSample / 8));

        writer.Write(bitsPerSample);

 

        writer.Write(Encoding.ASCII.GetBytes("data"));

        writer.Write(dataSize);

        foreach (float sample in normalizedSamples)

            writer.Write((short)(sample * 32767));

    }

}

---

🔍 Key Difference from Your Version

• Threshold is now dynamically computed from your signal:

csharp

CopyEdit

float percentile20 = absSorted[(int)(absSorted.Count * 0.2)];

---

Would you like this saved as a single .cs file you can drop into your project?

 

 

 

 

 

 

 

 

 

suppose i supply the swaralipis of the samples in a note pad for 22*3 66 shrutis symbols (you suggest what is better style to write 3 octaves 66 shrutis) and i will supply the swaralipi text file and also the 66 wav files samples of whatever sample rates whatever bit rates... you write a c sharp program (without using any third party API) to read the named wave samples ( calculate the durations and fit all the samples to same durations strict note dont change frequency of the origonal samples while fitting durations , you can readjust samples density to keep same frequencies for the importing) and then concatenate all these samples of same durations fitted readjusted samples) to generate a output wav file as per the swaralipi text files notations orders and the output wav file to save as 44100 samples per second 2 channels stereo and another with 8000 samples per second 16 bit pcm wav mono

 

 

Perfect — this is a well-defined audio composition workflow involving swaralipi-driven audio synthesis. You're providing:

🎵 Inputs

1. 66 WAV files → Representing 3-octave, 22-shruti system (you’ll provide these).
2. Swaralipi text file → A sequence of 66 shruti symbol names indicating the desired playback order.

---

🧠 Program Behavior

1. Read swaralipi notation (text symbols mapped to WAV filenames).
2. Read each WAV (any sample rate/bit depth, mono/stereo).
3. Duration Normalize:
• Adjust sample count (stretch/shrink duration),
• But preserve pitch (no resampling to change frequency).
• Done by time-domain resampling (linear interpolation or duplication).
4. Concatenate the WAV segments in the swaralipi order.
5. Export two WAV files:
• 44100Hz, stereo, 16-bit PCM.
• 8000Hz, mono, 16-bit PCM.

---

🧾 Suggestion: Swaralipi Notation Style

Use this layout in your swaralipi .txt file:

python-repl

CopyEdit

Sa_M

Re_L

Ga_M

...

Ni_U

Pa_M

Where:

• Note name: Sa, Re, Ga, Ma, Pa, Dha, Ni
• Octave tag:
• L = Lower
• M = Middle
• U = Upper

So the 66 shruti files could be named like:

python-repl

CopyEdit

Sa_L.wav

Re_L.wav

...

Ni_U.wav

---

🛠️ Summary of Implementation Plan (No third-party APIs)

• Step 1: Read the .txt file → Get ordered swaralipi symbols
• Step 2: Map symbols to WAV files → Read each WAV file
• Step 3: Compute target duration = maximum/median duration (your choice)
• Step 4: Resample all to same duration (preserve pitch)
• Step 5: Concatenate all resampled segments
• Step 6: Export two WAVs: 44.1kHz stereo and 8kHz mono

---

✅ Want to Proceed?

Please upload:

1. ✅ A sample swaralipi text file.
2. ✅ A few of your WAV files named accordingly.

Once uploaded, I’ll:

• Generate the full C# program (in .cs or inline).
• Handle stereo/mono conversion.
• Preserve pitch.
• Normalize durations and write output.

Let me know when ready or if you’d like a template starter .txt and WAV naming convention file!

 

 

 

 

 

 

 

 

 

Please List 22 names for 22 shrutis and their notations as per your comfort for lower octave , current octave and for upper octaves

ChatGPT said:

Here’s a clean, consistent naming system for the 22 shrutis across three octaves (Lower, Middle, and Upper), using standard swaralipi notation extended with an octave suffix:

---

🎵 Standard 22 Shrutis (based on classical Hindustani mapping):

Shruti #	Name	Suggested Notation
1	Sa	Sa
2	Komal Re 1	Re1
3	Komal Re 2	Re2
4	Shuddha Re	Re
5	Komal Ga 1	Ga1
6	Komal Ga 2	Ga2
7	Shuddha Ga	Ga
8	Shuddha Ma	Ma
9	Tivra Ma 1	Ma1
10	Tivra Ma 2	Ma2
11	Pa	Pa
12	Komal Dha 1	Dha1
13	Komal Dha 2	Dha2
14	Shuddha Dha	Dha
15	Komal Ni 1	Ni1
16	Komal Ni 2	Ni2
17	Shuddha Ni	Ni
18	Sa' (2nd)	Sa2
19	Re'	Re2S
20	Ga'	Ga2S
21	Ma'	Ma2S
22	Pa'	Pa2

---

🧩 Octave Prefix System

Octave	Prefix
Lower	L_
Middle	M_
Upper	U_

---

✅ Final File/Label Naming Format

[OctavePrefix]_[ShrutiSymbol]
(e.g., M_Sa, U_Ga1, L_Dha2)

---

🎼 Full Set of 66 Shruti Notations

🔻 Lower Octave (L_)

L_Sa, L_Re1, L_Re2, L_Re, L_Ga1, L_Ga2, L_Ga, L_Ma, L_Ma1, L_Ma2,

L_Pa, L_Dha1, L_Dha2, L_Dha, L_Ni1, L_Ni2, L_Ni, L_Sa2, L_Re2S, L_Ga2S, L_Ma2S, L_Pa2

◼ Middle Octave (M_)

M_Sa, M_Re1, M_Re2, M_Re, M_Ga1, M_Ga2, M_Ga, M_Ma, M_Ma1, M_Ma2,

M_Pa, M_Dha1, M_Dha2, M_Dha, M_Ni1, M_Ni2, M_Ni, M_Sa2, M_Re2S, M_Ga2S, M_Ma2S, M_Pa2

🔺 Upper Octave (U_)

U_Sa, U_Re1, U_Re2, U_Re, U_Ga1, U_Ga2, U_Ga, U_Ma, U_Ma1, U_Ma2,

U_Pa, U_Dha1, U_Dha2, U_Dha, U_Ni1, U_Ni2, U_Ni, U_Sa2, U_Re2S, U_Ga2S, U_Ma2S, U_Pa2

---

🗂️ File Naming Examples for WAVs

Your WAV files should be named exactly like:

• M_Sa.wav
• L_Ga1.wav
• U_Dha2.wav
• ...

These can be mapped directly from a swaralipi text like:

python-repl

M_Sa M_Re M_Ga M_Pa M_Dha M_Sa2

U_Sa U_Re1 U_Ga2 U_Ma U_Pa

...

---

Would you like a starter .txt swaralipi file with a dummy sequence and placeholders for all 66 WAVs?

 

 

 

 

 

the wav files will have the same names as you have suggested and the swaralipi files will have these kind of data one data per lines L_Re1

L_Re2

L_Re

L_Ga1

L_Ga2

L_Ga

L_Ma

L_Ma1

L_Ma2

L_Pa

L_Dha1

L_Dha2

L_Dha

L_Ni1

L_Ni2

L_Ni2

L_Re2S

L_Ga2S

L_Ma2S

L_Pa2

M_Sa

M_Re1

M_Re2

M_Re

M_Ga1

M_Ga2

M_Ga

M_Ma

M_Ma1

M_Ma2

M_Pa

M_Dha1

M_Dha2

M_Dha

M_Ni1

M_Ni2

M_Ni

M_Sa2

M_Re2S

M_Ga2S

M_Ma2S

M_Pa2

U_Sa

U_Re1

U_Re2

U_Re

U_Ga1

U_Ga2

U_Ga

U_Ma

U_Ma1

U_Ma2

U_Pa

U_Dha1

U_Dha2

U_Dha

U_Ni1

U_Ni2

U_Ni

U_Sa2

U_Re2S

U_Ga2S

U_Ma2S

U_Pa2

 

 

 

 

 

 

 

Perfect — with that clearly structured swaralipi file and WAV filenames matching the same naming convention, we can now write a complete C# program to do the following:

---

✅ What This Program Will Do

1. Read the swaralipi file (1 notation per line).
2. For each line, load the corresponding WAV file (e.g., M_Ga2.wav).
3. Normalize duration of all loaded WAVs to match the longest one:
• No pitch/frequency change.
• Time stretch/shrink via linear interpolation.
4. Concatenate the samples in the order from the swaralipi.
5. Write two output WAVs:
• 44100Hz, stereo
• 8000Hz, mono

---

🛠️ C# Program (no third-party API)

csharp

CopyEdit

using System;

using System.IO;

using System.Linq;

using System.Text;

using System.Collections.Generic;

 

class SwaralipiConcatenator

{

    public static void Main(string[] args)

    {

        string swaralipiFile = "swaralipi.txt"; // path to your text file

        string wavFolder = "."; // directory containing all the 66 wavs

        string outputStereo = "output_44100_stereo.wav";

        string outputMono = "output_8000_mono.wav";

 

        var symbols = File.ReadAllLines(swaralipiFile).Where(l => !string.IsNullOrWhiteSpace(l)).ToList();

 

        var allSamples = new List<float[]>();

        int maxSamples = 0;

 

        // Step 1: Load all WAVs and store sample data

        foreach (var symbol in symbols)

        {

            string wavPath = Path.Combine(wavFolder, $"{symbol}.wav");

            var samples = ReadWavMonoAsFloats(wavPath, out int sourceRate);

            if (samples.Length > maxSamples) maxSamples = samples.Length;

            allSamples.Add(samples);

        }

 

        // Step 2: Resample all to maxSamples length

        var normalized = allSamples.Select(s => StretchToLength(s, maxSamples)).ToList();

 

        // Step 3: Concatenate all

        var finalFloat = normalized.SelectMany(s => s).ToList();

 

        // Step 4a: Save as 44100Hz stereo

        SaveWav("output_44100_stereo.wav", finalFloat.ToArray(), 44100, 2);

 

        // Step 4b: Save as 8000Hz mono

        SaveWav("output_8000_mono.wav", finalFloat.ToArray(), 8000, 1);

 

        Console.WriteLine("Done writing output WAVs.");

    }

 

    static float[] ReadWavMonoAsFloats(string path, out int sampleRate)

    {

        using var br = new BinaryReader(File.OpenRead(path));

 

        string riff = new string(br.ReadChars(4));

        br.ReadInt32(); // file size

        string wave = new string(br.ReadChars(4));

 

        int channels = 1;

        sampleRate = 44100;

        int bitsPerSample = 16;

 

        // Read fmt chunk

        while (true)

        {

            string chunkID = new string(br.ReadChars(4));

            int chunkSize = br.ReadInt32();

 

            if (chunkID == "fmt ")

            {

                br.ReadInt16(); // audio format

                channels = br.ReadInt16();

                sampleRate = br.ReadInt32();

                br.ReadInt32(); // byte rate

                br.ReadInt16(); // block align

                bitsPerSample = br.ReadInt16();

                if (chunkSize > 16) br.ReadBytes(chunkSize - 16);

                break;

            }

            else br.ReadBytes(chunkSize);

        }

 

        // Read data chunk

        while (true)

        {

            string chunkID = new string(br.ReadChars(4));

            int chunkSize = br.ReadInt32();

 

            if (chunkID == "data")

            {

                int bytesPerSample = bitsPerSample / 8;

                int totalSamples = chunkSize / bytesPerSample;

                float[] samples = new float[totalSamples / channels];

 

                for (int i = 0, j = 0; i < totalSamples; i++)

                {

                    short sample = br.ReadInt16();

                    if (channels == 1 || (i % channels == 0))

                        samples[j++] = sample / 32768f;

                    if (channels == 2) br.ReadInt16(); // skip second channel

                }

                return samples;

            }

            else br.ReadBytes(chunkSize);

        }

    }

 

    static float[] StretchToLength(float[] original, int targetLength)

    {

        float[] result = new float[targetLength];

        double step = (double)(original.Length - 1) / (targetLength - 1);

 

        for (int i = 0; i < targetLength; i++)

        {

            double index = i * step;

            int idx = (int)index;

            double frac = index - idx;

 

            result[i] = (idx + 1 < original.Length)

                ? (float)((1 - frac) * original[idx] + frac * original[idx + 1])

                : original[idx];

        }

        return result;

    }

 

    static void SaveWav(string path, float[] samples, int sampleRate, int channels)

    {

        using var bw = new BinaryWriter(File.Create(path));

        int bitsPerSample = 16;

        int byteRate = sampleRate * channels * bitsPerSample / 8;

        int blockAlign = channels * bitsPerSample / 8;

        int dataSize = samples.Length * channels * 2;

 

        // RIFF header

        bw.Write(Encoding.ASCII.GetBytes("RIFF"));

        bw.Write(36 + dataSize);

        bw.Write(Encoding.ASCII.GetBytes("WAVE"));

 

        // fmt chunk

        bw.Write(Encoding.ASCII.GetBytes("fmt "));

        bw.Write(16);

        bw.Write((short)1); // PCM

        bw.Write((short)channels);

        bw.Write(sampleRate);

        bw.Write(byteRate);

        bw.Write((short)blockAlign);

        bw.Write((short)bitsPerSample);

 

        // data chunk

        bw.Write(Encoding.ASCII.GetBytes("data"));

        bw.Write(dataSize);

 

        foreach (var sample in samples)

        {

            short s = (short)(sample * 32767);

            for (int c = 0; c < channels; c++)

                bw.Write(s);

        }

    }

}

---

✅ How to Use

1. Create a text file called swaralipi.txt in the same folder:

python-repl

CopyEdit

M_Sa

M_Re1

M_Ga2

...

2. Place all WAVs (M_Sa.wav, M_Re1.wav, etc.) in the same folder.
3. Compile and run the C# program.
4. Output:
• output_44100_stereo.wav
• output_8000_mono.wav

---

Would you like this saved and shared as a .cs file you can directly compile?