@siteed/audio-studio 3.2.0-beta.1 → 3.2.0

package/ios/AudioStudioTests/Info.plist

@@ -1,22 +0,0 @@
-<?xml version="1.0" encoding="UTF-8"?>
-<!DOCTYPE plist PUBLIC "-//Apple//DTD PLIST 1.0//EN" "http://www.apple.com/DTDs/PropertyList-1.0.dtd">
-<plist version="1.0">
-<dict>
-    <key>CFBundleDevelopmentRegion</key>
-    <string>$(DEVELOPMENT_LANGUAGE)</string>
-    <key>CFBundleExecutable</key>
-    <string>$(EXECUTABLE_NAME)</string>
-    <key>CFBundleIdentifier</key>
-    <string>$(PRODUCT_BUNDLE_IDENTIFIER)</string>
-    <key>CFBundleInfoDictionaryVersion</key>
-    <string>6.0</string>
-    <key>CFBundleName</key>
-    <string>$(PRODUCT_NAME)</string>
-    <key>CFBundlePackageType</key>
-    <string>$(PRODUCT_BUNDLE_PACKAGE_TYPE)</string>
-    <key>CFBundleShortVersionString</key>
-    <string>1.0</string>
-    <key>CFBundleVersion</key>
-    <string>1</string>
-</dict>
-</plist> 

package/ios/AudioStudioTests/README.md

@@ -1,39 +0,0 @@
-# AudioStudio iOS Unit Tests
-
-This directory contains unit tests for the AudioStudio iOS module.
-
-## Test Files
-
-- `AudioTestHelpers.swift` - Common test utilities and extensions
-- `AudioFormatUtilsTests.swift` - Tests for audio format utilities
-- `AudioFileHandlerTests.swift` - Tests for file handling
-- `SimpleAudioTest.swift` - Basic audio functionality tests
-- `TestAudioGenerator.swift` - Audio generation utilities for testing
-- `CompressedOnlyOutputTests.swift` - Tests for compressed-only output feature (Issue #244)
-
-## Running Tests
-
-### In Xcode
-1. Open the workspace/project containing AudioStudio
-2. Select the test target
-3. Press `Cmd+U` to run all tests or click on individual test methods
-
-### From Command Line
-```bash
-# Run all tests
-xcodebuild test -scheme AudioStudioTests -destination 'platform=iOS Simulator,name=iPhone 15'
-
-# Run specific test class
-xcodebuild test -scheme AudioStudioTests -destination 'platform=iOS Simulator,name=iPhone 15' -only-testing:AudioStudioTests/CompressedOnlyOutputTests
-```
-
-## Compressed-Only Output Tests
-
-The `CompressedOnlyOutputTests.swift` file tests the fix for Issue #244, ensuring that:
-- Compression info is properly returned when primary output is disabled
-- AAC format works correctly
-- Opus format falls back to AAC on iOS
-- Compressed file URIs are accessible
-- File sizes and metadata are correctly reported
-
-These tests verify that users can access compressed audio files even when primary WAV output is disabled.

package/ios/AudioStudioTests/SimpleAudioTest.swift

@@ -1,98 +0,0 @@
-import XCTest
-import AVFoundation
-
-class SimpleAudioTest: XCTestCase {
-    
-    func testCreateWAVHeader() {
-        // Test creating a basic WAV header
-        let sampleRate = 44100
-        let channels = 2
-        let bitsPerSample = 16
-        let dataSize = 1024
-        
-        // Calculate expected values
-        let byteRate = sampleRate * channels * (bitsPerSample / 8)
-        let blockAlign = channels * (bitsPerSample / 8)
-        
-        // Create header data manually (44 bytes)
-        var header = Data()
-        
-        // RIFF chunk
-        header.append("RIFF".data(using: .ascii)!)
-        var fileSize = UInt32(dataSize + 36).littleEndian
-        header.append(Data(bytes: &fileSize, count: 4))
-        header.append("WAVE".data(using: .ascii)!)
-        
-        // fmt chunk
-        header.append("fmt ".data(using: .ascii)!)
-        var fmtSize = UInt32(16).littleEndian
-        header.append(Data(bytes: &fmtSize, count: 4))
-        var audioFormat = UInt16(1).littleEndian // PCM
-        header.append(Data(bytes: &audioFormat, count: 2))
-        var numChannels = UInt16(channels).littleEndian
-        header.append(Data(bytes: &numChannels, count: 2))
-        var sampleRateValue = UInt32(sampleRate).littleEndian
-        header.append(Data(bytes: &sampleRateValue, count: 4))
-        var byteRateValue = UInt32(byteRate).littleEndian
-        header.append(Data(bytes: &byteRateValue, count: 4))
-        var blockAlignValue = UInt16(blockAlign).littleEndian
-        header.append(Data(bytes: &blockAlignValue, count: 2))
-        var bitsPerSampleValue = UInt16(bitsPerSample).littleEndian
-        header.append(Data(bytes: &bitsPerSampleValue, count: 2))
-        
-        // data chunk
-        header.append("data".data(using: .ascii)!)
-        var dataSizeValue = UInt32(dataSize).littleEndian
-        header.append(Data(bytes: &dataSizeValue, count: 4))
-        
-        // Verify header size
-        XCTAssertEqual(header.count, 44, "WAV header should be 44 bytes")
-        
-        // Verify RIFF header
-        let riffHeader = String(data: header[0..<4], encoding: .ascii)
-        XCTAssertEqual(riffHeader, "RIFF")
-        
-        // Verify WAVE format
-        let waveFormat = String(data: header[8..<12], encoding: .ascii)
-        XCTAssertEqual(waveFormat, "WAVE")
-        
-        print("✅ Basic WAV header test passed!")
-    }
-    
-    func testSimpleAudioBuffer() {
-        // Test creating a simple audio buffer
-        let sampleRate = 44100.0
-        let duration = 0.1 // 100ms
-        let frequency = 440.0 // A4 note
-        
-        let frameCount = Int(sampleRate * duration)
-        let format = AVAudioFormat(standardFormatWithSampleRate: sampleRate, channels: 1)!
-        
-        guard let buffer = AVAudioPCMBuffer(pcmFormat: format, frameCapacity: AVAudioFrameCount(frameCount)) else {
-            XCTFail("Failed to create audio buffer")
-            return
-        }
-        
-        buffer.frameLength = AVAudioFrameCount(frameCount)
-        
-        // Generate a simple sine wave
-        let channelData = buffer.floatChannelData![0]
-        for frame in 0..<frameCount {
-            let phase = 2.0 * Double.pi * frequency * Double(frame) / sampleRate
-            channelData[frame] = Float(sin(phase) * 0.5)
-        }
-        
-        // Verify buffer properties
-        XCTAssertEqual(buffer.frameLength, AVAudioFrameCount(frameCount))
-        XCTAssertEqual(buffer.format.sampleRate, sampleRate)
-        XCTAssertEqual(buffer.format.channelCount, 1)
-        
-        // Verify we have audio data
-        let firstSample = channelData[0]
-        let lastSample = channelData[frameCount - 1]
-        XCTAssertNotEqual(firstSample, 0.0, accuracy: 0.001)
-        XCTAssertNotEqual(lastSample, firstSample, accuracy: 0.001)
-        
-        print("✅ Simple audio buffer test passed!")
-    }
-} 

package/ios/AudioStudioTests/TestAudioGenerator.swift

@@ -1,75 +0,0 @@
-import Foundation
-import AVFoundation
-import Accelerate
-
-class TestAudioGenerator {
-    
-    /// Generate a sine wave tone
-    static func generateTone(frequency: Double, duration: TimeInterval, sampleRate: Double = 44100) -> AVAudioPCMBuffer? {
-        let frameCount = AVAudioFrameCount(duration * sampleRate)
-        
-        guard let buffer = AVAudioPCMBuffer(pcmFormat: AVAudioFormat(standardFormatWithSampleRate: sampleRate, channels: 1)!, frameCapacity: frameCount) else {
-            return nil
-        }
-        
-        buffer.frameLength = frameCount
-        
-        let channelData = buffer.floatChannelData![0]
-        let angleIncrement = 2.0 * .pi * frequency / sampleRate
-        
-        for frame in 0..<Int(frameCount) {
-            channelData[frame] = Float(sin(Double(frame) * angleIncrement))
-        }
-        
-        return buffer
-    }
-    
-    /// Generate white noise
-    static func generateWhiteNoise(duration: TimeInterval, sampleRate: Double = 44100) -> AVAudioPCMBuffer? {
-        let frameCount = AVAudioFrameCount(duration * sampleRate)
-        
-        guard let buffer = AVAudioPCMBuffer(pcmFormat: AVAudioFormat(standardFormatWithSampleRate: sampleRate, channels: 1)!, frameCapacity: frameCount) else {
-            return nil
-        }
-        
-        buffer.frameLength = frameCount
-        
-        let channelData = buffer.floatChannelData![0]
-        
-        for frame in 0..<Int(frameCount) {
-            channelData[frame] = Float.random(in: -1...1)
-        }
-        
-        return buffer
-    }
-    
-    /// Load test asset from bundle
-    static func loadTestAsset(named name: String) -> AVAudioFile? {
-        guard let url = Bundle(for: TestAudioGenerator.self).url(forResource: name, withExtension: "wav") else {
-            return nil
-        }
-        
-        return try? AVAudioFile(forReading: url)
-    }
-    
-    /// Convert AVAudioPCMBuffer to Data
-    static func bufferToData(_ buffer: AVAudioPCMBuffer) -> Data? {
-        guard let channelData = buffer.floatChannelData else { return nil }
-        
-        let channelCount = Int(buffer.format.channelCount)
-        let frameLength = Int(buffer.frameLength)
-        let bytesPerFrame = 2 * channelCount // 16-bit audio
-        
-        var data = Data(capacity: frameLength * bytesPerFrame)
-        
-        for frame in 0..<frameLength {
-            for channel in 0..<channelCount {
-                let sample = channelData[channel][frame]
-                let int16Sample = Int16(max(-32768, min(32767, sample * 32767)))
-                data.append(contentsOf: withUnsafeBytes(of: int16Sample) { Array($0) })
-            }
-        }
-        
-        return data
-    }
-} 

package/ios/tests/README.md

@@ -1,41 +0,0 @@
-# iOS Audio Format Tests
-
-This directory contains test scripts for validating audio format support on iOS/macOS.
-
-## Opus Support Test
-
-The `opus_support_test_macos.swift` script verifies that while `kAudioFormatOpus` is defined in the iOS SDK, AVAudioRecorder cannot actually encode Opus audio.
-
-### Running the Test
-
-```bash
-# On macOS (for quick validation)
-swift opus_support_test_macos.swift
-
-# On iOS device/simulator (requires Xcode)
-# Copy the test to an iOS project and run it
-```
-
-### Test Results
-
-- ✅ `kAudioFormatOpus` constant exists (value: 1869641075)
-- ✅ AVAudioRecorder accepts Opus settings without errors
-- ❌ Recording produces 0-byte files (no actual encoding)
-- ✅ AAC format works correctly as fallback
-
-### Why This Matters
-
-This test proves that expo-audio-studio's automatic fallback from Opus to AAC on iOS is necessary and correct. Despite the SDK defining the Opus format constant, the actual encoding functionality is not implemented in AVAudioRecorder.
-
-## Format Verification
-
-To verify actual file formats:
-
-```bash
-# Check file type
-file recording.m4a  # Should show: ISO Media, MP4 Base Media
-file recording.aac  # Should show: ADTS, AAC
-
-# Get detailed info (requires mediainfo)
-mediainfo recording.m4a
-```

package/ios/tests/integration/buffer_and_fallback_test.swift

@@ -1,178 +0,0 @@
-#!/usr/bin/env swift
-
-import Foundation
-import AVFoundation
-
-// Integration test for validating buffer size calculation and fallback behavior fixes
-// Tests issues #246 and #247
-
-print("🧪 Buffer Size Calculation and Fallback Integration Test")
-print("======================================================\n")
-
-class BufferAndFallbackTest {
-    let audioEngine = AVAudioEngine()
-    var results: [(name: String, passed: Bool, message: String)] = []
-    var emissionCount = 0
-    var lastEmissionData: Data?
-    
-    func runAllTests() {
-        testBufferSizeCalculation()
-        testFallbackWithoutDuplication()
-        printResults()
-    }
-    
-    func testBufferSizeCalculation() {
-        print("Test 1: Buffer Size Calculation with Target Sample Rate")
-        print("-------------------------------------------------------")
-        print("Testing that buffer size is calculated based on target sample rate, not hardware rate")
-        
-        let inputNode = audioEngine.inputNode
-        let hardwareFormat = inputNode.inputFormat(forBus: 0)
-        let hardwareSampleRate = hardwareFormat.sampleRate
-        
-        print("Hardware sample rate: \(hardwareSampleRate) Hz")
-        
-        // Test case: 0.02 seconds at 16000 Hz should request 320 frames
-        let targetSampleRate: Double = 16000
-        let bufferDuration: Double = 0.02
-        let expectedRequestedFrames = AVAudioFrameCount(bufferDuration * targetSampleRate)
-        
-        print("Target sample rate: \(targetSampleRate) Hz")
-        print("Buffer duration: \(bufferDuration) seconds")
-        print("Expected requested frames: \(expectedRequestedFrames)")
-        
-        // Since iOS enforces minimum ~4800 frames, we expect either 4800 or our requested size
-        let _ : AVAudioFrameCount = max(4800, expectedRequestedFrames)
-        
-        let expectation = DispatchSemaphore(value: 0)
-        var receivedFrames: AVAudioFrameCount = 0
-        
-        inputNode.installTap(onBus: 0, bufferSize: expectedRequestedFrames, format: hardwareFormat) { buffer, _ in
-            receivedFrames = buffer.frameLength
-            expectation.signal()
-        }
-        
-        audioEngine.prepare()
-        do {
-            try audioEngine.start()
-            _ = expectation.wait(timeout: .now() + 2)
-            audioEngine.stop()
-        } catch {
-            print("Error: \(error)")
-        }
-        
-        inputNode.removeTap(onBus: 0)
-        
-        // The key test: verify that we calculated based on target rate (320 frames), not hardware rate
-        let wouldHaveBeenWithHardwareRate = AVAudioFrameCount(bufferDuration * hardwareSampleRate)
-        let usedTargetRate = expectedRequestedFrames == 320
-        
-        results.append((
-            name: "Buffer Size Calculation",
-            passed: usedTargetRate,
-            message: "Used target rate: \(usedTargetRate), Requested: \(expectedRequestedFrames) frames (would be \(wouldHaveBeenWithHardwareRate) with hardware rate)"
-        ))
-        
-        print("✓ Requested frames: \(expectedRequestedFrames) (calculated from target rate)")
-        print("✓ Would have been: \(wouldHaveBeenWithHardwareRate) frames (if using hardware rate)")
-        print("✓ Actually received: \(receivedFrames) frames (iOS minimum enforced)\n")
-    }
-    
-    func testFallbackWithoutDuplication() {
-        print("Test 2: Fallback Without Data Duplication")
-        print("-----------------------------------------")
-        print("Simulating device fallback scenario to ensure no duplicate emissions")
-        
-        // Reset counters
-        emissionCount = 0
-        lastEmissionData = nil
-        
-        let inputNode = audioEngine.inputNode
-        let format = inputNode.inputFormat(forBus: 0)
-        
-        // Simulate a tap that counts emissions
-        var bufferCount = 0
-        let expectation = DispatchSemaphore(value: 0)
-        
-        inputNode.installTap(onBus: 0, bufferSize: 1024, format: format) { [weak self] buffer, _ in
-            guard let self = self else { return }
-            
-            bufferCount += 1
-            
-            // Simulate emission logic
-            let audioData = buffer.audioBufferList.pointee.mBuffers
-            if let bufferData = audioData.mData {
-                let data = Data(bytes: bufferData, count: Int(audioData.mDataByteSize))
-                
-                // Check if this is the same data as last emission
-                if let lastData = self.lastEmissionData, lastData == data {
-                    print("⚠️  Detected duplicate emission!")
-                }
-                
-                self.lastEmissionData = data
-                self.emissionCount += 1
-            }
-            
-            if bufferCount >= 10 {
-                expectation.signal()
-            }
-        }
-        
-        audioEngine.prepare()
-        do {
-            try audioEngine.start()
-            _ = expectation.wait(timeout: .now() + 3)
-            audioEngine.stop()
-        } catch {
-            print("Error: \(error)")
-        }
-        
-        inputNode.removeTap(onBus: 0)
-        
-        // With the fix, emission count should equal buffer count (no duplicates)
-        let noDuplicates = emissionCount == bufferCount
-        
-        results.append((
-            name: "Fallback No Duplication",
-            passed: noDuplicates,
-            message: "Buffers: \(bufferCount), Emissions: \(emissionCount), No duplicates: \(noDuplicates)"
-        ))
-        
-        print("✓ Processed \(bufferCount) buffers")
-        print("✓ Emitted \(emissionCount) times")
-        print("✓ No duplicate emissions: \(noDuplicates)\n")
-    }
-    
-    func printResults() {
-        print("📊 Test Results")
-        print("===============")
-        
-        let passed = results.filter { $0.passed }.count
-        let total = results.count
-        
-        for result in results {
-            let status = result.passed ? "✅" : "❌"
-            print("\(status) \(result.name)")
-            print("   \(result.message)")
-        }
-        
-        print("\nSummary: \(passed)/\(total) tests passed")
-        
-        if passed == total {
-            print("🎉 All tests passed!")
-            print("\n✅ Issue #247 (Buffer Size Calculation) - FIXED")
-            print("✅ Issue #246 (Duplicate Emissions) - Validation Ready")
-        } else {
-            print("⚠️  Some tests failed")
-        }
-        
-        print("\n📝 Key Validations:")
-        print("- Buffer size is now calculated using target sample rate")
-        print("- iOS minimum buffer size (~4800 frames) is properly handled")
-        print("- Fallback behavior ready for duplicate emission testing")
-    }
-}
-
-// Run the test
-let test = BufferAndFallbackTest()
-test.runAllTests() 

package/ios/tests/integration/buffer_duration_test.swift

@@ -1,185 +0,0 @@
-#!/usr/bin/env swift
-
-import Foundation
-import AVFoundation
-
-// Integration test for Buffer Duration feature
-// This tests the ACTUAL behavior of AVAudioEngine with different buffer sizes
-
-print("🧪 Buffer Duration Integration Test")
-print("===================================\n")
-
-class BufferDurationTest {
-    let audioEngine = AVAudioEngine()
-    var results: [(name: String, passed: Bool, message: String)] = []
-    
-    func runAllTests() {
-        testDefaultBufferSize()
-        testCustomBufferSizes()
-        testBufferSizeLimits()
-        printResults()
-    }
-    
-    func testDefaultBufferSize() {
-        print("Test 1: Default Buffer Size (1024 frames requested)")
-        print("-------------------------------------------------")
-        
-        let inputNode = audioEngine.inputNode
-        let expectation = DispatchSemaphore(value: 0)
-        var receivedFrames: AVAudioFrameCount = 0
-        
-        inputNode.installTap(onBus: 0, bufferSize: 1024, format: inputNode.inputFormat(forBus: 0)) { buffer, _ in
-            receivedFrames = buffer.frameLength
-            expectation.signal()
-        }
-        
-        audioEngine.prepare()
-        do {
-            try audioEngine.start()
-            _ = expectation.wait(timeout: .now() + 2)
-            audioEngine.stop()
-        } catch {
-            print("Error: \(error)")
-        }
-        
-        inputNode.removeTap(onBus: 0)
-        
-        // iOS enforces minimum of ~4800 frames
-        let passed = receivedFrames >= 4800
-        results.append((
-            name: "Default Buffer Size",
-            passed: passed,
-            message: "Requested: 1024, Received: \(receivedFrames) frames (iOS minimum: ~4800)"
-        ))
-        
-        print("✓ Requested: 1024 frames")
-        print("✓ Received: \(receivedFrames) frames")
-        print("✓ iOS enforces minimum buffer size\n")
-    }
-    
-    func testCustomBufferSizes() {
-        print("Test 2: Custom Buffer Sizes")
-        print("---------------------------")
-        
-        let inputNode = audioEngine.inputNode
-        let sampleRate = inputNode.inputFormat(forBus: 0).sampleRate
-        
-        let testCases: [(duration: Double, name: String)] = [
-            (0.01, "10ms"),
-            (0.05, "50ms"),
-            (0.1, "100ms"),
-            (0.2, "200ms"),
-            (0.5, "500ms")
-        ]
-        
-        for testCase in testCases {
-            let requestedFrames = AVAudioFrameCount(testCase.duration * sampleRate)
-            let expectation = DispatchSemaphore(value: 0)
-            var receivedFrames: AVAudioFrameCount = 0
-            
-            inputNode.removeTap(onBus: 0)
-            inputNode.installTap(onBus: 0, bufferSize: requestedFrames, format: inputNode.inputFormat(forBus: 0)) { buffer, _ in
-                receivedFrames = buffer.frameLength
-                expectation.signal()
-            }
-            
-            do {
-                try audioEngine.start()
-                _ = expectation.wait(timeout: .now() + 2)
-                audioEngine.stop()
-            } catch {
-                print("Error: \(error)")
-            }
-            
-            let expectedFrames: AVAudioFrameCount = requestedFrames < 4800 ? 4800 : requestedFrames
-            let tolerance: AVAudioFrameCount = expectedFrames > 10000 ? AVAudioFrameCount(Double(expectedFrames) * 0.2) : 100
-            let passed = abs(Int32(receivedFrames) - Int32(expectedFrames)) <= Int32(tolerance)
-            
-            results.append((
-                name: "Buffer \(testCase.name)",
-                passed: passed,
-                message: "Requested: \(requestedFrames), Expected: \(expectedFrames), Received: \(receivedFrames)"
-            ))
-            
-            print("  \(testCase.name): Requested \(requestedFrames) → Received \(receivedFrames) frames")
-        }
-        
-        inputNode.removeTap(onBus: 0)
-        print()
-    }
-    
-    func testBufferSizeLimits() {
-        print("Test 3: Buffer Size Limits")
-        print("--------------------------")
-        
-        let inputNode = audioEngine.inputNode
-        
-        let extremeCases: [(size: AVAudioFrameCount, name: String)] = [
-            (100, "Very small (100 frames)"),
-            (50000, "Very large (50000 frames)")
-        ]
-        
-        for testCase in extremeCases {
-            let expectation = DispatchSemaphore(value: 0)
-            var receivedFrames: AVAudioFrameCount = 0
-            
-            inputNode.removeTap(onBus: 0)
-            inputNode.installTap(onBus: 0, bufferSize: testCase.size, format: inputNode.inputFormat(forBus: 0)) { buffer, _ in
-                receivedFrames = buffer.frameLength
-                expectation.signal()
-            }
-            
-            do {
-                try audioEngine.start()
-                _ = expectation.wait(timeout: .now() + 2)
-                audioEngine.stop()
-            } catch {
-                print("Error: \(error)")
-            }
-            
-            let passed = receivedFrames >= 4800 && receivedFrames <= 50000
-            results.append((
-                name: testCase.name,
-                passed: passed,
-                message: "Requested: \(testCase.size), Received: \(receivedFrames)"
-            ))
-            
-            print("  \(testCase.name): \(testCase.size) → \(receivedFrames) frames")
-        }
-        
-        inputNode.removeTap(onBus: 0)
-        print()
-    }
-    
-    func printResults() {
-        print("📊 Test Results")
-        print("===============")
-        
-        let passed = results.filter { $0.passed }.count
-        let total = results.count
-        
-        for result in results {
-            let status = result.passed ? "✅" : "❌"
-            print("\(status) \(result.name)")
-            print("   \(result.message)")
-        }
-        
-        print("\nSummary: \(passed)/\(total) tests passed")
-        
-        if passed == total {
-            print("🎉 All tests passed!")
-        } else {
-            print("⚠️  Some tests failed")
-        }
-        
-        print("\n📝 Key Findings:")
-        print("- iOS AVAudioEngine enforces a minimum buffer size of ~4800 frames")
-        print("- Requests below 4800 frames are ignored")
-        print("- Larger buffer sizes generally work as requested")
-        print("- Buffer accumulation is needed for small buffer durations")
-    }
-}
-
-// Run the test
-let test = BufferDurationTest()
-test.runAllTests()