10 Commits

Author SHA1 Message Date
Nick Payne 2040510e9e consuming output is already fairly well covered 2025-07-27 10:14:19 +01:00
Nick Payne 3449a9bb9c further README tweaks 2025-07-27 10:10:36 +01:00
Nick Payne ac0ae46cfa update readme 2025-07-27 10:06:01 +01:00
Nick Payne 8c3ee0f4e7 fix build error 2025-07-27 09:55:33 +01:00
Nick Payne abaa019bd2 only write PCM to stdout 2025-07-27 09:53:26 +01:00
Nick Payne 98e33b7bcb Merge pull request #3 from makeusabrew/memory-leak
Use ring buffer to avoid memory leak
2025-07-12 13:45:41 +01:00
Nick Payne ee8968e2d6 no need for buffer overflow error 2025-07-12 13:45:25 +01:00
Nick Payne 4bc36019c2 get rid of O(n) ops on hot audio packet path 2025-07-10 20:43:17 +01:00
Nick Payne 1b537eb395 use ring buffer to avoid memory leak 2025-07-10 14:08:52 +01:00
Nick Payne 80d7555b60 Merge pull request #2 from makeusabrew/readiness-test
Wait for aggregate device ID to be valid
2025-07-04 09:22:50 +01:00
14 changed files with 137 additions and 332 deletions
+52 -142
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@@ -1,10 +1,19 @@
# AudioTee
AudioTee captures your Mac's system audio output and writes it in PCM encoded chunks to `stdout` at regular intervals, either in base64-encoded JSON (good for humans, easy on terminals) or binary (good for other programs). It uses the [Core Audio taps](https://developer.apple.com/documentation/coreaudio/capturing-system-audio-with-core-audio-taps) API introduced in macOS 14.2 (released in December 2023). You can do whatever you want with this audio - stream it somewhere else, save it to disk, visualize it, etc.
**⚠️ API Instability Warning: The AudioTee API is unstable at present and subject to change without notice.**
By default, it taps the audio output from **all** running process and selects the most appropriate audio chunk output format to use based on the presence of a tty. Tap output is forced to `mono` (not yet configurable) and preserves your output device's sample rate (configurable via the `--sample-rate` flag). Only the default output device is currently supported.
AudioTee captures your Mac's system audio output and writes it in PCM encoded chunks to `stdout` at regular intervals. All logging and metadata information is written to `stderr`, meaning at its simplest you can
capture system audio to a file like this:
My original (and so far only) use case is streaming audio to a parent process which communicates with a realtime ASR service, so AudioTee makes some design decisions you might not agree with. Open an issue or a PR and we can talk about them. I'm also no Swift developer, so contributions improving codebase idioms and general hygiene are welcome.
```bash
/path/to/audiotee > output.pcm
```
System audio is captured using the [Core Audio taps](https://developer.apple.com/documentation/coreaudio/capturing-system-audio-with-core-audio-taps) API introduced in macOS 14.2 (released in December 2023). You can do whatever you want with this audio - stream it somewhere else, save it to disk, visualise it, etc.
By default, audiotee captures audio output from **all** running processes. Tap output is forced to `mono` (not yet configurable) and preserves your output device's sample rate (configurable via the `--sample-rate` flag). Only the default output device is currently supported.
My original (and so far only) use case is streaming audio to a parent process which communicates with a realtime ASR service, so AudioTee makes some design decisions you might not agree with. Open an issue or a PR and we can talk about them. I'm also no Swift developer, so contributions improving codebase idioms and general hygiene are welcome. I have internal variations (and, franky, improvements) of audiotee which allow recording mic input as well as system audio, and I'm open to making that part of the main API.
Recording system audio is harder than it should be on macOS, and folks often wrestle with outdated advice and poorly documented APIs. It's a boring problem which stands in the way of lots of fun applications. There's more code here than you need to solve this problem yourself: the main classes of interest are probably [`Core/AudioTapManager`](https://github.com/makeusabrew/audiotee/blob/main/Sources/Core/AudioTapManager.swift) and [`Core/AudioRecorder`](https://github.com/makeusabrew/audiotee/blob/main/Sources/Core/AudioRecorder.swift). Everything's wired together in [`CLI/AudioTee`](https://github.com/makeusabrew/audiotee/blob/main/Sources/CLI/AudioTee.swift). The rest is just CLI configuration support, output formatting logic, and some utility functions you could probably live without.
@@ -16,7 +25,7 @@ Recording system audio is harder than it should be on macOS, and folks often wre
## Quick start
The following will start capturing audio output from all running programs and write base64-encoded chunks of it to your terminal every 200ms:
The following will start capturing audio output from all running programs and write raw PCM audio data to your terminal:
```bash
git clone git@github.com:makeusabrew/audiotee.git
@@ -24,7 +33,17 @@ cd audiotee
swift run
```
If you're not playing audio when you run it, you'll just see packets full of `AAAAA...` - the base64 version of a bunch of zeroes.
More usefully, you can redirect `stdout` to a file:
```bash
swift run audiotee --sample-rate 16000 > output.pcm
```
Which you can play back using something like `ffplay`:
```bash
ffplay -f s16le -ar 16000 output.pcm
```
## Build
@@ -40,20 +59,23 @@ swift build -c release
Replace the path below with `.build/<arch>/<target>/audiotee`, e.g. `build/arm64-apple-macosx/release/audiotee` for a release build on Apple Silicon.
```bash
# Auto-detect output format (JSON in terminal, binary when piped)
# Write raw PCM audio to stdout (logs go to stderr)
./audiotee
# Always use JSON format (terminal-safe)
./audiotee --format json
# Redirect audio to a file
./audiotee > output.pcm
# Always use binary format (pipe-optimised)
./audiotee --format binary
# Pipe to another program
./audiotee | your_audio_processing_tool
# Redirect logs as well
./audiotee > captured_audio.pcm 2> audiotee.log
```
### Audio conversion
Note that performing sample rate conversion will also convert the output bit depth to
16-bit - assuming an original depth of 32-bit this results in a loss of dynamic range in exchange for half the output chunk size. For ASR services, 16-bit is sufficient, but in any case it's a behaviour worth being aware of.
16-bit - assuming an original depth of 32-bit this results in a loss of dynamic range in exchange for half the output chunk size. For ASR services, 16-bit is sufficient, but it's a behaviour worth being aware of.
```bash
# Convert to 16kHz mono (useful for ASR services)
@@ -94,147 +116,33 @@ Note that trying to include or exclude a PID which isn't currently playing audio
./audiotee --chunk-duration 0.1
```
## Output formats
## Output
AudioTee supports two output formats optimised for different use cases:
AudioTee writes raw PCM audio data directly to `stdout` in chunks. All logging, metadata, and status information is written to `stderr`, allowing for clean separation of audio data from program output.
### JSON format (`--format json` or auto in terminal)
### Audio format
JSON messages to stdout, one per line. Audio data is base64-encoded for terminal safety.
- **Format**: Raw PCM audio data
- **Channels**: Mono (1 channel)
- **Sample rate**: Matches your output device's sample rate by default (configurable)
- **Bit depth**: 32-bit float by default, or 16-bit when sample rate conversion is performed
- **Endianness**: Little-endian
- **Chunk duration**: 200ms by default (configurable)
### Binary format (`--format binary` or auto when piped)
### Logs and monitoring
JSON metadata lines followed by raw binary audio data. More efficient for piping to other processes.
All program logs are written to `stderr` and can be captured separately:
## Protocol
```bash
# Capture audio and logs separately
./audiotee > audio.pcm 2> audiotee.log
### Message types
All messages (except raw binary audio chunks) follow this envelope structure:
```json
{
"timestamp": "2024-03-21T15:30:45.123Z",
"message_type": "...",
"data": { ... }
}
# View logs in real-time while capturing audio
./audiotee > audio.pcm 2>&1 | grep "AudioTee"
```
#### 1. Metadata
Sent once at startup to describe the audio format:
```json
{
"timestamp": "2024-03-21T15:30:45.123Z",
"message_type": "metadata",
"data": {
"sample_rate": 48000,
"channels_per_frame": 1,
"bits_per_channel": 32,
"is_float": true,
"capture_mode": "audio",
"device_name": null,
"device_uid": null,
"encoding": "pcm_f32le"
}
}
```
#### 2. Stream start
Indicates audio data will follow:
```json
{
"timestamp": "2024-03-21T15:30:45.123Z",
"message_type": "stream_start",
"data": null
}
```
#### 3. Audio data
**JSON format:**
```json
{
"timestamp": "2024-03-21T15:30:45.123Z",
"message_type": "audio",
"data": {
"timestamp": "2024-03-21T15:30:45.123Z",
"duration": 0.2,
"peak_amplitude": 0.45,
"audio_data": "base64_encoded_raw_audio..."
}
}
```
**Binary format:**
```json
{
"timestamp": "2024-03-21T15:30:45.123Z",
"message_type": "audio",
"data": {
"timestamp": "2024-03-21T15:30:45.123Z",
"duration": 0.2,
"peak_amplitude": 0.45,
"audio_length": 9600
}
}
```
_Followed immediately by 9600 bytes of raw binary audio data_
#### 4. Stream stop
Sent when recording stops:
```json
{
"timestamp": "2024-03-21T15:30:45.123Z",
"message_type": "stream_stop",
"data": null
}
```
#### 5. Log messages
Info, error, and debug messages (useful for monitoring):
```json
{
"timestamp": "2024-03-21T15:30:45.123Z",
"message_type": "info",
"data": {
"message": "Starting AudioTee...",
"context": { "output_format": "auto" }
}
}
```
### Consuming output
**JSON format:**
1. Parse each line as JSON using the envelope structure
2. Use `metadata` message to understand the audio format
3. For `audio` messages, decode `audio_data` from base64 to get raw PCM data
4. Do something with each chunk of data
**Binary format:**
1. Parse JSON metadata lines using the envelope structure
2. Use `metadata` message to understand the audio format
3. For `audio` messages, read `audio_length` bytes of raw binary data after the JSON line
4. Do something with each chunk of data
**Note**: binary is actually a mixed mode; JSON during boot, JSON packet header information preceding each binary chunk.
## Command Line options
- `--format, -f`: Output format (`json`, `binary`, `auto`) [default: `auto`]
- `--include-processes`: Process IDs to tap (space-separated, empty = all processes)
- `--exclude-processes`: Process IDs to exclude (space-separated, empty = none)
- `--mute`: Mute processes being tapped
@@ -244,7 +152,9 @@ Info, error, and debug messages (useful for monitoring):
## Permissions
There is no provision in the code to pre-emptively check for the required `NSAudioCaptureUsageDescription` permission,
so you'll be prompted the first time AudioTee tries to record anything. If you want to check and/or request permissions ahead of time, check out [AudioCap's clever TCC probing approach](https://github.com/insidegui/AudioCap/blob/main/AudioCap/ProcessTap/AudioRecordingPermission.swift).
so you'll be prompted the first time AudioTee tries to record anything. If you want to check and/or request permissions ahead of time, check out [AudioCap's clever TCC probing approach](https://github.com/insidegui/AudioCap/blob/main/AudioCap/ProcessTap/AudioRecordingPermission.swift). Note that some terminal emulators like
iTerm don't always prompt for these permissions (the macOS builtin terminal definitely does), so you
might need to grant them ahead of time if audiotee looks like it's running but never records anything.
## References
-5
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@@ -189,11 +189,6 @@ class SimpleArgumentParser {
throw ArgumentParserError.invalidValue(optionName, value)
}
return doubleValue as! T
} else if type == OutputFormat.self {
guard let format = OutputFormat(rawValue: value) else {
throw ArgumentParserError.invalidValue(optionName, value)
}
return format as! T
}
throw ArgumentParserError.invalidValue(optionName, value)
+4 -26
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@@ -2,7 +2,6 @@ import CoreAudio
import Foundation
struct AudioTee {
var format: OutputFormat = .auto
var includeProcesses: [Int32] = []
var excludeProcesses: [Int32] = []
var mute: Bool = false
@@ -18,11 +17,6 @@ struct AudioTee {
discussion: """
AudioTee captures system audio using Core Audio taps and streams it as structured output.
Output formats:
• json: Base64-encoded audio in JSON messages (safe for terminals)
• binary: Raw binary audio with JSON metadata headers (efficient for pipes)
• auto: Automatically choose based on whether stdout is a terminal (default)
Process filtering:
• include-processes: Only tap specified process IDs (empty = all processes)
• exclude-processes: Tap all processes except specified ones
@@ -30,10 +24,8 @@ struct AudioTee {
Examples:
audiotee # Auto format, tap all processes
audiotee --format=json # Always use JSON format
audiotee --format=binary # Always use binary format
audiotee --sample-rate=16000 # Convert to 16kHz mono for ASR
audiotee --sample-rate=8000 # Convert to 8kHz for telephony
audiotee --sample-rate 16000 # Convert to 16kHz mono for ASR
audiotee --sample-rate 8000 # Convert to 8kHz for telephony
audiotee --include-processes 1234 # Only tap process 1234
audiotee --include-processes 1234 5678 9012 # Tap only these processes
audiotee --exclude-processes 1234 5678 # Tap everything except these
@@ -42,7 +34,6 @@ struct AudioTee {
)
// Configure arguments
parser.addOption(name: "format", shortName: "f", help: "Output format", defaultValue: "auto")
parser.addArrayOption(
name: "include-processes",
help: "Process IDs to include (space-separated, empty = all processes)")
@@ -62,7 +53,6 @@ struct AudioTee {
var audioTee = AudioTee()
// Extract values
audioTee.format = try parser.getValue("format", as: OutputFormat.self)
audioTee.includeProcesses = try parser.getArrayValue("include-processes", as: Int32.self)
audioTee.excludeProcesses = try parser.getArrayValue("exclude-processes", as: Int32.self)
audioTee.mute = parser.getFlag("mute")
@@ -102,7 +92,6 @@ struct AudioTee {
setupSignalHandlers()
Logger.info("Starting AudioTee...")
Logger.debug("Using output format: \(format)")
// Validate chunk duration
guard chunkDuration > 0 && chunkDuration <= 5.0 else {
@@ -143,7 +132,7 @@ struct AudioTee {
throw ExitCode.failure
}
let outputHandler = createOutputHandler(for: format)
let outputHandler = BinaryAudioOutputHandler()
let recorder = AudioRecorder(
deviceID: deviceID, outputHandler: outputHandler, convertToSampleRate: sampleRate,
chunkDuration: chunkDuration)
@@ -172,17 +161,6 @@ struct AudioTee {
}
}
private func createOutputHandler(for format: OutputFormat) -> AudioOutputHandler {
switch format {
case .json:
return JSONAudioOutputHandler()
case .binary:
return BinaryAudioOutputHandler()
case .auto:
return AutoAudioOutputHandler()
}
}
private func convertProcessFlags() -> ([Int32], Bool) {
if !includeProcesses.isEmpty {
// Include specific processes only
@@ -200,7 +178,7 @@ struct AudioTee {
// Helper for stderr output
var standardError = FileHandle.standardError
extension FileHandle: @retroactive TextOutputStream {
extension FileHandle: TextOutputStream {
public func write(_ string: String) {
let data = Data(string.utf8)
self.write(data)
-16
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@@ -1,16 +0,0 @@
enum OutputFormat: String, CaseIterable {
case json = "json"
case binary = "binary"
case auto = "auto"
var description: String {
switch self {
case .json:
return "Base64-encoded JSON (terminal-safe)"
case .binary:
return "Binary with JSON headers (pipe-optimised)"
case .auto:
return "Auto-detect based on TTY (default)"
}
}
}
+75 -29
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@@ -2,17 +2,64 @@ import CoreAudio
import Foundation
public class AudioBuffer {
private var buffer = Data()
private let targetChunkDuration: Double
private let streamFormat: AudioStreamBasicDescription
private var buffer: [UInt8]
private var writeIndex: Int = 0
private var readIndex: Int = 0
private var availableBytes: Int = 0
private let maxBufferSize: Int
private let bytesPerChunk: Int
private let chunkDuration: Double
public init(format: AudioStreamBasicDescription, chunkDuration: Double = 0.2) {
self.streamFormat = format
self.targetChunkDuration = chunkDuration
// Pre-calculate chunk parameters
let bytesPerFrame = Int(format.mBytesPerFrame)
let samplesPerChunk = Int(format.mSampleRate * chunkDuration)
self.bytesPerChunk = samplesPerChunk * bytesPerFrame
self.chunkDuration = Double(samplesPerChunk) / format.mSampleRate
// Calculate max buffer size to hold ~10 seconds of audio, way more than the maximum we allow
let bytesPerSecond = Int(format.mSampleRate) * bytesPerFrame
self.maxBufferSize = bytesPerSecond * 10
// Pre-allocated ring buffer
self.buffer = Array(repeating: 0, count: maxBufferSize)
}
public func append(_ data: Data) {
buffer.append(data)
guard availableBytes + data.count <= maxBufferSize else {
Logger.error(
"Audio buffer overflow",
context: [
"requested": String(data.count),
"available": String(maxBufferSize - availableBytes),
])
return
}
data.withUnsafeBytes { bytes in
let sourceBytes = bytes.bindMemory(to: UInt8.self)
let dataSize = sourceBytes.count
// Check if we can copy in one block (no wrap-around)
if writeIndex + dataSize <= maxBufferSize {
// only one write needed
buffer.replaceSubrange(writeIndex..<writeIndex + dataSize, with: sourceBytes)
writeIndex = (writeIndex + dataSize) % maxBufferSize
} else {
// two writes needed due to wrap-around
let firstChunkSize = maxBufferSize - writeIndex
let secondChunkSize = dataSize - firstChunkSize
buffer.replaceSubrange(writeIndex..<maxBufferSize, with: sourceBytes.prefix(firstChunkSize))
buffer.replaceSubrange(0..<secondChunkSize, with: sourceBytes.suffix(secondChunkSize))
writeIndex = secondChunkSize
}
}
availableBytes += data.count
}
public func processChunks() -> [AudioPacket] {
@@ -25,37 +72,36 @@ public class AudioBuffer {
return packets
}
public func flushRemaining() -> AudioPacket? {
guard !buffer.isEmpty else { return nil }
let packet = AudioPacket(
timestamp: Date(),
duration: 0.0, // Unknown duration for final chunk
peakAmplitude: 0.0,
rawAudioData: buffer
)
buffer.removeAll()
return packet
}
private func nextChunk() -> AudioPacket? {
let bytesPerFrame = Int(streamFormat.mBytesPerFrame)
let samplesPerChunk = Int(streamFormat.mSampleRate * targetChunkDuration)
let bytesPerChunk = samplesPerChunk * bytesPerFrame
// Check if we have enough data for a complete chunk
guard availableBytes >= bytesPerChunk else { return nil }
guard buffer.count >= bytesPerChunk else { return nil }
var chunkData = Data(capacity: bytesPerChunk)
let chunkData = buffer.prefix(bytesPerChunk)
// Check if we can copy in one block (no wrap-around)
if readIndex + bytesPerChunk <= maxBufferSize {
// one copy needed
chunkData.append(contentsOf: buffer[readIndex..<readIndex + bytesPerChunk])
readIndex = (readIndex + bytesPerChunk) % maxBufferSize
} else {
// two copies needed due to wrap-around
let firstChunkSize = maxBufferSize - readIndex
let secondChunkSize = bytesPerChunk - firstChunkSize
chunkData.append(contentsOf: buffer[readIndex..<maxBufferSize])
chunkData.append(contentsOf: buffer[0..<secondChunkSize])
readIndex = secondChunkSize
}
availableBytes -= bytesPerChunk
let packet = AudioPacket(
timestamp: Date(),
duration: Double(samplesPerChunk) / streamFormat.mSampleRate,
peakAmplitude: 0.0, // No analysis in raw mode
rawAudioData: Data(chunkData)
duration: chunkDuration,
rawAudioData: chunkData
)
buffer.removeFirst(bytesPerChunk)
return packet
}
}
-1
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@@ -119,7 +119,6 @@ public class AudioFormatConverter {
return AudioPacket(
timestamp: packet.timestamp,
duration: packet.duration,
peakAmplitude: packet.peakAmplitude,
rawAudioData: outputData
)
}
-3
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@@ -3,18 +3,15 @@ import Foundation
public struct AudioPacket {
public let timestamp: Date
public let duration: Double
public let peakAmplitude: Float // useful for level monitoring
public let rawAudioData: Data
public init(
timestamp: Date,
duration: Double,
peakAmplitude: Float,
rawAudioData: Data
) {
self.timestamp = timestamp
self.duration = duration
self.peakAmplitude = peakAmplitude
self.rawAudioData = rawAudioData
}
}
+4 -3
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@@ -74,7 +74,8 @@ public class AudioRecorder {
Logger.info("Audio device started successfully")
}
// FIXME: note to self, what about installTap? Would require audio engine and a node?
// Note to self, what about installTap? Would require audio engine and a node?
// No; AudioEngine.installTap() can only fire as often as 100ms. too slow for us
private func setupAndStartIOProc() {
Logger.debug("Creating IO proc")
var status = AudioDeviceCreateIOProcID(
@@ -126,8 +127,8 @@ public class AudioRecorder {
func stopRecording() {
// Send any remaining buffered audio, applying conversion if needed
if let finalPacket = audioBuffer?.flushRemaining() {
let processedPacket = converter?.transform(finalPacket) ?? finalPacket
audioBuffer?.processChunks().forEach { packet in
let processedPacket = converter?.transform(packet) ?? packet
outputHandler.handleAudioPacket(processedPacket)
}
@@ -1,31 +0,0 @@
import Foundation
/// Auto-detecting output handler based on TTY
public class AutoAudioOutputHandler: AudioOutputHandler {
private let handler: AudioOutputHandler
public init() {
// Auto-detect based on whether stdout is a terminal
if isatty(STDOUT_FILENO) != 0 {
handler = JSONAudioOutputHandler()
} else {
handler = BinaryAudioOutputHandler()
}
}
public func handleAudioPacket(_ packet: AudioPacket) {
handler.handleAudioPacket(packet)
}
public func handleMetadata(_ metadata: AudioStreamMetadata) {
handler.handleMetadata(metadata)
}
public func handleStreamStart() {
handler.handleStreamStart()
}
public func handleStreamStop() {
handler.handleStreamStop()
}
}
@@ -5,12 +5,6 @@ public class BinaryAudioOutputHandler: AudioOutputHandler {
public init() {}
public func handleAudioPacket(_ packet: AudioPacket) {
// Create metadata without the audio data
let metadata = BinaryPacketHeader(from: packet)
// Write JSON metadata line
Logger.writeMessage(.audio, data: metadata)
// Write raw binary audio data directly to stdout
FileHandle.standardOutput.write(packet.rawAudioData)
}
@@ -1,23 +0,0 @@
import Foundation
/// Base64-encoded JSON output (terminal-safe)
public class JSONAudioOutputHandler: AudioOutputHandler {
public init() {}
public func handleAudioPacket(_ packet: AudioPacket) {
let jsonPacket = JSONAudioPacket(from: packet)
Logger.writeMessage(.audio, data: jsonPacket)
}
public func handleMetadata(_ metadata: AudioStreamMetadata) {
Logger.writeMessage(.metadata, data: metadata)
}
public func handleStreamStart() {
Logger.writeMessage(.streamStart, data: Optional<String>.none)
}
public func handleStreamStop() {
Logger.writeMessage(.streamStop, data: Optional<String>.none)
}
}
-45
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@@ -1,45 +0,0 @@
import Foundation
/// JSON-serializable version of AudioPacket with base64-encoded audio data
public struct JSONAudioPacket: Codable {
public let timestamp: Date
public let duration: Double
public let peakAmplitude: Float
public let audioData: String // base64 encoded audio data
public enum CodingKeys: String, CodingKey {
case timestamp
case duration
case peakAmplitude = "peak_amplitude"
case audioData = "audio_data"
}
public init(from packet: AudioPacket) {
self.timestamp = packet.timestamp
self.duration = packet.duration
self.peakAmplitude = packet.peakAmplitude
self.audioData = packet.rawAudioData.base64EncodedString()
}
}
/// Metadata-only packet for binary output (without base64 audio data)
public struct BinaryPacketHeader: Codable {
public let timestamp: Date
public let duration: Double
public let peakAmplitude: Float
public let audioLength: Int // Length of raw audio data in bytes
public enum CodingKeys: String, CodingKey {
case timestamp
case duration
case peakAmplitude = "peak_amplitude"
case audioLength = "audio_length"
}
public init(from packet: AudioPacket) {
self.timestamp = packet.timestamp
self.duration = packet.duration
self.peakAmplitude = packet.peakAmplitude
self.audioLength = packet.rawAudioData.count
}
}
+2 -2
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@@ -24,8 +24,8 @@ public class Logger {
let message = Message(type: type, data: data)
do {
let jsonData = try jsonEncoder.encode(message)
FileHandle.standardOutput.write(jsonData)
FileHandle.standardOutput.write("\n".data(using: .utf8)!)
FileHandle.standardError.write(jsonData)
FileHandle.standardError.write("\n".data(using: .utf8)!)
} catch {
// TODO: handle at some point
}
BIN
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