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@@ -1,10 +1,19 @@
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# AudioTee
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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.
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**⚠️ API Instability Warning: The AudioTee API is unstable at present and subject to change without notice.**
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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.
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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
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capture system audio to a file like this:
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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.
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```bash
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/path/to/audiotee > output.pcm
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```
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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.
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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.
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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.
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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.
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@@ -16,7 +25,7 @@ Recording system audio is harder than it should be on macOS, and folks often wre
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## Quick start
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The following will start capturing audio output from all running programs and write base64-encoded chunks of it to your terminal every 200ms:
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The following will start capturing audio output from all running programs and write raw PCM audio data to your terminal:
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```bash
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git clone git@github.com:makeusabrew/audiotee.git
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@@ -24,7 +33,17 @@ cd audiotee
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swift run
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```
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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.
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More usefully, you can redirect `stdout` to a file:
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```bash
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swift run audiotee --sample-rate 16000 > output.pcm
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```
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Which you can play back using something like `ffplay`:
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```bash
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ffplay -f s16le -ar 16000 output.pcm
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```
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## Build
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@@ -40,20 +59,23 @@ swift build -c release
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Replace the path below with `.build/<arch>/<target>/audiotee`, e.g. `build/arm64-apple-macosx/release/audiotee` for a release build on Apple Silicon.
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```bash
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# Auto-detect output format (JSON in terminal, binary when piped)
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# Write raw PCM audio to stdout (logs go to stderr)
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./audiotee
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# Always use JSON format (terminal-safe)
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./audiotee --format json
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# Redirect audio to a file
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./audiotee > output.pcm
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# Always use binary format (pipe-optimised)
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./audiotee --format binary
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# Pipe to another program
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./audiotee | your_audio_processing_tool
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# Redirect logs as well
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./audiotee > captured_audio.pcm 2> audiotee.log
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```
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### Audio conversion
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Note that performing sample rate conversion will also convert the output bit depth to
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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.
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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.
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```bash
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# Convert to 16kHz mono (useful for ASR services)
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@@ -94,147 +116,33 @@ Note that trying to include or exclude a PID which isn't currently playing audio
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./audiotee --chunk-duration 0.1
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```
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## Output formats
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## Output
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AudioTee supports two output formats optimised for different use cases:
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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.
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### JSON format (`--format json` or auto in terminal)
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### Audio format
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JSON messages to stdout, one per line. Audio data is base64-encoded for terminal safety.
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- **Format**: Raw PCM audio data
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- **Channels**: Mono (1 channel)
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- **Sample rate**: Matches your output device's sample rate by default (configurable)
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- **Bit depth**: 32-bit float by default, or 16-bit when sample rate conversion is performed
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- **Endianness**: Little-endian
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- **Chunk duration**: 200ms by default (configurable)
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### Binary format (`--format binary` or auto when piped)
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### Logs and monitoring
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JSON metadata lines followed by raw binary audio data. More efficient for piping to other processes.
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All program logs are written to `stderr` and can be captured separately:
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## Protocol
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```bash
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# Capture audio and logs separately
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./audiotee > audio.pcm 2> audiotee.log
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### Message types
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All messages (except raw binary audio chunks) follow this envelope structure:
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```json
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{
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"timestamp": "2024-03-21T15:30:45.123Z",
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"message_type": "...",
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"data": { ... }
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}
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# View logs in real-time while capturing audio
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./audiotee > audio.pcm 2>&1 | grep "AudioTee"
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```
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#### 1. Metadata
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Sent once at startup to describe the audio format:
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```json
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{
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"timestamp": "2024-03-21T15:30:45.123Z",
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"message_type": "metadata",
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"data": {
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"sample_rate": 48000,
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"channels_per_frame": 1,
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"bits_per_channel": 32,
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"is_float": true,
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"capture_mode": "audio",
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"device_name": null,
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"device_uid": null,
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"encoding": "pcm_f32le"
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}
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}
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```
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#### 2. Stream start
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Indicates audio data will follow:
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```json
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{
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"timestamp": "2024-03-21T15:30:45.123Z",
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"message_type": "stream_start",
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"data": null
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}
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```
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#### 3. Audio data
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**JSON format:**
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```json
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{
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"timestamp": "2024-03-21T15:30:45.123Z",
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"message_type": "audio",
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"data": {
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"timestamp": "2024-03-21T15:30:45.123Z",
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"duration": 0.2,
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"peak_amplitude": 0.45,
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"audio_data": "base64_encoded_raw_audio..."
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}
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}
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```
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**Binary format:**
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||||
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||||
```json
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{
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"timestamp": "2024-03-21T15:30:45.123Z",
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"message_type": "audio",
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"data": {
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||||
"timestamp": "2024-03-21T15:30:45.123Z",
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"duration": 0.2,
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"peak_amplitude": 0.45,
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"audio_length": 9600
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}
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}
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```
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_Followed immediately by 9600 bytes of raw binary audio data_
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#### 4. Stream stop
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Sent when recording stops:
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```json
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{
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"timestamp": "2024-03-21T15:30:45.123Z",
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"message_type": "stream_stop",
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"data": null
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}
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```
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#### 5. Log messages
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||||
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Info, error, and debug messages (useful for monitoring):
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||||
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||||
```json
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||||
{
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"timestamp": "2024-03-21T15:30:45.123Z",
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"message_type": "info",
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"data": {
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"message": "Starting AudioTee...",
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"context": { "output_format": "auto" }
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||||
}
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||||
}
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||||
```
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||||
### Consuming output
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**JSON format:**
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||||
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1. Parse each line as JSON using the envelope structure
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2. Use `metadata` message to understand the audio format
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3. For `audio` messages, decode `audio_data` from base64 to get raw PCM data
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||||
4. Do something with each chunk of data
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**Binary format:**
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||||
|
||||
1. Parse JSON metadata lines using the envelope structure
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2. Use `metadata` message to understand the audio format
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||||
3. For `audio` messages, read `audio_length` bytes of raw binary data after the JSON line
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4. Do something with each chunk of data
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||||
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||||
**Note**: binary is actually a mixed mode; JSON during boot, JSON packet header information preceding each binary chunk.
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## Command Line options
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||||
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- `--format, -f`: Output format (`json`, `binary`, `auto`) [default: `auto`]
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- `--include-processes`: Process IDs to tap (space-separated, empty = all processes)
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- `--exclude-processes`: Process IDs to exclude (space-separated, empty = none)
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- `--mute`: Mute processes being tapped
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@@ -244,7 +152,9 @@ Info, error, and debug messages (useful for monitoring):
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## Permissions
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||||
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There is no provision in the code to pre-emptively check for the required `NSAudioCaptureUsageDescription` permission,
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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).
|
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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
|
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iTerm don't always prompt for these permissions (the macOS builtin terminal definitely does), so you
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might need to grant them ahead of time if audiotee looks like it's running but never records anything.
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## References
|
||||
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@@ -189,11 +189,6 @@ class SimpleArgumentParser {
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throw ArgumentParserError.invalidValue(optionName, value)
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}
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return doubleValue as! T
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} else if type == OutputFormat.self {
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guard let format = OutputFormat(rawValue: value) else {
|
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throw ArgumentParserError.invalidValue(optionName, value)
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}
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return format as! T
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}
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|
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throw ArgumentParserError.invalidValue(optionName, value)
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@@ -2,7 +2,6 @@ import CoreAudio
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import Foundation
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|
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struct AudioTee {
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var format: OutputFormat = .auto
|
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var includeProcesses: [Int32] = []
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var excludeProcesses: [Int32] = []
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var mute: Bool = false
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@@ -18,11 +17,6 @@ struct AudioTee {
|
||||
discussion: """
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AudioTee captures system audio using Core Audio taps and streams it as structured output.
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|
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Output formats:
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• json: Base64-encoded audio in JSON messages (safe for terminals)
|
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• binary: Raw binary audio with JSON metadata headers (efficient for pipes)
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• auto: Automatically choose based on whether stdout is a terminal (default)
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|
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Process filtering:
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• include-processes: Only tap specified process IDs (empty = all processes)
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• exclude-processes: Tap all processes except specified ones
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@@ -30,10 +24,8 @@ struct AudioTee {
|
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|
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Examples:
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audiotee # Auto format, tap all processes
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audiotee --format=json # Always use JSON format
|
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audiotee --format=binary # Always use binary format
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audiotee --sample-rate=16000 # Convert to 16kHz mono for ASR
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audiotee --sample-rate=8000 # Convert to 8kHz for telephony
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audiotee --sample-rate 16000 # Convert to 16kHz mono for ASR
|
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audiotee --sample-rate 8000 # Convert to 8kHz for telephony
|
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audiotee --include-processes 1234 # Only tap process 1234
|
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audiotee --include-processes 1234 5678 9012 # Tap only these processes
|
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audiotee --exclude-processes 1234 5678 # Tap everything except these
|
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@@ -42,7 +34,6 @@ struct AudioTee {
|
||||
)
|
||||
|
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// Configure arguments
|
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parser.addOption(name: "format", shortName: "f", help: "Output format", defaultValue: "auto")
|
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parser.addArrayOption(
|
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name: "include-processes",
|
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help: "Process IDs to include (space-separated, empty = all processes)")
|
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@@ -62,7 +53,6 @@ struct AudioTee {
|
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var audioTee = AudioTee()
|
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|
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// Extract values
|
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audioTee.format = try parser.getValue("format", as: OutputFormat.self)
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audioTee.includeProcesses = try parser.getArrayValue("include-processes", as: Int32.self)
|
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audioTee.excludeProcesses = try parser.getArrayValue("exclude-processes", as: Int32.self)
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audioTee.mute = parser.getFlag("mute")
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@@ -102,7 +92,6 @@ struct AudioTee {
|
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setupSignalHandlers()
|
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|
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Logger.info("Starting AudioTee...")
|
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Logger.debug("Using output format: \(format)")
|
||||
|
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// Validate chunk duration
|
||||
guard chunkDuration > 0 && chunkDuration <= 5.0 else {
|
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@@ -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)
|
||||
|
||||
@@ -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)"
|
||||
}
|
||||
}
|
||||
}
|
||||
@@ -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
|
||||
}
|
||||
}
|
||||
|
||||
@@ -119,7 +119,6 @@ public class AudioFormatConverter {
|
||||
return AudioPacket(
|
||||
timestamp: packet.timestamp,
|
||||
duration: packet.duration,
|
||||
peakAmplitude: packet.peakAmplitude,
|
||||
rawAudioData: outputData
|
||||
)
|
||||
}
|
||||
|
||||
@@ -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
|
||||
}
|
||||
}
|
||||
|
||||
@@ -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)
|
||||
}
|
||||
}
|
||||
@@ -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
|
||||
}
|
||||
}
|
||||
@@ -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
Binary file not shown.
Reference in New Issue
Block a user