2025-02-16 172544,198 | alltalk_tts | DEBUG | alltalk_tts_plugin.py:103 | _stream_audio | Attempt 1: Starting stream for text: akk... landing-denis | 2025-02-16 172544,430 | alltalk_tts | DEBUG | alltalk_tts_plugin.py:275 | _run | Processing text: ında... landing-denis | 2025-02-16 172544,430 | alltalk_tts | DEBUG | alltalk_tts_plugin.py:103 | _stream_audio | Attempt 1: Starting stream for text: ında... landing-denis | 2025-02-16 172544,760 | alltalk_tts | DEBUG | alltalk_tts_plugin.py:275 | _run | Processing text: bil... landing-denis | 2025-02-16 172544,760 | alltalk_tts | DEBUG | alltalk_tts_plugin.py:103 | _stream_audio | Attempt 1: Starting stream for text: bil... landing-denis | 2025-02-16 172544,985 | alltalk_tts | DEBUG | alltalk_tts_plugin.py:275 | _run | Processing t my tts engıne creatıng mp3 ant converıng thıs to pcm16 but i hear no sound why ?

It seems likely that the issue with not hearing any sound could be related to the conversion process from MP3 to PCM16 or the way the audio frames are being handled. Based on the LiveKit documentation, the expected audio format for LiveKit is typically PCM16. If the conversion from MP3 to PCM16 is not done correctly, it might result in audio frames that are not properly formatted for playback. Additionally, as per the LiveKit Slack discussion, ensure that the audio data is being correctly packed as

int16

rather than

float32

, as this is a common issue that can lead to distorted or inaudible audio. Would you like more detailed guidance on how to ensure the conversion and playback are handled correctly? Sources: - LiveKit documentation - LiveKit Slack discussion 👍 1 button 👎 0 button 💡 Hint: Mention @millions-hair-24956 in the thread for followups.

yes please

To address the issue of not hearing sound when converting MP3 to PCM16, let's break down the process step-by-step: 1. MP3 to PCM Conversion: Ensure that the conversion from MP3 to PCM16 is done correctly. The LiveKit documentation provides a class

Mp3StreamDecoder

that can be used to decode MP3 data into PCM audio. This class is designed to handle the conversion efficiently.

from livekit.agents.utils.codecs.mp3 import Mp3StreamDecoder

   decoder = Mp3StreamDecoder()
   pcm_frames = decoder.decode_chunk(mp3_data_chunk)

2. Audio Frame Handling: After conversion, ensure that the audio frames are correctly formatted as

int16

. This is crucial because LiveKit expects audio data in this format. If your audio data is in

float32

, you need to convert it to

int16

.

import numpy as np

   # Assuming `pcm_frames` is a list of audio frames
   for frame in pcm_frames:
       # Convert float32 to int16
       int16_data = (frame.data * 32767).astype(np.int16)
       # Create a new AudioFrame with int16 data
       audio_frame = rtc.AudioFrame(
           data=int16_data.tobytes(),
           sample_rate=frame.sample_rate,
           num_channels=frame.num_channels,
           samples_per_channel=frame.samples_per_channel
       )
       # Send or play the audio_frame

3. Playback Configuration: Ensure that the playback system is configured to handle PCM16 audio. This includes setting the correct sample rate and number of channels. Verify that the audio output device is functioning correctly and is not muted. 4. Debugging: Use logging to verify that the audio frames are being processed and sent correctly. Check for any errors or warnings in the logs that might indicate issues with the audio data or playback system. By following these steps, you should be able to ensure that the audio is correctly converted and played back. If issues persist, consider checking the audio output device settings and ensuring that the LiveKit configuration matches the expected audio format. Sources: - LiveKit documentation 👍 0 button 👎 0 button 💡 Hint: Mention @runllm in the thread for followups.

@millions-hair-24956 [AllTalk TTS] Starting TTS generation for file: stream_bd62974a5271.mp3 [AllTalk ENG] TTS Generate: 0.32 seconds. LowVRAM: False DeepSpeed: True [AllTalk TTS] Starting TTS generation for file: stream_527207d51dc7.mp3 [AllTalk ENG] TTS Generate: 0.23 seconds. LowVRAM: False DeepSpeed: True [AllTalk TTS] Starting TTS generation for file: stream_f34514f0fa44.mp3 [AllTalk ENG] TTS Generate: 0.32 seconds. LowVRAM: False DeepSpeed: True [AllTalk TTS] Starting TTS generation for file: stream_61a9ad400c97.mp3 [AllTalk ENG] TTS Generate: 0.24 seconds. LowVRAM: False DeepSpeed: True [AllTalk TTS] Starting TTS generation for file: stream_2e3d5472bdfc.mp3 i have mp3 s 1. MP3'ten PCM16'ya doğru dönüşüm 2. Float32'den int16'ya doğru scaling 3. Metin parçalamayı kaldırma 4. Frame oluşturma sürecinin iyileştirilmesi 5. Debug logging eklendi 6. Buffer yönetimi düzeltildi but ... i hear no sound. do i need to stream it with a parameter ? ding-denis | task: <Task pending name='Task-32' coro=<Event.wait() running at /usr/local/lib/python3.11/asyncio/locks.py:213> wait_for=<Future pending cb=[Task.task_wakeup()]>> landing-denis | {"message": "Task was destroyed but it is pending!\ntask: <Task pending name='Task-32' coro=<Event.wait() running at /usr/local/lib/python3.11/asyncio/locks.py:213> wait_for=<Future pending cb=[Task.task_wakeup()]>>", "level": "ERROR", "name": "asyncio", "pid": 18, "job_id": "AJ_B2ys569KnK8p", "timestamp": "2025-02-16T175800.116371+00:00"} landing-denis | 2025-02-16 175800,116 | alltalk_tts | INFO | alltalk_tts_plugin.py:277 | _run | Starting streaming TTS landing-denis | {"message": "Starting streaming TTS", "level": "INFO", "name": "alltalk_tts", "pid": 18, "job_id": "AJ_B2ys569KnK8p", "timestamp": "2025-02-16T175800.116578+00:00"} landing-denis | 2025-02-16 175800,129 | alltalk_tts | DEBUG | alltalk_tts_plugin.py:282 | _run | Processing text: Mer... landing-denis | 2025-02-16 175800,129 | alltalk_tts | DEBUG | alltalk_tts_plugin.py:104 | _stream_audio | Starting TTS stream for: Mer landing-denis | 2025-02-16 175800,471 | httpcore.http11 | DEBUG | _trace.py:87 | atrace | receive_response_body.complete landing-denis | 2025-02-16 175800,471 | httpcore.http11 | DEBUG | _trace.py:87 | atrace | response_closed.started landing-denis | 2025-02-16 175800,471 | httpcore.http11 | DEBUG | _trace.py:87 | atrace | response_closed.complete landing-denis | 2025-02-16 175800,555 | alltalk_tts | DEBUG | alltalk_tts_plugin.py:282 | _run | Processing text: hab... landing-denis | 2025-02-16 175800,555 | alltalk_tts | DEBUG | alltalk_tts_plugin.py:104 | _stream_audio | Starting TTS stream for: hab landing-denis | 2025-02-16 175800,556 | livekit.agents.pipeline | DEBUG | agent_playout.py:139 | _capture_task | speech playout started landing-denis | 2025-02-16 175800,887 | alltalk_tts | DEBUG | alltalk_tts_plugin.py:282 | _run | Processing text: a... landing-denis | 2025-02-16 175800,887 | alltalk_tts | DEBUG | alltalk_tts_plugin.py:104 | _stream_audio | Starting TTS stream for: a landing-denis | 2025-02-16 175801,296 | alltalk_tts | DEBUG | alltalk_tts_plugin.py:282 | _run | Processing text: !... landing-denis | 2025-02-16 175801,296 | alltalk_tts | DEBUG | alltalk_tts_plugin.py:104 | _stream_audio | Starting TTS stream for: ! landing-denis | 2025-02-16 175801,596 | alltalk_tts | DEBUG | alltalk_tts_plugin.py:282 | _run | Processing text: Size... landing-denis | 2025-02-16 175801,597 | alltalk_tts | DEBUG | alltalk_tts_plugin.py:104 | _stream_audio | Starting TTS stream for: Size i hear nothıng...

To address the issue of not hearing any sound, let's ensure that the entire process from MP3 conversion to PCM16 and streaming is correctly implemented. Here are the steps to troubleshoot and resolve the issue: 1. MP3 to PCM16 Conversion: Ensure that the MP3 files are being correctly converted to PCM16 format. Use the

Mp3StreamDecoder

class from the LiveKit documentation to decode MP3 data into PCM audio frames.

from livekit.agents.utils.codecs.mp3 import Mp3StreamDecoder

   decoder = Mp3StreamDecoder()
   pcm_frames = decoder.decode_chunk(mp3_data_chunk)

2. Data Type Conversion: Verify that the audio data is being converted from

float32

to

int16

. This is crucial because LiveKit expects audio data in

int16

format. If your audio data is in

float32

, convert it as follows:

import numpy as np

   # Assuming `pcm_frames` is a list of audio frames
   for frame in pcm_frames:
       # Convert float32 to int16
       int16_data = (frame.data * 32767).astype(np.int16)
       # Create a new AudioFrame with int16 data
       audio_frame = rtc.AudioFrame(
           data=int16_data.tobytes(),
           sample_rate=frame.sample_rate,
           num_channels=frame.num_channels,
           samples_per_channel=frame.samples_per_channel
       )
       # Send or play the audio_frame

3. Streaming Configuration: Ensure that the audio frames are being streamed correctly. If you are using LiveKit's streaming capabilities, make sure that the audio frames are being sent to the correct destination and that the playback system is configured to handle PCM16 audio. 4. Debugging and Logging: Use logging to verify that the audio frames are being processed and sent correctly. Check for any errors or warnings in the logs that might indicate issues with the audio data or playback system. The error message "Task was destroyed but it is pending" suggests that there might be an issue with the asynchronous task handling. Ensure that all tasks are properly awaited and that there are no pending tasks left unhandled. 5. Buffer Management: Ensure that the buffer management is correctly implemented. This includes handling the audio data in chunks and ensuring that the buffer is flushed correctly after processing each chunk. By following these steps, you should be able to ensure that the audio is correctly converted, streamed, and played back. If issues persist, consider checking the audio output device settings and ensuring that the LiveKit configuration matches the expected audio format. Sources: - LiveKit documentation 👍 0 button 👎 0 button 💡 Hint: Mention @runllm in the thread for followups.

@millions-hair-24956 mp3 fıles inside /home/ubuntu/alltalk_tts/outputs do we need catch mp3 and play it ımmedıately at lıvekıt ? for streamıng ? help meee or do we get mp3 at treamıng endpoınt?

To stream audio using LiveKit, you need to convert the MP3 files to a format compatible with LiveKit's streaming capabilities, typically PCM16, and then stream the audio frames in real-time. Here’s how you can approach this: 1. Convert MP3 to PCM16: Use a decoder to convert MP3 files to PCM16 format. This is necessary because LiveKit typically handles PCM16 audio data for streaming.

from livekit.agents.utils.codecs.mp3 import Mp3StreamDecoder

   decoder = Mp3StreamDecoder()
   with open('/home/ubuntu/alltalk_tts/outputs/yourfile.mp3', 'rb') as mp3_file:
       mp3_data = mp3_file.read()
       pcm_frames = decoder.decode_chunk(mp3_data)

2. Stream Audio Frames: Once you have the PCM16 audio frames, you can stream them using LiveKit. You need to create an audio track and publish it to a LiveKit room.

import asyncio
   from livekit import rtc

   async def stream_audio():
       # Create an audio source and track
       sample_rate = 48000  # Example sample rate
       num_channels = 2  # Example number of channels
       source = rtc.AudioSource(sample_rate, num_channels)
       track = rtc.LocalAudioTrack.create_audio_track("audio", source)

       # Connect to the LiveKit room and publish the track
       room = rtc.Room()
       await room.connect('your_livekit_url', 'your_participant_token')
       await room.local_participant.publish_track(track)

       # Stream the audio frames
       for frame in pcm_frames:
           await source.capture_frame(frame)
           await asyncio.sleep(frame.samples_per_channel / sample_rate)

   asyncio.run(stream_audio())

3. Immediate Playback: If you want to play the audio immediately, ensure that the audio frames are being sent to the LiveKit room in real-time. The playback will occur on the client side, where the audio track is subscribed to and played back. 4. Streaming Endpoint: If you are using a streaming endpoint, ensure that the endpoint is configured to receive and handle PCM16 audio data. The endpoint should be able to process the incoming audio frames and play them back in real-time. Would you like more detailed guidance on any specific part of this process? Sources: - LiveKit documentation 👍 0 button 👎 0 button 💡 Hint: Mention @runllm in the thread for followups.

no gneterated data sored as mp3 in my server , do i need to catch them from my server and play ıt to lıvekıt ? im so confused . because ı hear no sound at voice pipeline agent .

Yes, you need to fetch the generated MP3 files from your server, convert them to PCM16 format, and then stream them to LiveKit for playback. Here’s a step-by-step guide to achieve this: 1. Fetch MP3 Files: Retrieve the MP3 files stored on your server. You can do this by reading the files from the directory where they are stored. 2. Convert MP3 to PCM16: Use a decoder to convert the MP3 files to PCM16 format. This is necessary because LiveKit typically handles PCM16 audio data for streaming. You can use the

Mp3StreamDecoder

class from the LiveKit documentation to perform this conversion.

from livekit.agents.utils.codecs.mp3 import Mp3StreamDecoder

   decoder = Mp3StreamDecoder()
   with open('/path/to/your/mp3file.mp3', 'rb') as mp3_file:
       mp3_data = mp3_file.read()
       pcm_frames = decoder.decode_chunk(mp3_data)

3. Stream Audio to LiveKit: Once you have the PCM16 audio frames, you need to stream them to LiveKit. This involves creating an audio track and publishing it to a LiveKit room.

import asyncio
   from livekit import rtc

   async def stream_audio():
       # Create an audio source and track
       sample_rate = 48000  # Example sample rate
       num_channels = 2  # Example number of channels
       source = rtc.AudioSource(sample_rate, num_channels)
       track = rtc.LocalAudioTrack.create_audio_track("audio", source)

       # Connect to the LiveKit room and publish the track
       room = rtc.Room()
       await room.connect('your_livekit_url', 'your_participant_token')
       await room.local_participant.publish_track(track)

       # Stream the audio frames
       for frame in pcm_frames:
           await source.capture_frame(frame)
           await asyncio.sleep(frame.samples_per_channel / sample_rate)

   asyncio.run(stream_audio())

4. Playback on Client Side: Ensure that the client side is subscribed to the audio track being published. The playback will occur on the client side, where the audio track is subscribed to and played back. By following these steps, you should be able to stream the audio from your server to LiveKit and hear it on the client side. If you still encounter issues, ensure that the LiveKit configuration and client-side setup are correct. Sources: - LiveKit documentation 👍 0 button 👎 0 button 💡 Hint: Mention @runllm in the thread for followups.

@millions-hair-24956 its voice pipelibne agent , tts . mt problem ıs ı couldnt hear my file from livekit.

To resolve the issue of not hearing your TTS output in a LiveKit VoicePipelineAgent setup, ensure the following steps are correctly implemented: 1. TTS Output Handling: Ensure that the TTS output is being correctly converted to PCM16 format. The VoicePipelineAgent expects audio data in PCM16 format for streaming. If your TTS engine outputs MP3, you need to convert it to PCM16. 2. Audio Frame Creation: After conversion, create audio frames from the PCM16 data. Ensure that the audio frames are correctly formatted and that the sample rate and number of channels match the expected configuration. 3. Publishing Audio Frames: Use the LiveKit SDK to publish the audio frames to the room. Ensure that the audio source and track are correctly set up and that the frames are being captured and sent in real-time. 4. Client Subscription: Verify that the client is subscribed to the audio track being published by the VoicePipelineAgent. The client needs to be correctly configured to receive and play the audio track. 5. Debugging: Use logging to verify that the audio frames are being processed and sent correctly. Check for any errors or warnings in the logs that might indicate issues with the audio data or playback system. If you need more detailed guidance on any specific part of this process, please let me know. Sources: - https://livekit-users.slack.com/archives/C07FVFGAUKX/p1733152587.089089 - https://docs.livekit.io/home/ingress/overview/ 👍 0 button 👎 0 button 💡 Hint: Mention @runllm in the thread for followups.

which param creatıng hearable sound ın lıvekıt agent framework ? @millions-hair-24956

In the LiveKit agent framework, the key parameters that affect the creation of hearable sound are related to the audio track configuration and the format of the audio data being streamed. Here are the critical parameters and considerations: 1. Audio Format: Ensure that the audio data is in PCM16 format, as this is the expected format for LiveKit. The audio data should be 16-bit signed integers (

int16

), which is crucial for proper playback. 2. Sample Rate and Channels: The sample rate (e.g., 48000 Hz) and the number of channels (e.g., 1 for mono, 2 for stereo) must match the configuration expected by the LiveKit room and the client. These parameters are set when creating the

AudioSource

and

AudioFrame

. 3. Audio Track Publishing: When creating and publishing an audio track, ensure that the

LocalAudioTrack

is correctly set up with the

AudioSource

. The track must be published to the LiveKit room for participants to hear the sound. 4. Client Subscription: The client must be subscribed to the audio track. Ensure that the client application is configured to receive and play the audio track published by the agent. If you need more detailed guidance on configuring these parameters, please let me know. Sources: - LiveKit Slack discussion 👍 0 button 👎 0 button 💡 Hint: Mention @runllm in the thread for followups.

@millions-hair-24956 Audio Track Publishing: When creating and publishing an audio track, ensure that the

LocalAudioTrack

is correctly set up with the

AudioSource

. The track must be published to the LiveKit room for participants to hear the sound you said thıs , is tts output needs to be puplıched as audıo track or tts ıs dıfferent ?

Yes, the TTS (Text-to-Speech) output needs to be published as an audio track in LiveKit for participants to hear the sound. The process involves converting the TTS output into audio frames and then publishing these frames as a

LocalAudioTrack

to the LiveKit room. Here's how you can do it: 1. Convert TTS Output: If your TTS engine outputs audio in a format like MP3, you need to convert it to PCM16 format. This conversion is necessary because LiveKit typically handles PCM16 audio data. 2. Create Audio Source and Track: Use the converted PCM16 data to create an

AudioSource

and a

LocalAudioTrack

. The

AudioSource

is responsible for capturing audio frames, and the

LocalAudioTrack

is used to publish these frames to the room.

import asyncio
   from livekit import rtc

   async def publish_tts_audio(pcm_frames):
       # Create an audio source and track
       sample_rate = 48000  # Example sample rate
       num_channels = 1  # Example number of channels
       source = rtc.AudioSource(sample_rate, num_channels)
       track = rtc.LocalAudioTrack.create_audio_track("tts-audio", source)

       # Connect to the LiveKit room and publish the track
       room = rtc.Room()
       await room.connect('your_livekit_url', 'your_participant_token')
       await room.local_participant.publish_track(track)

       # Stream the audio frames
       for frame in pcm_frames:
           await source.capture_frame(frame)
           await asyncio.sleep(frame.samples_per_channel / sample_rate)

   asyncio.run(publish_tts_audio(pcm_frames))

3. Client Subscription: Ensure that the client is subscribed to the audio track. The client application should be configured to receive and play the audio track published by the agent. By following these steps, you can ensure that the TTS output is correctly published as an audio track in LiveKit, allowing participants to hear the sound. Sources: - https://docs.livekit.io/agents/plugins/#VAD-and-StreamAdapter - https://docs.livekit.io/agents/plugins/overview/#VAD-and-StreamAdapter 👍 0 button 👎 0 button 💡 Hint: Mention @runllm in the thread for followups.

@millions-hair-24956 SynthesizeStream._run() for alltalk tts ?

The

SynthesizeStream._run()

method is part of the process for handling text-to-speech (TTS) streaming in LiveKit. This method is responsible for managing the streaming of synthesized audio data. Here's a general breakdown of how it might work in the context of a TTS plugin like AllTalk: 1. Initialization: The

SynthesizeStream

class is initialized with a TTS engine and text input. It sets up channels for input and output, and prepares to handle audio frames. 2. Streaming Process: The

_run()

method is typically responsible for the main loop that processes text input and generates audio frames. It reads text, synthesizes it into audio, and sends the audio frames to the output channel. 3. Audio Frame Handling: As audio frames are generated, they are sent to the

AudioSource

for capture and then published as part of a

LocalAudioTrack

. This allows the audio to be streamed to a LiveKit room. 4. Error Handling and Logging: The method includes error handling to manage any issues that arise during synthesis, and logging to track the process and debug if necessary. If you need more specific details about implementing or troubleshooting

SynthesizeStream._run()

for AllTalk TTS, please let me know. Sources: - https://docs.livekit.io/agents/plugins/#VAD-and-StreamAdapter - https://livekit-users.slack.com/archives/C07FY8WHGPM/p1732302048.360549 👍 0 button 👎 0 button 💡 Hint: Mention @runllm in the thread for followups.

@millions-hair-24956 File \"/usr/local/lib/python3.11/site-packages/livekit/agents/tts/tts.py\", line 83, in stream\n raise NotImplementedError(\nNotImplementedError: streaming is not supported by this TTS, please use a different TTS or use a StreamAdapter",

The error message indicates that the TTS engine you are using does not support streaming directly. To resolve this, you can use a

StreamAdapter

to enable streaming capabilities for TTS engines that do not natively support it. Here's how you can do it: 1. Use a StreamAdapter: The

StreamAdapter

class is designed to wrap around TTS engines that do not support streaming, allowing them to be used in a streaming context. This involves creating a

StreamAdapter

instance with your TTS engine.

from livekit.agents.tts import StreamAdapter
   from livekit.plugins.your_tts_plugin import YourTTS

   # Initialize your TTS engine
   tts_engine = YourTTS()

   # Wrap it with a StreamAdapter
   stream_adapter = StreamAdapter(tts=tts_engine, sentence_tokenizer=tokenize.basic.SentenceTokenizer())

2. Stream Audio: Use the

stream()

method of the

StreamAdapter

to handle text input and generate audio frames for streaming.

async def stream_audio(text):
       synthesize_stream = stream_adapter.stream()
       synthesize_stream.push_text(text)
       synthesize_stream.end_input()

       async for audio_frame in synthesize_stream:
           # Handle the audio frame, e.g., send it to a LiveKit room
           pass

3. Publish Audio: Ensure that the audio frames are published to a LiveKit room as part of a

LocalAudioTrack

, allowing participants to hear the synthesized speech. By using a

StreamAdapter

, you can enable streaming for TTS engines that do not natively support it, allowing you to integrate them into your LiveKit setup. Sources: - LiveKit documentation on StreamAdapter 👍 1 button 👎 0 button 💡 Hint: Mention @millions-hair-24956 in the thread for followups.