Add the en-fr config.
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README.md
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README.md
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@ -3,8 +3,8 @@ Delayed Streams Modeling (DSM) is a flexible formulation for streaming, multimod
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## Speech To Text
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## Speech To Text
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### Leaderboard model
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### English only model
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The leaderboard model handles english only, it has ~2.6B parameters.
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The main model handles english only, it has ~2.6B parameters.
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#### PyTorch implementation
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#### PyTorch implementation
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[[Hugging Face]](https://huggingface.co/kyutai/stt-2.6B-en)
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[[Hugging Face]](https://huggingface.co/kyutai/stt-2.6B-en)
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@ -55,6 +55,19 @@ The script simulates some real-time processing of the audio. Faster processing
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can be triggered by setting the real-time factor, e.g. `--rtf 500` will process
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can be triggered by setting the real-time factor, e.g. `--rtf 500` will process
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the data as fast as possible.
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the data as fast as possible.
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### English + French model
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This model has ~1B parameters and supports both English and French.
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#### Rust implementation
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[[Hugging Face]](https://huggingface.co/kyutai/stt-1B-en_fr-candle)
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The only difference with the en only model is the config file used when
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launching the server.
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```bash
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moshi-server worker --config configs/config-stt-enfr-hf.toml
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```
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## Text To Speech
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## Text To Speech
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We're in the process of open-sourcing our TTS models. Check back for updates!
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We're in the process of open-sourcing our TTS models. Check back for updates!
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configs/config-stt-enfr-hf.toml
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configs/config-stt-enfr-hf.toml
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@ -0,0 +1,46 @@
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static_dir = "./static/"
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log_dir = "$HOME/tmp/tts-logs"
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instance_name = "tts"
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authorized_ids = ["open_token"]
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[modules.asr]
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path = "/api/asr-streaming"
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type = "BatchedAsr"
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lm_model_file = "hf://kyutai/stt-1B-en_fr-candle/model.safetensors"
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text_tokenizer_file = "hf://kyutai/stt-1B-en_fr-candle/tokenizer_en_fr_audio_8000.model"
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audio_tokenizer_file = "hf://kyutai/stt-1B-en_fr-candle/mimi-pytorch-e351c8d8@125.safetensors"
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asr_delay_in_tokens = 6
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batch_size = 64
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conditioning_learnt_padding = true
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temperature = 0.0
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[modules.asr.model]
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audio_vocab_size = 2049
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text_in_vocab_size = 8001
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text_out_vocab_size = 8000
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audio_codebooks = 32
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[modules.asr.model.transformer]
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d_model = 2048
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num_heads = 16
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num_layers = 16
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dim_feedforward = 8192
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causal = true
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norm_first = true
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bias_ff = false
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bias_attn = false
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context = 750
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max_period = 100000
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use_conv_block = false
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use_conv_bias = true
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gating = "silu"
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norm = "RmsNorm"
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positional_embedding = "Rope"
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conv_layout = false
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conv_kernel_size = 3
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kv_repeat = 1
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max_seq_len = 40960
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[modules.asr.model.extra_heads]
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num_heads = 4
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dim = 6
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