huggingface
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         title: Bark
       - local: model_doc/clap
         title: CLAP
+      - local: model_doc/csm
+        title: CSM
       - local: model_doc/dac
         title: dac
       - local: model_doc/encodec
 
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+<!--Copyright 2025 The HuggingFace Team. All rights reserved.
+
+Licensed under the Apache License, Version 2.0 (the "License"); you may not use this file except in compliance with
+the License. You may obtain a copy of the License at
+
+http://www.apache.org/licenses/LICENSE-2.0
+
+Unless required by applicable law or agreed to in writing, software distributed under the License is distributed on
+an "AS IS" BASIS, WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. See the License for the
+specific language governing permissions and limitations under the License.
+
+⚠️ Note that this file is in Markdown but contain specific syntax for our doc-builder (similar to MDX) that may not be
+rendered properly in your Markdown viewer.
+
+-->
+
+# Csm
+
+## Overview
+
+The Conversational Speech Model (CSM) is the first open-source contextual text-to-speech model [released by Sesame](https://www.sesame.com/research/crossing_the_uncanny_valley_of_voice). It is designed to generate natural-sounding speech with or without conversational context. This context typically consists of multi-turn dialogue between speakers, represented as sequences of text and corresponding spoken audio.
+
+**Model Architecture:**
+CSM is composed of two LLaMA-style auto-regressive transformer decoders: a backbone decoder that predicts the first codebook token and a depth decoder that generates the remaining tokens. It uses the pretrained codec model [Mimi](./mimi.md), introduced by Kyutai, to encode speech into discrete codebook tokens and decode them back into audio.
+
+The original csm-1b checkpoint is available under the [Sesame](https://huggingface.co/sesame/csm-1b) organization on Hugging Face.
+
+<div class="flex justify-center">
+    <img src="https://huggingface.co/datasets/eustlb/documentation-images/resolve/main/csm_architecture.png"/>
+</div>
+
+## Usage Tips
+
+### Without Conversational Context
+
+CSM can be used to simply generate speech from a text prompt:
+
+```python
+import torch
+from transformers import CsmForConditionalGeneration, AutoProcessor
+
+model_id = "eustlb/csm-1b"
+device = "cuda" if torch.cuda.is_available() else "cpu"
+
+# load the model and the processor
+processor = AutoProcessor.from_pretrained(model_id)
+model = CsmForConditionalGeneration.from_pretrained(model_id, device_map=device)
+
+# prepare the inputs
+text = "[0]The past is just a story we tell ourselves." # `[0]` for speaker id 0
+inputs = processor(text, add_special_tokens=True).to(device)
+
+# another equivalent way to prepare the inputs
+conversation = [
+    {"role": "0", "content": [{"type": "text", "text": "The past is just a story we tell ourselves."}]},
+]
+inputs = processor.apply_chat_template(
+    conversation,
+    tokenize=True,
+    return_dict=True,
+).to(device)
+
+# infer the model
+audio = model.generate(**inputs, output_audio=True)
+processor.save_audio(audio, "example_without_context.wav")
+```
+
+### With Conversational Context
+
+CSM can be used to generate speech given a conversation, allowing consistency in the voices and content-aware generation:
+
+```python
+import torch
+from transformers import CsmForConditionalGeneration, AutoProcessor
+from datasets import load_dataset, Audio
+
+model_id = "eustlb/csm-1b"
+device = "cuda" if torch.cuda.is_available() else "cpu"
+
+# load the model and the processor
+processor = AutoProcessor.from_pretrained(model_id)
+model = CsmForConditionalGeneration.from_pretrained(model_id, device_map=device)
+
+# prepare the inputs
+ds = load_dataset("hf-internal-testing/dailytalk-dummy", split="train")
+# ensure the audio is 24kHz
+ds = ds.cast_column("audio", Audio(sampling_rate=24000))
+conversation = []
+
+# 1. context
+for text, audio, speaker_id in zip(ds[:4]["text"], ds[:4]["audio"], ds[:4]["speaker_id"]):
+    conversation.append(
+        {
+            "role": f"{speaker_id}",
+            "content": [{"type": "text", "text": text}, {"type": "audio", "path": audio["array"]}],
+        }
+    )
+
+# 2. text prompt
+conversation.append({"role": f"{ds[4]['speaker_id']}", "content": [{"type": "text", "text": ds[4]["text"]}]})
+
+inputs = processor.apply_chat_template(
+    conversation,
+    tokenize=True,
+    return_dict=True,
+).to(device)
+
+# infer the model
+audio = model.generate(**inputs, output_audio=True)
+processor.save_audio(audio, "example_with_context.wav")
+```
+
+### Batched Inference
+
+CSM supports batched inference!
+
+```python
+import torch
+from transformers import CsmForConditionalGeneration, AutoProcessor
+from datasets import load_dataset, Audio
+
+model_id = "eustlb/csm-1b"
+device = "cuda" if torch.cuda.is_available() else "cpu"
+
+# load the model and the processor
+processor = AutoProcessor.from_pretrained(model_id)
+model = CsmForConditionalGeneration.from_pretrained(model_id, device_map=device)
+
+# prepare the inputs 
+ds = load_dataset("hf-internal-testing/dailytalk-dummy", split="train")
+# ensure the audio is 24kHz
+ds = ds.cast_column("audio", Audio(sampling_rate=24000))
+# here a batch with two prompts
+conversation = [
+    [
+        {
+            "role": f"{ds[0]['speaker_id']}",
+            "content": [
+                {"type": "text", "text": ds[0]["text"]},
+                {"type": "audio", "path": ds[0]["audio"]["array"]},
+            ],
+        },
+        {
+            "role": f"{ds[1]['speaker_id']}",
+            "content": [
+                {"type": "text", "text": ds[1]["text"]},
+            ],
+        },
+    ],
+    [
+        {
+            "role": f"{ds[0]['speaker_id']}",
+            "content": [
+                {"type": "text", "text": ds[0]["text"]},
+            ],
+        }
+    ],
+]
+inputs = processor.apply_chat_template(
+    conversation,
+    tokenize=True,
+    return_dict=True,
+).to(device)
+
+audio = model.generate(**inputs, output_audio=True)
+processor.save_audio(audio, [f"speech_batch_idx_{i}.wav" for i in range(len(audio))])
+```
+
+### Making The Model Go Brrr
+
+CSM supports full-graph compilation with CUDA graphs!
+
+```python
+import torch
+import copy
+from transformers import CsmForConditionalGeneration, AutoProcessor
+from datasets import load_dataset
+
+model_id = "eustlb/csm-1b"
+device = "cuda"
+
+# set logs to ensure no recompilation and graph breaks
+torch._logging.set_logs(graph_breaks=True, recompiles=True, cudagraphs=True)
+
+# load the model and the processor
+processor = AutoProcessor.from_pretrained(model_id)
+model = CsmForConditionalGeneration.from_pretrained(model_id, device_map=device)
+
+# use static cache, enabling automatically torch compile with fullgraph and reduce-overhead
+model.generation_config.max_length = 250 # big enough to avoid recompilation
+model.generation_config.max_new_tokens = None # would take precedence over max_length
+model.generation_config.cache_implementation = "static"
+model.depth_decoder.generation_config.cache_implementation = "static"
+
+# generation kwargs
+gen_kwargs = {
+    "do_sample": False,
+    "depth_decoder_do_sample": False,
+    "temperature": 1.0,
+    "depth_decoder_temperature": 1.0,
+}
+
+# Define a timing decorator
+class TimerContext:
+    def __init__(self, name="Execution"):
+        self.name = name
+        self.start_event = None
+        self.end_event = None
+        
+    def __enter__(self):
+        # Use CUDA events for more accurate GPU timing
+        self.start_event = torch.cuda.Event(enable_timing=True)
+        self.end_event = torch.cuda.Event(enable_timing=True)
+        self.start_event.record()
+        return self
+
+    def __exit__(self, *args):
+        self.end_event.record()
+        torch.cuda.synchronize()
+        elapsed_time = self.start_event.elapsed_time(self.end_event) / 1000.0
+        print(f"{self.name} time: {elapsed_time:.4f} seconds")
+
+# prepare the inputs 
+ds = load_dataset("hf-internal-testing/dailytalk-dummy", split="train")
+
+conversation = [
+    {
+        "role": f"{ds[0]['speaker_id']}",
+        "content": [
+            {"type": "text", "text": ds[0]["text"]},
+            {"type": "audio", "path": ds[0]["audio"]["array"]},
+        ],
+    },
+    {
+        "role": f"{ds[1]['speaker_id']}",
+        "content": [
+            {"type": "text", "text": ds[1]["text"]},
+            {"type": "audio", "path": ds[1]["audio"]["array"]},
+        ],
+    },
+    {
+        "role": f"{ds[2]['speaker_id']}",
+        "content": [
+            {"type": "text", "text": ds[2]["text"]},
+        ],
+    },
+]
+
+padded_inputs_1 = processor.apply_chat_template(
+    conversation,
+    tokenize=True,
+    return_dict=True,
+).to(device)
+
+print("\n" + "="*50)
+print("First generation - compiling and recording CUDA graphs...")
+with TimerContext("First generation"):
+    _ = model.generate(**padded_inputs_1, **gen_kwargs)
+print("="*50)
+
+print("\n" + "="*50)
+print("Second generation - fast !!!")
+with TimerContext("Second generation"):
+    _ = model.generate(**padded_inputs_1, **gen_kwargs)
+print("="*50)
+
+# now with different inputs
+conversation = [
+    {
+        "role": f"{ds[0]['speaker_id']}",
+        "content": [
+            {"type": "text", "text": ds[2]["text"]},
+            {"type": "audio", "path": ds[2]["audio"]["array"]},
+        ],
+    },
+    {
+        "role": f"{ds[1]['speaker_id']}",
+        "content": [
+            {"type": "text", "text": ds[3]["text"]},
+            {"type": "audio", "path": ds[3]["audio"]["array"]},
+        ],
+    },
+    {
+        "role": f"{ds[2]['speaker_id']}",
+        "content": [
+            {"type": "text", "text": ds[4]["text"]},
+        ],
+    },
+]
+padded_inputs_2 = processor.apply_chat_template(
+    conversation,
+    tokenize=True,
+    return_dict=True,
+).to(device)
+
+print("\n" + "="*50)
+print("Generation with other inputs!")
+with TimerContext("Generation with different inputs"):
+    _ = model.generate(**padded_inputs_2, **gen_kwargs)
+print("="*50)
+```
+
+### Training
+
+CSM Transformers integration supports training!
+
+```python
+from transformers import CsmForConditionalGeneration, AutoProcessor
+from datasets import load_dataset, Audio
+
+model_id = "eustlb/csm-1b"
+device = "cuda"
+
+# load the model and the processor
+processor = AutoProcessor.from_pretrained(model_id)
+model = CsmForConditionalGeneration.from_pretrained(model_id, device_map=device)
+model.train()
+
+ds = load_dataset("hf-internal-testing/dailytalk-dummy", split="train")
+# ensure the audio is 24kHz
+ds = ds.cast_column("audio", Audio(sampling_rate=24000))
+conversation = []
+
+# context
+for text, audio, speaker_id in zip(ds[:4]["text"], ds[:4]["audio"], ds[:4]["speaker_id"]):
+    conversation.append(
+        {
+            "role": f"{speaker_id}",
+            "content": [{"type": "text", "text": text}, {"type": "audio", "path": audio["array"]}],
+        }
+    )
+
+inputs = processor.apply_chat_template(
+    conversation,
+    tokenize=True,
+    return_dict=True,
+    output_labels=True,
+).to(device)
+
+out = model(**inputs)
+out.loss.backward()
+```
+
+This model was contributed by [Eustache Le Bihan](https://huggingface.co/eustlb).
+The original code can be found [here](https://github.com/SesameAILabs/csm).
+
+
+## CsmConfig
+
+[[autodoc]] CsmConfig
+
+## CsmDepthDecoderConfig
+
+[[autodoc]] CsmDepthDecoderConfig
+
+## CsmProcessor
+
+[[autodoc]] CsmProcessor
+    - __call__
+
+## CsmForConditionalGeneration
+
+[[autodoc]] CsmForConditionalGeneration
+    - forward
+    - generate
+
+## CsmDepthDecoderForCausalLM
+
+[[autodoc]] CsmDepthDecoderForCausalLM
+
+## CsmDepthDecoderModel
+
+[[autodoc]] CsmDepthDecoderModel
+
+## CsmBackboneModel
+
+[[autodoc]] CsmBackboneModel