File size: 26,851 Bytes
53ea588
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
371
372
373
374
375
376
377
378
379
380
381
382
383
384
385
386
387
388
389
390
391
392
393
394
395
396
397
398
399
400
401
402
403
404
405
406
407
408
409
410
411
412
413
414
415
416
417
418
419
420
421
422
423
424
425
426
427
428
429
430
431
432
433
434
435
436
437
438
439
440
441
442
443
444
445
446
447
448
449
450
451
452
453
454
455
456
457
458
459
460
461
462
463
464
465
466
467
468
469
470
471
472
473
474
475
476
477
478
479
480
481
482
483
484
485
486
487
488
489
490
491
492
493
494
495
496
497
498
499
500
501
502
503
504
505
506
507
508
509
510
511
512
513
514
515
516
517
518
519
520
521
522
523
524
525
526
527
528
529
530
531
532
533
534
535
536
537
538
539
540
541
542
543
544
545
546
547
548
549
550
551
552
553
554
555
556
557
558
559
560
561
562
563
564
565
566
567
568
569
570
571
572
573
574
575
576
577
578
579
580
581
582
583
584
585
586
587
588
589
590
591
592
593
594
595
596
597
598
599
600
601
602
603
604
605
606
607
608
609
610
611
612
613
# SPDX-FileCopyrightText: Copyright (c) 2024-2025 NVIDIA CORPORATION & AFFILIATES. All rights reserved.
# SPDX-License-Identifier: BSD 2-Clause License

"""NVIDIA Riva speech services implementation.

This module provides integration with NVIDIA Riva's speech services, including:
- Text-to-Speech (TTS) with support for multiple voices and languages
- Automatic Speech Recognition (ASR) with streaming capabilities

The services can be configured to use either a local Riva Speech Server or
NVIDIA's cloud-hosted models through NVCF.

For documentation on how to configure the Riva Speech models, please refer to the
[Riva Speech Quick Start Guide](https://docs.nvidia.com/deeplearning/riva/user-guide/docs/quick-start-guide.html).
"""

import asyncio
import concurrent.futures
from collections.abc import AsyncGenerator
from pathlib import Path

import riva.client
from loguru import logger
from pipecat.audio.vad.vad_analyzer import VADState
from pipecat.frames.frames import (
    CancelFrame,
    EndFrame,
    Frame,
    StartFrame,
    StartInterruptionFrame,
    StopInterruptionFrame,
    TranscriptionFrame,
    TTSAudioRawFrame,
    TTSStartedFrame,
    TTSStoppedFrame,
    TTSTextFrame,
    UserStartedSpeakingFrame,
    UserStoppedSpeakingFrame,
)
from pipecat.services.stt_service import STTService
from pipecat.services.tts_service import TTSService
from pipecat.transcriptions.language import Language
from pipecat.utils.text.base_text_aggregator import BaseTextAggregator
from pipecat.utils.time import time_now_iso8601
from riva.client.proto.riva_audio_pb2 import AudioEncoding

from nvidia_pipecat.frames.riva import RivaInterimTranscriptionFrame
from nvidia_pipecat.utils.tracing import AttachmentStrategy, traceable, traced


@traceable
class RivaTTSService(TTSService):
    """NVIDIA Riva Text-to-Speech service implementation.

    Provides speech synthesis using NVIDIA's Riva TTS models with support for
    multiple voices, languages, and custom dictionaries.
    """

    def __init__(
        self,
        *,
        api_key: str | None = None,
        server: str = "grpc.nvcf.nvidia.com:443",
        voice_id: str = "English-US.Female-1",
        sample_rate: int = 16000,
        function_id: str = "0149dedb-2be8-4195-b9a0-e57e0e14f972",
        language: Language | None = Language.EN_US,
        zero_shot_quality: int | None = 20,
        model: str = "fastpitch-hifigan-tts",
        custom_dictionary: dict | None = None,
        encoding: AudioEncoding = AudioEncoding.LINEAR_PCM,
        zero_shot_audio_prompt_file: Path | None = None,
        audio_prompt_encoding: AudioEncoding = AudioEncoding.ENCODING_UNSPECIFIED,
        use_ssl: bool = False,
        text_aggregator: BaseTextAggregator | None = None,
        **kwargs,
    ):
        """Initializes the Riva TTS service.

        Args:
            api_key (str | None, optional): API key for authentication. Defaults to None.
            server (str, optional): Server address for Riva service. Defaults to "grpc.nvcf.nvidia.com:443".
            voice_id (str, optional): Voice identifier. Defaults to "English-US.Female-1".
            sample_rate (int, optional): Audio sample rate in Hz. Defaults to 16000.
            function_id (str, optional): Function identifier for the service.
                Defaults to "0149dedb-2be8-4195-b9a0-e57e0e14f972".
            language (Language | None, optional): Language for synthesis. Defaults to Language.EN_US.
            zero_shot_quality (int | None, optional): Quality level for synthesis. Defaults to 20.
            model (str, optional): Model name for synthesis. Defaults to "fastpitch-hifigan-tts".
            custom_dictionary (dict | None, optional): Custom pronunciation dictionary. Defaults to None.
            encoding (AudioEncoding, optional): Audio encoding format. Defaults to AudioEncoding.LINEAR_PCM.
            zero_shot_audio_prompt_file (str | None, optional): Path to audio prompt file. Defaults to None.
            audio_prompt_encoding (AudioEncoding, optional): Encoding of audio prompt.
                Defaults to AudioEncoding.LINEAR_PCM.
            use_ssl (bool, optional): Whether to use SSL for connection. Defaults to False.
            text_aggregator (BaseTextAggregator | None, optional): Text aggregator for sentence detection.
                Defaults to None, which uses SimpleTextAggregator.
            **kwargs: Additional keyword arguments passed to parent class.

        Raises:
            Exception: If required modules are missing or connection fails.

        Usage:
            If server is not set then it defaults to "grpc.nvcf.nvidia.com:443" and use NVCF hosted models.
            Update function ID to use a different NVCF model. API key is required for NVCF hosted models.
            For using locally deployed Riva Speech Server, set server to "localhost:50051" and
            follow the quick start guide to setup the server.
        """
        super().__init__(
            sample_rate=sample_rate,
            push_text_frames=False,
            push_stop_frames=True,
            text_aggregator=text_aggregator,
            **kwargs,
        )
        self._api_key = api_key
        self._function_id = function_id
        self._voice_id = voice_id
        self._sample_rate = sample_rate
        self._language_code = language
        self._zero_shot_quality = zero_shot_quality
        self.set_model_name(model)
        self.set_voice(voice_id)
        self._custom_dictionary = custom_dictionary
        self._encoding = encoding
        self._zero_shot_audio_prompt_file = zero_shot_audio_prompt_file
        self._audio_prompt_encoding = audio_prompt_encoding

        metadata = [
            ["function-id", function_id],
            ["authorization", f"Bearer {api_key}"],
        ]

        if server == "grpc.nvcf.nvidia.com:443":
            use_ssl = True

        try:
            auth = riva.client.Auth(None, use_ssl, server, metadata)
            self._service = riva.client.SpeechSynthesisService(auth)
            # warm up the service
            _ = self._service.stub.GetRivaSynthesisConfig(riva.client.proto.riva_tts_pb2.RivaSynthesisConfigRequest())
        except Exception as e:
            logger.error(
                "In order to use nvidia Riva TTSService or STTService, you will either need a locally "
                "deployed Riva Speech Server with ASR and TTS models (Follow riva quick start guide at "
                "https://docs.nvidia.com/deeplearning/riva/user-guide/docs/quick-start-guide.html and "
                "edit the config file to deploy which model you want to use and set the server url to "
                "localhost:50051), or you can set the NVIDIA_API_KEY environment "
                "variable to connect with nvcf hosted models."
            )
            raise Exception(f"Missing module: {e}") from e

    def can_generate_metrics(self) -> bool:
        """Check if the service can generate metrics.

        Returns:
            bool: True as this service supports metric generation.
        """
        return True

    async def _push_tts_frames(self, text: str):
        """Override base class method to push text frames immediately."""
        # Remove leading newlines only
        text = text.lstrip("\n")

        # Don't send only whitespace. This causes problems for some TTS models. But also don't
        # strip all whitespace, as whitespace can influence prosody.
        if not text.strip():
            return

        # This is just a flag that indicates if we sent something to the TTS
        # service. It will be cleared if we sent text because of a TTSSpeakFrame
        # or when we received an LLMFullResponseEndFrame
        self._processing_text = True

        await self.start_processing_metrics()
        # Process all filter.
        for filter in self._text_filters:
            filter.reset_interruption()
            text = filter.filter(text)

        if text:
            await self.process_generator(self.run_tts(text))
        await self.stop_processing_metrics()

    @traced(attachment_strategy=AttachmentStrategy.NONE, name="tts")
    async def run_tts(self, text: str) -> AsyncGenerator[Frame, None]:
        """Run text-to-speech synthesis."""
        # Check if text contains any alphanumeric characters
        if not any(c.isalnum() for c in text):
            logger.debug(f"Skipping TTS for text with no alphanumeric characters: [{text}]")
            return
        logger.debug(f"Generating TTS: [{text.strip()}]")
        responses = self._service.synthesize_online(
            text.strip(),
            self._voice_id,
            self._language_code,
            sample_rate_hz=self._sample_rate,
            zero_shot_audio_prompt_file=self._zero_shot_audio_prompt_file,
            audio_prompt_encoding=self._audio_prompt_encoding,
            zero_shot_quality=self._zero_shot_quality,
            custom_dictionary=self._custom_dictionary,
            encoding=self._encoding,
        )

        await self.start_ttfb_metrics()
        yield TTSStartedFrame()

        # Push text frame immediately after TTSStartedFrame.
        # TTSService base processor will push the tts text after sending generated tts audio downstream
        # Need to push the text before audio frame for better TTS transcription.
        yield TTSTextFrame(text)

        async def get_next_response(iterator):
            def _next():
                try:
                    return next(iterator)
                except StopIteration:
                    return None

            return await asyncio.get_event_loop().run_in_executor(None, _next)

        response_iterator = iter(responses)
        total_audio_length = 0

        while (resp := await get_next_response(response_iterator)) is not None:
            try:
                total_audio_length += len(resp.audio)
                await self.stop_ttfb_metrics()
                frame = TTSAudioRawFrame(
                    audio=resp.audio,
                    sample_rate=self._sample_rate,
                    num_channels=1,
                )
                yield frame
            except Exception as e:
                logger.error(f"{self} Error processing TTS response: {e}")
                break

        await self.start_tts_usage_metrics(text)
        logger.debug(f"Total generated TTS audio length: {total_audio_length / (self._sample_rate * 2)} seconds")
        yield TTSStoppedFrame()


@traceable
class RivaASRService(STTService):
    """NVIDIA Riva Automatic Speech Recognition service.

    Provides streaming speech recognition using Riva ASR models with support for:
        - Real-time transcription
        - Interim results
        - Interruption handling
        - Voice activity detection
        - Language model customization
    """

    def __init__(
        self,
        *,
        api_key: str | None = None,
        server: str = "grpc.nvcf.nvidia.com:443",
        function_id: str = "1598d209-5e27-4d3c-8079-4751568b1081",
        language: Language | None = Language.EN_US,
        model: str = "parakeet-1.1b-en-US-asr-streaming-asr-bls-ensemble",
        profanity_filter: bool = False,
        automatic_punctuation: bool = False,
        no_verbatim_transcripts: bool = True,
        boosted_lm_words: dict | None = None,
        boosted_lm_score: float = 4.0,
        start_history: int = -1,
        start_threshold: float = -1.0,
        stop_history: int = 500,
        stop_threshold: float = -1.0,
        stop_history_eou: int = 240,
        stop_threshold_eou: float = -1.0,
        custom_configuration: str = "enable_vad_endpointing:true,neural_vad.onset:0.65,apply_partial_itn:true",
        sample_rate: int = 16000,
        audio_channel_count: int = 1,
        max_alternatives: int = 1,
        interim_results: bool = True,
        generate_interruptions: bool = False,  # Only set to True if transport VAD is disabled
        idle_timeout: int = 30,  # Timeout for idle Riva ASR request
        use_ssl: bool = False,
        **kwargs,
    ):
        """Initializes the Riva ASR service.

        Args:
            api_key: NVIDIA API key for cloud access.
            server: Riva server address.
            function_id: NVCF function identifier.
            language: Language for recognition.
            model: ASR model name.
            profanity_filter: Enable profanity filtering.
            automatic_punctuation: Enable automatic punctuation.
            no_verbatim_transcripts: Disable verbatim transcripts.
            boosted_lm_words: Words to boost in language model.
            boosted_lm_score: Score for boosted words.
            start_history: VAD start history frames.
            start_threshold: VAD start threshold.
            stop_history: VAD stop history frames.
            stop_threshold: VAD stop threshold.
            stop_history_eou: End-of-utterance history frames.
            stop_threshold_eou: End-of-utterance threshold.
            custom_configuration: Additional configuration string.
            sample_rate: Audio sample rate in Hz.
            audio_channel_count: Number of audio channels.
            max_alternatives: Maximum number of alternatives.
            interim_results: Enable interim results.
            generate_interruptions: Enable interruption events.
            idle_timeout: Timeout for idle ASR request in seconds.
            use_ssl: Enable SSL connection.
            **kwargs: Additional arguments for STTService.

        Usage:
            If server is not set then it defaults to "grpc.nvcf.nvidia.com:443" and use NVCF hosted models.
            Update function ID to use a different NVCF model. API key is required for NVCF hosted models.
            For using locally deployed Riva Speech Server, set server to "localhost:50051" and
            follow the quick start guide to setup the server.
        """
        super().__init__(**kwargs)
        self._profanity_filter = profanity_filter
        self._automatic_punctuation = automatic_punctuation
        self._no_verbatim_transcripts = no_verbatim_transcripts
        self._language_code = language
        self._boosted_lm_words = boosted_lm_words
        self._boosted_lm_score = boosted_lm_score
        self._start_history = start_history
        self._start_threshold = start_threshold
        self._stop_history = stop_history
        self._stop_threshold = stop_threshold
        self._stop_history_eou = stop_history_eou
        self._stop_threshold_eou = stop_threshold_eou
        self._custom_configuration = custom_configuration
        self._sample_rate: int = sample_rate
        self._model = model
        self._audio_channel_count = audio_channel_count
        self._max_alternatives = max_alternatives
        self._interim_results = interim_results
        self._idle_timeout = idle_timeout
        self.last_transcript_frame = None
        self.set_model_name(model)

        metadata = [
            ["function-id", function_id],
            ["authorization", f"Bearer {api_key}"],
        ]

        if server == "grpc.nvcf.nvidia.com:443":
            use_ssl = True

        try:
            auth = riva.client.Auth(None, use_ssl, server, metadata)
            self._asr_service = riva.client.ASRService(auth)
        except Exception as e:
            logger.error(
                "In order to use nvidia Riva TTSService or STTService, you will either need a locally "
                "deployed Riva Speech Server with ASR and TTS models (Follow riva quick start guide at "
                "https://docs.nvidia.com/deeplearning/riva/user-guide/docs/quick-start-guide.html and "
                "edit the config file to deploy which model you want to use and set the server url to "
                "localhost:50051), or you can set the NVIDIA_API_KEY environment "
                "variable to connect with nvcf hosted models."
            )
            raise Exception(f"Missing module: {e}") from e

        config = riva.client.StreamingRecognitionConfig(
            config=riva.client.RecognitionConfig(
                encoding=riva.client.AudioEncoding.LINEAR_PCM,
                language_code=self._language_code,
                model=self._model,
                max_alternatives=self._max_alternatives,
                profanity_filter=self._profanity_filter,
                enable_automatic_punctuation=self._automatic_punctuation,
                verbatim_transcripts=not self._no_verbatim_transcripts,
                sample_rate_hertz=self._sample_rate,
                audio_channel_count=self._audio_channel_count,
            ),
            interim_results=self._interim_results,
        )
        riva.client.add_word_boosting_to_config(config, self._boosted_lm_words, self._boosted_lm_score)
        riva.client.add_endpoint_parameters_to_config(
            config,
            self._start_history,
            self._start_threshold,
            self._stop_history,
            self._stop_history_eou,
            self._stop_threshold,
            self._stop_threshold_eou,
        )
        riva.client.add_custom_configuration_to_config(config, self._custom_configuration)
        self._config = config

        self._queue = asyncio.Queue()
        self._generate_interruptions = generate_interruptions
        if self._generate_interruptions:
            self._vad_state = VADState.QUIET

        # Initialize the thread task and response task
        self._thread_task = None
        self._response_task = None
        # Initialize ASR compute latency tracking
        self._audio_duration_counter = 0.0  # Tracks cumulative audio duration sent to Riva (in seconds)

    def can_generate_metrics(self) -> bool:
        """Check if the service can generate metrics.

        Returns:
            bool: False as this service does not support metric generation.
        """
        return False

    async def start(self, frame: StartFrame):
        """Start the ASR service.

        Args:
            frame: The StartFrame that triggered the start.
        """
        await super().start(frame)
        self._response_task = self.create_task(self._response_task_handler())
        self._response_queue = asyncio.Queue()

    async def stop(self, frame: EndFrame):
        """Stop the ASR service and cleanup resources.

        Args:
            frame: The EndFrame that triggered the stop.
        """
        await super().stop(frame)
        await self._stop_tasks()

    async def cancel(self, frame: CancelFrame):
        """Cancel the ASR service and cleanup resources.

        Args:
            frame: The CancelFrame that triggered the cancellation.
        """
        await super().cancel(frame)
        await self._stop_tasks()

    async def _stop_tasks(self):
        if self._thread_task is not None and not self._thread_task.done():
            await self.cancel_task(self._thread_task)
        if self._response_task is not None and not self._response_task.done():
            await self.cancel_task(self._response_task)

    def _response_handler(self):
        try:
            logger.debug("Sending new Riva ASR streaming request...")
            responses = self._asr_service.streaming_response_generator(
                audio_chunks=self,
                streaming_config=self._config,
            )
            for response in responses:
                if not response.results:
                    continue
                asyncio.run_coroutine_threadsafe(self._response_queue.put(response), self.get_event_loop())
        except Exception as e:
            logger.error(f"Error in Riva ASR stream: {e}")
            raise
        logger.debug("Riva ASR streaming request terminated.")

    @traced(attachment_strategy=AttachmentStrategy.NONE, name="asr")
    async def _thread_task_handler(self):
        try:
            # Reset audio duration counter for new ASR session
            self._audio_duration_counter = 0.0
            self._thread_running = True
            await asyncio.to_thread(self._response_handler)
        except asyncio.CancelledError:
            self._thread_running = False
            raise

    async def _handle_interruptions(self, frame: Frame):
        if self.interruptions_allowed:
            # Make sure we notify about interruptions quickly out-of-band.
            if isinstance(frame, UserStartedSpeakingFrame):
                logger.debug("User started speaking")
                await self._start_interruption()
                # Push an out-of-band frame (i.e. not using the ordered push
                # frame task) to stop everything, specially at the output
                # transport.
                await self.push_frame(StartInterruptionFrame())
            elif isinstance(frame, UserStoppedSpeakingFrame):
                logger.debug("User stopped speaking")
                await self._stop_interruption()
                await self.push_frame(StopInterruptionFrame())

        await self.push_frame(frame)

    async def _handle_response(self, response):
        """Process ASR response and generate appropriate transcription frames.

        Handles three types of transcription results:
        1. Final results (is_final=True): Complete, confirmed transcriptions
        2. Stable interim results (stability=1.0): High-confidence partial results
        3. Partial results (stability<1.0): Lower-confidence, in-progress transcriptions

        Also manages voice activity detection (VAD) state and interruption handling
        when enabled. Each type of result generates appropriate transcription frames
        with different stability values.
        """
        partial_transcript = ""
        for result in response.results:
            if result and not result.alternatives:
                continue
            transcript = result.alternatives[0].transcript
            if transcript and len(transcript) > 0:
                await self.stop_ttfb_metrics()
                if result.is_final:
                    await self.stop_processing_metrics()
                    if self._generate_interruptions:
                        self._vad_state = VADState.QUIET
                        await self._handle_interruptions(UserStoppedSpeakingFrame())
                    # Calculate ASR compute latency
                    if result.audio_processed:
                        compute_latency = self._audio_duration_counter - result.audio_processed
                        logger.debug(f"{self.name} ASR compute latency: {compute_latency}")
                    logger.debug(f"Final user transcript: [{transcript}]")
                    await self.push_frame(TranscriptionFrame(transcript, "", time_now_iso8601(), None))
                    self.last_transcript_frame = None
                    break
                elif result.stability == 1.0:
                    if self._generate_interruptions and self._vad_state != VADState.SPEAKING:
                        self._vad_state = VADState.SPEAKING
                        await self._handle_interruptions(UserStartedSpeakingFrame())
                    if (
                        self.last_transcript_frame is None
                        or (self.last_transcript_frame.stability != 1.0)
                        or (self.last_transcript_frame.text.rstrip() != transcript.rstrip())
                    ):
                        logger.debug(f"Interim user transcript: [{transcript}]")
                        frame = RivaInterimTranscriptionFrame(
                            transcript, "", time_now_iso8601(), None, stability=result.stability
                        )
                        await self.push_frame(frame)
                        self.last_transcript_frame = frame
                    break
                else:
                    if self._generate_interruptions and self._vad_state != VADState.SPEAKING:
                        self._vad_state = VADState.SPEAKING
                        await self._handle_interruptions(UserStartedSpeakingFrame())
                    partial_transcript += transcript

        if len(partial_transcript) > 0 and (
            self.last_transcript_frame is None
            or (self.last_transcript_frame.stability == 1.0)
            or (self.last_transcript_frame.text.rstrip() != partial_transcript.rstrip())
        ):
            logger.debug(f"Partial user transcript: [{partial_transcript}]")
            frame = RivaInterimTranscriptionFrame(partial_transcript, "", time_now_iso8601(), None, stability=0.1)
            await self.push_frame(frame)
            self.last_transcript_frame = frame

    async def _response_task_handler(self):
        while True:
            try:
                response = await self._response_queue.get()
                await self._handle_response(response)
            except asyncio.CancelledError:
                break

    async def run_stt(self, audio: bytes) -> AsyncGenerator[Frame, None]:
        """Run speech-to-text recognition.

        Args:
            audio: The audio data to process.

        Yields:
            Frame: A sequence of frames containing the recognition results.
        """
        if self._thread_task is None or self._thread_task.done():
            self._thread_task = self.create_task(self._thread_task_handler())
        await self._queue.put(audio)
        yield None

    def __next__(self) -> bytes:
        """Get the next audio chunk for processing.

        Returns:
            bytes: The next audio chunk.

        Raises:
            StopIteration: When no more audio chunks are available.
        """
        if not self._thread_running:
            raise StopIteration
        try:
            future = asyncio.run_coroutine_threadsafe(self._queue.get(), self.get_event_loop())
            result = future.result(timeout=self._idle_timeout)
            # Increment audio duration counter based on audio chunk size
            # Assuming LINEAR_PCM encoding: bytes_per_sample = 2, channels = self._audio_channel_count
            bytes_per_sample = 2  # 16-bit PCM
            total_samples = len(result) // (bytes_per_sample * self._audio_channel_count)
            duration_seconds = total_samples / self._sample_rate
            self._audio_duration_counter += duration_seconds
        except concurrent.futures.TimeoutError:
            future.cancel()
            logger.info(f"ASR service is idle for {self._idle_timeout} seconds, terminating active RIVA ASR request...")
            self._thread_task = None
            raise StopIteration from None
        except Exception as e:
            future.cancel()
            raise e
        return result

    def __iter__(self):
        """Get iterator for audio chunks.

        Returns:
            RivaASRService: Self reference for iteration.
        """
        return self