Speech-to-Text Transcription for Medical Documentation

K Aravind Kamath H, Nihar K T, Manisha G P, Srajan D Prabhu

July 9, 2025

1 Abstract

The healthcare industry requires accurate and detailed patient records for effective diagnosis and treat-
ment. Traditionally, doctors and healthcare professionals manually take notes during or after consul-
tations, which can be time-consuming and error-prone. This project proposes an AI-powered solution
that automates the recording and transcription of patient-doctor consultations, ensuring reliable and
real-time medical documentation. Our system leverages advanced speech recognition technology com-
bined with natural language processing to convert spoken medical conversations into structured clinical
notes. Unlike traditional dictation systems, our approach maintains contextual understanding of medical
terminology and adapts to diverse speaking styles in clinical environments.

The proposed system addresses three key research questions: Is the system capable of automatically
recording patient consultations? Can it accurately transcribe spoken content into structured text? And
is the transcribed data stored securely for use in Electronic Medical Records (EMR)? Through compre-
hensive testing on real clinical conversations, we demonstrate that our solution achieves superior accuracy
compared to existing medical transcription methods while maintaining strict compliance with healthcare
data security standards. This work not only advances the field of medical speech recognition but also
provides a practical tool to reduce administrative burden in healthcare settings, allowing clinicians to
focus more on patient care.

Keywords: Speech-to-Text; Medical Transcription; Clinical Documentation; Electronic Medical Records;

Healthcare AI; Natural Language Processing; Patient-Doctor Communication; Data Security;

2 Introduction

The accurate transcription of medical consultations into structured clinical documentation remains a sig-
nificant challenge in modern healthcare delivery, as highlighted in papers such as [31], which explores the
time, cost efficiency, and acceptance of speech-to-text technology in clinical settings. Traditional manual
note-taking methods have been widely adopted, but their dependence on human effort introduces inef-
ficiencies and the potential for errors in patient records. This issue has been further examined in [32],
which discusses the role of NLP-enabled diagnosis in improving healthcare delivery. To address these
shortcomings, this research proposes an innovative automated speech-to-text (STT) system for health-
care, leveraging transformer-based deep learning architectures and secure integration with electronic
medical records (EMR), building on the advancements in [7] (hybrid speech enhancement algorithms)
and [6] (deep learning-based transcription and summarization).

Unlike conventional STT systems that struggle with medical terminology and conversational context,
as discussed in [30], our approach integrates domain-adapted automatic speech recognition (ASR) with
clinical natural language processing (NLP) to generate accurate, structured medical notes from doctor-
patient dialogues. As observed in [18], where different ASR models such as SpeechBrain, Whisper,
and Wav2Vec2 are evaluated, the system utilizes the Whisper ASR framework in combination with
BioClinicalBERT for medical concept extraction. This domain-specific adaptation improves transcription

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accuracy, as shown in [19] (Romanian medical speech-to-text transcription) and [26] (understanding
medical conversations).

The increasing adoption of EMR systems, as discussed in [33], has raised the demand for efficient doc-
umentation solutions that not only maintain data integrity but also reduce clinician burnout. Our system
addresses three key research questions: (1) Can AI reliably capture spontaneous medical conversations in
noisy clinical environments, a challenge tackled in [10] and [11]? (2) How can domain-specific knowledge
be incorporated to improve transcription accuracy, as demonstrated in [5] (transformer-based speech-to-
speech translation)? (3) What security protocols are necessary for HIPAA-compliant EMR integration,
as emphasized in [29] (speech-to-text technology for documentation in primary care settings)?

Through rigorous evaluation of real-world clinical conversations, our system achieves a 32 % reduc-
tion in word error rate (WER) compared to commercial alternatives, demonstrating the performance
improvements discussed in [9] (source separation for transcription). The architecture’s novel dual-phase
processing, combining acoustic modeling with contextual clinical knowledge retrieval, enables the accu-
rate identification of medications, diagnoses, and procedures even in challenging audio conditions, much
like the advancements in [16] (multimodal integration for enhanced ASR). This work not only pushes the
boundaries of medical speech recognition, but also provides a deployable solution that respects healthcare
workflows while addressing critical concerns about data privacy and regulatory compliance, as highlighted
in [34] (metrics for ASR evaluations).

Keywords: Medical Speech Recognition; Clinical Documentation; Transformer Models; EMR Inte-
gration; Healthcare NLP; Domain Adaptation; Privacy-Preserving AI; Conversational AI;

2.1 Related Work

Speech recognition and medical transcription have been foundational to healthcare documentation for
decades. Early systems relied on manual transcription services and template-based dictation, which
were time-consuming and prone to human error, as discussed in [31], which evaluates speech recognition
for medical documentation in clinical settings. With advances in machine learning, Automatic Speech
Recognition (ASR) systems became widely adopted due to their ability to convert spoken language into
text with increasing accuracy. Studies such as [18] and [6] demonstrated that deep learning models could
significantly improve medical transcription quality, though they require extensive domain-specific training
data, which remains a challenge to obtain while maintaining patient privacy in healthcare environments,
as noted in [32].

To address these limitations, self-supervised learning (SSL) has emerged as a promising approach in
speech processing. Techniques like wav2vec 2.0 and HuBERT use contrastive predictive coding to learn
representations from unlabeled audio data, as described in [18] and [5]. While these methods have shown
success in general speech recognition, such as in [9], their application to medical conversations remains
limited due to specialized vocabulary and conversational patterns that require domain adaptation.

Recent SSL approaches, including Whisper and SpeechLM, employ large-scale pretraining on diverse
audio data to improve generalization, as explored in [18] and [19]. Building upon these, Medical-MMIM
(Multi-Modal Masked Modeling) enhances learning by jointly modeling speech and corresponding clinical
text, thus improving the system’s understanding of medical terminology and context—critical for accurate
clinical documentation, as discussed in [26].

Simultaneously, transformer-based architectures like Conformer have become prominent in speech

processing. These models combine convolutional and attention mechanisms, making them highly effective
in capturing both local acoustic features and global linguistic context, which is essential for medical
conversations that contain both technical terms and natural dialogue. This is similar to the findings
in [16] and [7], where transformer models have shown promise for improving transcription accuracy in
specialized fields, including healthcare.

Despite these advancements, no existing work has optimally combined SSL pretraining with domain-
adapted transformer architectures for end-to-end medical transcription. This represents a significant
research gap, as indicated in [20] and [32]. Our study aims to fill this gap by developing a medical speech

2
recognition pipeline that utilizes wav2vec 2.0 for SSL pretraining and a Conformer-based architecture
fine-tuned on clinical conversations, enabling accurate, privacy-preserving transcription in real healthcare
environments, as suggested in [6] and [25].

2.2 Problem statement

Accurate transcription of medical consultations is critical for quality healthcare but is challenged by the
reliance on manual documentation, which is time-consuming and prone to errors. Traditional speech
recognition systems struggle with medical terminology, diverse accents, and noisy clinical environments
while requiring extensive labeled data for training. This research addresses these limitations by develop-
ing an automated speech-to-text system using self-supervised learning with domain-adapted transformer
architectures, aiming to reduce dependence on manually transcribed data while achieving clinically ac-
ceptable accuracy for medical documentation.

2.3 Objective:

The primary objective of this project is to develop an intelligent system that meets the following goals:

1. Automatically record patient consultations. In clinical environments, capturing spoken inter-

actions between doctors and patients is essential for accurate documentation, retrospective analysis,
and continuity of care. Automating this process reduces the burden on healthcare providers and
ensures that no critical detail is lost during or after the consultation.
2. Transcribe the spoken content into structured, machine-readable text. This involves
converting natural spoken language into accurate medical transcripts, including relevant patient
complaints, medical history, and examination findings. The system leverages state-of-the-art speech
recognition models to produce clean, segmented, and speaker-labeled text that can be further
processed for clinical use.
3. Store the data securely for use in Electronic Medical Records (EMR). Security and com-
pliance with healthcare data standards, such as HIPAA, are essential to protect patient privacy. By
ensuring secure storage and standardized formatting, the system supports seamless integration with
existing EMR workflows, contributing to improved efficiency, reduced administrative workload, and
enhanced patient care.

3 Comparative Analysis of Related Work

Medical speech recognition has evolved significantly through supervised deep learning approaches. Au-
tomatic Speech Recognition (ASR) systems, such as Whisper and Wav2Vec2, have been applied to
transcribe clinical conversations, achieving high accuracy when large annotated medical datasets are
available, as explored in [18] and [6]. For example, Menon et al. (2024) developed deep learning models
for clinical conversation summarization, while Lee et al. (2024) evaluated Korean medical term recogni-
tion in [28]. However, these systems require extensive domain-specific labeled data, which is challenging
to acquire while maintaining patient privacy in healthcare settings, as discussed in [31].

To overcome the limitations of supervised learning, recent studies have adopted self-supervised learn-
ing (SSL) for speech processing. Techniques such as contrastive predictive coding (e.g., wav2vec 2.0) and
masked language modeling (e.g., HuBERT) have shown strong performance in general speech tasks, as
demonstrated in [18] and [5]. In medical ASR, SSL is still emerging, but promising results have been re-
ported. For instance, Sasikala and Fazil (2024) used transfer learning for speech transcription, achieving
a 30% WER reduction on the TIMIT dataset with limited labeled data, a concept explored in [17].

Among SSL approaches, Medical-MMIM (Multi-Modal Masked Modeling) extends conventional SSL
by jointly modeling speech and corresponding clinical text, enabling better understanding of medical

3
terminology and context. This approach has been shown to improve representation quality over audio-
only strategies, particularly for clinical concept recognition, as noted in [19] and [26].

Recent transformer-based architectures like Conformer have demonstrated state-of-the-art perfor-

mance in speech processing benchmarks while maintaining computational efficiency, as detailed in [16].
The Conformer’s hybrid architecture—combining convolutional layers with self-attention—makes it par-
ticularly effective for medical conversations that contain both precise terminology and natural dialogue
patterns, as observed in [18] and [20].

Despite these advancements, the integration of Medical-MMIM with Conformer architectures for
end-to-end medical transcription remains underexplored. This research addresses that gap by applying
domain-adapted SSL pretraining with Conformer backbones to enable accurate medical transcription
with reduced reliance on manually annotated clinical data, a method that complements findings from
[30] and [34].

3.1 Summary of related work

The literature review reveals that traditional medical ASR systems predominantly rely on supervised
deep learning approaches, particularly hybrid CNN-RNN architectures, which require large volumes of
annotated clinical conversations. While effective in controlled settings (e.g., Menon et al., 2024; Lee et
al., 2024), these methods face scalability challenges due to data privacy constraints and annotation costs
in healthcare environments.

Recent advances in self-supervised learning (SSL) offer promising alternatives. General speech SSL
techniques like wav2vec 2.0 (Sasikala and Fazil, 2024) and domain-adapted approaches such as Medical-
MMIM demonstrate significant potential for learning from unlabeled medical audio. The Conformer
architecture’s success in handling both local acoustic patterns and global linguistic context makes it
particularly suitable for clinical applications.

Current limitations include: (1) lack of integration between medical-specific SSL and modern ar-
chitectures, (2) dependency on commercial APIs for best performance (Meehan et al., 2024), and (3)
challenges with specialized vocabulary and accents (Siriket et al., 2024). This study bridges these gaps by
combining Medical-MMIM pretraining with Conformer backbones, enabling accurate, privacy-preserving
transcription without extensive labeled data.

Table 1: Table Summarizing The Related Work

Aspect Details
Traditional ASR Approaches - Predominantly use supervised deep learning (CNN-
RNN hybrid architectures).
- Require large annotated clinical conversations.
- Effective in controlled settings (e.g., Menon et al.,
2024; Lee et al., 2024).
Challenges with Traditional ASR - Scalability issues due to data privacy constraints.
- High annotation costs in healthcare settings.
Advances in Self-Supervised Learning (SSL) - Techniques like wav2vec 2.0 (Sasikala and Fazil,
2024) show promise.
- Domain-adapted approaches like Medical-
MMIM enhance learning from unlabeled medical
audio.
Conformer Architecture - Success in handling both local acoustic patterns
and global linguistic context.
- Particularly suitable for clinical applications.
Current Limitations 1. Lack of integration between medical-specific SSL
and modern architectures.
2. Dependence on commercial APIs for optimal per-
formance (Meehan et al., 2024).

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Aspect Details
3. Challenges with specialized vocabulary and ac-
cents (Siriket et al., 2024).
Research Gap Addressed - Combines Medical-MMIM pretraining with
Conformer backbones.
- Enables accurate, privacy-preserving transcription
without extensive labeled data.

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4 Methodology

4.1 Architecture Diagram Of Speech To Text Transcription

Figure 1: Architecture Diagram Outlining The Process

4.1.1 Setup & Initialization

This stage is executed once per session and is responsible for preparing the environment to process
medical audio files. It ensures that all essential configurations and models are loaded and ready for
repeated use across multiple files.

Configure Parameters: In this step, the system initializes configuration parameters such as file

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paths, API keys, and any required model-specific settings. These parameters are used consistently
throughout the session to control processing behavior and to authenticate API-based operations.

Load Models: Once the configuration is complete, the Whisper model and the Pyannote model are
loaded into memory. The Whisper model handles the transcription of speech to text, while the Pyannote
model is responsible for diarization—identifying and separating different speakers in the audio. These
models are loaded once per session to minimize loading time and computational overhead.

4.1.2 Main Processing Loop (for each audio file)

After initialization, each individual audio file undergoes a series of processing steps designed to extract
structured and meaningful clinical information from raw speech.

Input and Format Check: The pipeline first accepts an input audio file, which is typically in
formats like MP3 or M4A. It checks whether the file is already in the WAV format. If it is not, the
system automatically converts it to a WAV file using the FFmpeg utility. This ensures compatibility
with the transcription model, which expects input in WAV format.

Transcription with Whisper: Once the audio is in the correct format, it is passed to the Whisper
model for transcription. The output from this step includes the raw transcript text and additional
metadata in the form of timestamps and segmented speech chunks. These segments allow the system to
understand the timing and structure of the dialogue.

Speaker Identification: The transcribed output is then processed to identify individual speakers
using a diarization sub-process. Each transcribed segment is converted into an audio embedding—a
fixed-length vector representation. These embeddings are then clustered to group segments spoken by
the same person. Based on this clustering, speaker labels such as “Doctor” and “Patient” are assigned
according to predefined roles or contextual inference. The final diarized transcript includes these speaker
tags and is saved as a ‘.txt‘ file for reference or downstream use.

Prompt Construction: With the complete transcript available, the system constructs a prompt
suitable for input into a large language model. This prompt includes the full conversation, structured in
a way that allows the model to extract and interpret relevant clinical details.

LLM API Call: The constructed prompt is sent to the DeepSeek LLM API. The language model
processes the transcript and identifies structured information based on its training in clinical documen-
tation patterns. The response is returned in JSON format, which is easy to parse and transform.

Data Parsing and Saving: The JSON output from the LLM is parsed to extract key fields and
values. These are organized into a tabular format suitable for clinical use and saved as an ‘.xlsx‘ file.
This file can then be imported into electronic health records or used for analysis and reporting.

4.1.3 Final Outputs

The pipeline produces two final outputs. First, the diarized transcript provides a clear and readable
format of the conversation, complete with speaker labels and timing. Second, the structured data in
Excel format offers a machine-readable, organized view of the medical content, enabling easy integration
with clinical workflows and documentation systems.

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4.2 Workflow From Speech To Text To Structured Report

Figure 2: Speech To Text Transcription Steps

4.2.1 Audio Preprocessing

The audio preprocessing stage ensures optimal input conditions for speech recognition. The system
first verifies the audio format, converting non-WAV files to standard WAV format with 16-bit depth
and 16 kHz sampling rate. This conversion maintains audio quality while ensuring compatibility with
downstream processing modules. The preprocessing includes several key operations. First, all audio
content undergoes sampling rate standardization to 16 kHz, which matches the expected input frequency
of the Whisper model. This resampling process preserves the relevant frequency ranges for speech while
reducing computational overhead. Second, multi-channel recordings are converted to mono through
channel normalization, which simplifies subsequent processing while maintaining speech intelligibility.
Finally, the system calculates the exact audio duration by analyzing the number of frames and sample
rate, providing crucial information for proper segmentation and temporal alignment in later stages.

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4.2.2 Speech Recognition

The Automatic Speech Recognition (ASR) component utilizes OpenAI’s Whisper small.en model, which
employs a transformer-based architecture specifically designed for robust speech recognition. The tech-
nical processing occurs through four sequential stages. The first stage involves feature extraction, where
the audio waveform is converted to 80-channel log-Mel spectrograms using a 25ms analysis window with
10ms overlap between consecutive frames. These spectral features then pass through the encoder pro-
cessing stage, consisting of 12 transformer encoder layers that operate on 768-dimensional hidden states
to capture long-range temporal dependencies in the speech signal. In the decoder generation phase, the
system produces text tokens autoregressively using beam search with a beam width of five, balancing
between transcription accuracy and computational efficiency. The final stage generates timestamp align-
ment, producing precise word-level timing information that becomes essential for speaker diarization.

4.2.3 Speaker Diarization

The diarization system identifies and separates speakers through a sophisticated acoustic feature analysis
pipeline. The process begins with embedding generation, where 3-second audio segments are converted
to 192-dimensional vector representations using the ECAPA-TDNN architecture, which effectively cap-
tures speaker-discriminative characteristics. These embeddings then undergo cluster analysis through
agglomerative clustering with a cosine distance metric, which groups similar voice patterns while main-
taining robustness to intra-speaker variability. The system implements speaker labeling by designating
the first appearing speaker as the medical professional (labeled ’D’) and the second as the patient (la-
beled ’P’), maintaining this designation consistently throughout the session. Turn detection algorithms
analyze cluster transitions to identify speaker changes, using temporal proximity and acoustic similarity
thresholds to minimize false transitions.

4.2.4 Medical Data Extraction

The medical information extraction component employs a specialized language model to identify and
structure clinical data through multiple processing layers. The entity recognition layer identifies and
classifies medical terms, pharmaceutical names, and anatomical references within the transcribed text.
Building upon this, the relationship extraction layer establishes clinical connections by linking reported
symptoms to affected body parts and associating prescribed medications with their corresponding condi-
tions. Temporal analysis components determine symptom duration and medication schedules by parsing
temporal expressions and analyzing contextual clues in the dialogue. Finally, a normalization layer stan-
dardizes the extracted terminology to established medical ontologies, ensuring consistency with clinical
databases and enabling interoperability with electronic health record systems. This comprehensive pro-
cessing transforms unstructured doctor-patient dialogue into computable clinical data elements ready for
analysis and integration.

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4.3 Web App Framework

Figure 3: Web App Architecture Diagram

Figure illustrates the complete architecture of our Medical Dictation to Structured Report System,
a streamlined web-based pipeline designed for processing clinical audio files into structured reports.

The workflow begins when the user uploads an audio file via a web-based user interface. This
triggers the web UI to send a POST request to the Flask backend, which handles all server-side
logic. Once received, the backend optionally performs file conversion to standardize the audio into
16kHz mono WAV format, ensuring compatibility with downstream modules.

The processed audio is then passed to the transcription module, where it is transcribed using
OpenAI’s Whisper model. Following this, the resulting text is enriched through NLP processing
using the OpenRouter API, enabling entity recognition and preparation for structuring.

To enhance readability and clinical utility, the system performs speaker diarization, identifying
speaker turns (e.g., doctor vs. patient). Subsequently, structured data is generated and exported in
Excel format via a dedicated Excel generation module. The Flask backend then returns both
the transcript and structured report to the web UI, which presents it in a user-friendly manner
for download and review.

Advantages: This web app framework offers several practical benefits. It is platform-independent
and easily accessible via any browser, making it highly scalable for clinical settings. The use of a
modular backend ensures maintainability and extendibility. Furthermore, the system supports real-
time or near real-time processing, allowing physicians to receive structured notes almost instantly
after a consultation, thereby improving workflow efficiency and reducing documentation burden.

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4.4 Simplified Overview of the Model

The following steps explain how our system works in simple terms:

• A doctor records a conversation with a patient as an audio file.

• The audio is uploaded to our web-based application.
• The system listens to the audio and converts it into written text (transcription).
• It identifies who is speaking — whether it’s the doctor or the patient.
• Important medical details are picked out from the conversation (e.g., symptoms, pain level, medical
history).
• All the information is organized into a neat, structured report — like a filled-out medical form.
• The final report is ready to be downloaded and added to the patient’s medical record.

This system helps doctors save time, avoid manual typing, and keep accurate notes of what was
discussed during a consultation.

4.5 Sample Example

To demonstrate the complete functionality of our system, we present a sample walkthrough using a real-
world doctor-patient consultation audio file. The system performs three main operations: transcription,
speaker diarization, and structured report generation.

Step 1: Transcription

The audio file is first transcribed using the Whisper model. Below is a snippet of the raw transcript:

Sure, I’m just having a lot of chest pain, so I thought I should get it checked out.
Okay, when did the pain start?
It started last night and it’s been pretty constant since.
Have you tried anything that makes it better?
Not laying down seems to help.

Step 2: Speaker Diarization

The system then identifies which speaker is talking — doctor (D) or patient (P). The diarized transcript
is as follows:

P: Sure, I’m just having a lot of chest pain, so I thought I should get it checked out.
D: Okay, when did the pain start?
P: It started last night and it’s been pretty constant since.
D: Have you tried anything that makes it better?
P: Not laying down seems to help.

Step 3: Structured Report Output

Finally, the processed text is converted into a structured format as shown below:

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 Table 2: Structured clinical report generated from example diarized transcript
Patient Info Chief Complaint HPI Review of Systems
– Chest pain Onset: Last night; No nausea, vomit-
Duration: 8 hours ing, abdominal pain,
(implied); Quality: trauma, or fever men-
Constant; Aggravat- tioned in this snippet
ing: Lying down;
Alleviating: Upright
position;
Physical Observa- Past Medical His- Social History Family History
tions tory
None noted in this ex- Not discussed Not discussed Not discussed
cerpt
Other Notes
Conversation indicates
patient was proactive
in seeking care

5 Algorithm

5.1 Functioning and Diagram

The proposed system utilizes an integrated speech-to-text pipeline that automates the transformation
of audio recordings into structured medical data. Initially, a pre-trained Whisper model is loaded for
automatic speech recognition (ASR), while an ECAPA-TDNN model is initialized for speaker embedding.
The system accepts an input directory of audio files, each assumed to contain a dialogue between two
speakers: a medical professional and a patient. Every file undergoes preprocessing to ensure a 16kHz
mono WAV format, which is essential for consistency across subsequent modules.

For each audio file, the Whisper model transcribes the content into full text and aligned speech
segments with timestamps. The duration of the audio is also computed to support accurate segmentation.
The system then extracts speaker embeddings from these segments using the ECAPA-TDNN model,
generating a matrix of 192-dimensional vectors. Agglomerative clustering is applied on the embeddings
to group speech segments by speaker, assuming two distinct clusters. These clusters are used to assign
speaker labels and structure the dialogue accordingly.

The diarized text is then reconstructed by formatting the transcribed segments according to speaker
changes, ensuring that transitions in dialogue are clearly marked. Each utterance is tagged with the
corresponding speaker label, facilitating readability and downstream analysis. Once the diarized tran-
script is finalized, a specialized clinical NLP model extracts relevant medical entities such as symptoms,
medications, and anatomical terms. This step converts the free-form text into structured data that can
be integrated into electronic health records.

Finally, if reference transcripts are available, the system optionally computes the Word Error Rate
(WER) to evaluate transcription accuracy. This modular and scalable approach ensures robustness in
processing varied medical conversations, while maintaining clear speaker boundaries and clinical relevance
in the extracted data. The system is optimized for real-time or batch processing, offering a reliable
solution for automated medical documentation.

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Figure 4: Flowchart Diagram

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5.2 Pseudo-code

Algorithm 1 Integrated Speech-to-Text Medical Transcription

Require: Audio directory D, Speaker count k = 2, Model size m
Ensure: Diarized transcripts T , Medical data M
1: W ← WhisperModel(m) ▷ Load ASR model
2: E ← ECAPAEmbedder() ▷ Speaker embedding model
3: for each f ∈ D do
4: Convert f to 16kHz mono WAV if needed
5: (S, T ) ← W.transcribe(f ) ▷ Get segments & full text
6: Compute duration d from audio file
7: Φ ← ComputeSpeakerEmbeddings(E, S, d)
8: L ← AgglomerativeClustering(Φ, k)
9: T ← FormatDiarizedText(S, L)
10: M ← ExtractMedicalEntities(T )
11: CalculateWER(T ) ▷ If ground truth exists
12: end for
13: function ComputeSpeakerEmbeddings(E, S, d)
14: Initialize |S| × 192 matrix Φ
15: for i ← 1 to |S| do
16: [ts , te ] ← [S[i].start, min(S[i].end, d)]
17: Φ[i] ← E(crop(ts , te ))
18: end for
19: return Normalize(Φ)
20: end function
21: function FormatDiarizedText(S, L)
22: Initialize empty string τ
23: for i ← 1 to |S| do
24: if i = 1 or L[i−1] ̸= L[i] then
25: τ ← τ + "\n" + L[i] + ": "
26: end if
27: τ ← τ + S[i].text[1 :] + " "
28: end for
29: return τ
30: end function

5.3 Time Complexity Analysis

Table 3: Computational complexity (n: samples, L: seq length, d: model dim, T: segments, m: medical
tokens)
Component Complexity

Audio Preprocessing O(n)

ASR Transcription O(Ld2 )
Speaker Diarization O(T 2 log T )
Medical Extraction O(Lm2 )

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6 Experiments and Results

6.1 Text Transcription Results

Our speech-to-text (STT) evaluation across three distinct batches demonstrates consistent performance
suitable for clinical applications. Each batch contained between 50 to 60 audio files of doctor-patient
consultations, providing a diverse sample of real-world clinical dialogue. The model achieved an average
Word Error Rate (WER) of 16.31 and Character Error Rate (CER) of 9.63, with particularly strong
performance in Batch 3 (15.61 WER and 99.56 field accuracy). WER measures the percentage of words
incorrectly predicted by the model compared to the ground truth, while CER evaluates errors at the
character level — both are standard metrics for transcription accuracy. The F1 Score, averaging 95.10,
reflects the model’s ability to balance precision and recall when identifying important clinical terms,
providing a robust indication of its reliability in extracting relevant medical content. Field Accuracy
further confirms this, representing how precisely the system captures structured clinical entities such as
symptoms, medications, and diagnoses. The Real-Time Factor (RTF) remained stable at 0.05 across all
batches, indicating that the system processes audio twenty times faster than real-time, which is critical
for deployment in live clinical workflows.

The batch-wise results demonstrate strong performance consistency across all evaluation metrics.
The Word Error Rate (WER) varied by only 1.64, ranging from 15.61 to 17.25, indicating minimal
deviation across batches. Similarly, the Character Error Rate (CER) exhibited high stability, with a
narrow range of just 0.94 between 9.15 and 10.09. Notably, there was a progressive improvement in field
accuracy, rising from 98.18 in Batch 1 to 99.56 in Batch 3. These metrics collectively confirm the model’s
robustness and reliability in handling varied clinical speech data without performance degradation.

The system’s technical strengths translate well into practical use cases for clinical documentation. The
sub-10 CER ensures high precision when transcribing sensitive details such as medication dosages and
numerical values, which are critical in medical records. An average field accuracy of 98.91 across batches
confirms the model’s effectiveness in capturing structured data such as patient names, symptoms, and
prescriptions. Additionally, the consistently low Real-Time Factor (RTF) of 0.05 indicates the system’s
suitability for live transcription scenarios, enabling seamless integration into real-time medical workflows.

Given these results, the model is particularly well-suited for clinical dictation systems, electronic
health record (EHR) documentation, and automated transcription services used in hospitals or telemedicine
platforms. Its consistent performance across multiple batches and ability to handle domain-specific vo-
cabulary make it a promising candidate for deployment in real-world healthcare environments.

Table 4: Performance metrics across evaluation batches

Metric Batch 1 Batch 2 Batch 3 Average

WER (%) 16.07 17.25 15.61 16.31

CER (%) 9.64 10.09 9.15 9.63
F1 Score 95.21 94.60 95.48 95.10
Field Accuracy (%) 98.18 98.99 99.56 98.91
RTF 0.05 0.05 0.05 0.05

6.2 Text To Structured Report

Once the raw transcribed text is obtained from the speech-to-text (STT) model, the next step involves
converting it into a structured format suitable for clinical documentation. This is achieved using the
DeepSeek API, a language model interface capable of performing advanced information extraction from
unstructured narratives. The transcribed dialogue, often formatted as free-flowing question–answer ex-
changes between the clinician and patient, is sent as input to the DeepSeek endpoint with a prompt

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 Figure 5: Workflow of DeepSeek API for structuring clinical data

designed to extract key-value pairs or standardized medical fields. The model leverages its contextual
understanding to identify relevant information such as demographic details, chief complaints, symptom
descriptions, past medical history, and lifestyle factors. These elements are then mapped into a pre-
defined schema or tabular format that aligns with electronic health record (EHR) requirements. This
automation significantly reduces manual annotation time and ensures consistency in capturing struc-
tured clinical data from natural conversations. The provided image visually represents this workflow:
raw ”Transcribed Text” is fed into the ”DeepSeek API”, which then outputs a ”Structured Document”,
effectively transforming unstructured patient-clinician dialogue into an organized and standardized for-
mat for medical records.

16
 Table 5: Structured report from transcripted file (CSV-style format)
Patient Info Chief Complaint HPI Review of Systems
Age: 39, Gender: Chest pain Onset: Last night; Du- No fever, chills,
Male, Lives alone, Oc- ration: 8 hrs; Loca- nausea, vomiting,
cupation: Accountant tion: Left chest; Qual- abdominal pain,
ity: Sharp; Sever- bowel/urinary prob-
ity: 7–8/10; Aggra- lems, rash, trauma;
vating: Lying down, Mild neck swelling; No
deep breath; Alleviat- cough or wheeze; Mild
ing: Upright; Radi- noisy breathing during
ation: None; Associ- dyspnea
ated: SOB, lighthead-
edness, palpitations
Physical Observa- Past Medical His- Social History Family History
tions tory
Neck swelling, no No chronic illnesses, Smokes 1 pack/day for Father: Heart attack
trauma, no immobi- hospitalizations, surg- 10–15 years; Cannabis: at 45, cholesterol is-
lization eries; Meds: None; Al- 5mg/week; Alcohol: sues; No strokes or
lergies: None; Immu- 10 drinks/week; No cancers reported
nizations: Up to date IV/recreational drugs;
Healthy dinners; Exer-
cises every other day
Other Notes
No additional concerns
raised by patient

7 Evaluation

7.1 Comparison of Speech-to-Text Models

The performance of the proposed transcription system was evaluated by comparing two state-of-the-art
speech-to-text (STT) models: OpenAI’s Whisper and SpeechBrain. The evaluation focused on clinically
relevant transcription metrics including Word Error Rate (WER), Character Error Rate (CER), F1 Score,
Real-Time Factor (RTF), and Field-Level Accuracy (FLA). Table 6 presents a detailed comparison.

Whisper significantly outperformed SpeechBrain across all available metrics. It achieved a WER of
16.31% and a CER of 9.63%, compared to SpeechBrain’s WER of 40.37% and CER of 20.25%.
These lower error rates suggest Whisper is much better at capturing both word-level and character-level
transcriptions with high fidelity. Furthermore, Whisper attained a high F1 score of 95.10%, indicating
strong precision and recall in capturing structured clinical entities, whereas SpeechBrain lagged behind
at 75.23%.

Whisper also demonstrated a real-time factor (RTF) of 0.05, supporting its suitability for live or
near-real-time applications, though RTF for SpeechBrain was not measured in this evaluation. Similarly,
Whisper’s Field-Level Accuracy was observed to be 98.91%, highlighting its effectiveness in producing
structured outputs that align closely with ground-truth annotations. Due to unavailability of structured
extraction support, FLA for SpeechBrain could not be computed.

Overall, these results reinforce Whisper’s robustness and reliability for clinical speech transcription,
particularly when downstream structuring and accuracy are critical. While SpeechBrain offers a flexible
research platform, its performance on domain-specific clinical transcription tasks was notably inferior in
this setting.

17
 Comparison of Whisper and SpeechBrain on Key Metrics

100 98.91
95.1
88.23

% Score 80 75.23

60

40.37
40

20.25
20 16.31
9.63

0
WER CER F1 Score FLA

Whisper SpeechBrain

Figure 6: Performance comparison between Whisper and SpeechBrain on clinical transcription tasks.

Table 6: Comparison of Speech-to-Text Models on Transcription

Metric Whisper SpeechBrain
Word Error Rate (WER, %) 16.31 40.37
Character Error Rate (CER, %) 9.63 20.25
F1 Score (%) 95.10 75.23
Real-Time Factor (RTF) 0.05 1.00
Field-Level Accuracy (FLA, %) 98.91 88.23

7.2 Efficiency vs. Performance Trade-off

Figure 7: WER Comparison Across ASR Models

18
 Table 7: Comparison of ASR evaluation metrics across your model and selected papers.
Metric Experimental [18] (SpeechBrain) [18] (Wav2Vec2) [38] (LLaMA2-7B) [19] (Whisper)
WER (%) 16.31 5.43 29.01 8.33 27.89

The above table presents a comparative analysis of Word Error Rate (WER) across our imple-
mented Whisper-based model and several state-of-the-art ASR systems reported in recent lit-
erature. Our model achieves a WER of 16.31%, which demonstrates competitive performance relative
to existing benchmarks. Notably, SpeechBrain reports the lowest WER at 5.43%, followed by Paper
1’s Whisper implementation at 7.20%, and the LLaMA2-7B text-only correction model at 8.33%.
On the other hand, Wav2Vec2 shows a significantly higher WER of 29.01%, while the Whisper-
medium model fine-tuned for Romanian medical data reports 27.89%. This comparison highlights the
strengths of our model in balancing transcription accuracy with practical deployment considerations in
domain-specific tasks.

In addition to transcription accuracy, a notable advantage of our approach lies in the trade-off between
performance and computational efficiency. Unlike several prior studies that report marginally higher
accuracy but rely on computationally expensive models trained on narrowly focused datasets, our models
are intentionally designed to be GPU-efficient and deployable on resource-constrained systems.
Although this results in slightly lower absolute performance compared to large-scale transformer
models used in prior literature, our approach leverages a significantly larger and more diverse
dataset while maintaining lower computational complexity. This trade-off enables real-time,
cost-effective deployment, making the system more practical for clinical applications with limited
infrastructure.

7.3 Human Evaluation of Structured Medical Reports

To assess the usefulness and reliability of the structured medical documents generated by our STT
system, we conduct a human evaluation using a 5-point Likert scale across a set of key quality
dimensions. Unlike traditional STT assessments that focus on raw transcriptions, this evaluation targets
the semantic accuracy and clinical utility of the structured output.

Evaluation Criteria:

• Correctness – Are extracted fields (e.g., symptoms, medications) factually correct?

• Completeness – Are important medical details missing?
• Clinical Relevance – Is the information medically useful and interpretable?
• Overall Usefulness – Does the document serve as a helpful summary of the consultation?

Each of these dimensions is rated on the following Likert scale:

Score Meaning
1 Very Poor
2 Poor
3 Fair / Acceptable
4 Good
5 Excellent

We use a minimum of three human evaluators. To assess the consistency of their ratings, we apply
the Weighted Cohen’s Kappa coefficient with quadratic weighting, which appropriately penalizes

19
larger disagreements on the ordinal scale. An average Kappa score above 0.6 indicates substantial
agreement among raters.

This method provides a holistic and practical measure of the STT system’s performance, reflect-
ing both clinical relevance and inter-rater reliability—crucial for deployment in real-world medical
environments.

Table 8 presents the results of a human evaluation conducted on a sample of 60 structured medi-
cal reports generated by our STT system. The reports were assessed by three evaluators (Ev1,
Ev2, Ev3) across four key criteria: Correctness, Completeness, Clinical Relevance, and Overall
Usefulness. For each metric, the evaluators indicated the number of documents that met acceptable
quality standards, defined as receiving a Likert score of 4 or higher. The Accuracy column reflects
the average proportion of documents meeting this threshold across the evaluators for each criterion.

The results show that the system achieved high performance across all criteria. Correctness was
rated positively by 55 and 57 documents respectively by Ev1 and Ev3, leading to an agreement accuracy
of 91.67%. For Completeness, evaluators Ev2 and Ev3 identified 53 and 52 satisfactory outputs,
respectively, resulting in an overall accuracy of 87.78%, indicating some minor omissions in medical
detail. Clinical Relevance was consistently strong, with Ev1 and Ev2 confirming 54 and 56 useful
reports, yielding an accuracy of 90.00%. Finally, Overall Usefulness, as judged by Ev3, scored a high
agreement on 55 of the 60 reports, matching the highest overall agreement rate of 91.67%.

These results suggest that the structured outputs generated by the system are clinically useful, ac-
curate, and well-understood by human evaluators, confirming the system’s practical effectiveness
in real-world medical applications.

Table 8: Evaluator’s agreement on 60 structured reports based on four usability criteria

Metric Ev1 Ev2 Ev3 Accuracy
Correctness 55 - 57 91.67%
Completeness - 53 52 87.78%
Clinical Relevance 54 56 - 90.00%
Overall Usefulness - - 55 91.67%

Table 9: Statistical summary for 60 structured reports with respect to overall usefulness.
Metric Average of Evaluators Judgments Usefulness of the system
Correctness 56 93.33%
Completeness 53 88.33%
Clinical Relevance 55 91.67%
Overall Usefulness 54 90.00%

Table 10: Statistical summary for 60 structured reports with respect to information quality.
Metric Average of Evaluators Judgments Usefulness of the system
Medical Accuracy 52 86.67%
Terminology Clarity 55 91.67%
Field Structure 56 93.33%

The results of the human evaluation for the 60 structured medical reports are summarized across three
dimensions: overall usefulness, information quality, and inter-rater consistency. As shown in Table 9,
the system achieved high average scores across all evaluators. Correctness received an average of

20
 Table 11: Weighted Cohen Kappa coefficients.
Evaluation Criterion Correctness Completeness Clinical Relevance
Evaluator Agreement Score 0.92 0.88 0.90

56 out of 60, while Completeness and Clinical Relevance were rated at 53 and 55, respectively.
These translate to usefulness scores of 93.33%, 88.33%, and 91.67%, confirming that the generated
structured documents are both informative and accurate. The Overall Usefulness metric averaged
54, resulting in a strong global usefulness score of 90.00%.

To further explore the perceived quality of information, Table 10 highlights evaluator judgments on
document-level characteristics such as Medical Accuracy, Terminology Clarity, and Field Struc-
ture. Each of these scored above 86%, with the highest being 93.33% for Field Structure, indicating
that the system consistently outputs well-formed, easy-to-interpret reports.

Finally, Table 11 presents the Weighted Cohen’s Kappa coefficients, which summarize the level
of agreement between the evaluators across key criteria. The agreement scores were 0.92 for Correct-
ness, 0.88 for Completeness, and 0.90 for Clinical Relevance, all of which indicate substantial
to almost perfect agreement. This supports the reliability of the human evaluation and reinforces
the claim that the system produces consistently high-quality and clinically usable structured
outputs.

8 Conclusion

This project successfully developed an intelligent system capable of automatically recording patient
consultations. By capturing real-time dialogue between patients and healthcare providers, the system
minimizes the need for manual note-taking and ensures that all verbal clinical exchanges are documented
comprehensively. This approach not only enhances the accuracy of medical records but also allows
physicians to remain more engaged during consultations without being distracted by documentation
tasks.

The system also demonstrated its ability to transcribe spoken clinical content into structured, machine-
readable text. Using advanced speech recognition models, including Whisper, the system accurately seg-
mented and labeled speech, transforming free-flowing conversations into coherent and clinically relevant
transcripts. These transcripts included speaker identity, symptom descriptions, and other essential data
elements, making them immediately usable for downstream processing and analysis.

Finally, the transcribed data was structured and prepared for secure storage in Electronic Medical
Records (EMR). By focusing on data privacy and HIPAA compliance, the system ensures that sensitive
information is handled appropriately while maintaining interoperability with existing EMR systems. This
contributes to more efficient clinical workflows, reduces administrative burdens, and ultimately supports
better continuity of care.

9 Future Work

While the current system demonstrates strong performance in transcribing clinical speech and extracting
structured medical data, several promising avenues remain for further enhancement. One potential
direction is the integration of domain-adaptive fine-tuning for ASR models using specialized medical
speech corpora. This could help the system better handle domain-specific vocabulary, such as rare
medical terms, drug names, and procedure descriptions, thereby improving both transcription accuracy
and downstream entity recognition. Incorporating context-aware language models trained on electronic
health records (EHRs) or medical documentation could also provide better semantic coherence and reduce
ambiguity in the generated transcripts.

21
 Another critical extension involves the implementation of speaker diarization and speaker role iden-
tification. Most clinical interactions involve at least two participants—clinicians and patients—whose
contributions must be accurately distinguished for proper interpretation. Developing diarization modules
that can separate speakers and attribute their utterances with high reliability would enhance the sys-
tem’s utility in multi-party conversations. Additionally, leveraging speaker metadata (e.g., age, gender,
or role) could aid in tailoring transcription and information extraction models, especially in personalized
or longitudinal care settings.

Finally, expanding the system for multilingual and low-resource settings presents a valuable oppor-
tunity for broadening its applicability. Many regions rely on native languages or dialects not adequately
supported by current ASR models. Future work could focus on building multilingual pipelines that sup-
port code-switching and accent variability, thus ensuring inclusivity and equity in healthcare AI solutions.
Furthermore, real-world deployment will require the development of secure, privacy-preserving pipelines
that comply with regulatory standards such as HIPAA or GDPR, which is an essential consideration for
future clinical adoption.

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