refactor: extract audio buffer operations to domain layer

- Add abogen/domain/audio_buffer.py with core audio operations:
  - create_silence(): create silence audio buffer
  - mix_audio(): mix source into target buffer with auto-resize
  - normalize_audio(): normalize to prevent clipping
  - ensure_buffer_size(): extend buffer to minimum size
  - concatenate_audio(): join multiple audio buffers
  - audio_duration(): calculate duration from samples
  - samples_for_duration(): calculate samples from duration
  - SAMPLE_RATE constant (24000)

- Update abogen/pyqt/conversion.py:
  - Import and use create_silence for chapter silence
  - Use mix_audio for subtitle file mixing
  - Use normalize_audio for clipping prevention
  - Use create_silence for padding in subtitle processing

- Update abogen/webui/conversion_runner.py:
  - Import and use create_silence in append_silence
  - Replace np.zeros with domain function

- Add tests/test_audio_buffer.py with comprehensive unit tests
This commit is contained in:
Artem Akymenko
2026-07-15 20:02:33 +03:00
parent 7fef9c1d93
commit 0d46076bf6
4 changed files with 494 additions and 36 deletions
+15 -33
View File
@@ -29,6 +29,12 @@ from abogen.domain.output_paths import (
sanitize_output_stem,
)
from abogen.domain.audio_helpers import build_ffmpeg_command, to_float32
from abogen.domain.audio_buffer import (
create_silence,
mix_audio,
normalize_audio,
SAMPLE_RATE,
)
import abogen.hf_tracker as hf_tracker
import static_ffmpeg
import threading # for efficient waiting
@@ -1285,10 +1291,7 @@ class ConversionThread(QThread):
# Add silence between chapters for merged output (except after the last chapter)
if merge_chapters_at_end and chapter_idx < total_chapters:
silence_samples = int(
self.silence_duration * 24000
) # Silence duration at 24,000 Hz
silence_audio = np.zeros(silence_samples, dtype="float32")
silence_audio = create_silence(self.silence_duration)
silence_bytes = silence_audio.tobytes()
if merged_out_file:
@@ -1596,9 +1599,8 @@ class ConversionThread(QThread):
max_end_time = max(
(end for _, end, _ in subtitles if end is not None), default=0
)
audio_buffer = np.zeros(
int(max_end_time * rate) + rate, dtype="float32"
)
buffer_samples = int(max_end_time * rate) + rate
audio_buffer = np.zeros(buffer_samples, dtype="float32")
# Process each subtitle and mix into buffer
self.etr_start_time = time.time()
@@ -1799,33 +1801,14 @@ class ConversionThread(QThread):
# Pad or trim to subtitle duration
target_samples = int(subtitle_duration * rate)
if len(full_audio) < target_samples:
full_audio = np.concatenate(
[
full_audio,
np.zeros(
target_samples - len(full_audio), dtype="float32"
),
]
)
padding_duration = (target_samples - len(full_audio)) / rate
full_audio = np.concatenate([full_audio, create_silence(padding_duration)])
elif len(full_audio) > target_samples:
full_audio = full_audio[:target_samples]
# Mix audio into buffer at the correct position (handles overlaps)
start_sample = int(start_time * rate)
end_sample = start_sample + len(full_audio)
if end_sample > len(audio_buffer):
# Extend buffer if needed
audio_buffer = np.concatenate(
[
audio_buffer,
np.zeros(
end_sample - len(audio_buffer), dtype="float32"
),
]
)
# Mix (add) the audio - this handles overlaps by combining them
audio_buffer[start_sample:end_sample] += full_audio
mix_audio(audio_buffer, full_audio, start_sample)
# Write subtitle
if subtitle_file:
@@ -1860,12 +1843,11 @@ class ConversionThread(QThread):
self.progress_updated.emit(percent, etr_str)
# Normalize audio buffer to prevent clipping from mixed overlaps
max_amplitude = np.abs(audio_buffer).max()
if max_amplitude > 1.0:
if np.abs(audio_buffer).max() > 1.0:
self.log_updated.emit(
f"\n -> Normalizing audio (peak: {max_amplitude:.2f})"
f"\n -> Normalizing audio (peak: {np.abs(audio_buffer).max():.2f})"
)
audio_buffer = audio_buffer / max_amplitude
audio_buffer = normalize_audio(audio_buffer)
# Write the complete audio buffer
self.log_updated.emit(("\nFinalizing audio. Please wait...", "grey"))