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Merge pull request #3 from ace-step/fix_decode_audio

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Fix decode audio
This commit is contained in:
Sayo
2026-03-01 16:38:36 +08:00
committed by GitHub
parent b041a32ab9 2242646b25
commit dfc08a8df9
1 changed files with 48 additions and 54 deletions
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nodes.py
+48 -54
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@@ -95,28 +95,54 @@ def _encode_audio_b64(audio) -> str:
return _tensor_to_wav_b64(waveform, audio["sample_rate"])
def _parse_wav_bytes(audio_bytes: bytes):
"""Parse WAV bytes manually to support PCM (fmt=1) and float32 (fmt=3)."""
if audio_bytes[:4] != b"RIFF" or audio_bytes[8:12] != b"WAVE":
raise RuntimeError("Not a valid WAV file")
pos = 12
fmt_tag = n_channels = sample_rate = sampwidth = 0
data_offset = data_size = 0
while pos < len(audio_bytes) - 8:
chunk_id = audio_bytes[pos:pos+4]
chunk_size = struct.unpack_from("<I", audio_bytes, pos+4)[0]
pos += 8
if chunk_id == b"fmt ":
fmt_tag = struct.unpack_from("<H", audio_bytes, pos)[0]
n_channels = struct.unpack_from("<H", audio_bytes, pos+2)[0]
sample_rate = struct.unpack_from("<I", audio_bytes, pos+4)[0]
sampwidth = struct.unpack_from("<H", audio_bytes, pos+14)[0] // 8
elif chunk_id == b"data":
data_offset = pos
data_size = chunk_size
break
pos += chunk_size
if data_offset == 0:
raise RuntimeError("WAV data chunk not found")
raw = audio_bytes[data_offset:data_offset + data_size]
n_frames = data_size // (n_channels * sampwidth)
if fmt_tag == 1: # PCM int
if sampwidth == 2:
samples = np.frombuffer(raw, dtype=np.int16).astype(np.float32) / 32768.0
elif sampwidth == 4:
samples = np.frombuffer(raw, dtype=np.int32).astype(np.float32) / 2147483648.0
else:
raise RuntimeError(f"Unsupported PCM sample width: {sampwidth}")
elif fmt_tag == 3: # IEEE float32
samples = np.frombuffer(raw, dtype=np.float32).copy()
else:
raise RuntimeError(f"Unsupported WAV format tag: {fmt_tag}")
samples = samples.reshape(n_frames, n_channels).T
tensor = torch.from_numpy(samples).unsqueeze(0)
return tensor, sample_rate
def _decode_audio_data_url(data_url: str):
if "," in data_url:
b64_data = data_url.split(",", 1)[1]
else:
b64_data = data_url
audio_bytes = base64.b64decode(b64_data)
buf = io.BytesIO(audio_bytes)
with wave.open(buf, "rb") as wf:
n_channels = wf.getnchannels()
sampwidth = wf.getsampwidth()
sample_rate = wf.getframerate()
n_frames = wf.getnframes()
raw = wf.readframes(n_frames)
if sampwidth == 2:
samples = np.frombuffer(raw, dtype=np.int16).astype(np.float32) / 32768.0
elif sampwidth == 4:
samples = np.frombuffer(raw, dtype=np.int32).astype(np.float32) / 2147483648.0
else:
raise RuntimeError(f"Unsupported WAV sample width: {sampwidth}")
samples = samples.reshape(n_frames, n_channels).T
tensor = torch.from_numpy(samples).unsqueeze(0)
return tensor, sample_rate
return _parse_wav_bytes(audio_bytes)
def _parse_audio_response(result: dict):
@@ -243,7 +269,7 @@ class AceStepAudioCodes:
"""Editable audio codes passthrough.
Paste codes manually, or receive from Text2Music Server generation output."""
CATEGORY = "ACE-Step"
CATEGORY = "api node/audio/ACE-Step"
FUNCTION = "process"
RETURN_TYPES = ("STRING",)
RETURN_NAMES = ("audio_codes",)
@@ -281,7 +307,7 @@ class AceStepAudioCodes:
class AceStepText2MusicGenParams:
"""Generation parameters for text2music / cover / remix / repaint."""
CATEGORY = "ACE-Step"
CATEGORY = "api node/audio/ACE-Step"
FUNCTION = "build"
RETURN_TYPES = ("ACESTEP_GEN_PARAMS",)
RETURN_NAMES = ("gen_params",)
@@ -429,7 +455,7 @@ class AceStepText2MusicGenParams:
class AceStepSettings:
"""Inference settings: LM + DiT parameters."""
CATEGORY = "ACE-Step"
CATEGORY = "api node/audio/ACE-Step"
FUNCTION = "build"
RETURN_TYPES = ("ACESTEP_SETTINGS",)
RETURN_NAMES = ("settings",)
@@ -633,7 +659,7 @@ class AceStepText2MusicServer:
Inputs: serve_config fields + gen_params + settings.
Outputs: audio + info + audio_codes."""
CATEGORY = "ACE-Step"
CATEGORY = "api node/audio/ACE-Step"
FUNCTION = "generate"
RETURN_TYPES = ("AUDIO", "STRING", "STRING")
RETURN_NAMES = ("audio", "info", "audio_codes")
@@ -664,47 +690,15 @@ class AceStepText2MusicServer:
headers = _make_headers(api_key)
body = _build_request_body(gen_params, settings)
actual_task = body.get("task_type", "text2music")
result = _post_json(f"{base}/v1/chat/completions", body, headers)
audio, text_content = _parse_audio_response(result)
info_parts = []
gp = gen_params
param_summary = (
f"task: {actual_task}\n"
f"caption: {gp.get('caption', '')}\n"
f"lyrics: {gp.get('lyrics', '')[:80]}{'...' if len(gp.get('lyrics', '')) > 80 else ''}\n"
f"bpm: {gp.get('bpm', 0)} key: {gp.get('key_scale', '')} "
f"duration: {gp.get('duration', -1.0)} time_sig: {gp.get('time_signature', '')}\n"
f"vocal_language: {gp.get('vocal_language', '')} "
f"instrumental: {gp.get('instrumental', False)}"
)
if actual_task == "cover":
param_summary += (
f"\ncover_strength: {gp.get('cover_strength', 0.0)} "
f"remix_strength: {gp.get('remix_strength', 1.0)}"
)
if gp.get("audio_codes"):
param_summary += f"\naudio_codes: {len(gp['audio_codes'])} chars"
if actual_task == "repaint":
param_summary += (
f"\nrepaint_start: {gp.get('repaint_start', 0.0)} "
f"repaint_end: {gp.get('repaint_end', 0.0)}"
)
info_parts.append(f"=== Gen Params ===\n{param_summary}")
if text_content:
info_parts.append(f"\n=== Server Response ===\n{text_content}")
out_codes = ""
choices = result.get("choices", [])
if choices:
msg = choices[0].get("message", {})
out_codes = msg.get("audio_codes", "") or ""
if out_codes:
info_parts.append(f"\n=== Audio Codes ({len(out_codes)} chars) ===")
info = "\n".join(info_parts)
return (audio, info, out_codes)
return (audio, text_content, out_codes)
# =============================================================================
@@ -714,7 +708,7 @@ class AceStepText2MusicServer:
class AceStepShowText:
"""Display any STRING input as text."""
CATEGORY = "ACE-Step"
CATEGORY = "api node/audio/ACE-Step"
FUNCTION = "show"
RETURN_TYPES = ("STRING",)
RETURN_NAMES = ("text",)