sonusai 0.16.0__tar.gz → 0.17.0__tar.gz

{sonusai-0.16.0 → sonusai-0.17.0}/PKG-INFO

@@ -1,6 +1,6 @@
 Metadata-Version: 2.1
 Name: sonusai
-Version: 0.16.0
+Version: 0.17.0
 Summary: Framework for building deep neural network models for sound, speech, and voice AI
 Home-page: https://aaware.com
 License: GPL-3.0-only

{sonusai-0.16.0 → sonusai-0.17.0}/pyproject.toml

@@ -1,6 +1,6 @@
 [tool.poetry]
 name = "sonusai"
-version = "0.16.0"
+version = "0.17.0"
 description = "Framework for building deep neural network models for sound, speech, and voice AI"
 authors = ["Chris Eddington <chris@aaware.com>", "Jason Calderwood <jason@aaware.com>"]
 maintainers = ["Chris Eddington <chris@aaware.com>", "Jason Calderwood <jason@aaware.com>"]
@@ -40,10 +40,13 @@ torchaudio = "~2.2"
 tqdm = "^4.66.1"
 
 [tool.poetry.group.dev.dependencies]
+einops = "^0.8.0"
 icecream = "^2.1.3"
 mypy = "^1.6.0"
 mypy-extensions = "^1.0.0"
 pytest = "^8.1.1"
+sonusai-asr-cloud = "^0.1.0"
+sonusai-torchl = "^0.1.0"
 types-pyyaml = "^6.0.12.12"
 types-requests = "^2.31.0.8"
 

{sonusai-0.16.0 → sonusai-0.17.0}/sonusai/__init__.py

@@ -19,6 +19,7 @@ commands_doc = """
    onnx_predict                 Run ONNX predict on a trained model
    plot                         Plot mixture data
    post_spenh_targetf           Run post-processing for speech enhancement targetf data
+   summarize_metric_spenh       Summarize speech enhancement and analysis results
    tplot                        Plot truth data
    vars                         List custom SonusAI variables
 """

{sonusai-0.16.0 → sonusai-0.17.0}/sonusai/audiofe.py

@@ -12,7 +12,7 @@ options:
     -m MODEL, --model MODEL         PL model .py file path.
     -k CKPT, --checkpoint CKPT      PL checkpoint file with weights.
     -a ASR, --asr ASR               ASR method to use.
-    -w WMODEL, --whisper WMODEL     Whisper model used in aixplain_whisper and whisper methods. [default: tiny].
+    -w WMODEL, --whisper WMODEL     Model used in whisper, aixplain_whisper and faster_whisper methods. [default: tiny].
 
 Aaware SonusAI Audio Front End.
 
@@ -24,47 +24,43 @@ audiofe_capture_<TIMESTAMP>.wav.
 If a model is specified, run prediction on audio data from this model. Then compute the inverse transform of the
 prediction result and save to audiofe_predict_<TIMESTAMP>.wav.
 
+Also, if a model is specified, save plots of the capture data (time-domain signal and feature) to
+audiofe_capture_<TIMESTAMP>.png and predict data (time-domain signal and feature) to
+audiofe_predict_<TIMESTAMP>.png.
+
 If an ASR is specified, run ASR on the captured audio and print the results. In addition, if a model was also specified,
-run ASR on the predict audio and print the results.
+run ASR on the predict audio and print the results.  Examples: faster_whisper, google,
 
 If the debug option is enabled, write capture audio, feature, reconstruct audio, predict, and predict audio to
 audiofe_<TIMESTAMP>.h5.
 
 """
-from os.path import exists
-from select import select
-from sys import stdin
+import signal
 
-import h5py
 import numpy as np
-import pyaudio
-import torch
-from docopt import docopt
-from docopt import printable_usage
-
-import sonusai
-from sonusai import create_file_handler
-from sonusai import initial_log_messages
-from sonusai import logger
-from sonusai import update_console_handler
+
 from sonusai.mixture import AudioT
-from sonusai.mixture import CHANNEL_COUNT
-from sonusai.mixture import SAMPLE_RATE
-from sonusai.mixture import get_audio_from_feature
-from sonusai.mixture import get_feature_from_audio
-from sonusai.mixture import read_audio
-from sonusai.utils import calc_asr
-from sonusai.utils import create_timestamp
-from sonusai.utils import get_input_device_index_by_name
-from sonusai.utils import get_input_devices
-from sonusai.utils import load_torchl_ckpt_model
-from sonusai.utils import trim_docstring
-from sonusai.utils import write_wav
+
+
+def signal_handler(_sig, _frame):
+    import sys
+
+    from sonusai import logger
+
+    logger.info('Canceled due to keyboard interrupt')
+    sys.exit(1)
+
+
+signal.signal(signal.SIGINT, signal_handler)
 
 
 def main() -> None:
+    from docopt import docopt
+
+    import sonusai
+    from sonusai.utils import trim_docstring
+
     args = docopt(trim_docstring(__doc__), version=sonusai.__version__, options_first=True)
-    ts = create_timestamp()
 
     verbose = args['--verbose']
     length = float(args['--length'])
@@ -76,13 +72,63 @@ def main() -> None:
     debug = args['--debug']
     show = args['--show']
 
-    capture_name = f'audiofe_capture_{ts}.wav'
-    predict_name = f'audiofe_predict_{ts}.wav'
+    from os.path import exists
+
+    import h5py
+    import pyaudio
+    import torch
+    from docopt import printable_usage
+    from sonusai_torchl.utils import load_torchl_ckpt_model
+    from sonusai.utils.onnx_utils import load_ort_session
+
+    from sonusai import create_file_handler
+    from sonusai import initial_log_messages
+    from sonusai import logger
+    from sonusai import update_console_handler
+    from sonusai.mixture import SAMPLE_RATE
+    from sonusai.mixture import get_audio_from_feature
+    from sonusai.mixture import get_feature_from_audio
+    from sonusai.utils import calc_asr
+    from sonusai.utils import create_timestamp
+    from sonusai.utils import get_input_devices
+    from sonusai.utils import trim_docstring
+    from sonusai.utils import write_wav
+
+    ts = create_timestamp()
+    capture_name = f'audiofe_capture_{ts}'
+    capture_wav = capture_name + '.wav'
+    capture_png = capture_name + '.png'
+    predict_name = f'audiofe_predict_{ts}'
+    predict_wav = predict_name + '.wav'
+    predict_png = predict_name + '.png'
     h5_name = f'audiofe_{ts}.h5'
 
-    if model_name is not None and ckpt_name is None:
-        print(printable_usage(trim_docstring(__doc__)))
-        exit(1)
+    if model_name is not None:
+        from os.path import splitext
+        if splitext(model_name)[1] == '.onnx':
+            session, options, model_root, hparams, sess_inputs, sess_outputs = load_ort_session(model_name)
+            if hparams is None:
+                logger.error(f'Error: onnx model does not have required SonusAI hyper-parameters, can not proceed.')
+                raise SystemExit(1)
+            feature_mode = hparams["feature"]
+            model_is_onnx = True
+            in0name = sess_inputs[0].name
+            in0type = sess_inputs[0].type
+            out0name = sess_outputs[0].name
+            out_names = [n.name for n in session.get_outputs()]
+            if in0type.find('float16') != -1:
+                model_is_fp16 = True
+                logger.info(f'Detected input of float16, converting all feature inputs to that type.')
+            else:
+                model_is_fp16 = False
+        else:
+            model_is_onnx = False
+            if ckpt_name is None:
+                print(printable_usage(trim_docstring(__doc__)))
+                exit(1)
+            model = load_torchl_ckpt_model(model_name=model_name, ckpt_name=ckpt_name)
+            feature_mode = model.hparams.feature
+            model.eval()
 
     # Setup logging file
     create_file_handler('audiofe.log')
@@ -107,26 +153,28 @@ def main() -> None:
         except ValueError as e:
             logger.exception(e)
             return
+        # Only write if capture, not for file input
+        write_wav(capture_wav, capture_audio, SAMPLE_RATE)
+        logger.info('')
+        logger.info(f'Wrote capture audio with shape {capture_audio.shape} to {capture_wav}')
 
-    write_wav(capture_name, capture_audio, SAMPLE_RATE)
-    logger.info('')
-    logger.info(f'Wrote capture audio with shape {capture_audio.shape} to {capture_name}')
     if debug:
         with h5py.File(h5_name, 'a') as f:
             if 'capture_audio' in f:
                 del f['capture_audio']
             f.create_dataset('capture_audio', data=capture_audio)
-        logger.info(f'Wrote capture audio with shape {capture_audio.shape} to {h5_name}')
+        logger.info(f'Wrote capture feature data with shape {capture_audio.shape} to {h5_name}')
 
     if asr_name is not None:
+        logger.info(f'Running ASR on captured audio with {asr_name} ...')
         capture_asr = calc_asr(capture_audio, engine=asr_name, whisper_model_name=whisper_name).text
         logger.info(f'Capture audio ASR: {capture_asr}')
 
     if model_name is not None:
-        model = load_torchl_ckpt_model(model_name=model_name, ckpt_name=ckpt_name)
-        model.eval()
+        feature = get_feature_from_audio(audio=capture_audio, feature_mode=feature_mode)  #frames x stride x feat_params
+        save_figure(capture_png, capture_audio, feature)
+        logger.info(f'Wrote capture plots to {capture_png}')
 
-        feature = get_feature_from_audio(audio=capture_audio, feature_mode=model.hparams.feature)
         if debug:
             with h5py.File(h5_name, 'a') as f:
                 if 'feature' in f:
@@ -134,23 +182,20 @@ def main() -> None:
                 f.create_dataset('feature', data=feature)
             logger.info(f'Wrote feature with shape {feature.shape} to {h5_name}')
 
-        # if debug:
-        #     reconstruct_name = f'audiofe_reconstruct_{ts}.wav'
-        #     reconstruct_audio = get_audio_from_feature(feature=feature, feature_mode=model.hparams.feature)
-        #     samples = min(len(capture_audio), len(reconstruct_audio))
-        #     max_err = np.max(np.abs(capture_audio[:samples] - reconstruct_audio[:samples]))
-        #     logger.info(f'Maximum error between capture and reconstruct: {max_err}')
-        #     write_wav(reconstruct_name, reconstruct_audio, SAMPLE_RATE)
-        #     logger.info(f'Wrote reconstruct audio with shape {reconstruct_audio.shape} to {reconstruct_name}')
-        #     with h5py.File(h5_name, 'a') as f:
-        #         if 'reconstruct_audio' in f:
-        #             del f['reconstruct_audio']
-        #         f.create_dataset('reconstruct_audio', data=reconstruct_audio)
-        #     logger.info(f'Wrote reconstruct audio with shape {reconstruct_audio.shape} to {h5_name}')
-
-        with torch.no_grad():
-            # model wants batch x timesteps x feature_parameters
-            predict = model(torch.tensor(feature).permute((1, 0, 2))).permute(1, 0, 2).numpy()
+        if model_is_onnx:
+            # run ort session, wants i.e. batch x tsteps x feat_params, outputs numpy BxTxFP or BxFP
+            # Note full reshape not needed here since we assume speech enhanement type model, so a transpose suffices
+            if model_is_fp16:
+                feature = np.float16(feature)
+            # run inference, ort session wants i.e. batch x tsteps x feat_params, outputs numpy BxTxFP or BxFP
+            predict = np.transpose(session.run(out_names, {in0name: np.transpose(feature,(1,0,2))})[0],(1,0,2))
+        else:
+            with torch.no_grad():
+                # model wants batch x timesteps x feature_parameters
+                predict = model(torch.tensor(feature).permute((1, 0, 2))).permute(1, 0, 2).numpy()
+
+
+
         if debug:
             with h5py.File(h5_name, 'a') as f:
                 if 'predict' in f:
@@ -158,9 +203,9 @@ def main() -> None:
                 f.create_dataset('predict', data=predict)
             logger.info(f'Wrote predict with shape {predict.shape} to {h5_name}')
 
-        predict_audio = get_audio_from_feature(feature=predict, feature_mode=model.hparams.feature)
-        write_wav(predict_name, predict_audio, SAMPLE_RATE)
-        logger.info(f'Wrote predict audio with shape {predict_audio.shape} to {predict_name}')
+        predict_audio = get_audio_from_feature(feature=predict, feature_mode=feature_mode)
+        write_wav(predict_wav, predict_audio, SAMPLE_RATE)
+        logger.info(f'Wrote predict audio with shape {predict_audio.shape} to {predict_wav}')
         if debug:
             with h5py.File(h5_name, 'a') as f:
                 if 'predict_audio' in f:
@@ -168,12 +213,27 @@ def main() -> None:
                 f.create_dataset('predict_audio', data=predict_audio)
             logger.info(f'Wrote predict audio with shape {predict_audio.shape} to {h5_name}')
 
+        save_figure(predict_png, predict_audio, predict)
+        logger.info(f'Wrote predict plots to {predict_png}')
+
         if asr_name is not None:
+            logger.info(f'Running ASR on model-enhanced audio with {asr_name} ...')
             predict_asr = calc_asr(predict_audio, engine=asr_name, whisper_model_name=whisper_name).text
             logger.info(f'Predict audio ASR: {predict_asr}')
 
 
 def get_frames_from_device(input_name: str | None, length: float, chunk: int = 1024) -> AudioT:
+    from select import select
+    from sys import stdin
+
+    import pyaudio
+
+    from sonusai import logger
+    from sonusai.mixture import CHANNEL_COUNT
+    from sonusai.mixture import SAMPLE_RATE
+    from sonusai.utils import get_input_device_index_by_name
+    from sonusai.utils import get_input_devices
+
     p = pyaudio.PyAudio()
 
     input_devices = get_input_devices(p)
@@ -224,6 +284,10 @@ def get_frames_from_device(input_name: str | None, length: float, chunk: int = 1
 
 
 def get_frames_from_file(input_name: str, length: float) -> AudioT:
+    from sonusai import logger
+    from sonusai.mixture import SAMPLE_RATE
+    from sonusai.mixture import read_audio
+
     logger.info(f'Capturing from {input_name}')
     frames = read_audio(input_name)
     if length != -1:
@@ -233,5 +297,37 @@ def get_frames_from_file(input_name: str, length: float) -> AudioT:
     return frames
 
 
+def save_figure(name: str, audio: np.ndarray, feature: np.ndarray) -> None:
+    import matplotlib.pyplot as plt
+    from scipy.interpolate import CubicSpline
+
+    from sonusai.mixture import SAMPLE_RATE
+    from sonusai.utils import unstack_complex
+
+    spectrum = 20 * np.log(np.abs(np.squeeze(unstack_complex(feature)).transpose()))
+    frames = spectrum.shape[1]
+    samples = (len(audio) // frames) * frames
+    length_in_s = samples / SAMPLE_RATE
+    interp = samples // frames
+
+    ts = np.arange(0.0, length_in_s, interp / SAMPLE_RATE)
+    t = np.arange(0.0, length_in_s, 1 / SAMPLE_RATE)
+
+    spectrum = CubicSpline(ts, spectrum, axis=-1)(t)
+
+    fig, (ax1, ax2) = plt.subplots(nrows=2)
+    ax1.set_title(name)
+    ax1.plot(t, audio[:samples])
+    ax1.set_ylabel('Signal')
+    ax1.set_xlim(0, length_in_s)
+    ax1.set_ylim(-1, 1)
+
+    ax2.imshow(spectrum, origin='lower', aspect='auto')
+    ax2.set_xticks([])
+    ax2.set_ylabel('Feature')
+
+    plt.savefig(name, dpi=300)
+
+
 if __name__ == '__main__':
     main()