openCom-Companion/sbapp/sideband/audioproc.py

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import os
import io
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import sh
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import math
import time
import struct
import numpy as np
import RNS
import LXMF
if RNS.vendor.platformutils.is_android():
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from pyogg import OpusFile, OpusBufferedEncoder, OggOpusWriter
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from pydub import AudioSegment
else:
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if RNS.vendor.platformutils.is_linux():
from sbapp.pyogg import OpusFile, OpusBufferedEncoder, OggOpusWriter
else:
from pyogg import OpusFile, OpusBufferedEncoder, OggOpusWriter
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from sbapp.pydub import AudioSegment
codec2_modes = {
# LXMF.AM_CODEC2_450PWB: ???, # No bindings
# LXMF.AM_CODEC2_450: ???, # No bindings
LXMF.AM_CODEC2_700C: 700,
LXMF.AM_CODEC2_1200: 1200,
LXMF.AM_CODEC2_1300: 1300,
LXMF.AM_CODEC2_1400: 1400,
LXMF.AM_CODEC2_1600: 1600,
LXMF.AM_CODEC2_2400: 2400,
LXMF.AM_CODEC2_3200: 3200,
}
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def samples_from_ogg(file_path=None, output_rate=8000):
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if file_path != None and os.path.isfile(file_path):
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opus_file = OpusFile(file_path)
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audio = AudioSegment(
bytes(opus_file.as_array()),
frame_rate=opus_file.frequency,
sample_width=opus_file.bytes_per_sample,
channels=opus_file.channels)
audio = audio.split_to_mono()[0]
audio = audio.apply_gain(-audio.max_dBFS)
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audio = audio.set_frame_rate(output_rate)
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audio = audio.set_sample_width(2)
return audio.get_array_of_samples()
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def resample(samples, width, channels, input_rate, output_rate, normalize):
audio = AudioSegment(
samples,
frame_rate=input_rate,
sample_width=width,
channels=channels)
if normalize:
audio = audio.apply_gain(-audio.max_dBFS)
resampled_audio = audio.set_frame_rate(output_rate)
return resampled_audio.get_array_of_samples().tobytes()
def samples_to_ogg(samples=None, file_path=None, normalize=False, input_channels=1, input_sample_width=2, input_rate=8000, output_rate=12000, profile="audio"):
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try:
if file_path != None and samples != None:
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if input_rate != output_rate or normalize:
samples = resample(samples, input_sample_width, input_channels, input_rate, output_rate, normalize)
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pcm_data = io.BytesIO(samples)
channels = input_channels; samples_per_second = output_rate; bytes_per_sample = 2
frame_duration_ms = 60
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opus_buffered_encoder = OpusBufferedEncoder()
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opus_buffered_encoder.set_application(profile)
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opus_buffered_encoder.set_sampling_frequency(samples_per_second)
opus_buffered_encoder.set_channels(channels)
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opus_buffered_encoder.set_frame_size(frame_duration_ms)
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ogg_opus_writer = OggOpusWriter(file_path, opus_buffered_encoder)
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frame_duration = frame_duration_ms/1000.0
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frame_size = int(frame_duration * samples_per_second)
bytes_per_frame = frame_size*bytes_per_sample
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ogg_opus_writer.write(memoryview(bytearray(samples)))
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ogg_opus_writer.close()
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return True
except Exception as e:
RNS.trace_exception(e)
return False
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def samples_to_wav(samples=None, file_path=None):
if samples != None and file_path != None:
import wave
with wave.open(file_path, "wb") as wf:
wf.setnchannels(1)
wf.setsampwidth(2)
wf.setframerate(8000)
wf.writeframes(samples)
return True
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def voice_processing(input_path):
try:
ffmpeg = None
ffmpeg = sh.ffmpeg
if ffmpeg:
filters = "highpass=f=250, lowpass=f=3000,speechnorm=e=12.5:r=0.0001:l=1"
output_bitrate = "12k"
opus_apptype = "audio"
output_path = input_path.replace(".ogg","")+".p.ogg"
args = [
"-i", input_path, "-filter:a", filters,
"-c:a", "libopus", "-application", opus_apptype,
"-vbr", "on","-b:a", output_bitrate, output_path]
try:
try:
os.unlink(output_path)
except:
pass
ffmpeg(*args)
return output_path
except Exception as e:
RNS.log("Could not process audio with ffmpeg", RNS.LOG_ERROR)
RNS.trace_exception(e)
return None
except Exception as e:
return None
def detect_codec2():
try:
import pycodec2
return True
except Exception as e:
RNS.log("Could not import codec2 module, libcodec2 is probably not installed or available", RNS.LOG_ERROR)
RNS.trace_exception(e)
return False
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# Samples must be 8KHz, 16-bit, 1 channel
def encode_codec2(samples, mode):
ap_start = time.time()
import pycodec2
if not mode in codec2_modes:
return None
c2 = pycodec2.Codec2(codec2_modes[mode])
SPF = c2.samples_per_frame()
PACKET_SIZE = SPF * 2 # 16-bit samples
STRUCT_FORMAT = '{}h'.format(SPF)
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F_FRAMES = len(samples)/SPF
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N_FRAMES = math.floor(len(samples)/SPF)
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# TODO: Add padding to align to whole frames
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frames = np.array(samples[0:N_FRAMES*SPF], dtype=np.int16)
encoded = b""
for pi in range(0, N_FRAMES):
pstart = pi*SPF
pend = (pi+1)*SPF
frame = frames[pstart:pend]
encoded_packet = c2.encode(frame)
encoded += encoded_packet
ap_duration = time.time() - ap_start
RNS.log("Codec2 encoding complete in "+RNS.prettytime(ap_duration)+", bytes out: "+str(len(encoded)), RNS.LOG_DEBUG)
return encoded
def decode_codec2(encoded_bytes, mode):
ap_start = time.time()
import pycodec2
if not mode in codec2_modes:
return None
c2 = pycodec2.Codec2(codec2_modes[mode])
SPF = c2.samples_per_frame()
BPF = c2.bytes_per_frame()
STRUCT_FORMAT = '{}h'.format(SPF)
N_FRAMES = math.floor(len(encoded_bytes)/BPF)
decoded = b""
for pi in range(0, N_FRAMES):
pstart = pi*BPF
pend = (pi+1)*BPF
encoded_packet = encoded_bytes[pstart:pend]
decoded_frame = c2.decode(encoded_packet)
decoded += struct.pack(STRUCT_FORMAT, *decoded_frame)
ap_duration = time.time() - ap_start
RNS.log("Codec2 decoding complete in "+RNS.prettytime(ap_duration)+", samples out: "+str(len(decoded)), RNS.LOG_DEBUG)
return decoded