[fix] doc and bpm work

This commit is contained in:
alban 2020-09-29 00:56:54 +02:00
parent 7f42b10b95
commit 9887f62202
2 changed files with 183 additions and 75 deletions

152
README.md
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@ -1,14 +1,108 @@
# Redilysis = Redis + Audio Analysis
# (Audio Analysis | redis ) == <3
Redilysis sends audio analysis to a redis server.
Redilysis sends audio analysis to a redis server.
The idea is to share a single audio analysis to many Visual Jockey filters, in our case for lasers.
Two modes exist for now, you need to run two processes to get the complete experience!
**Two modes are available, so you might need to run two processes for full analysis.**
### Spectrum Mode
This is the default mode.
### Redis Keys and Contents
Each **word in bold** is a key which you can query the redis server for. Ex:
```
$ redis-cli get spectrum_120
"[2.21, 0.56, 0.51, 0.32, 0.27, 0.21, 0.18, 0.17, 0.18, 0.23]"
```
**rms**
* **Mode** spectrum
* **Type** float number
* **Length** scalar
* **Meaning** Represents the root-mean-square -a mean value- for all frequencies between ```C0``` and ```C9```, e.g. between 12Hz and 8,372Hz.
* **Use** A fairly basic information about the scene audio volume.
* **Example**
* ```"0.12"```
* The audio volume for the scene is pretty low.
* It is obtained by averaging the RMS of every audio frame during the capture.
**spectrum_10**
* **Mode** spectrum
* **Type** array of float numbers (0.0-10.0)
* **Length** 10
* **Meaning** Represents the audio volume for the 10 **octaves** between ```C0``` and ```C9```, e.g. between 12Hz and 8,372Hz.
* **Use** A simple and useful way to get a global idea of the sound landscape.
* **Example**
* ```"[2.21, 0.56, 0.51, 0.32, 0.27, 0.21, 0.18, 0.17, 0.18, 0.23]"```
* The audio volume for the `C4` octave is `spectrum_10[4]`.
* That value being `0.27` is pretty low meaning almost no audio volume for that octave.
* It is calculated by averaging the volume of the octave's notes, e.g. `C4, D4, D#4, E4, F4, F#4, G4, G#4, A4, A#4, B4`.
**spectrum_120**
* **Mode** spectrum
* **Type** array of float numbers (0.0-10.0)
* **Length** 120
* **Meaning** Represents the audio volume for the 120 **notes** between ```C0``` and ```C9```, e.g. between 12Hz and 8,372Hz.
* **Use** More detailed than spectrum_10, it allows to find the standing out notes of the audio landscape.
* **Example**
* ```"[5.55, 2.61, 2.49, 1.79, 2.09, 4.35, 1.99, 1.57, 1.47, 0.77, 0.91, 0.89, 0.85, 0.56, 0.53, 0.73, 0.53, 0.46, 0.43, 0.44, 0.27, 0.45, 0.7, 0.81, 0.98, 0.7, 0.71, 0.6, 0.83, 0.51, 0.32, 0.31, 0.33, 0.24, 0.25, 0.33, 0.39, 0.43, 0.51, 0.28, 0.27, 0.25, 0.38, 0.25, 0.27, 0.3, 0.2, 0.27, 0.35, 0.29, 0.34, 0.3, 0.27, 0.27, 0.22, 0.21, 0.21, 0.29, 0.22, 0.28, 0.18, 0.19, 0.25, 0.26, 0.25, 0.24, 0.2, 0.21, 0.19, 0.18, 0.19, 0.17, 0.2, 0.17, 0.18, 0.17, 0.15, 0.17, 0.19, 0.18, 0.21, 0.16, 0.16, 0.18, 0.15, 0.13, 0.14, 0.16, 0.2, 0.17, 0.17, 0.2, 0.18, 0.16, 0.18, 0.15, 0.15, 0.16, 0.16, 0.19, 0.19, 0.19, 0.17, 0.18, 0.17, 0.19, 0.23, 0.23, 0.2, 0.23, 0.24, 0.36, 0.34, 0.23, 0.22, 0.2, 0.19, 0.18, 0.21, 0.21]"```
* The audio volume for the `C2` note is `spectrum_10[23]` (12x2 - 1).
* That value being `0.81` is average meaning there is some audio volume for that octave.
bpm
* **Mode** bpm
* **Type**
* **Length**
* **Meaning** Represents
* **Use**
* **Example**
bpm_sample_interval
* **Mode** bpm
* **Type**
* **Length**
* **Meaning** Represents
* **Example**
bpm_delay
* **Mode** bpm
* **Type**
* **Length**
* **Meaning** Represents
* **Example**
beats
* **Mode** bpm
* **Type**
* **Length**
* **Meaning** Represents
* **Example**
### Requirements and installation
* python 2.7
* audio card
* redis server
#### Installation
```python
sudo apt install python-pyaudio python
git clone https://git.interhacker.space/tmplab/redilysis.git
cd redilysis
pip install -r requirements.txt
python redilysis.py --help
```
### Running in Spectrum Mode
```
python redilysis.py -m spectrum
```
This is the default mode.
It performs some frequency analysis (Fast Fourier Transform) to detect "energy" in the human audition bandwidths.
@ -18,53 +112,15 @@ It can run at sub-second frequency (100ms) with no problem.
It reports realistic data: spectrum analysis is the easy part.
### BPM Mode
### Running in BPM Mode
This mode is more experimental.
It attempts to detect beats based on the
## Keys and contents in Redis
bpm_time : (milliseconds integer timestamp) last update time
onset
bpm
beats
spectrum_time
## Installation
```python
sudo apt install python-pyaudio python3
git clone https://git.interhacker.space/tmplab/redilysis.git
cd redilysis
pip install -r requirements.txt
python3 redilysis.py --help
```
python redilysis.py -m bpm -s 0.5
```
## Guide
This mode is experimental.
There are two available modes.
**One is the slow mode with BPM recognition:**
python3 redilysis.py -m bpm -s 1 -f 44100
Pushes following keys in redis:
* onset
* bpm
* beats
It attempts to detect beats based on complex parameters.
**The other is a fast mode with spectrogram analysis**
python3 redilysis.py -m spectrum -s 0.1 -f 4410
Pushes following keys in redis:
* rms
* spectrum
* tuning

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@ -34,42 +34,52 @@ _FRAMES_PER_BUFFER = 4410
_N_FFT = 4096
_RATE = 44100
_SAMPLING_FREQUENCY = 0.1
_BPM_MIN=10
_BPM_MAX=400
# Argument parsing
# Audio Args
parser = argparse.ArgumentParser(prog='realtime_redis')
# Audio Capture Args
parser.add_argument('--list-devices','-L', action='store_true', help='Which devices are detected by pyaudio')
parser.add_argument('--mode','-m', required=False, default='spectrum', choices=['spectrum', 'bpm'], type=str, help='Which mode to use. Default=spectrum')
parser.add_argument('--device','-d', required=False, type=int, help='Which pyaudio device to use')
#parser.add_argument('--frames','-f', required=False, default=4410, type=int, help='How many frames per buffer. Default={}'.format(_FRAMES_PER_BUFFER))
parser.add_argument('--sampling-frequency','-s', required=False, default=0.1, type=float, help='Which frequency, in seconds. Default={}f '.format(_SAMPLING_FREQUENCY))
parser.add_argument('--channels','-c', required=False, default=_CHANNELS, type=int, help='How many channels. Default={} '.format(_CHANNELS))
parser.add_argument('--rate','-r', required=False, default=44100, type=int, help='Which rate. Default={} '.format(_RATE))
parser.add_argument('--rate','-r', required=False, default=44100, type=int, help='The audio capture rate in Hz. Default={} '.format(_RATE))
#parser.add_argument('--frames','-f', required=False, default=4410, type=int, help='How many frames per buffer. Default={}'.format(_FRAMES_PER_BUFFER))
# BPM Mode Args
parser.add_argument('--bpm-min', required=False, default=_BPM_MIN, type=int, help='BPM mode only. The low BPM threshold. Default={} '.format(_BPM_MIN))
parser.add_argument('--bpm-max', required=False, default=_BPM_MAX, type=int, help='BPM mode only. The high BPM threshold. Default={} '.format(_BPM_MAX))
# Redis Args
parser.add_argument("-i","--ip",help="IP address of the Redis server ",default="127.0.0.1",type=str)
parser.add_argument("-p","--port",help="Port of the Redis server ",default="6379",type=str)
# Stardard Args
# Standard Args
parser.add_argument("-v","--verbose",action="store_true",help="Verbose")
args = parser.parse_args()
# global
bpm = 120.0
start = 0
# Set real variables
F_LO = librosa.note_to_hz('C0')
F_HI = librosa.note_to_hz('C10')
BAND_TONES = _BAND_TONES
CHANNELS = args.channels
DEVICE = args.device
FRAMES_PER_BUFFER = int(args.rate * args.sampling_frequency )
LIST_DEVICES = args.list_devices
MODE = args.mode
N_FFT = _N_FFT
RATE = args.rate
SAMPLING_FREQUENCY = args.sampling_frequency
ip = args.ip
port = args.port
verbose = args.verbose
BAND_TONES = _BAND_TONES
N_FFT = _N_FFT
CHANNELS = args.channels
DEVICE = args.device
FRAMES_PER_BUFFER = int(args.rate * args.sampling_frequency )
LIST_DEVICES = args.list_devices
MODE = args.mode
RATE = args.rate
SAMPLING_FREQUENCY = args.sampling_frequency
bpm_min = args.bpm_min
bpm_max = args.bpm_max
ip = args.ip
port = args.port
verbose = args.verbose
if( MODE == "bpm" and SAMPLING_FREQUENCY < 0.5 ):
debug( "You should use a --sampling_frequency superior to 0.5 in BPM mode...")
@ -106,32 +116,73 @@ p = pyaudio.PyAudio()
def m_bpm(audio_data):
"""
This function saves slow analysis to redis
* onset
* bpm
* beat
"""
global bpm
global start
if( bpm <= 10):
bpm = 10
onset = librosa.onset.onset_detect(
y = audio_data,
sr = RATE
)
bpm_delay = SAMPLING_FREQUENCY + start - time.time()
# Detect tempo / bpm
new_bpm, beats = librosa.beat.beat_track(
y = audio_data,
sr = RATE,
trim = False,
start_bpm = bpm,
#start_bpm = bpm,
units = "time"
)
'''
new_bpm = librosa.beat.tempo(y = audio_data, sr=RATE)[0]
'''
# Correct the eventual octave error
if new_bpm < bpm_min or new_bpm > bpm_max:
octaveErrorList = [ 0.5, 2, 0.3333, 3 ]
for key,factor in enumerate(octaveErrorList):
correction = new_bpm * factor
if correction > bpm_min and correction < bpm_max:
debug( "Corrected bpm to:{}".format(correction))
new_bpm = correction
break
if new_bpm < bpm_min :
new_bpm = bpm_min
else :
new_bpm = bpm_max
'''
|start end|
Capture |........................|
BPM detect+Redis set ||
Client Redis get |
Time |........................||.............|
---SAMPLING_FREQUENCY----
- < TIME-START
Read Delay --------------- < 2*SAMPLING_FREQUENCY - PTTL
Delay -----------------------------------------
Beats |last beat
. known ...b....b....b....b....b.
. passed (...b....b....b.)
. guessed (..b....b....b....b...
Next Beat Calculation b....b....b....b.|..b
=> (Delay - last beat) + x*BPM/60 (with x >= read_delay/BPM/60)
Redis:
bpm_sample_interval
|........................|
bpm_delay
|.........................|
'''
bpm = new_bpm
# Save to Redis
r.set( 'onset', json.dumps( onset.tolist() ) )
r.set( 'bpm', json.dumps( new_bpm ) )
r.set( 'bpm', new_bpm, px=( 2* int(SAMPLING_FREQUENCY * 1000)))
r.set( 'bpm_sample_interval', SAMPLING_FREQUENCY )
r.set( 'bpm_delay', bpm_delay )
r.set( 'beats', json.dumps( beats.tolist() ) )
bpm = new_bpm
debug( "bpm:{} onset:{} beats:{}".format(bpm,onset,beats) )
debug( "bpm:{} bpm_delay:{} beats:{}".format(bpm,bpm_delay,beats) )
return True
def m_spectrum(audio_data):
@ -179,6 +230,7 @@ def m_spectrum(audio_data):
def callback(in_data, frame_count, time_info, status):
audio_data = numpy.fromstring(in_data, dtype=numpy.float32)
global start
start = time.time()
if MODE == 'spectrum':
m_spectrum(audio_data)