[fix] doc and bpm work
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README.md
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README.md
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# Redilysis = Redis + Audio Analysis
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# (Audio Analysis | redis ) == <3
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Redilysis sends audio analysis to a redis server.
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Redilysis sends audio analysis to a redis server.
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The idea is to share a single audio analysis to many Visual Jockey filters, in our case for lasers.
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Two modes exist for now, you need to run two processes to get the complete experience!
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**Two modes are available, so you might need to run two processes for full analysis.**
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### Spectrum Mode
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This is the default mode.
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### Redis Keys and Contents
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Each **word in bold** is a key which you can query the redis server for. Ex:
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```
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$ redis-cli get spectrum_120
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"[2.21, 0.56, 0.51, 0.32, 0.27, 0.21, 0.18, 0.17, 0.18, 0.23]"
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```
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**rms**
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* **Mode** spectrum
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* **Type** float number
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* **Length** scalar
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* **Meaning** Represents the root-mean-square -a mean value- for all frequencies between ```C0``` and ```C9```, e.g. between 12Hz and 8,372Hz.
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* **Use** A fairly basic information about the scene audio volume.
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* **Example**
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* ```"0.12"```
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* The audio volume for the scene is pretty low.
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* It is obtained by averaging the RMS of every audio frame during the capture.
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**spectrum_10**
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* **Mode** spectrum
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* **Type** array of float numbers (0.0-10.0)
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* **Length** 10
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* **Meaning** Represents the audio volume for the 10 **octaves** between ```C0``` and ```C9```, e.g. between 12Hz and 8,372Hz.
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* **Use** A simple and useful way to get a global idea of the sound landscape.
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* **Example**
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* ```"[2.21, 0.56, 0.51, 0.32, 0.27, 0.21, 0.18, 0.17, 0.18, 0.23]"```
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* The audio volume for the `C4` octave is `spectrum_10[4]`.
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* That value being `0.27` is pretty low meaning almost no audio volume for that octave.
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* 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`.
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**spectrum_120**
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* **Mode** spectrum
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* **Type** array of float numbers (0.0-10.0)
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* **Length** 120
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* **Meaning** Represents the audio volume for the 120 **notes** between ```C0``` and ```C9```, e.g. between 12Hz and 8,372Hz.
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* **Use** More detailed than spectrum_10, it allows to find the standing out notes of the audio landscape.
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* **Example**
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* ```"[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]"```
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* The audio volume for the `C2` note is `spectrum_10[23]` (12x2 - 1).
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* That value being `0.81` is average meaning there is some audio volume for that octave.
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bpm
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* **Mode** bpm
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* **Type**
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* **Length**
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* **Meaning** Represents
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* **Use**
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* **Example**
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bpm_sample_interval
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* **Mode** bpm
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* **Type**
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* **Length**
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* **Meaning** Represents
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* **Example**
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bpm_delay
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* **Mode** bpm
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* **Type**
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* **Length**
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* **Meaning** Represents
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* **Example**
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beats
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* **Mode** bpm
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* **Type**
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* **Length**
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* **Meaning** Represents
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* **Example**
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### Requirements and installation
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* python 2.7
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* audio card
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* redis server
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#### Installation
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```python
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sudo apt install python-pyaudio python
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git clone https://git.interhacker.space/tmplab/redilysis.git
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cd redilysis
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pip install -r requirements.txt
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python redilysis.py --help
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```
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### Running in Spectrum Mode
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```
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python redilysis.py -m spectrum
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```
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This is the default mode.
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It performs some frequency analysis (Fast Fourier Transform) to detect "energy" in the human audition bandwidths.
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@ -18,53 +112,15 @@ It can run at sub-second frequency (100ms) with no problem.
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It reports realistic data: spectrum analysis is the easy part.
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### BPM Mode
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### Running in BPM Mode
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This mode is more experimental.
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It attempts to detect beats based on the
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## Keys and contents in Redis
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bpm_time : (milliseconds integer timestamp) last update time
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onset
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bpm
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beats
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spectrum_time
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## Installation
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```python
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sudo apt install python-pyaudio python3
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git clone https://git.interhacker.space/tmplab/redilysis.git
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cd redilysis
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pip install -r requirements.txt
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python3 redilysis.py --help
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```
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python redilysis.py -m bpm -s 0.5
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```
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## Guide
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This mode is experimental.
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There are two available modes.
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**One is the slow mode with BPM recognition:**
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python3 redilysis.py -m bpm -s 1 -f 44100
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Pushes following keys in redis:
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* onset
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* bpm
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* beats
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It attempts to detect beats based on complex parameters.
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**The other is a fast mode with spectrogram analysis**
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python3 redilysis.py -m spectrum -s 0.1 -f 4410
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Pushes following keys in redis:
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* rms
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* spectrum
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* tuning
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redilysis.py
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redilysis.py
@ -34,42 +34,52 @@ _FRAMES_PER_BUFFER = 4410
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_N_FFT = 4096
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_RATE = 44100
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_SAMPLING_FREQUENCY = 0.1
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_BPM_MIN=10
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_BPM_MAX=400
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# Argument parsing
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# Audio Args
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parser = argparse.ArgumentParser(prog='realtime_redis')
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# Audio Capture Args
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parser.add_argument('--list-devices','-L', action='store_true', help='Which devices are detected by pyaudio')
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parser.add_argument('--mode','-m', required=False, default='spectrum', choices=['spectrum', 'bpm'], type=str, help='Which mode to use. Default=spectrum')
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parser.add_argument('--device','-d', required=False, type=int, help='Which pyaudio device to use')
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#parser.add_argument('--frames','-f', required=False, default=4410, type=int, help='How many frames per buffer. Default={}'.format(_FRAMES_PER_BUFFER))
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parser.add_argument('--sampling-frequency','-s', required=False, default=0.1, type=float, help='Which frequency, in seconds. Default={}f '.format(_SAMPLING_FREQUENCY))
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parser.add_argument('--channels','-c', required=False, default=_CHANNELS, type=int, help='How many channels. Default={} '.format(_CHANNELS))
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parser.add_argument('--rate','-r', required=False, default=44100, type=int, help='Which rate. Default={} '.format(_RATE))
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parser.add_argument('--rate','-r', required=False, default=44100, type=int, help='The audio capture rate in Hz. Default={} '.format(_RATE))
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#parser.add_argument('--frames','-f', required=False, default=4410, type=int, help='How many frames per buffer. Default={}'.format(_FRAMES_PER_BUFFER))
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# BPM Mode Args
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parser.add_argument('--bpm-min', required=False, default=_BPM_MIN, type=int, help='BPM mode only. The low BPM threshold. Default={} '.format(_BPM_MIN))
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parser.add_argument('--bpm-max', required=False, default=_BPM_MAX, type=int, help='BPM mode only. The high BPM threshold. Default={} '.format(_BPM_MAX))
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# Redis Args
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parser.add_argument("-i","--ip",help="IP address of the Redis server ",default="127.0.0.1",type=str)
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parser.add_argument("-p","--port",help="Port of the Redis server ",default="6379",type=str)
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# Stardard Args
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# Standard Args
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parser.add_argument("-v","--verbose",action="store_true",help="Verbose")
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args = parser.parse_args()
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# global
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bpm = 120.0
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start = 0
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# Set real variables
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F_LO = librosa.note_to_hz('C0')
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F_HI = librosa.note_to_hz('C10')
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BAND_TONES = _BAND_TONES
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CHANNELS = args.channels
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DEVICE = args.device
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FRAMES_PER_BUFFER = int(args.rate * args.sampling_frequency )
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LIST_DEVICES = args.list_devices
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MODE = args.mode
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N_FFT = _N_FFT
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RATE = args.rate
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SAMPLING_FREQUENCY = args.sampling_frequency
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ip = args.ip
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port = args.port
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verbose = args.verbose
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BAND_TONES = _BAND_TONES
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N_FFT = _N_FFT
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CHANNELS = args.channels
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DEVICE = args.device
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FRAMES_PER_BUFFER = int(args.rate * args.sampling_frequency )
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LIST_DEVICES = args.list_devices
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MODE = args.mode
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RATE = args.rate
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SAMPLING_FREQUENCY = args.sampling_frequency
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bpm_min = args.bpm_min
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bpm_max = args.bpm_max
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ip = args.ip
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port = args.port
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verbose = args.verbose
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if( MODE == "bpm" and SAMPLING_FREQUENCY < 0.5 ):
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debug( "You should use a --sampling_frequency superior to 0.5 in BPM mode...")
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@ -106,32 +116,73 @@ p = pyaudio.PyAudio()
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def m_bpm(audio_data):
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"""
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This function saves slow analysis to redis
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* onset
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* bpm
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* beat
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"""
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global bpm
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global start
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if( bpm <= 10):
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bpm = 10
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onset = librosa.onset.onset_detect(
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y = audio_data,
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sr = RATE
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)
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bpm_delay = SAMPLING_FREQUENCY + start - time.time()
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# Detect tempo / bpm
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new_bpm, beats = librosa.beat.beat_track(
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y = audio_data,
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sr = RATE,
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trim = False,
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start_bpm = bpm,
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#start_bpm = bpm,
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units = "time"
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)
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'''
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new_bpm = librosa.beat.tempo(y = audio_data, sr=RATE)[0]
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'''
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# Correct the eventual octave error
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if new_bpm < bpm_min or new_bpm > bpm_max:
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octaveErrorList = [ 0.5, 2, 0.3333, 3 ]
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for key,factor in enumerate(octaveErrorList):
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correction = new_bpm * factor
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if correction > bpm_min and correction < bpm_max:
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debug( "Corrected bpm to:{}".format(correction))
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new_bpm = correction
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break
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if new_bpm < bpm_min :
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new_bpm = bpm_min
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else :
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new_bpm = bpm_max
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'''
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|start end|
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Capture |........................|
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BPM detect+Redis set ||
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Client Redis get |
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Time |........................||.............|
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---SAMPLING_FREQUENCY----
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- < TIME-START
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Read Delay --------------- < 2*SAMPLING_FREQUENCY - PTTL
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Delay -----------------------------------------
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Beats |last beat
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. known ...b....b....b....b....b.
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. passed (...b....b....b.)
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. guessed (..b....b....b....b...
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Next Beat Calculation b....b....b....b.|..b
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=> (Delay - last beat) + x*BPM/60 (with x >= read_delay/BPM/60)
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Redis:
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bpm_sample_interval
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|........................|
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bpm_delay
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|.........................|
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'''
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bpm = new_bpm
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# Save to Redis
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r.set( 'onset', json.dumps( onset.tolist() ) )
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r.set( 'bpm', json.dumps( new_bpm ) )
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r.set( 'bpm', new_bpm, px=( 2* int(SAMPLING_FREQUENCY * 1000)))
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r.set( 'bpm_sample_interval', SAMPLING_FREQUENCY )
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r.set( 'bpm_delay', bpm_delay )
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r.set( 'beats', json.dumps( beats.tolist() ) )
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bpm = new_bpm
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debug( "bpm:{} onset:{} beats:{}".format(bpm,onset,beats) )
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debug( "bpm:{} bpm_delay:{} beats:{}".format(bpm,bpm_delay,beats) )
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return True
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def m_spectrum(audio_data):
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@ -179,6 +230,7 @@ def m_spectrum(audio_data):
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def callback(in_data, frame_count, time_info, status):
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audio_data = numpy.fromstring(in_data, dtype=numpy.float32)
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global start
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start = time.time()
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if MODE == 'spectrum':
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m_spectrum(audio_data)
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