While trying to use PyAV to encode live mono audio from a microphone to a compressed audio stream (using mp2 or flac as encoder), the program kept raising an exception ValueError: [Errno 22] Invalid argument.
To remove the live microphone source as a cause of the problem, and to make the problematic code easier for others to run/test, I have removed the mic source and now just generate a pure tone as a sequence of input buffers.
All attempts to figure out the missing or mismatched or incorrect argument have just resulted in seeing documentation and examples that are the same as my code.
I would like to know from someone who has used PyAV successfully for mono audio what the correct method and parameters are for encoding mono frames into the mono stream.
The package used is av 10.0.0 installed with
pip3 install av --no-binary av
so it uses my package-manager provided ffmpeg library, which is version 4.2.7.
The problematic python code is:
#!/usr/bin/env python3
# -*- coding: utf-8 -*-
"""
Recreating an error 22 when encoding sound with PyAV.
Created on Sun Feb 19 08:10:29 2023
@author: andrewm
"""
import typing
import sys
import math
import fractions
import av
from av import AudioFrame
""" Ensure some PyAudio constants are still defined without changing
the PyAudio recording callback function and without depending
on PyAudio simply for reproducing the PyAV bug [Errno 22] thrown in
File "av/filter/context.pyx", line 89, in av.filter.context.FilterContext.push
"""
class PA_Stub():
paContinue = True
paComplete= False
pyaudio = PA_Stub()
"""Generate pure tone at given frequency with amplitude 0...1.0 at
sampling frewuency fs and beginning at phase offset 'phase'.
Returns the new phase after the sinusoid has cycled over the
sampling window length.
"""
def generate_tone(
freq:int, phase:float, amp:float, fs, samp_fmt, buffer:bytearray
) -> float:
assert samp_fmt == "s16", "Only s16 supported atm"
samp_size_bytes = 2
n_samples = int(len(buffer)/samp_size_bytes)
window = [int(0) for i in range(n_samples)]
theta = phase
phase_inc = 2*math.pi * freq / fs
for i in range(n_samples):
v = amp * math.sin(theta)
theta += phase_inc
s = int((2**15-1)*v)
window[i] = s
for sample_i in range(len(window)):
byte_i = sample_i * samp_size_bytes
enc = window[sample_i].to_bytes(
2, byteorder=sys.byteorder, signed=True
)
buffer[byte_i] = enc[0]
buffer[byte_i+1] = enc[1]
return theta
channels = 1
fs = 44100 # Record at 44100 samples per second
fft_size_samps = 256
chunk_samps = fft_size_samps * 10 # Record in chunks that are multiples of fft windows.
# print(f"fft_size_samps={fft_size_samps}\nchunk_samps={chunk_samps}")
seconds = 3.0
out_filename = "testoutput.wav"
# Store data in chunks for 3 seconds
sample_limit = int(fs * seconds)
sample_len = 0
frames = [] # Initialize array to store frames
ffmpeg_codec_name = 'mp2' # flac, mp3, or libvorbis make same error.
sample_size_bytes = 2
buffer = bytearray(int(chunk_samps*sample_size_bytes))
chunkperiod = chunk_samps / fs
total_chunks = int(math.ceil(seconds / chunkperiod))
phase = 0.0
### uncomment if you want to see the synthetic data being used as a mic input.
# with open("test.raw","wb") as raw_out:
# for ci in range(total_chunks):
# phase = generate_tone(2600, phase, 0.8, fs, "s16", buffer)
# raw_out.write(buffer)
# print("finished gen test")
# sys.exit(0)
# #----
# Using mp2 or mkv as the container format gets the same error.
with av.open(out_filename+'.mp2', "w", format="mp2") as output_con:
output_con.metadata["title"] = "My title"
output_con.metadata["key"] = "value"
channel_layout = "mono"
sample_fmt = "s16p"
ostream = output_con.add_stream(ffmpeg_codec_name, fs, layout=channel_layout)
assert ostream is not None, "No stream!"
cctx = ostream.codec_context
cctx.sample_rate = fs
cctx.time_base = fractions.Fraction(numerator=1,denominator=fs)
cctx.format = sample_fmt
cctx.channels = channels
cctx.layout = channel_layout
print(cctx, f"layout#{cctx.channel_layout}")
# Define PyAudio-style callback for recording plus PyAV transcoding.
def rec_callback(in_data, frame_count, time_info, status):
global sample_len
global ostream
frames.append(in_data)
nsamples = int(len(in_data) / (channels*sample_size_bytes))
frame = AudioFrame(format=sample_fmt, layout=channel_layout, samples=nsamples)
frame.sample_rate = fs
frame.time_base = fractions.Fraction(numerator=1,denominator=fs)
frame.pts = sample_len
frame.planes[0].update(in_data)
print(frame, len(in_data))
for out_packet in ostream.encode(frame):
output_con.mux(out_packet)
for out_packet in ostream.encode(None):
output_con.mux(out_packet)
sample_len += nsamples
retflag = pyaudio.paContinue if sample_len<sample_limit else pyaudio.paComplete
return (in_data, retflag)
print('Beginning')
### some e.g. PyAudio code which starts the recording process normally.
# istream = p.open(
# format=sample_format,
# channels=channels,
# rate=fs,
# frames_per_buffer=chunk_samps,
# input=True,
# stream_callback=rec_callback
# )
# print(istream)
# Normally at this point you just sleep the main thread while
# PyAudio calls back with mic data, but here it is all generated.
for ci in range(total_chunks):
phase = generate_tone(2600, phase, 0.8, fs, "s16", buffer)
ret_data, ret_flag = rec_callback(buffer, ci, {}, 1)
print('.', end='')
print(" closing.")
# Stop and close the istream
# istream.stop_stream()
# istream.close()
If you uncomment the RAW output part you will find the generated data can be imported as PCM s16 Mono 44100Hz into Audacity and plays the expected tone, so the generated audio data does not seem to be the problem.
The normal program console output up until the exception is:
<av.AudioCodecContext audio/mp2 at 0x7f8e38202cf0> layout#4
Beginning
<av.AudioFrame 0, pts=0, 2560 samples at 44100Hz, mono, s16p at 0x7f8e38202eb0> 5120
.<av.AudioFrame 0, pts=2560, 2560 samples at 44100Hz, mono, s16p at 0x7f8e382025f0> 5120
The stack trace is:
Traceback (most recent call last):
File "Dev/multichan_recording/av_encode.py", line 147, in <module>
ret_data, ret_flag = rec_callback(buffer, ci, {}, 1)
File "Dev/multichan_recording/av_encode.py", line 121, in rec_callback
for out_packet in ostream.encode(frame):
File "av/stream.pyx", line 153, in av.stream.Stream.encode
File "av/codec/context.pyx", line 484, in av.codec.context.CodecContext.encode
File "av/audio/codeccontext.pyx", line 42, in av.audio.codeccontext.AudioCodecContext._prepare_frames_for_encode
File "av/audio/resampler.pyx", line 101, in av.audio.resampler.AudioResampler.resample
File "av/filter/graph.pyx", line 211, in av.filter.graph.Graph.push
File "av/filter/context.pyx", line 89, in av.filter.context.FilterContext.push
File "av/error.pyx", line 336, in av.error.err_check
ValueError: [Errno 22] Invalid argument
edit: It's interesting that the error happens on the 2nd AudioFrame, as apparently the first one was encoded okay, because they are given the same attribute values aside from the Presentation Time Stamp (pts), but leaving this out and letting PyAV/ffmpeg generate the PTS by itself does not fix the error, so an incorrect PTS does not seem the cause.
After a brief glance in av/filter/context.pyx the exception must come from a bad return value from res = lib.av_buffersrc_write_frame(self.ptr, frame.ptr)
Trying to dig into av_buffersrc_write_frame from the ffmpeg source it is not clear what could be causing this error. The only obvious one is a mismatch between channel layouts, but my code is setting the layout the same in the Stream and the Frame. That problem had been found by an old question pyav - cannot save stream as mono and their answer (that one parameter required is undocumented) is the only reason the code now has the layout='mono' argument when making the stream.
The program output shows layout #4 is being used, and from https://github.com/FFmpeg/FFmpeg/blob/release/4.2/libavutil/channel_layout.h you can see this is the value for symbol AV_CH_FRONT_CENTER which is the only channel in the MONO layout.
The mismatch is surely some other object property or an undocumented parameter requirement.
How do you encode mono audio to a compressed stream with PyAV?
The issue is related to ostream.encode(None).
The following loop is used for "flushing the encoder", and should be placed only at the end:
for out_packet in ostream.encode(None):
output_con.mux(out_packet)
Here is a reference (for example).
Your implementation executes ostream.encode(None) at the end of every iteration, so the first iteration succeeds, and the second iteration fails.
Execute the for out_packet in ostream.encode(None) loop only once at the end.
...
for ci in range(total_chunks):
phase = generate_tone(2600, phase, 0.8, fs, "s16", buffer)
ret_data, ret_flag = rec_callback(buffer, ci, {}, 1)
print('.', end='')
# Pass None to the encoder at the end - flush last packets
for out_packet in ostream.encode(None):
output_con.mux(out_packet)
Updated code:
#!/usr/bin/env python3
# -*- coding: utf-8 -*-
"""
Recreating an error 22 when encoding sound with PyAV.
Created on Sun Feb 19 08:10:29 2023
@author: andrewm
"""
import typing
import sys
import math
import fractions
import av
from av import AudioFrame
""" Ensure some PyAudio constants are still defined without changing
the PyAudio recording callback function and without depending
on PyAudio simply for reproducing the PyAV bug [Errno 22] thrown in
File "av/filter/context.pyx", line 89, in av.filter.context.FilterContext.push
"""
class PA_Stub():
paContinue = True
paComplete= False
pyaudio = PA_Stub()
"""Generate pure tone at given frequency with amplitude 0...1.0 at
sampling frewuency fs and beginning at phase offset 'phase'.
Returns the new phase after the sinusoid has cycled over the
sampling window length.
"""
def generate_tone(
freq:int, phase:float, amp:float, fs, samp_fmt, buffer:bytearray
) -> float:
assert samp_fmt == "s16", "Only s16 supported atm"
samp_size_bytes = 2
n_samples = int(len(buffer)/samp_size_bytes)
window = [int(0) for i in range(n_samples)]
theta = phase
phase_inc = 2*math.pi * freq / fs
for i in range(n_samples):
v = amp * math.sin(theta)
theta += phase_inc
s = int((2**15-1)*v)
window[i] = s
for sample_i in range(len(window)):
byte_i = sample_i * samp_size_bytes
enc = window[sample_i].to_bytes(
2, byteorder=sys.byteorder, signed=True
)
buffer[byte_i] = enc[0]
buffer[byte_i+1] = enc[1]
return theta
channels = 1
fs = 44100 # Record at 44100 samples per second
fft_size_samps = 256
chunk_samps = fft_size_samps * 10 # Record in chunks that are multiples of fft windows.
# print(f"fft_size_samps={fft_size_samps}\nchunk_samps={chunk_samps}")
seconds = 3.0
out_filename = "testoutput.wav"
# Store data in chunks for 3 seconds
sample_limit = int(fs * seconds)
sample_len = 0
frames = [] # Initialize array to store frames
ffmpeg_codec_name = 'mp2' # flac, mp3, or libvorbis make same error.
sample_size_bytes = 2
buffer = bytearray(int(chunk_samps*sample_size_bytes))
chunkperiod = chunk_samps / fs
total_chunks = int(math.ceil(seconds / chunkperiod))
phase = 0.0
### uncomment if you want to see the synthetic data being used as a mic input.
# with open("test.raw","wb") as raw_out:
# for ci in range(total_chunks):
# phase = generate_tone(2600, phase, 0.8, fs, "s16", buffer)
# raw_out.write(buffer)
# print("finished gen test")
# sys.exit(0)
# #----
# Encode test.raw as MP2 with MP2 codec using FFmpeg CLI (Rotem):
#https://stackoverflow.com/questions/11986279/can-ffmpeg-convert-audio-from-raw-pcm-to-wav
#https://stackoverflow.com/questions/68161835/why-is-ffmpeg-warning-guessed-channel-layout-for-input-stream-0-0-mono
#ffmpeg -y -f s16le -ar 44100 -ac 1 -channel_layout mono -i test.raw -acodec mp2 -ac 1 -ar 44100 -channel_layout mono testoutput_cli.mp2
# Using mp2 or mkv as the container format gets the same error.
with av.open(out_filename+'.mp2', "w", format="mp2") as output_con:
output_con.metadata["title"] = "My title"
output_con.metadata["key"] = "value"
channel_layout = "mono"
sample_fmt = "s16p"
ostream = output_con.add_stream(ffmpeg_codec_name, fs, layout=channel_layout)
assert ostream is not None, "No stream!"
cctx = ostream.codec_context
cctx.sample_rate = fs
cctx.time_base = fractions.Fraction(numerator=1,denominator=fs)
cctx.format = sample_fmt
cctx.channels = channels
cctx.layout = channel_layout
print(cctx, f"layout#{cctx.channel_layout}")
# Define PyAudio-style callback for recording plus PyAV transcoding.
def rec_callback(in_data, frame_count, time_info, status):
global sample_len
global ostream
frames.append(in_data)
nsamples = int(len(in_data) / (channels*sample_size_bytes))
frame = AudioFrame(format=sample_fmt, layout=channel_layout, samples=nsamples)
frame.sample_rate = fs
frame.time_base = fractions.Fraction(numerator=1,denominator=fs)
frame.pts = sample_len
frame.planes[0].update(in_data)
print(frame, len(in_data))
for out_packet in ostream.encode(frame):
output_con.mux(out_packet)
#for out_packet in ostream.encode(None):
# output_con.mux(out_packet)
sample_len += nsamples
retflag = pyaudio.paContinue if sample_len<sample_limit else pyaudio.paComplete
return (in_data, retflag)
print('Beginning')
### some e.g. PyAudio code which starts the recording process normally.
# istream = p.open(
# format=sample_format,
# channels=channels,
# rate=fs,
# frames_per_buffer=chunk_samps,
# input=True,
# stream_callback=rec_callback
# )
# print(istream)
# Normally at this point you just sleep the main thread while
# PyAudio calls back with mic data, but here it is all generated.
for ci in range(total_chunks):
phase = generate_tone(2600, phase, 0.8, fs, "s16", buffer)
ret_data, ret_flag = rec_callback(buffer, ci, {}, 1)
print('.', end='')
# Pass None to the encoder at the end - flush last packets
for out_packet in ostream.encode(None):
output_con.mux(out_packet)
print(" closing.")
# Stop and close the istream
# istream.stop_stream()
# istream.close()