Comments (15)
juha-h
commented on July 20, 2024
I dumped at the receiving end a sample of audio that baresip android sent using opus codec:
https://box.tutpro.com/tmp/dump-2018-05-11-15-57-52-enc.wav
As you can hear, the first second or so is OK and after that the trouble starts.
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juha-h
commented on July 20, 2024
I also tried with speex codec and again voice quality was fine. So it is only opus that has the issue.
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alfredh
commented on July 20, 2024
We should first try to establish if the problem is in the audio driver
or in the opus codec.
The auloop module will create a local audio loop where the recorded input
is just played out, and no audio codecs are involved. Can you please test
the /auloop command on your device with the sample rates listed in auloop.c:
static const struct {
uint32_t srate;
uint32_t ch;
} configv[] = {
{ 8000, 1},
{16000, 1},
{32000, 1},
{44100, 1},
{48000, 1},
{ 8000, 2},
{16000, 2},
{32000, 2},
{44100, 2},
{48000, 2},
};
Regarding the building of libopus, does it matter which Android NDK was used ?
Perhaps also try to compile it with fixed point, by adding this:
--enable-fixed-point compile without floating point (for machines without
a fast enough FPU)
Alfred
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juha-h
commented on July 20, 2024
Alfred E. Heggestad writes:
Regarding the building of libopus, does it matter which Android NDK was used ?
Perhaps also try to compile it with fixed point, by adding this:
```
--enable-fixed-point compile without floating point (for machines without
a fast enough FPU)
```
I only have one NDK (r16) installed. I tried with --enable-fixed-point
and perhaps audio quality was slightly better than without it, but still
much worse than G7221 or speex.
Regarding /auloop, I need to first implement it in my Android app,
…-- Juha
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juha-h
commented on July 20, 2024
Alfred E. Heggestad writes:
We should first try to establish if the problem is in the audio driver
or in the opus codec.
The auloop module will create a local audio loop where the recorded input
is just played out, and no audio codecs are involved. Can you please test
the `/auloop` command on your device with the sample rates listed in
auloop.c:
```
static const struct {
uint32_t srate;
uint32_t ch;
} configv[] = {
{ 8000, 1},
{16000, 1},
{32000, 1},
{44100, 1},
{48000, 1},
{ 8000, 2},
{16000, 2},
{32000, 2},
{44100, 2},
{48000, 2},
};
```
How does one select the rate and channel count or how to go to the next
one? It starts with 8000/1, which sounds OK on my Android device.
…-- Juha
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juha-h
commented on July 20, 2024
Found out how to go to the next rate/ch: give auloop command again.
…-- Juha
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juha-h
commented on July 20, 2024
Alfred E. Heggestad writes:
The auloop module will create a local audio loop where the recorded input
is just played out, and no audio codecs are involved. Can you please test
the `/auloop` command on your device with the sample rates listed in
auloop.c:
```
static const struct {
uint32_t srate;
uint32_t ch;
} configv[] = {
{ 8000, 1},
{16000, 1},
{32000, 1},
{44100, 1},
{48000, 1},
{ 8000, 2},
{16000, 2},
{32000, 2},
{44100, 2},
{48000, 2},
};
```
Did all the tests and audio was OK at all rates and ch counts.
Based on source code of auloop module, it is possible to also set
auloop_codec in config file. Will try that next.
…-- Juha
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juha-h
commented on July 20, 2024
Forgot to mention that when I compile opus lib, I get this kind of note:
CC src/opus_decoder.lo
src/opus_decoder.c:37:10: note: #pragma message: You appear to be compiling without optimization, if so opus will be very slow.
# pragma message "You appear to be compiling without optimization, if so opus will be very slow."
^
I searched the net and got the impression that the note is nothing to
worry about, since the optimization only applies to x86 architecture.
…-- Juha
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juha-h
commented on July 20, 2024
juha-h writes:
Based on source code of auloop module, it is possible to also set
auloop_codec in config file. Will try that next.
I added to the config
auloop_codec OPUS/48000/2
and run auloop:
05-13 13:13:52.319 27349 27366 D Baresip : Audio-loop started: 44100Hz, 2ch
05-13 13:13:53.757 27349 27349 D Baresip : processing command 'auloop'
05-13 13:13:53.757 27349 27366 D Baresip : auloop: could not find codec: OPUS/48000/2
05-13 13:13:53.838 27349 27366 D Baresip : Audio-loop: 48000Hz, 2ch
05-13 13:13:53.867 27349 27366 D Baresip :
05-13 13:13:53.867 27349 27366 D Baresip : Audio-loop started: 48000Hz, 2ch
48000Hz 2ch S16LE n_read=1156 n_write=1292 rw_ratio=0.89 codec='OPUS/48000/2'
48000Hz 2ch S16LE n_read=1166 n_write=1302 rw_ratio=0.89 codec='OPUS/48000/2'
...
48000Hz 2ch S16LE n_read=4060 n_write=4196 rw_ratio=0.96 codec='OPUS/48000/2'
48000Hz 2ch S16LE n_read=4070 n_write=4206 rw_ratio=0.96 codec='OPUS/48000/2'
05-13 13:14:26.508 27349 27349 D Baresip : processing command 'auloop'
Voice quality was much better in the loop than in actual call to another
baresip, where voice is sounds broken.
…-- Juha
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alfredh
commented on July 20, 2024
I did a test using the aufile module as audio input, reading from a WAV file,
excluding errors in the opensles driver.
NDK: r16b
API: 21
Device: Samsung J5
Opus: version 1.2.1
The result is good audio, no crackling.
The Baresip output is here:
shell@j5xnlte:/data/local/tmp $ ./baresip
baresip v0.5.9 Copyright (C) 2010 - 2018 Alfred E. Heggestad et al.
Local network address: IPv4=:10.0.0.104
aucodec: opus/48000/2
auplay: opensles
ausrc: opensles
ausrc: aubridge
auplay: aubridge
ausrc: aufile
medianat: stun
medianat: turn
medianat: ice
Populated 1 account
Populated 2 contacts
Populated 1 audio codec
Populated 0 audio filters
Populated 0 video codecs
Populated 0 video filters
baresip is ready.
[email protected]: {0/UDP/v4} 200 OK (callflow v0.1.0 (arm6/linux)) [1 binding]
All 1 useragent registered successfully! (14 ms)
answering call from sip:[email protected] with 200
audio: Set audio decoder: opus 48000Hz 2ch
audio: player started with sample format S16LE
audio: Set audio encoder: opus 48000Hz 2ch
aufile: loading input file 'lr.wav'
aufile: lr.wav: 48000 Hz, 2 channels
aufile: audio ptime=20 sampc=1920 aubuf=[192000:3840000]
aufile: end of file
aufile: loaded 7442240 bytes
audio: source started with sample format S16LE
audio tx pipeline: aufile ---> opus
audio rx pipeline: opensles <--- opus
stream: incoming rtp for 'audio' established, receiving from 10.0.0.109:11546
[email protected]: Call established: sip:[email protected]
aufile: end of file34/31320 (bit/s)
aufile: player thread exited
sip:[email protected]: Call with sip:[email protected] terminated (duration: 39 secs)
audio Transmit: Receive:
packets: 1951 1888
avg. bitrate: 32.2 31.5 (kbit/s)
errors: 0 0
pkt.report: 1754 1694
lost: 0 15
jitter: 2.3 3.6 (ms)
Quit
ua: stop all (forced=0)
Stopping 1 useragent..
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juha-h
commented on July 20, 2024
Alfred E. Heggestad writes:
I did a test using the `aufile` module as audio input, reading from a
WAV file, excluding errors in the opensles driver.
NDK: r16b
API: 21
Device: Samsung J5
Opus: version 1.2.1
The result is good audio, no crackling.
I can try that too. Would it be possible for me to try with your
libopus.a?
…-- Juha
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alfredh
commented on July 20, 2024
it is possible that the opus encoder is using too much CPU. if this happens
the whole audio input will be blocked, leading to lost PCM samples inside Baresip.
couple of more things to try:
-
After running a call for >10 seconds press
Ato see the audio debug
(pay attention to transmit aubuf and overruns/underruns) -
Test again with aufile module and run Baresip with -v, using this commit:
- Try
audio_txmode thread
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juha-h
commented on July 20, 2024
Alfred,
Since opus audio is fine when I tried with your libopus.a, I started the
make process from scratch in order to find out if I had changed
something in the Makefile. I also downloaded android-ndk-r16b.
The result was that opus does not compile:
make[3]: Entering directory '/usr/src/baresip-android/opus-1.2.1'
CC celt/bands.lo
CC celt/celt.lo
In file included from celt/celt.c:48:0:
celt/float_cast.h:129:10: error: #warning "Don't have the functions lrint() and lrintf ()." [-Werror=cpp]
#warning "Don't have the functions lrint() and lrintf ()."
^
celt/float_cast.h:130:10: error: #warning "Replacing these functions with a standard C cast." [-Werror=cpp]
#warning "Replacing these functions with a standard C cast."
^
cc1: all warnings being treated as errors
It does compile when I'm using the ndk that comes with Android Studio.
So perhaps the whole mess is caused by ndk in Android Studio (which I
assumed is r16).
Anyway, how did you overcome the above compilation error? Have you
added some extra CFLAGS to your Makefile?
…-- Juha
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juha-h
commented on July 20, 2024
juha-h writes:
It does compile when I'm using the ndk that comes with Android Studio.
So perhaps the whole mess is caused by ndk in Android Studio (which I
assumed is r16).
Anyway, how did you overcome the above compilation error? Have you
added some extra CFLAGS to your Makefile?
Alfred,
I added -Wno-error flag to the opus section of Makefile and was able to
build libopus.a. Then I gave it a try in my Android app and the result
was that opus audio worked fine!
So this whole mess was caused by the NDK bundled in Android Studio.
Thanks for your help. I'll close this issue.
…-- Juha
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juha-h
commented on July 20, 2024
Opus audio quality is good when the lib is built using android-ndk-r16b instead of the ndk bundled in Android Studio.
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