Sound feedback
ricardojlrufino opened this issue · comments
It would be cool has sound feedback for reading.
I tried to put a beep, but logic works only on some terminals.
Line 334 in c51b58a
The beep doesn't work.
In the past, I tried to use alsa but no success.
I'm not an alsa expert.
The hw device is busy because rmm opens it and doesn't release it.
I don't know how to solve this problem.
Maybe using mixer? Reconfiguring alsa?
I don't know.
How microphone audio is captured?
Thanks to another user, alsa-lib is hacked.
The pcm audio is extracted and sent to a fifo.
Its possible play audio without rmm running
// A simple C example to play a mono or stereo, 16-bit 44KHz
// WAVE file using ALSA. This goes directly to the first
// audio card (ie, its first set of audio out jacks). It
// uses the snd_pcm_writei() mode of outputting waveform data,
// blocking.
//
// Compile as so to create "alsawave":
// gcc -o alsawave alsawave.c -lasound
//
// Run it from a terminal, specifying the name of a WAVE file to play:
// ./alsawave MyWaveFile.wav
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <stdint.h>
// Include the ALSA .H file that defines ALSA functions/data
#include <alsa/asoundlib.h>
#pragma pack (1)
/////////////////////// WAVE File Stuff /////////////////////
// An IFF file header looks like this
typedef struct _FILE_head
{
unsigned char ID[4]; // could be {'R', 'I', 'F', 'F'} or {'F', 'O', 'R', 'M'}
unsigned int Length; // Length of subsequent file (including remainder of header). This is in
// Intel reverse byte order if RIFF, Motorola format if FORM.
unsigned char Type[4]; // {'W', 'A', 'V', 'E'} or {'A', 'I', 'F', 'F'}
} FILE_head;
// An IFF chunk header looks like this
typedef struct _CHUNK_head
{
unsigned char ID[4]; // 4 ascii chars that is the chunk ID
unsigned int Length; // Length of subsequent data within this chunk. This is in Intel reverse byte
// order if RIFF, Motorola format if FORM. Note: this doesn't include any
// extra byte needed to pad the chunk out to an even size.
} CHUNK_head;
// WAVE fmt chunk
typedef struct _FORMAT {
short wFormatTag;
unsigned short wChannels;
unsigned int dwSamplesPerSec;
unsigned int dwAvgBytesPerSec;
unsigned short wBlockAlign;
unsigned short wBitsPerSample;
// Note: there may be additional fields here, depending upon wFormatTag
} FORMAT;
#pragma pack()
// Size of the audio card hardware buffer. Here we want it
// set to 1024 16-bit sample points. This is relatively
// small in order to minimize latency. If you have trouble
// with underruns, you may need to increase this, and PERIODSIZE
// (trading off lower latency for more stability)
#define BUFFERSIZE (2*1024)
// How many sample points the ALSA card plays before it calls
// our callback to fill some more of the audio card's hardware
// buffer. Here we want ALSA to call our callback after every
// 64 sample points have been played
#define PERIODSIZE (2*64)
// Handle to ALSA (audio card's) playback port
snd_pcm_t *PlaybackHandle;
// Handle to our callback thread
snd_async_handler_t *CallbackHandle;
// Points to loaded WAVE file's data
unsigned char *WavePtr;
// Size (in frames) of loaded WAVE file's data
snd_pcm_uframes_t WaveSize;
// Sample rate
unsigned short WaveRate;
// Bit resolution
unsigned char WaveBits;
// Number of channels in the wave file
unsigned char WaveChannels;
// The name of the ALSA port we output to. In this case, we're
// directly writing to hardware card 0,0 (ie, first set of audio
// outputs on the first audio card)
static const char SoundCardPortName[] = "plughw:0,0";
// For WAVE file loading
static const unsigned char Riff[4] = { 'R', 'I', 'F', 'F' };
static const unsigned char Wave[4] = { 'W', 'A', 'V', 'E' };
static const unsigned char Fmt[4] = { 'f', 'm', 't', ' ' };
static const unsigned char Data[4] = { 'd', 'a', 't', 'a' };
/********************** compareID() *********************
* Compares the passed ID str (ie, a ptr to 4 Ascii
* bytes) with the ID at the passed ptr. Returns TRUE if
* a match, FALSE if not.
*/
static unsigned char compareID(const unsigned char * id, unsigned char * ptr)
{
register unsigned char i = 4;
while (i--)
{
if ( *(id)++ != *(ptr)++ ) return(0);
}
return(1);
}
/********************** waveLoad() *********************
* Loads a WAVE file.
*
* fn = Filename to load.
*
* RETURNS: 0 if success, non-zero if not.
*
* NOTE: Sets the global "WavePtr" to an allocated buffer
* containing the wave data, and "WaveSize" to the size
* in sample points.
*/
static unsigned char waveLoad(const char *fn)
{
const char *message;
FILE_head head;
register int inHandle;
if ((inHandle = open(fn, O_RDONLY)) == -1)
message = "didn't open";
// Read in IFF File header
else
{
if (read(inHandle, &head, sizeof(FILE_head)) == sizeof(FILE_head))
{
// Is it a RIFF and WAVE?
if (!compareID(&Riff[0], &head.ID[0]) || !compareID(&Wave[0], &head.Type[0]))
{
message = "is not a WAVE file";
goto bad;
}
// Read in next chunk header
while (read(inHandle, &head, sizeof(CHUNK_head)) == sizeof(CHUNK_head))
{
// ============================ Is it a fmt chunk? ===============================
if (compareID(&Fmt[0], &head.ID[0]))
{
FORMAT format;
// Read in the remainder of chunk
if (read(inHandle, &format.wFormatTag, sizeof(FORMAT)) != sizeof(FORMAT)) break;
// Can't handle compressed WAVE files
if (format.wFormatTag != 1)
{
message = "compressed WAVE not supported";
goto bad;
}
WaveBits = (unsigned char)format.wBitsPerSample;
WaveRate = (unsigned short)format.dwSamplesPerSec;
WaveChannels = format.wChannels;
}
// ============================ Is it a data chunk? ===============================
else if (compareID(&Data[0], &head.ID[0]))
{
// Size of wave data is head.Length. Allocate a buffer and read in the wave data
if (!(WavePtr = (unsigned char *)malloc(head.Length)))
{
message = "won't fit in RAM";
goto bad;
}
if (read(inHandle, WavePtr, head.Length) != head.Length)
{
free(WavePtr);
break;
}
// Store size (in frames)
WaveSize = (head.Length * 8) / ((unsigned int)WaveBits * (unsigned int)WaveChannels);
close(inHandle);
return(0);
}
// ============================ Skip this chunk ===============================
else
{
if (head.Length & 1) ++head.Length; // If odd, round it up to account for pad byte
lseek(inHandle, head.Length, SEEK_CUR);
}
}
}
message = "is a bad WAVE file";
bad: close(inHandle);
}
printf("%s %s\n", fn, message);
return(1);
}
/********************** play_audio() **********************
* Plays the loaded waveform.
*
* NOTE: ALSA sound card's handle must be in the global
* "PlaybackHandle". A pointer to the wave data must be in
* the global "WavePtr", and its size of "WaveSize".
*/
static void play_audio(void)
{
register snd_pcm_uframes_t count, frames;
// Output the wave data
count = 0;
do
{
frames = snd_pcm_writei(PlaybackHandle, WavePtr + count, WaveSize - count);
// If an error, try to recover from it
if (frames < 0)
frames = snd_pcm_recover(PlaybackHandle, frames, 0);
if (frames < 0)
{
printf("Error playing wave: %s\n", snd_strerror(frames));
break;
}
// Update our pointer
count += frames;
} while (count < WaveSize);
// Wait for playback to completely finish
if (count == WaveSize)
snd_pcm_drain(PlaybackHandle);
}
/*********************** free_wave_data() *********************
* Frees any wave data we loaded.
*
* NOTE: A pointer to the wave data be in the global
* "WavePtr".
*/
static void free_wave_data(void)
{
if (WavePtr) free(WavePtr);
WavePtr = 0;
}
void list_devices(){
char **hints;
/* Enumerate sound devices */
int err = snd_device_name_hint(-1, "pcm", (void***)&hints);
if (err != 0)
return;//Error! Just return
char** n = hints;
while (*n != NULL) {
char *name = snd_device_name_get_hint(*n, "NAME");
if (name != NULL && 0 != strcmp("null", name)) {
//Copy name to another buffer and then free it
fprintf(stderr, "Name: %s \n", name);
free(name);
}
n++;
}//End of while
//Free hint buffer too
snd_device_name_free_hint((void**)hints);
}
int main(int argc, char **argv)
{
// No wave data loaded yet
WavePtr = 0;
list_devices();
if (argc < 2)
printf("You must supply the name of a 16-bit mono WAVE file to play\n");
// Load the wave file
else if (!waveLoad(argv[1]))
{
register int err;
// Open audio card we wish to use for playback
if ((err = snd_pcm_open(&PlaybackHandle, &SoundCardPortName[0], SND_PCM_STREAM_PLAYBACK, 0)) < 0)
printf("Can't open audio %s: %s\n", &SoundCardPortName[0], snd_strerror(err));
else
{
switch (WaveBits)
{
case 8:
err = SND_PCM_FORMAT_U8;
break;
case 16:
err = SND_PCM_FORMAT_S16;
break;
case 24:
err = SND_PCM_FORMAT_S24;
break;
case 32:
err = SND_PCM_FORMAT_S32;
break;
}
// Set the audio card's hardware parameters (sample rate, bit resolution, etc)
if ((err = snd_pcm_set_params(PlaybackHandle, err, SND_PCM_ACCESS_RW_INTERLEAVED, WaveChannels, WaveRate, 1, 500000)) < 0)
printf("Can't set sound parameters: %s\n", snd_strerror(err));
// Play the waveform
else
play_audio();
// Close sound card
snd_pcm_close(PlaybackHandle);
}
}
// Free the WAVE data
free_wave_data();
return(0);
}
http://freewavesamples.com/files/Roland-JD-990-Windchimes.wav
Rmm running i get
root@yi-cam-01:/tmp/sd/my# ./audioplay /tmp/sd/record/Roland-JD-990-Windchimes.wav
Name: default:CARD=sun8iw19codec
Name: sysdefault:CARD=sun8iw19codec
Name: default:CARD=snddaudio0
Name: sysdefault:CARD=snddaudio0
Can't open audio plughw:0,0: Resource busy
If you stop rmm you can play audio.
For example with aplay or speaker-test from alsa-utils.
But if you stop rmm you can't receive h264 stream.
its possible to create a virtual alsa device ?!
In my frist attempt to decompile a binary, i found the alsa "CARD NUMBER", and changed to 2 ..
but I don’t know how to create a virtual device ... I don’t even know if it’s possible to do this on the current platform
changin rmm, to use: hw:0,0 , get this erro:
E0326 19:03:24.987167 443 alsa_interface.c:447] <alsaOpenPcm> audio open error: Resource busy
E0326 19:03:24.989010 443 audio_hw.c:1761] <audioHw_AO_Enable> audioHw_AO_Enable,l:1761,open_pcm failed
rmm_init_adec_ao(3731) MPI_AO_Enable ao_dev:0 fail! ret:0xffffffff
but my example only works with hw: 0.0 ..
PS: I'm not running at the same time ...
Use this new hacked library:
libasound.so.2.0.0.gz
It will create a new fifo /tmp/audio_in_fifo
If you send to this fifo a raw pcm the cam should play it.
The file must be a raw pcm (you can export it with audacity) with the following parameters:
- 16 KHz
- 16 bit
- mono
- signed
Copy to the fifo with a normal cat.
Ah...
Change system.sh to run rmm with this library:
...
if [[ $(get_config DISABLE_CLOUD) == "no" ]] ; then
(
cd /home/app
LD_LIBRARY_PATH="/home/yi-hack/lib:/lib:/usr/lib:/home/lib:/home/qigan/lib:/home/app/locallib" ./rmm &
sleep 4
dd if=/tmp/audio_fifo of=/dev/null bs=1 count=8192
# dd if=/dev/zero of=/tmp/audio_in_fifo bs=1 count=1024
./mp4record &
./cloud &
./p2p_tnp &
./oss &
if [ -f ./oss_fast ]; then
./oss_fast &
fi
if [ -f ./oss_lapse ]; then
./oss_lapse &
fi
./watch_process &
)
else
(
cd /home/app
LD_LIBRARY_PATH="/home/yi-hack/lib:/lib:/usr/lib:/home/lib:/home/qigan/lib:/home/app/locallib" ./rmm &
sleep 4
dd if=/tmp/audio_fifo of=/dev/null bs=1 count=8192
# dd if=/dev/zero of=/tmp/audio_in_fifo bs=1 count=1024
# Trick to start circular buffer filling
start_buffer
if [[ $(get_config REC_WITHOUT_CLOUD) == "yes" ]] ; then
./mp4record &
fi
mkdir /tmp/etc
cp -R /etc/* /tmp/etc
mount --bind /tmp/etc /etc
echo "127.0.0.1 api.eu.xiaoyi.com" >> /etc/hosts
)
fi
...
I tested it and the sound is very distorted, it seems that it is kind of accelerated
Im using this file
StarWars3.zip
I started a big wav music using:
root@yi-cam-01:/tmp/sd/my# ./aplay OficinaG3.wav
mode 0 - bits 0 - channels 0
mode 0 - bits 0 - channels 0
mode 0 - bits 0 - channels 0
Playing WAVE 'OficinaG3.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo
underrun!!! (at least 0.079 ms long)
underrun!!! (at least 0.086 ms long)
then:
cat StarWars3.wav > /tmp/audio_in_fifo
sorry, running RMM works... !!! amazing !!
There are only a few minor distortions when I went to test on notification sounds, which would be nice for confirming the reading of the qrcode
I'm sending you to test ..
notifications-sounds.zip
.
Sometimes i get: Segmentation fault (core dumped), running aplay
PS: RMM application (STOPPED) to test sound quality without it
root@yi-cam-01:/tmp/sd/my# ./aplay job-done-501-16khz.wav
mode 0 - bits 0 - channels 0
mode 0 - bits 0 - channels 0
mode 0 - bits 0 - channels 0
Playing WAVE 'job-done-501-16khz.wav' : Signed 16 bit Little Endian, Rate 16000 Hz, Mono
root@yi-cam-01:/tmp/sd/my# ./aplay job-done-501-16khz.wav
mode 0 - bits 0 - channels 0
mode 0 - bits 0 - channels 0
mode 0 - bits 0 - channels 0
Playing WAVE 'job-done-501-16khz.wav' : Signed 16 bit Little Endian, Rate 16000 Hz, Mono
Segmentation fault (core dumped)