Created
September 6, 2017 22:53
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#include <stdio.h> | |
#include <unistd.h> | |
#include <fcntl.h> | |
#include <sys/ioctl.h> | |
#include <linux/soundcard.h> | |
#include <webrtc_audio_processing/module_common_types.h> | |
#include <webrtc_audio_processing/audio_processing.h> | |
int main() | |
{ | |
int errno; | |
int format = AFMT_S16_NE; /* native endian */ | |
int speed = 32000; | |
int blocksize = speed / 100; | |
int channels = 1; | |
int fd = open("/dev/dsp", O_RDWR); | |
if (!fd) { | |
printf("Could not open sound device\n"); | |
return -1; | |
} | |
//if (0 != ioctl(fd, FIONBIO, &nonblock)) | |
// return errno; | |
if (0 != ioctl(fd, SNDCTL_DSP_SETFMT, &format)) | |
return errno; | |
if (0 != ioctl(fd, SNDCTL_DSP_CHANNELS, &channels)) | |
return errno; | |
if (2 == channels) { | |
int stereo = 1; | |
if (0 != ioctl(fd, SNDCTL_DSP_STEREO, &stereo)) | |
return errno; | |
} | |
if (0 != ioctl(fd, SNDCTL_DSP_SPEED, &speed)) | |
return errno; | |
printf("oss: init: %d Hz %d ch, blocksize=%d\n", | |
speed, channels, blocksize); | |
blocksize = speed / 100; | |
char buf[320]; | |
// BEGIN WEBRTC TESTBENCH | |
webrtc::AudioProcessing* apm = webrtc::AudioProcessing::Create(0); | |
apm->set_sample_rate_hz(speed); | |
apm->set_num_channels(1, 1); | |
apm->set_num_reverse_channels(1); | |
apm->echo_cancellation()->Enable(true); | |
apm->echo_cancellation()->set_device_sample_rate_hz(32000); | |
apm->set_stream_delay_ms(0); | |
while (true) { | |
webrtc::AudioFrame frame; | |
frame._audioChannel = 1;// ss->channels | |
frame._frequencyInHz = 32000; | |
frame._payloadDataLengthInSamples = blocksize; | |
read(fd, frame._payloadData, blocksize); | |
errno = apm->AnalyzeReverseStream(&frame); | |
if (errno) { | |
printf("AnalyzeReverseStream() = %d\n", errno); | |
} | |
apm->set_stream_delay_ms(0); | |
errno = apm->ProcessStream(&frame); | |
if (errno) { | |
printf("ProcessStream() = %d\n", errno); | |
} | |
write(fd, frame._payloadData, blocksize); | |
} | |
webrtc::AudioProcessing::Destroy(apm); | |
} |
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