All streams have rate 8000 (rate of umts sound card), 1 channel and 16bit per
sample (SND_PCM_FORMAT_S16_LE).
-We set buffer_size of sound card to 1000.
+We set buffer_size of sound card to 1024.
The sound card buffer consists of 4 periods.
-Period has 250 frames
+Period has 256 frames
Frame has just one sample (one channel) and both sample and frame are 2 bytes
-We always play/record one period (250 samples, 500 bytes). At rate 8000Hz if
-we had 8000 samples it would be 1s, with 250 samples it makes 31.25ms long
+We always play/record one period (256 samples, 512 bytes). At rate 8000Hz if
+we had 8000 samples it would be 1s, with 256 samples it makes 31.25ms long
period - this is our latency.
*/
/* Use the newer ALSA API */
#define ALSA_PCM_NEW_HW_PARAMS_API
+/* Define this to enable speex acoustic echo cancellation */
+// #define USE_SPEEX_AEC
+
+/* Define this to use walkie talkie echo reduction */
+// #define USE_WALKIE_TALKIE_AEC
+
#include <unistd.h>
#include <string.h>
#include <time.h>
#include <alsa/asoundlib.h>
+#ifdef USE_SPEEX_AEC
+#include <speex/speex_echo.h>
+#endif
+
#define ERR_PCM_OPEN_FAILED -1
#define ERR_HW_PARAMS_ANY -2
#define ERR_HW_PARAMS_SET_ACCESS -3
snd_pcm_stream_t stream; // in: SND_PCM_STREAM_PLAYBACK / SND_PCM_STREAM_CAPTURE
snd_pcm_uframes_t start_threshold; // in: start treshold or 0 to keep default
snd_pcm_uframes_t stop_threshold; // in: stop treshold or 0 to keep default
- snd_pcm_uframes_t buffer_size; // in/out: hw buffer size, e.g. 1000 frames
- snd_pcm_uframes_t period_size; // in/out: period size, e.g. 250 frames
+ snd_pcm_uframes_t buffer_size; // in/out: hw buffer size, e.g. 1024 frames
+ snd_pcm_uframes_t period_size; // in/out: period size, e.g. 256 frames
snd_pcm_t *handle; // out: pcm handle
snd_pcm_hw_params_t *hwparams; // out:
snd_pcm_sw_params_t *swparams; // out:
- int period_buffer_size; // out: size 2000 (250 frames=250 samples, one sample=2bytes)
+ int period_buffer_size; // out: size 2000 (256 frames=256 samples, one sample=2bytes)
char *period_buffer; // out: allocated buffer for playing/recording
};
ERR_HW_PARAMS_SET_RATE);
}
- /* Period size in frames (e.g. 250) */
+ /* Period size in frames (e.g. 256) */
rc = snd_pcm_hw_params_set_period_size(s->handle, s->hwparams,
s->period_size, 0);
if (rc < 0) {
ERR_HW_PARAMS_SET_PERIOD_SIZE);
}
- /* Buffer size in frames (e.g. 1000) */
+ /* Buffer size in frames (e.g. 1024) */
rc = snd_pcm_hw_params_set_buffer_size(s->handle, s->hwparams,
s->buffer_size);
if (rc < 0) {
fflush(logfile);*/
}
+#ifdef USE_WALKIE_TALKIE_AEC
+
static void vol_up(char *buf, int buf_size)
{
int i;
s16 *val = (short *)(buf);
- for(i = 0; i < buf_size; i += 2)
- {
+ for (i = 0; i < buf_size; i += 2) {
*val = *val * 2;
}
}
{
int i;
s16 *val = (short *)(buf);
- for(i = 0; i < buf_size; i += 2)
- {
+ for (i = 0; i < buf_size; i += 2) {
*val = *val / 2;
}
}
We use integer for computing volumes, this should be fine if the buffer size
is not too big (65535 is max).
*/
-//int row = 0;
void reduce_echo(char *buf_a, char *buf_b, int buf_size)
{
int i;
unsigned int sum_a = 0;
unsigned int sum_b = 0;
+ //static int row = 0;
s16 *a;
s16 *b;
a = (short *)(buf_a);
b = (short *)(buf_b);
- for(i = 0; i < buf_size; i+=2)
- {
+ for (i = 0; i < buf_size; i += 2) {
sum_a += abs(*a);
sum_b += abs(*b);
a++;
b++;
}
- if(sum_a > 10000) {
+ if (sum_a > 10000) {
//fprintf(stderr, "listening %d ", sum_a);
-
+
vol_up(buf_a, buf_size);
vol_down(buf_b, buf_size);
} else {
//fprintf(stderr, "talking %d ", sum_a);
-
+
vol_up(buf_b, buf_size);
vol_down(buf_a, buf_size);
}
//printf("%d, %d, %d\n", row++, sum_a, sum_b);
}
+#endif
int main()
{
int rc;
int started = 0;
char *logfilename;
+#ifdef USE_SPEEX_AEC
+ SpeexEchoState *echo_state;
+#endif
logfile = stderr;
logfilename = getenv("GSM_VOICE_ROUTING_LOGFILE");
if (rc != -20) {
fprintf(logfile, "nice() failed\n");
}
-
+#ifdef USE_SPEEX_AEC
+ /* 256=frame (period size), 3000 is filter length (recommended is 1/3 of
+ reverbation time - for 1s it's 8000 / 3 */
+ echo_state = speex_echo_state_init(256, 8000);
+#endif
+
struct route_stream p0 = {
.id = "p0",
.pcm_name = "default",
.stream = SND_PCM_STREAM_PLAYBACK,
- .start_threshold = 1000,
- .stop_threshold = 1000,
- .buffer_size = 1000,
- .period_size = 250,
+ .start_threshold = 1024,
+ .stop_threshold = 1024,
+ .buffer_size = 1024,
+ .period_size = 256,
.handle = 0,
.period_buffer = 0
};
.stream = SND_PCM_STREAM_CAPTURE,
.start_threshold = 0,
.stop_threshold = 0,
- .buffer_size = 1000,
- .period_size = 250,
+ .buffer_size = 1024,
+ .period_size = 256,
.handle = 0,
.period_buffer = 0
};
.id = "p1",
.pcm_name = "hw:1,0",
.stream = SND_PCM_STREAM_PLAYBACK,
- .start_threshold = 1000,
- .stop_threshold = 1000,
- .buffer_size = 1000,
- .period_size = 250,
+ .start_threshold = 1024,
+ .stop_threshold = 1024,
+ .buffer_size = 1024,
+ .period_size = 256,
.handle = 0,
.period_buffer = 0
};
.stream = SND_PCM_STREAM_CAPTURE,
.start_threshold = 0,
.stop_threshold = 0,
- .buffer_size = 1000,
- .period_size = 250,
+ .buffer_size = 1024,
+ .period_size = 256,
.handle = 0,
.period_buffer = 0
};
started = 1;
}
-
memmove(p0.period_buffer, r1.period_buffer, r1.period_buffer_size);
memmove(p1.period_buffer, r0.period_buffer, r0.period_buffer_size);
- // reduce_echo(p0.period_buffer, p1.period_buffer, p0.period_size * 2);
-
+#ifdef USE_WALKIE_TALKIE_AEC
+ reduce_echo(p0.period_buffer, p1.period_buffer, p0.period_size * 2);
+#endif
+
+#ifdef USE_SPEEX_AEC
+ speex_echo_playback(echo_state, (spx_int16_t *) p0.period_buffer);
+ speex_echo_capture(echo_state, (spx_int16_t *) r0.period_buffer,
+ (spx_int16_t *) p1.period_buffer);
+#endif
+
route_stream_write(&p0);
route_stream_write(&p1);
}
+#ifdef USE_SPEEX_AEC
+ speex_echo_state_destroy(echo_state);
+#endif
+
close_route_stream(&p0);
close_route_stream(&p1);
close_route_stream(&r0);
git / gta04-gsm-voice-routing.git / commitdiff