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echo-cancel: Allow enabling the extended filter in webrtc AEC

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This creates a longer filter that is more complex and less sensitive to
incorrect delay reporting from the hardware. There is also a
delay-agnostic mode that can eventually be enabled if required.

In some very quick testing, not enabling this seems to provide better
results during double-talk.
This commit is contained in:
Arun Raghavan 2016-02-17 19:46:54 +05:30 • committed by Tanu Kaskinen
parent f8beaae238
commit a14db55c2b
1 changed files with 14 additions and 2 deletions
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16
src/modules/echo-cancel/webrtc.cc
View file

@ -46,6 +46,7 @@ PA_C_DECL_END
#define DEFAULT_ROUTING_MODE "speakerphone" #define DEFAULT_ROUTING_MODE "speakerphone"
#define DEFAULT_COMFORT_NOISE true #define DEFAULT_COMFORT_NOISE true
#define DEFAULT_DRIFT_COMPENSATION false #define DEFAULT_DRIFT_COMPENSATION false
#define DEFAULT_EXTENDED_FILTER false
static const char* const valid_modargs[] = { static const char* const valid_modargs[] = {
"high_pass_filter", "high_pass_filter",
@ -56,6 +57,7 @@ static const char* const valid_modargs[] = {
"routing_mode", "routing_mode",
"comfort_noise", "comfort_noise",
"drift_compensation", "drift_compensation",
"extended_filter",
NULL NULL
}; };
@ -81,7 +83,8 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
uint32_t *nframes, const char *args) { uint32_t *nframes, const char *args) {
webrtc::AudioProcessing *apm = NULL; webrtc::AudioProcessing *apm = NULL;
webrtc::ProcessingConfig pconfig; webrtc::ProcessingConfig pconfig;
bool hpf, ns, agc, dgc, mobile, cn; webrtc::Config config;
bool hpf, ns, agc, dgc, mobile, cn, ext_filter;
int rm = -1; int rm = -1;
pa_modargs *ma; pa_modargs *ma;
@ -154,7 +157,16 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
} }
} }
apm = webrtc::AudioProcessing::Create(); ext_filter = DEFAULT_EXTENDED_FILTER;
if (pa_modargs_get_value_boolean(ma, "extended_filter", &ext_filter) < 0) {
pa_log("Failed to parse extended_filter value");
goto fail;
}
if (ext_filter)
config.Set<webrtc::ExtendedFilter>(new webrtc::ExtendedFilter(true));
apm = webrtc::AudioProcessing::Create(config);
out_ss->format = PA_SAMPLE_S16NE; out_ss->format = PA_SAMPLE_S16NE;
*play_ss = *out_ss; *play_ss = *out_ss;