[pulseaudio-discuss] [PATCH v3 03/24] echo-cancel: Add support for the webrtc intelligibility enhancer
Tanu Kaskinen
tanuk at iki.fi
Sun Jan 24 08:30:32 PST 2016
On Mon, 2016-01-18 at 13:06 +0530, arun at accosted.net wrote:
> From: Arun Raghavan <git at arunraghavan.net>
>
> Just exposing this, disabled by default. It's not used by Chromium at
> the moment.
> ---
> src/modules/echo-cancel/webrtc.cc | 14 ++++++++++++--
> 1 file changed, 12 insertions(+), 2 deletions(-)
>
> diff --git a/src/modules/echo-cancel/webrtc.cc b/src/modules/echo-cancel/webrtc.cc
> index f4f1395..bbfa43f 100644
> --- a/src/modules/echo-cancel/webrtc.cc
> +++ b/src/modules/echo-cancel/webrtc.cc
> @@ -47,6 +47,7 @@ PA_C_DECL_END
> #define DEFAULT_COMFORT_NOISE true
> #define DEFAULT_DRIFT_COMPENSATION false
> #define DEFAULT_EXTENDED_FILTER false
> +#define DEFAULT_INTELLIGIBILITY_ENHANCER false
>
> static const char* const valid_modargs[] = {
> "high_pass_filter",
> @@ -58,6 +59,7 @@ static const char* const valid_modargs[] = {
> "comfort_noise",
> "drift_compensation",
> "extended_filter",
> + "intelligibility_enhancer",
> NULL
> };
>
> @@ -84,7 +86,7 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
> webrtc::AudioProcessing *apm = NULL;
> webrtc::ProcessingConfig pconfig;
> webrtc::Config config;
> - bool hpf, ns, agc, dgc, mobile, cn, ext_filter;
> + bool hpf, ns, agc, dgc, mobile, cn, ext_filter, intelligibility;
> int rm = -1;
> pa_modargs *ma;
>
> @@ -163,8 +165,16 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
> goto fail;
> }
>
> + intelligibility = DEFAULT_INTELLIGIBILITY_ENHANCER;
> + if (pa_modargs_get_value_boolean(ma, "intelligibility_enhancer", &intelligibility) < 0) {
> + pa_log("Failed to parse intelligibility_enhancer value");
> + goto fail;
> + }
> +
> if (ext_filter)
> config.Set(new webrtc::ExtendedFilter(true));
> + if (intelligibility)
> + config.Set<webrtc::Intelligibility>(new webrtc::Intelligibility(true));
The same comment as in the previous patch: could the if statement be
eliminated?
>
> apm = webrtc::AudioProcessing::Create(config);
>
> @@ -253,7 +263,7 @@ void pa_webrtc_ec_play(pa_echo_canceller *ec, const uint8_t *play) {
> pa_assert(play_frame.samples_per_channel_ <= webrtc::AudioFrame::kMaxDataSizeSamples);
> memcpy(play_frame.data_, play, ec->params.priv.webrtc.blocksize);
>
> - apm->AnalyzeReverseStream(&play_frame);
> + apm->ProcessReverseStream(&play_frame);
This looks like a potentially unrelated change. Why is this change
done?
--
Tanu
More information about the pulseaudio-discuss
mailing list