[pulseaudio-discuss] [PATCH v3 11/24] echo-cancel: Start capture at a sane value if we're doing webrtc AGC
Tanu Kaskinen
tanuk at iki.fi
Mon Feb 1 03:18:02 PST 2016
On Mon, 2016-01-18 at 13:06 +0530, arun at accosted.net wrote:
> From: Arun Raghavan <git at arunraghavan.net>
>
> This is required to make sure the capture output has sufficient energy
> for the AGC to do its job.
I think the word "value" in the first line of the commit message should
be replaced with "volume". After reading just the commit message, I
didn't know what the patch was about.
> ---
> src/modules/echo-cancel/echo-cancel.h | 1 +
> src/modules/echo-cancel/webrtc.cc | 14 +++++++++++++-
> 2 files changed, 14 insertions(+), 1 deletion(-)
>
> diff --git a/src/modules/echo-cancel/echo-cancel.h b/src/modules/echo-cancel/echo-cancel.h
> index 142f8ac..b570095 100644
> --- a/src/modules/echo-cancel/echo-cancel.h
> +++ b/src/modules/echo-cancel/echo-cancel.h
> @@ -68,6 +68,7 @@ struct pa_echo_canceller_params {
> pa_sample_spec sample_spec;
> bool agc;
> bool trace;
> + bool first;
> } webrtc;
> #endif
> /* each canceller-specific structure goes here */
> diff --git a/src/modules/echo-cancel/webrtc.cc b/src/modules/echo-cancel/webrtc.cc
> index c2585a8..6431651 100644
> --- a/src/modules/echo-cancel/webrtc.cc
> +++ b/src/modules/echo-cancel/webrtc.cc
> @@ -52,6 +52,7 @@ PA_C_DECL_END
> #define DEFAULT_TRACE false
>
> #define WEBRTC_AGC_MAX_VOLUME 255
> +#define WEBRTC_AGC_START_VOLUME 85
>
> static const char* const valid_modargs[] = {
> "high_pass_filter",
> @@ -297,6 +298,7 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
> ec->params.priv.webrtc.sample_spec = *out_ss;
> ec->params.priv.webrtc.blocksize = (uint64_t)pa_bytes_per_second(out_ss) * BLOCK_SIZE_US / PA_USEC_PER_SEC;
> *nframes = ec->params.priv.webrtc.blocksize / pa_frame_size(out_ss);
> + ec->params.priv.webrtc.first = true;
>
> pa_modargs_free(ma);
> return true;
> @@ -354,7 +356,17 @@ void pa_webrtc_ec_record(pa_echo_canceller *ec, const uint8_t *rec, uint8_t *out
> apm->ProcessStream(&out_frame);
>
> if (ec->params.priv.webrtc.agc) {
> - new_volume = apm->gain_control()->stream_analog_level();
> + if (PA_UNLIKELY(ec->params.priv.webrtc.first)) {
> + /* We start at a sane default volume (taken from the Chromium
> + * condition on the experimental AGC in audio_processing.h). This is
> + * needed to make sure that there's enough energy in the capture
> + * signal for the AGC to work */
> + ec->params.priv.webrtc.first = false;
> + new_volume = WEBRTC_AGC_START_VOLUME;
This is done only on initial loading of the module. Does AGC work ok
after resuming a suspended source?
--
Tanu
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