diff options
author | Arun Raghavan <git@arunraghavan.net> | 2016-02-17 19:47:02 +0530 |
---|---|---|
committer | Arun Raghavan <git@arunraghavan.net> | 2016-02-25 09:09:12 +0530 |
commit | 19fb2481ea389d0f3e5f45e4bf5b94a7ac1fd399 (patch) | |
tree | daf69f982373f41af30cd4f87aa6494ca4d14758 | |
parent | a84d65d74805367f35048fa1734dfd1436246042 (diff) | |
download | pulseaudio-19fb2481ea389d0f3e5f45e4bf5b94a7ac1fd399.tar.gz |
echo-cancel: Start capture at a sane volume if we're doing webrtc AGC
This is required to make sure the capture output has sufficient energy
for the AGC to do its job.
-rw-r--r-- | src/modules/echo-cancel/echo-cancel.h | 1 | ||||
-rw-r--r-- | src/modules/echo-cancel/webrtc.cc | 14 |
2 files changed, 14 insertions, 1 deletions
diff --git a/src/modules/echo-cancel/echo-cancel.h b/src/modules/echo-cancel/echo-cancel.h index 2a0dee27f..cc554d53f 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; void *trace_callback; bool agc; + 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 d818fc0e8..be13d7519 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", @@ -299,6 +300,7 @@ 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; @@ -363,7 +365,17 @@ 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; + } else { + new_volume = apm->gain_control()->stream_analog_level(); + } + if (old_volume != new_volume) { pa_cvolume_set(&v, ss->channels, webrtc_volume_to_pa(new_volume)); pa_echo_canceller_set_capture_volume(ec, &v); |