mirrors/pulseaudio
1
0
Fork 0
mirror of https://gitlab.freedesktop.org/pulseaudio/pulseaudio.git synced 2025-11-02 09:01:46 -05:00
Code Issues Projects Releases Packages Wiki Activity Actions

echo-cancel: Deal with volume limit breakage in webrtc AGC

Browse source 
The AGC code no longer seems to honour the analog volume limits we set,
and internally uses 0-255 as the volume range. So we switch to use that
(keeping the old API usage as is in case this gets fixed upstream).
This commit is contained in:
Arun Raghavan 2016-02-17 19:47:01 +05:30
parent 426c98acbb
commit a84d65d748
1 changed files with 21 additions and 4 deletions
Download patch file Download diff file Expand all files Collapse all files

25
src/modules/echo-cancel/webrtc.cc
View file

@ -51,6 +51,8 @@ PA_C_DECL_END
#define DEFAULT_INTELLIGIBILITY_ENHANCER false #define DEFAULT_INTELLIGIBILITY_ENHANCER false
#define DEFAULT_TRACE false #define DEFAULT_TRACE false
#define WEBRTC_AGC_MAX_VOLUME 255
static const char* const valid_modargs[] = { static const char* const valid_modargs[] = {
"high_pass_filter", "high_pass_filter",
"noise_suppression", "noise_suppression",
@ -95,6 +97,16 @@ class PaWebrtcTraceCallback : public webrtc::TraceCallback {
} }
}; };
static int webrtc_volume_from_pa(pa_volume_t v)
{
return (v * WEBRTC_AGC_MAX_VOLUME) / PA_VOLUME_NORM;
}
static pa_volume_t webrtc_volume_to_pa(int v)
{
return (v * PA_VOLUME_NORM) / WEBRTC_AGC_MAX_VOLUME;
}
static void pa_webrtc_ec_fixate_spec(pa_sample_spec *rec_ss, pa_channel_map *rec_map, static void pa_webrtc_ec_fixate_spec(pa_sample_spec *rec_ss, pa_channel_map *rec_map,
pa_sample_spec *play_ss, pa_channel_map *play_map, pa_sample_spec *play_ss, pa_channel_map *play_map,
pa_sample_spec *out_ss, pa_channel_map *out_map) pa_sample_spec *out_ss, pa_channel_map *out_map)
@ -271,7 +283,7 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
ec->params.priv.webrtc.agc = false; ec->params.priv.webrtc.agc = false;
} else { } else {
apm->gain_control()->set_mode(webrtc::GainControl::kAdaptiveAnalog); apm->gain_control()->set_mode(webrtc::GainControl::kAdaptiveAnalog);
if (apm->gain_control()->set_analog_level_limits(0, PA_VOLUME_NORM-1) != apm->kNoError) { if (apm->gain_control()->set_analog_level_limits(0, WEBRTC_AGC_MAX_VOLUME) != apm->kNoError) {
pa_log("Failed to initialise AGC"); pa_log("Failed to initialise AGC");
goto fail; goto fail;
} }
@ -330,6 +342,7 @@ void pa_webrtc_ec_record(pa_echo_canceller *ec, const uint8_t *rec, uint8_t *out
webrtc::AudioFrame out_frame; webrtc::AudioFrame out_frame;
const pa_sample_spec *ss = &ec->params.priv.webrtc.sample_spec; const pa_sample_spec *ss = &ec->params.priv.webrtc.sample_spec;
pa_cvolume v; pa_cvolume v;
int old_volume, new_volume;
out_frame.num_channels_ = ss->channels; out_frame.num_channels_ = ss->channels;
out_frame.sample_rate_hz_ = ss->rate; out_frame.sample_rate_hz_ = ss->rate;
@ -342,15 +355,19 @@ void pa_webrtc_ec_record(pa_echo_canceller *ec, const uint8_t *rec, uint8_t *out
if (ec->params.priv.webrtc.agc) { if (ec->params.priv.webrtc.agc) {
pa_cvolume_init(&v); pa_cvolume_init(&v);
pa_echo_canceller_get_capture_volume(ec, &v); pa_echo_canceller_get_capture_volume(ec, &v);
apm->gain_control()->set_stream_analog_level(pa_cvolume_avg(&v)); old_volume = webrtc_volume_from_pa(pa_cvolume_avg(&v));
apm->gain_control()->set_stream_analog_level(old_volume);
} }
apm->set_stream_delay_ms(0); apm->set_stream_delay_ms(0);
apm->ProcessStream(&out_frame); apm->ProcessStream(&out_frame);
if (ec->params.priv.webrtc.agc) { if (ec->params.priv.webrtc.agc) {
pa_cvolume_set(&v, ss->channels, apm->gain_control()->stream_analog_level()); new_volume = apm->gain_control()->stream_analog_level();
pa_echo_canceller_set_capture_volume(ec, &v); 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);
}
} }
memcpy(out, out_frame.data_, ec->params.priv.webrtc.blocksize); memcpy(out, out_frame.data_, ec->params.priv.webrtc.blocksize);