echo-cancel: add webrtc AEC3 support

Drop a number of now unsupported features, and add new parameters for
pre-/post-amplification.

Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/395>

1 files changed, 113 insertions, 320 deletions

diff --git a/src/modules/echo-cancel/webrtc.cc b/src/modules/echo-cancel/webrtc.cc
index 56daab0fd..ed4bb65a5 100644
--- a/src/modules/echo-cancel/webrtc.cc
+++ b/src/modules/echo-cancel/webrtc.cc
@@ -3,8 +3,8 @@
 
     Copyright 2011 Collabora Ltd.
               2015 Aldebaran SoftBank Group
-
-    Contributor: Arun Raghavan <mail@arunraghavan.net>
+              2020 Arun Raghavan <arun@asymptotic.io>
+              2020 Eero Nurkkala <eero.nurkkala@offcode.fi>
 
     PulseAudio is free software; you can redistribute it and/or modify
     it under the terms of the GNU Lesser General Public License as published
@@ -34,80 +34,47 @@ PA_C_DECL_BEGIN
 #include "echo-cancel.h"
 PA_C_DECL_END
 
-#include <webrtc/modules/audio_processing/include/audio_processing.h>
-#include <webrtc/modules/interface/module_common_types.h>
-#include <webrtc/system_wrappers/include/trace.h>
+#define WEBRTC_APM_DEBUG_DUMP 0
+
+#include <modules/audio_processing/include/audio_processing.h>
 
 #define BLOCK_SIZE_US 10000
 
 #define DEFAULT_HIGH_PASS_FILTER true
 #define DEFAULT_NOISE_SUPPRESSION true
+#define DEFAULT_TRANSIENT_NOISE_SUPPRESSION true
 #define DEFAULT_ANALOG_GAIN_CONTROL true
 #define DEFAULT_DIGITAL_GAIN_CONTROL false
 #define DEFAULT_MOBILE false
-#define DEFAULT_ROUTING_MODE "speakerphone"
 #define DEFAULT_COMFORT_NOISE true
 #define DEFAULT_DRIFT_COMPENSATION false
-#define DEFAULT_VAD true
-#define DEFAULT_EXTENDED_FILTER false
-#define DEFAULT_INTELLIGIBILITY_ENHANCER false
-#define DEFAULT_EXPERIMENTAL_AGC false
+#define DEFAULT_VAD false
 #define DEFAULT_AGC_START_VOLUME 85
-#define DEFAULT_BEAMFORMING false
-#define DEFAULT_TRACE false
+#define DEFAULT_POSTAMP_ENABLE false
+#define DEFAULT_POSTAMP_GAIN_DB 0
+#define DEFAULT_PREAMP_ENABLE false
+#define DEFAULT_PREAMP_GAIN_DB 0
 
 #define WEBRTC_AGC_MAX_VOLUME 255
+#define WEBRTC_POSTAMP_GAIN_MAX_DB 90
+#define WEBRTC_PREAMP_GAIN_MAX_DB 90
 
 static const char* const valid_modargs[] = {
-    "high_pass_filter",
-    "noise_suppression",
+    "agc_start_volume",
     "analog_gain_control",
     "digital_gain_control",
+    "high_pass_filter",
     "mobile",
-    "routing_mode",
-    "comfort_noise",
-    "drift_compensation",
+    "noise_suppression",
+    "post_amplifier",
+    "post_amplifier_gain",
+    "pre_amplifier",
+    "pre_amplifier_gain",
+    "transient_noise_suppression",
     "voice_detection",
-    "extended_filter",
-    "intelligibility_enhancer",
-    "experimental_agc",
-    "agc_start_volume",
-    "beamforming",
-    "mic_geometry", /* documented in parse_mic_geometry() */
-    "target_direction", /* documented in parse_mic_geometry() */
-    "trace",
     NULL
 };
 
-static int routing_mode_from_string(const char *rmode) {
-    if (pa_streq(rmode, "quiet-earpiece-or-headset"))
-        return webrtc::EchoControlMobile::kQuietEarpieceOrHeadset;
-    else if (pa_streq(rmode, "earpiece"))
-        return webrtc::EchoControlMobile::kEarpiece;
-    else if (pa_streq(rmode, "loud-earpiece"))
-        return webrtc::EchoControlMobile::kLoudEarpiece;
-    else if (pa_streq(rmode, "speakerphone"))
-        return webrtc::EchoControlMobile::kSpeakerphone;
-    else if (pa_streq(rmode, "loud-speakerphone"))
-        return webrtc::EchoControlMobile::kLoudSpeakerphone;
-    else
-        return -1;
-}
-
-class PaWebrtcTraceCallback : public webrtc::TraceCallback {
-    void Print(webrtc::TraceLevel level, const char *message, int length)
-    {
-        if (level & webrtc::kTraceError || level & webrtc::kTraceCritical)
-            pa_log("%s", message);
-        else if (level & webrtc::kTraceWarning)
-            pa_log_warn("%s", message);
-        else if (level & webrtc::kTraceInfo)
-            pa_log_info("%s", message);
-        else
-            pa_log_debug("%s", message);
-    }
-};
-
 static int webrtc_volume_from_pa(pa_volume_t v)
 {
     return (v * WEBRTC_AGC_MAX_VOLUME) / PA_VOLUME_NORM;
@@ -120,8 +87,7 @@ static pa_volume_t webrtc_volume_to_pa(int v)
 
 static void 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 *out_ss, pa_channel_map *out_map,
-                                  bool beamforming)
+                                  pa_sample_spec *out_ss, pa_channel_map *out_map)
 {
     rec_ss->format = PA_SAMPLE_FLOAT32NE;
     play_ss->format = PA_SAMPLE_FLOAT32NE;
@@ -139,110 +105,22 @@ static void webrtc_ec_fixate_spec(pa_sample_spec *rec_ss, pa_channel_map *rec_ma
     *out_ss = *rec_ss;
     *out_map = *rec_map;
 
-    if (beamforming) {
-        /* The beamformer gives us a single channel */
-        out_ss->channels = 1;
-        pa_channel_map_init_mono(out_map);
-    }
-
     /* Playback stream rate needs to be the same as capture */
     play_ss->rate = rec_ss->rate;
 }
 
-static bool parse_point(const char **point, float (&f)[3]) {
-    int ret, length;
-
-    ret = sscanf(*point, "%g,%g,%g%n", &f[0], &f[1], &f[2], &length);
-    if (ret != 3)
-        return false;
-
-    /* Consume the bytes we've read so far */
-    *point += length;
-
-    return true;
-}
-
-static bool parse_mic_geometry(const char **mic_geometry, std::vector<webrtc::Point>& geometry) {
-    /* The microphone geometry is expressed as cartesian point form:
-     *   x1,y1,z1,x2,y2,z2,...
-     *
-     * Where x1,y1,z1 is the position of the first microphone with regards to
-     * the array's "center", x2,y2,z2 the position of the second, and so on.
-     *
-     * 'x' is the horizontal coordinate, with positive values being to the
-     * right from the mic array's perspective.
-     *
-     * 'y' is the depth coordinate, with positive values being in front of the
-     * array.
-     *
-     * 'z' is the vertical coordinate, with positive values being above the
-     * array.
-     *
-     * All distances are in meters.
-     */
-
-    /* The target direction is expected to be in spherical point form:
-     *   a,e,r
-     *
-     * Where 'a' is the azimuth of the target point relative to the center of
-     * the array, 'e' its elevation, and 'r' the radius.
-     *
-     * 0 radians azimuth is to the right of the array, and positive angles
-     * move in a counter-clockwise direction.
-     *
-     * 0 radians elevation is horizontal w.r.t. the array, and positive
-     * angles go upwards.
-     *
-     * radius is distance from the array center in meters.
-     */
-
-    long unsigned int i;
-    float f[3];
-
-    for (i = 0; i < geometry.size(); i++) {
-        if (!parse_point(mic_geometry, f)) {
-            pa_log("Failed to parse channel %lu in mic_geometry", i);
-            return false;
-        }
-
-        /* Except for the last point, we should have a trailing comma */
-        if (i != geometry.size() - 1) {
-            if (**mic_geometry != ',') {
-                pa_log("Failed to parse channel %lu in mic_geometry", i);
-                return false;
-            }
-
-            (*mic_geometry)++;
-        }
-
-        pa_log_debug("Got mic #%lu position: (%g, %g, %g)", i, f[0], f[1], f[2]);
-
-        geometry[i].c[0] = f[0];
-        geometry[i].c[1] = f[1];
-        geometry[i].c[2] = f[2];
-    }
-
-    if (**mic_geometry != '\0') {
-        pa_log("Failed to parse mic_geometry value: more parameters than expected");
-        return false;
-    }
-
-    return true;
-}
-
 bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
                        pa_sample_spec *rec_ss, pa_channel_map *rec_map,
                        pa_sample_spec *play_ss, pa_channel_map *play_map,
                        pa_sample_spec *out_ss, pa_channel_map *out_map,
                        uint32_t *nframes, const char *args) {
-    webrtc::AudioProcessing *apm = NULL;
+    webrtc::AudioProcessing *apm = webrtc::AudioProcessingBuilder().Create();
     webrtc::ProcessingConfig pconfig;
-    webrtc::Config config;
-    bool hpf, ns, agc, dgc, mobile, cn, vad, ext_filter, intelligibility, experimental_agc, beamforming;
-    int rm = -1, i;
-    uint32_t agc_start_volume;
+    webrtc::AudioProcessing::Config config;
+    bool hpf, ns, tns, agc, dgc, mobile, pre_amp, vad, post_amp;
+    int i;
+    uint32_t agc_start_volume, pre_amp_gain, post_amp_gain;
     pa_modargs *ma;
-    bool trace = false;
 
     if (!(ma = pa_modargs_new(args, valid_modargs))) {
         pa_log("Failed to parse submodule arguments.");
@@ -261,6 +139,12 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
         goto fail;
     }
 
+    tns = DEFAULT_TRANSIENT_NOISE_SUPPRESSION;
+    if (pa_modargs_get_value_boolean(ma, "transient_noise_suppression", &tns) < 0) {
+        pa_log("Failed to parse transient_noise_suppression value");
+        goto fail;
+    }
+
     agc = DEFAULT_ANALOG_GAIN_CONTROL;
     if (pa_modargs_get_value_boolean(ma, "analog_gain_control", &agc) < 0) {
         pa_log("Failed to parse analog_gain_control value");
@@ -278,62 +162,47 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
         goto fail;
     }
 
-    mobile = DEFAULT_MOBILE;
-    if (pa_modargs_get_value_boolean(ma, "mobile", &mobile) < 0) {
-        pa_log("Failed to parse mobile value");
+    pre_amp = DEFAULT_PREAMP_ENABLE;
+    if (pa_modargs_get_value_boolean(ma, "pre_amplifier", &pre_amp) < 0) {
+        pa_log("Failed to parse pre_amplifier value");
         goto fail;
     }
-
-    ec->params.drift_compensation = DEFAULT_DRIFT_COMPENSATION;
-    if (pa_modargs_get_value_boolean(ma, "drift_compensation", &ec->params.drift_compensation) < 0) {
-        pa_log("Failed to parse drift_compensation value");
+    pre_amp_gain = DEFAULT_PREAMP_GAIN_DB;
+    if (pa_modargs_get_value_u32(ma, "pre_amplifier_gain", &pre_amp_gain) < 0) {
+        pa_log("Failed to parse pre_amplifier_gain value");
         goto fail;
     }
-
-    if (mobile) {
-        if (ec->params.drift_compensation) {
-            pa_log("Can't use drift_compensation in mobile mode");
-            goto fail;
-        }
-
-        if ((rm = routing_mode_from_string(pa_modargs_get_value(ma, "routing_mode", DEFAULT_ROUTING_MODE))) < 0) {
-            pa_log("Failed to parse routing_mode value");
-            goto fail;
-        }
-
-        cn = DEFAULT_COMFORT_NOISE;
-        if (pa_modargs_get_value_boolean(ma, "comfort_noise", &cn) < 0) {
-            pa_log("Failed to parse cn value");
-            goto fail;
-        }
-    } else {
-        if (pa_modargs_get_value(ma, "comfort_noise", NULL) || pa_modargs_get_value(ma, "routing_mode", NULL)) {
-            pa_log("The routing_mode and comfort_noise options are only valid with mobile=true");
-            goto fail;
-        }
+    if (pre_amp_gain > WEBRTC_PREAMP_GAIN_MAX_DB) {
+        pa_log("Preamp gain must not exceed %u", WEBRTC_PREAMP_GAIN_MAX_DB);
+        goto fail;
     }
 
-    vad = DEFAULT_VAD;
-    if (pa_modargs_get_value_boolean(ma, "voice_detection", &vad) < 0) {
-        pa_log("Failed to parse voice_detection value");
+    post_amp = DEFAULT_POSTAMP_ENABLE;
+    if (pa_modargs_get_value_boolean(ma, "post_amplifier", &post_amp) < 0) {
+        pa_log("Failed to parse post_amplifier value");
         goto fail;
     }
-
-    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");
+    post_amp_gain = DEFAULT_POSTAMP_GAIN_DB;
+    if (pa_modargs_get_value_u32(ma, "post_amplifier_gain", &post_amp_gain) < 0) {
+        pa_log("Failed to parse post_amplifier_gain value");
+        goto fail;
+    }
+    if (post_amp_gain > WEBRTC_POSTAMP_GAIN_MAX_DB) {
+        pa_log("Postamp gain must not exceed %u", WEBRTC_POSTAMP_GAIN_MAX_DB);
         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");
+    mobile = DEFAULT_MOBILE;
+    if (pa_modargs_get_value_boolean(ma, "mobile", &mobile) < 0) {
+        pa_log("Failed to parse mobile value");
         goto fail;
     }
 
-    experimental_agc = DEFAULT_EXPERIMENTAL_AGC;
-    if (pa_modargs_get_value_boolean(ma, "experimental_agc", &experimental_agc) < 0) {
-        pa_log("Failed to parse experimental_agc value");
+    ec->params.drift_compensation = DEFAULT_DRIFT_COMPENSATION;
+
+    vad = DEFAULT_VAD;
+    if (pa_modargs_get_value_boolean(ma, "voice_detection", &vad) < 0) {
+        pa_log("Failed to parse voice_detection value");
         goto fail;
     }
 
@@ -348,82 +217,7 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
     }
     ec->params.webrtc.agc_start_volume = agc_start_volume;
 
-    beamforming = DEFAULT_BEAMFORMING;
-    if (pa_modargs_get_value_boolean(ma, "beamforming", &beamforming) < 0) {
-        pa_log("Failed to parse beamforming value");
-        goto fail;
-    }
-
-    if (ext_filter)
-        config.Set<webrtc::ExtendedFilter>(new webrtc::ExtendedFilter(true));
-    if (intelligibility)
-        pa_log_warn("The intelligibility enhancer is not currently supported");
-    if (experimental_agc)
-        config.Set<webrtc::ExperimentalAgc>(new webrtc::ExperimentalAgc(true, ec->params.webrtc.agc_start_volume));
-
-    trace = DEFAULT_TRACE;
-    if (pa_modargs_get_value_boolean(ma, "trace", &trace) < 0) {
-        pa_log("Failed to parse trace value");
-        goto fail;
-    }
-
-    if (trace) {
-        webrtc::Trace::CreateTrace();
-        webrtc::Trace::set_level_filter(webrtc::kTraceAll);
-        ec->params.webrtc.trace_callback = new PaWebrtcTraceCallback();
-        webrtc::Trace::SetTraceCallback((PaWebrtcTraceCallback *) ec->params.webrtc.trace_callback);
-    }
-
-    webrtc_ec_fixate_spec(rec_ss, rec_map, play_ss, play_map, out_ss, out_map, beamforming);
-
-    /* We do this after fixate because we need the capture channel count */
-    if (beamforming) {
-        std::vector<webrtc::Point> geometry(rec_ss->channels);
-        webrtc::SphericalPointf direction(0.0f, 0.0f, 0.0f);
-        const char *mic_geometry, *target_direction;
-
-        if (!(mic_geometry = pa_modargs_get_value(ma, "mic_geometry", NULL))) {
-            pa_log("mic_geometry must be set if beamforming is enabled");
-            goto fail;
-        }
-
-        if (!parse_mic_geometry(&mic_geometry, geometry)) {
-            pa_log("Failed to parse mic_geometry value");
-            goto fail;
-        }
-
-        if ((target_direction = pa_modargs_get_value(ma, "target_direction", NULL))) {
-            float f[3];
-
-            if (!parse_point(&target_direction, f)) {
-                pa_log("Failed to parse target_direction value");
-                goto fail;
-            }
-
-            if (*target_direction != '\0') {
-                pa_log("Failed to parse target_direction value: more parameters than expected");
-                goto fail;
-            }
-
-#define IS_ZERO(f) ((f) < 0.000001 && (f) > -0.000001)
-
-            if (!IS_ZERO(f[1]) || !IS_ZERO(f[2])) {
-                pa_log("The beamformer currently only supports targeting along the azimuth");
-                goto fail;
-            }
-
-            direction.s[0] = f[0];
-            direction.s[1] = f[1];
-            direction.s[2] = f[2];
-        }
-
-        if (!target_direction)
-            config.Set<webrtc::Beamforming>(new webrtc::Beamforming(true, geometry));
-        else
-            config.Set<webrtc::Beamforming>(new webrtc::Beamforming(true, geometry, direction));
-    }
-
-    apm = webrtc::AudioProcessing::Create(config);
+    webrtc_ec_fixate_spec(rec_ss, rec_map, play_ss, play_map, out_ss, out_map);
 
     pconfig = {
         webrtc::StreamConfig(rec_ss->rate, rec_ss->channels, false), /* input stream */
@@ -436,46 +230,60 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
         goto fail;
     }
 
+    if (pre_amp) {
+       config.pre_amplifier.enabled = true;
+       config.pre_amplifier.fixed_gain_factor = (float)pre_amp_gain;
+    } else
+       config.pre_amplifier.enabled = false;
+
     if (hpf)
-        apm->high_pass_filter()->Enable(true);
-
-    if (!mobile) {
-        apm->echo_cancellation()->enable_drift_compensation(ec->params.drift_compensation);
-        apm->echo_cancellation()->Enable(true);
-    } else {
-        apm->echo_control_mobile()->set_routing_mode(static_cast<webrtc::EchoControlMobile::RoutingMode>(rm));
-        apm->echo_control_mobile()->enable_comfort_noise(cn);
-        apm->echo_control_mobile()->Enable(true);
-    }
+        config.high_pass_filter.enabled = true;
+    else
+        config.high_pass_filter.enabled = false;
 
-    if (ns) {
-        apm->noise_suppression()->set_level(webrtc::NoiseSuppression::kHigh);
-        apm->noise_suppression()->Enable(true);
-    }
+    config.echo_canceller.enabled = true;
 
-    if (agc || dgc) {
-        if (mobile && rm <= webrtc::EchoControlMobile::kEarpiece) {
-            /* Maybe this should be a knob, but we've got a lot of knobs already */
-            apm->gain_control()->set_mode(webrtc::GainControl::kFixedDigital);
-            ec->params.webrtc.agc = false;
-        } else if (dgc) {
-            apm->gain_control()->set_mode(webrtc::GainControl::kAdaptiveDigital);
-            ec->params.webrtc.agc = false;
-        } else {
-            apm->gain_control()->set_mode(webrtc::GainControl::kAdaptiveAnalog);
-            if (apm->gain_control()->set_analog_level_limits(0, WEBRTC_AGC_MAX_VOLUME) !=
-                    webrtc::AudioProcessing::kNoError) {
-                pa_log("Failed to initialise AGC");
-                goto fail;
-            }
-            ec->params.webrtc.agc = true;
-        }
+    if (!mobile)
+        config.echo_canceller.mobile_mode = false;
+    else
+        config.echo_canceller.mobile_mode = true;
+
+    if (ns)
+       config.noise_suppression.enabled = true;
+    else
+       config.noise_suppression.enabled = false;
 
-        apm->gain_control()->Enable(true);
+    if (tns)
+       config.transient_suppression.enabled = true;
+    else
+       config.transient_suppression.enabled = false;
+
+    if (dgc) {
+        ec->params.webrtc.agc = false;
+        config.gain_controller1.enabled = true;
+        if (mobile)
+            config.gain_controller1.mode = webrtc::AudioProcessing::Config::GainController1::kFixedDigital;
+        else
+            config.gain_controller1.mode = webrtc::AudioProcessing::Config::GainController1::kAdaptiveDigital;
+    } else if (agc) {
+        ec->params.webrtc.agc = true;
+        config.gain_controller1.enabled = true;
+        config.gain_controller1.mode = webrtc::AudioProcessing::Config::GainController1::kAdaptiveAnalog;
+        config.gain_controller1.analog_level_minimum = 0;
+        config.gain_controller1.analog_level_maximum = WEBRTC_AGC_MAX_VOLUME;
     }
 
     if (vad)
-        apm->voice_detection()->Enable(true);
+        config.voice_detection.enabled = true;
+    else
+        config.voice_detection.enabled = false;
+
+    if (post_amp) {
+        config.gain_controller2.enabled = true;
+        config.gain_controller2.fixed_digital.gain_db = (float)post_amp_gain;
+        config.gain_controller2.adaptive_digital.enabled = false;
+    } else
+        config.gain_controller2.enabled = false;
 
     ec->params.webrtc.apm = apm;
     ec->params.webrtc.rec_ss = *rec_ss;
@@ -485,6 +293,8 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
     *nframes = ec->params.webrtc.blocksize;
     ec->params.webrtc.first = true;
 
+    apm->ApplyConfig(config);
+
     for (i = 0; i < rec_ss->channels; i++)
         ec->params.webrtc.rec_buffer[i] = pa_xnew(float, *nframes);
     for (i = 0; i < play_ss->channels; i++)
@@ -496,10 +306,7 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
 fail:
     if (ma)
         pa_modargs_free(ma);
-    if (ec->params.webrtc.trace_callback) {
-        webrtc::Trace::ReturnTrace();
-        delete ((PaWebrtcTraceCallback *) ec->params.webrtc.trace_callback);
-    } if (apm)
+    if (apm)
         delete apm;
 
     return false;
@@ -515,12 +322,6 @@ void pa_webrtc_ec_play(pa_echo_canceller *ec, const uint8_t *play) {
     pa_deinterleave(play, (void **) buf, ss->channels, pa_sample_size(ss), n);
 
     pa_assert_se(apm->ProcessReverseStream(buf, config, config, buf) == webrtc::AudioProcessing::kNoError);
-
-    /* FIXME: If ProcessReverseStream() makes any changes to the audio, such as
-     * applying intelligibility enhancement, those changes don't have any
-     * effect. This function is called at the source side, but the processing
-     * would have to be done in the sink to be able to feed the processed audio
-     * to speakers. */
 }
 
 void pa_webrtc_ec_record(pa_echo_canceller *ec, const uint8_t *rec, uint8_t *out) {
@@ -538,7 +339,7 @@ void pa_webrtc_ec_record(pa_echo_canceller *ec, const uint8_t *rec, uint8_t *out
     if (ec->params.webrtc.agc) {
         pa_volume_t v = pa_echo_canceller_get_capture_volume(ec);
         old_volume = webrtc_volume_from_pa(v);
-        apm->gain_control()->set_stream_analog_level(old_volume);
+        apm->set_stream_analog_level(old_volume);
     }
 
     apm->set_stream_delay_ms(0);
@@ -553,7 +354,7 @@ void pa_webrtc_ec_record(pa_echo_canceller *ec, const uint8_t *rec, uint8_t *out
             ec->params.webrtc.first = false;
             new_volume = ec->params.webrtc.agc_start_volume;
         } else {
-            new_volume = apm->gain_control()->stream_analog_level();
+            new_volume = apm->recommended_stream_analog_level();
         }
 
         if (old_volume != new_volume)
@@ -564,9 +365,6 @@ void pa_webrtc_ec_record(pa_echo_canceller *ec, const uint8_t *rec, uint8_t *out
 }
 
 void pa_webrtc_ec_set_drift(pa_echo_canceller *ec, float drift) {
-    webrtc::AudioProcessing *apm = (webrtc::AudioProcessing*)ec->params.webrtc.apm;
-
-    apm->echo_cancellation()->set_stream_drift_samples(drift * ec->params.webrtc.blocksize);
 }
 
 void pa_webrtc_ec_run(pa_echo_canceller *ec, const uint8_t *rec, const uint8_t *play, uint8_t *out) {
@@ -577,11 +375,6 @@ void pa_webrtc_ec_run(pa_echo_canceller *ec, const uint8_t *rec, const uint8_t *
 void pa_webrtc_ec_done(pa_echo_canceller *ec) {
     int i;
 
-    if (ec->params.webrtc.trace_callback) {
-        webrtc::Trace::ReturnTrace();
-        delete ((PaWebrtcTraceCallback *) ec->params.webrtc.trace_callback);
-    }
-
     if (ec->params.webrtc.apm) {
         delete (webrtc::AudioProcessing*)ec->params.webrtc.apm;
         ec->params.webrtc.apm = NULL;