Merge branch 'support-old-webrtc' into 'master'

echo-cancel: Support old webrtc-audio-processing library

See merge request pulseaudio/pulseaudio!836

meson.build

@@ -726,9 +726,15 @@ if get_option('daemon')
     cdata.set('HAVE_SOXR', 1)
   endif
 
-  webrtc_dep = dependency('webrtc-audio-processing-1', version : '>= 1.0', required : get_option('webrtc-aec'))
+  webrtc_dep = dependency('webrtc-audio-processing-1', version : '>= 1.0', required : false)
   if webrtc_dep.found()
     cdata.set('HAVE_WEBRTC', 1)
+    cdata.set('HAVE_WEBRTC1', 1)
+  else
+    webrtc_dep = dependency('webrtc-audio-processing', version : ['>= 0.2', '< 1.0'], required : get_option('webrtc-aec'))
+    if webrtc_dep.found()
+      cdata.set('HAVE_WEBRTC', 1)
+    endif
   endif
 
   systemd_dep = dependency('systemd', required : get_option('systemd'))

src/modules/echo-cancel/meson.build

@@ -9,9 +9,15 @@
 
 add_languages('cpp')
 
+if cdata.has('HAVE_WEBRTC1')
   libwebrtc_util_sources = [
     'webrtc.cc'
   ]
+else
+  libwebrtc_util_sources = [
+    'webrtc-old.cc'
+  ]
+endif
 
 if host_machine.system() == 'darwin'
   ignore_unresolved_symbols_link_args = ['-Wl,-undefined,dynamic_lookup']

src/modules/echo-cancel/webrtc-old.cc

@@ -0,0 +1,594 @@
+/***
+    This file is part of PulseAudio.
+
+    Copyright 2011 Collabora Ltd.
+              2015 Aldebaran SoftBank Group
+
+    Contributor: Arun Raghavan <mail@arunraghavan.net>
+
+    PulseAudio is free software; you can redistribute it and/or modify
+    it under the terms of the GNU Lesser General Public License as published
+    by the Free Software Foundation; either version 2.1 of the License,
+    or (at your option) any later version.
+
+    PulseAudio is distributed in the hope that it will be useful, but
+    WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+    General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public License
+    along with PulseAudio; if not, see <http://www.gnu.org/licenses/>.
+***/
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <pulse/cdecl.h>
+
+PA_C_DECL_BEGIN
+#include <pulsecore/core-util.h>
+#include <pulsecore/modargs.h>
+
+#include <pulse/timeval.h>
+#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 BLOCK_SIZE_US 10000
+
+#define DEFAULT_HIGH_PASS_FILTER true
+#define DEFAULT_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_AGC_START_VOLUME 85
+#define DEFAULT_BEAMFORMING false
+#define DEFAULT_TRACE false
+
+#define WEBRTC_AGC_MAX_VOLUME 255
+
+static const char* const valid_modargs[] = {
+    "high_pass_filter",
+    "noise_suppression",
+    "analog_gain_control",
+    "digital_gain_control",
+    "mobile",
+    "routing_mode",
+    "comfort_noise",
+    "drift_compensation",
+    "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;
+}
+
+static pa_volume_t webrtc_volume_to_pa(int v)
+{
+    return (v * PA_VOLUME_NORM) / WEBRTC_AGC_MAX_VOLUME;
+}
+
+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)
+{
+    rec_ss->format = PA_SAMPLE_FLOAT32NE;
+    play_ss->format = PA_SAMPLE_FLOAT32NE;
+
+    /* AudioProcessing expects one of the following rates */
+    if (rec_ss->rate >= 48000)
+        rec_ss->rate = 48000;
+    else if (rec_ss->rate >= 32000)
+        rec_ss->rate = 32000;
+    else if (rec_ss->rate >= 16000)
+        rec_ss->rate = 16000;
+    else
+        rec_ss->rate = 8000;
+
+    *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::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;
+    pa_modargs *ma;
+    bool trace = false;
+
+    if (!(ma = pa_modargs_new(args, valid_modargs))) {
+        pa_log("Failed to parse submodule arguments.");
+        goto fail;
+    }
+
+    hpf = DEFAULT_HIGH_PASS_FILTER;
+    if (pa_modargs_get_value_boolean(ma, "high_pass_filter", &hpf) < 0) {
+        pa_log("Failed to parse high_pass_filter value");
+        goto fail;
+    }
+
+    ns = DEFAULT_NOISE_SUPPRESSION;
+    if (pa_modargs_get_value_boolean(ma, "noise_suppression", &ns) < 0) {
+        pa_log("Failed to parse 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");
+        goto fail;
+    }
+
+    dgc = agc ? false : DEFAULT_DIGITAL_GAIN_CONTROL;
+    if (pa_modargs_get_value_boolean(ma, "digital_gain_control", &dgc) < 0) {
+        pa_log("Failed to parse digital_gain_control value");
+        goto fail;
+    }
+
+    if (agc && dgc) {
+        pa_log("You must pick only one between analog and digital gain control");
+        goto fail;
+    }
+
+    mobile = DEFAULT_MOBILE;
+    if (pa_modargs_get_value_boolean(ma, "mobile", &mobile) < 0) {
+        pa_log("Failed to parse mobile 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");
+        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;
+        }
+    }
+
+    vad = DEFAULT_VAD;
+    if (pa_modargs_get_value_boolean(ma, "voice_detection", &vad) < 0) {
+        pa_log("Failed to parse voice_detection 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");
+        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;
+    }
+
+    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");
+        goto fail;
+    }
+
+    agc_start_volume = DEFAULT_AGC_START_VOLUME;
+    if (pa_modargs_get_value_u32(ma, "agc_start_volume", &agc_start_volume) < 0) {
+        pa_log("Failed to parse agc_start_volume value");
+        goto fail;
+    }
+    if (agc_start_volume > WEBRTC_AGC_MAX_VOLUME) {
+        pa_log("AGC start volume must not exceed %u", WEBRTC_AGC_MAX_VOLUME);
+        goto fail;
+    }
+    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);
+
+    pconfig = {
+        webrtc::StreamConfig(rec_ss->rate, rec_ss->channels, false), /* input stream */
+        webrtc::StreamConfig(out_ss->rate, out_ss->channels, false), /* output stream */
+        webrtc::StreamConfig(play_ss->rate, play_ss->channels, false), /* reverse input stream */
+        webrtc::StreamConfig(play_ss->rate, play_ss->channels, false), /* reverse output stream */
+    };
+    if (apm->Initialize(pconfig) != webrtc::AudioProcessing::kNoError) {
+        pa_log("Error initialising audio processing module");
+        goto fail;
+    }
+
+    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);
+    }
+
+    if (ns) {
+        apm->noise_suppression()->set_level(webrtc::NoiseSuppression::kHigh);
+        apm->noise_suppression()->Enable(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;
+        }
+
+        apm->gain_control()->Enable(true);
+    }
+
+    if (vad)
+        apm->voice_detection()->Enable(true);
+
+    ec->params.webrtc.apm = apm;
+    ec->params.webrtc.rec_ss = *rec_ss;
+    ec->params.webrtc.play_ss = *play_ss;
+    ec->params.webrtc.out_ss = *out_ss;
+    ec->params.webrtc.blocksize = (uint64_t) out_ss->rate * BLOCK_SIZE_US / PA_USEC_PER_SEC;
+    *nframes = ec->params.webrtc.blocksize;
+    ec->params.webrtc.first = true;
+
+    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++)
+        ec->params.webrtc.play_buffer[i] = pa_xnew(float, *nframes);
+
+    pa_modargs_free(ma);
+    return true;
+
+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)
+        delete apm;
+
+    return false;
+}
+
+void pa_webrtc_ec_play(pa_echo_canceller *ec, const uint8_t *play) {
+    webrtc::AudioProcessing *apm = (webrtc::AudioProcessing*)ec->params.webrtc.apm;
+    const pa_sample_spec *ss = &ec->params.webrtc.play_ss;
+    int n = ec->params.webrtc.blocksize;
+    float **buf = ec->params.webrtc.play_buffer;
+    webrtc::StreamConfig config(ss->rate, ss->channels, false);
+
+    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) {
+    webrtc::AudioProcessing *apm = (webrtc::AudioProcessing*)ec->params.webrtc.apm;
+    const pa_sample_spec *rec_ss = &ec->params.webrtc.rec_ss;
+    const pa_sample_spec *out_ss = &ec->params.webrtc.out_ss;
+    float **buf = ec->params.webrtc.rec_buffer;
+    int n = ec->params.webrtc.blocksize;
+    int old_volume, new_volume;
+    webrtc::StreamConfig rec_config(rec_ss->rate, rec_ss->channels, false);
+    webrtc::StreamConfig out_config(out_ss->rate, out_ss->channels, false);
+
+    pa_deinterleave(rec, (void **) buf, rec_ss->channels, pa_sample_size(rec_ss), n);
+
+    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_delay_ms(0);
+    pa_assert_se(apm->ProcessStream(buf, rec_config, out_config, buf) == webrtc::AudioProcessing::kNoError);
+
+    if (ec->params.webrtc.agc) {
+        if (PA_UNLIKELY(ec->params.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.webrtc.first = false;
+            new_volume = ec->params.webrtc.agc_start_volume;
+        } else {
+            new_volume = apm->gain_control()->stream_analog_level();
+        }
+
+        if (old_volume != new_volume)
+            pa_echo_canceller_set_capture_volume(ec, webrtc_volume_to_pa(new_volume));
+    }
+
+    pa_interleave((const void **) buf, out_ss->channels, out, pa_sample_size(out_ss), n);
+}
+
+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) {
+    pa_webrtc_ec_play(ec, play);
+    pa_webrtc_ec_record(ec, rec, out);
+}
+
+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;
+    }
+
+    for (i = 0; i < ec->params.webrtc.rec_ss.channels; i++)
+        pa_xfree(ec->params.webrtc.rec_buffer[i]);
+    for (i = 0; i < ec->params.webrtc.play_ss.channels; i++)
+        pa_xfree(ec->params.webrtc.play_buffer[i]);
+}