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

PulseAudio requires webrtc-audio-processing-1 >= 1.0 since v17.0, but
Debian hasn't managed to package that version yet. The dependency is
only used for the echo-cancel module, and the API exposed by the webrtc
component doesn't appear to be changed.

It looks like we can revert the changes to allow building with an older
version, but it would be better to have the code support both versions
to make it easier to transition from early library versions to later
versions. As a step towards that, check in an older copy of the webrtc
component from commit b16b107171, and build with that when we have the
old library version.

Signed-off-by: Alper Nebi Yasak <alpernebiyasak@gmail.com>

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]);
+}