[PATCH v2 16/25] echo-cancel: Use anonymous unions for echo canceller params

[arun@xxxxxxxxxxxx (Arun Raghavan)]

From: Arun Raghavan <git@xxxxxxxxxxxxxxxx>

Makes this part of the code just a little less verbose.
---
 src/modules/echo-cancel/adrian.c      | 18 +++++-----
 src/modules/echo-cancel/echo-cancel.h |  3 +-
 src/modules/echo-cancel/null.c        |  4 +--
 src/modules/echo-cancel/speex.c       | 50 +++++++++++++--------------
 src/modules/echo-cancel/webrtc.cc     | 64 +++++++++++++++++------------------
 5 files changed, 70 insertions(+), 69 deletions(-)

diff --git a/src/modules/echo-cancel/adrian.c b/src/modules/echo-cancel/adrian.c
index 60a2b66..3c47fae 100644
--- a/src/modules/echo-cancel/adrian.c
+++ b/src/modules/echo-cancel/adrian.c
@@ -78,16 +78,16 @@ bool pa_adrian_ec_init(pa_core *c, pa_echo_canceller *ec,
 
     rate = out_ss->rate;
     *nframes = (rate * frame_size_ms) / 1000;
-Â Â Â Â ec->params.priv.adrian.blocksize = (*nframes) * pa_frame_size(out_ss);
+Â Â Â Â ec->params.adrian.blocksize = (*nframes) * pa_frame_size(out_ss);
 
-Â Â Â Â pa_log_debug ("Using nframes %d, blocksize %u, channels %d, rate %d", *nframes, ec->params.priv.adrian.blocksize, out_ss->channels, out_ss->rate);
+Â Â Â Â pa_log_debug ("Using nframes %d, blocksize %u, channels %d, rate %d", *nframes, ec->params.adrian.blocksize, out_ss->channels, out_ss->rate);
 
     /* For now we only support SSE */
     if (c->cpu_info.cpu_type == PA_CPU_X86 && (c->cpu_info.flags.x86 & PA_CPU_X86_SSE))
         have_vector = 1;
 
-Â Â Â Â ec->params.priv.adrian.aec = AEC_init(rate, have_vector);
-Â Â Â Â if (!ec->params.priv.adrian.aec)
+Â Â Â Â ec->params.adrian.aec = AEC_init(rate, have_vector);
+Â Â Â Â if (!ec->params.adrian.aec)
         goto fail;
 
     pa_modargs_free(ma);
@@ -102,17 +102,17 @@ fail:
 void pa_adrian_ec_run(pa_echo_canceller *ec, const uint8_t *rec, const uint8_t *play, uint8_t *out) {
     unsigned int i;
 
-Â Â Â Â for (i = 0; i < ec->params.priv.adrian.blocksize; i += 2) {
+Â Â Â Â for (i = 0; i < ec->params.adrian.blocksize; i += 2) {
         /* We know it's S16NE mono data */
         int r = *(int16_t *)(rec + i);
         int p = *(int16_t *)(play + i);
-Â Â Â Â Â Â Â Â *(int16_t *)(out + i) = (int16_t) AEC_doAEC(ec->params.priv.adrian.aec, r, p);
+Â Â Â Â Â Â Â Â *(int16_t *)(out + i) = (int16_t) AEC_doAEC(ec->params.adrian.aec, r, p);
     }
 }
 
 void pa_adrian_ec_done(pa_echo_canceller *ec) {
-Â Â Â Â if (ec->params.priv.adrian.aec) {
-Â Â Â Â Â Â Â Â AEC_done(ec->params.priv.adrian.aec);
-Â Â Â Â Â Â Â Â ec->params.priv.adrian.aec = NULL;
+Â Â Â Â if (ec->params.adrian.aec) {
+Â Â Â Â Â Â Â Â AEC_done(ec->params.adrian.aec);
+Â Â Â Â Â Â Â Â ec->params.adrian.aec = NULL;
     }
 }
diff --git a/src/modules/echo-cancel/echo-cancel.h b/src/modules/echo-cancel/echo-cancel.h
index b570095..37f99c0 100644
--- a/src/modules/echo-cancel/echo-cancel.h
+++ b/src/modules/echo-cancel/echo-cancel.h
@@ -69,10 +69,11 @@ struct pa_echo_canceller_params {
             bool agc;
             bool trace;
             bool first;
+Â Â Â Â Â Â Â Â Â Â Â Â unsigned int agc_start_volume;
         } webrtc;
 #endif
         /* each canceller-specific structure goes here */
-Â Â Â Â } priv;
+Â Â Â Â };
 
     /* Set this if canceller can do drift compensation. Also see set_drift()
      * below */
diff --git a/src/modules/echo-cancel/null.c b/src/modules/echo-cancel/null.c
index 673b14f..c8ecf27 100644
--- a/src/modules/echo-cancel/null.c
+++ b/src/modules/echo-cancel/null.c
@@ -34,7 +34,7 @@ bool pa_null_ec_init(pa_core *c, pa_echo_canceller *ec,
     char strss_sink[PA_SAMPLE_SPEC_SNPRINT_MAX];
 
     *nframes = 256;
-Â Â Â Â ec->params.priv.null.out_ss = *out_ss;
+Â Â Â Â ec->params.null.out_ss = *out_ss;
 
     *rec_ss = *out_ss;
     *rec_map = *out_map;
@@ -49,7 +49,7 @@ bool pa_null_ec_init(pa_core *c, pa_echo_canceller *ec,
 void pa_null_ec_run(pa_echo_canceller *ec, const uint8_t *rec, const uint8_t *play, uint8_t *out) {
     /* The null implementation simply copies the recorded buffer to the output
        buffer and ignores the play buffer. */
-Â Â Â Â memcpy(out, rec, 256 * pa_frame_size(&ec->params.priv.null.out_ss));
+Â Â Â Â memcpy(out, rec, 256 * pa_frame_size(&ec->params.null.out_ss));
 }
 
 void pa_null_ec_done(pa_echo_canceller *ec) {
diff --git a/src/modules/echo-cancel/speex.c b/src/modules/echo-cancel/speex.c
index 11e53b3..08c1027 100644
--- a/src/modules/echo-cancel/speex.c
+++ b/src/modules/echo-cancel/speex.c
@@ -111,26 +111,26 @@ static bool pa_speex_ec_preprocessor_init(pa_echo_canceller *ec, pa_sample_spec
             goto fail;
         }
 
-Â Â Â Â Â Â Â Â ec->params.priv.speex.pp_state = speex_preprocess_state_init(nframes, out_ss->rate);
+Â Â Â Â Â Â Â Â ec->params.speex.pp_state = speex_preprocess_state_init(nframes, out_ss->rate);
 
         tmp = agc;
-Â Â Â Â Â Â Â Â speex_preprocess_ctl(ec->params.priv.speex.pp_state, SPEEX_PREPROCESS_SET_AGC, &tmp);
+Â Â Â Â Â Â Â Â speex_preprocess_ctl(ec->params.speex.pp_state, SPEEX_PREPROCESS_SET_AGC, &tmp);
 
         tmp = denoise;
-Â Â Â Â Â Â Â Â speex_preprocess_ctl(ec->params.priv.speex.pp_state, SPEEX_PREPROCESS_SET_DENOISE, &tmp);
+Â Â Â Â Â Â Â Â speex_preprocess_ctl(ec->params.speex.pp_state, SPEEX_PREPROCESS_SET_DENOISE, &tmp);
 
         if (echo_suppress) {
             if (echo_suppress_attenuation)
-Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â speex_preprocess_ctl(ec->params.priv.speex.pp_state, SPEEX_PREPROCESS_SET_ECHO_SUPPRESS,
+Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â speex_preprocess_ctl(ec->params.speex.pp_state, SPEEX_PREPROCESS_SET_ECHO_SUPPRESS,
                                      &echo_suppress_attenuation);
 
             if (echo_suppress_attenuation_active) {
-Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â speex_preprocess_ctl(ec->params.priv.speex.pp_state, SPEEX_PREPROCESS_SET_ECHO_SUPPRESS_ACTIVE,
+Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â speex_preprocess_ctl(ec->params.speex.pp_state, SPEEX_PREPROCESS_SET_ECHO_SUPPRESS_ACTIVE,
                                      &echo_suppress_attenuation_active);
             }
 
-Â Â Â Â Â Â Â Â Â Â Â Â speex_preprocess_ctl(ec->params.priv.speex.pp_state, SPEEX_PREPROCESS_SET_ECHO_STATE,
-Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â ec->params.priv.speex.state);
+Â Â Â Â Â Â Â Â Â Â Â Â speex_preprocess_ctl(ec->params.speex.pp_state, SPEEX_PREPROCESS_SET_ECHO_STATE,
+Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â ec->params.speex.state);
         }
 
         pa_log_info("Loaded speex preprocessor with params: agc=%s, denoise=%s, echo_suppress=%s", pa_yes_no(agc),
@@ -176,12 +176,12 @@ bool pa_speex_ec_init(pa_core *c, pa_echo_canceller *ec,
     *nframes = pa_echo_canceller_blocksize_power2(rate, frame_size_ms);
 
     pa_log_debug ("Using nframes %d, channels %d, rate %d", *nframes, out_ss->channels, out_ss->rate);
-Â Â Â Â ec->params.priv.speex.state = speex_echo_state_init_mc(*nframes, (rate * filter_size_ms) / 1000, out_ss->channels, out_ss->channels);
+Â Â Â Â ec->params.speex.state = speex_echo_state_init_mc(*nframes, (rate * filter_size_ms) / 1000, out_ss->channels, out_ss->channels);
 
-Â Â Â Â if (!ec->params.priv.speex.state)
+Â Â Â Â if (!ec->params.speex.state)
         goto fail;
 
-Â Â Â Â speex_echo_ctl(ec->params.priv.speex.state, SPEEX_ECHO_SET_SAMPLING_RATE, &rate);
+Â Â Â Â speex_echo_ctl(ec->params.speex.state, SPEEX_ECHO_SET_SAMPLING_RATE, &rate);
 
     if (!pa_speex_ec_preprocessor_init(ec, out_ss, *nframes, ma))
         goto fail;
@@ -192,34 +192,34 @@ bool pa_speex_ec_init(pa_core *c, pa_echo_canceller *ec,
 fail:
     if (ma)
         pa_modargs_free(ma);
-Â Â Â Â if (ec->params.priv.speex.pp_state) {
-Â Â Â Â Â Â Â Â speex_preprocess_state_destroy(ec->params.priv.speex.pp_state);
-Â Â Â Â Â Â Â Â ec->params.priv.speex.pp_state = NULL;
+Â Â Â Â if (ec->params.speex.pp_state) {
+Â Â Â Â Â Â Â Â speex_preprocess_state_destroy(ec->params.speex.pp_state);
+Â Â Â Â Â Â Â Â ec->params.speex.pp_state = NULL;
     }
-Â Â Â Â if (ec->params.priv.speex.state) {
-Â Â Â Â Â Â Â Â speex_echo_state_destroy(ec->params.priv.speex.state);
-Â Â Â Â Â Â Â Â ec->params.priv.speex.state = NULL;
+Â Â Â Â if (ec->params.speex.state) {
+Â Â Â Â Â Â Â Â speex_echo_state_destroy(ec->params.speex.state);
+Â Â Â Â Â Â Â Â ec->params.speex.state = NULL;
     }
     return false;
 }
 
 void pa_speex_ec_run(pa_echo_canceller *ec, const uint8_t *rec, const uint8_t *play, uint8_t *out) {
-Â Â Â Â speex_echo_cancellation(ec->params.priv.speex.state, (const spx_int16_t *) rec, (const spx_int16_t *) play,
+Â Â Â Â speex_echo_cancellation(ec->params.speex.state, (const spx_int16_t *) rec, (const spx_int16_t *) play,
                             (spx_int16_t *) out);
 
     /* preprecessor is run after AEC. This is not a mistake! */
-Â Â Â Â if (ec->params.priv.speex.pp_state)
-Â Â Â Â Â Â Â Â speex_preprocess_run(ec->params.priv.speex.pp_state, (spx_int16_t *) out);
+Â Â Â Â if (ec->params.speex.pp_state)
+Â Â Â Â Â Â Â Â speex_preprocess_run(ec->params.speex.pp_state, (spx_int16_t *) out);
 }
 
 void pa_speex_ec_done(pa_echo_canceller *ec) {
-Â Â Â Â if (ec->params.priv.speex.pp_state) {
-Â Â Â Â Â Â Â Â speex_preprocess_state_destroy(ec->params.priv.speex.pp_state);
-Â Â Â Â Â Â Â Â ec->params.priv.speex.pp_state = NULL;
+Â Â Â Â if (ec->params.speex.pp_state) {
+Â Â Â Â Â Â Â Â speex_preprocess_state_destroy(ec->params.speex.pp_state);
+Â Â Â Â Â Â Â Â ec->params.speex.pp_state = NULL;
     }
 
-Â Â Â Â if (ec->params.priv.speex.state) {
-Â Â Â Â Â Â Â Â speex_echo_state_destroy(ec->params.priv.speex.state);
-Â Â Â Â Â Â Â Â ec->params.priv.speex.state = NULL;
+Â Â Â Â if (ec->params.speex.state) {
+Â Â Â Â Â Â Â Â speex_echo_state_destroy(ec->params.speex.state);
+Â Â Â Â Â Â Â Â ec->params.speex.state = NULL;
     }
 }
diff --git a/src/modules/echo-cancel/webrtc.cc b/src/modules/echo-cancel/webrtc.cc
index abca811..693bf88 100644
--- a/src/modules/echo-cancel/webrtc.cc
+++ b/src/modules/echo-cancel/webrtc.cc
@@ -249,13 +249,13 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
     if (experimental_agc)
         config.Set<webrtc::ExperimentalAgc>(new webrtc::ExperimentalAgc(true, WEBRTC_AGC_START_VOLUME));
 
-Â Â Â Â ec->params.priv.webrtc.trace = DEFAULT_TRACE;
-Â Â Â Â if (pa_modargs_get_value_boolean(ma, "trace", &ec->params.priv.webrtc.trace) < 0) {
+Â Â Â Â ec->params.webrtc.trace = DEFAULT_TRACE;
+Â Â Â Â if (pa_modargs_get_value_boolean(ma, "trace", &ec->params.webrtc.trace) < 0) {
         pa_log("Failed to parse trace value");
         goto fail;
     }
 
-Â Â Â Â if (ec->params.priv.webrtc.trace) {
+Â Â Â Â if (ec->params.webrtc.trace) {
         webrtc::Trace::CreateTrace();
         webrtc::Trace::set_level_filter(webrtc::kTraceAll);
         webrtc::Trace::SetTraceCallback(new PaWebrtcTraceCallback());
@@ -294,17 +294,17 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
         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);
-Â Â Â Â Â Â Â Â Â Â Â Â ec->params.priv.webrtc.agc = false;
+Â Â Â Â Â Â Â Â Â Â Â Â ec->params.webrtc.agc = false;
         } else if (dgc) {
             apm->gain_control()->set_mode(webrtc::GainControl::kAdaptiveDigital);
-Â Â Â Â Â Â Â Â Â Â Â Â ec->params.priv.webrtc.agc = false;
+Â Â Â Â Â Â Â Â Â Â Â Â 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) != apm->kNoError) {
                 pa_log("Failed to initialise AGC");
                 goto fail;
             }
-Â Â Â Â Â Â Â Â Â Â Â Â ec->params.priv.webrtc.agc = true;
+Â Â Â Â Â Â Â Â Â Â Â Â ec->params.webrtc.agc = true;
         }
 
         apm->gain_control()->Enable(true);
@@ -313,11 +313,11 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
     if (vad)
         apm->voice_detection()->Enable(true);
 
-Â Â Â Â ec->params.priv.webrtc.apm = apm;
-Â Â Â Â ec->params.priv.webrtc.sample_spec = *out_ss;
-Â Â Â Â ec->params.priv.webrtc.blocksize = (uint64_t)pa_bytes_per_second(out_ss) * BLOCK_SIZE_US / PA_USEC_PER_SEC;
-Â Â Â Â *nframes = ec->params.priv.webrtc.blocksize / pa_frame_size(out_ss);
-Â Â Â Â ec->params.priv.webrtc.first = true;
+Â Â Â Â ec->params.webrtc.apm = apm;
+Â Â Â Â ec->params.webrtc.sample_spec = *out_ss;
+Â Â Â Â ec->params.webrtc.blocksize = (uint64_t)pa_bytes_per_second(out_ss) * BLOCK_SIZE_US / PA_USEC_PER_SEC;
+Â Â Â Â *nframes = ec->params.webrtc.blocksize / pa_frame_size(out_ss);
+Â Â Â Â ec->params.webrtc.first = true;
 
     pa_modargs_free(ma);
     return true;
@@ -325,7 +325,7 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
 fail:
     if (ma)
         pa_modargs_free(ma);
-Â Â Â Â if (ec->params.priv.webrtc.trace)
+Â Â Â Â if (ec->params.webrtc.trace)
         webrtc::Trace::ReturnTrace();
     if (apm)
         delete apm;
@@ -334,37 +334,37 @@ fail:
 }
 
 void pa_webrtc_ec_play(pa_echo_canceller *ec, const uint8_t *play) {
-Â Â Â Â webrtc::AudioProcessing *apm = (webrtc::AudioProcessing*)ec->params.priv.webrtc.apm;
+Â Â Â Â webrtc::AudioProcessing *apm = (webrtc::AudioProcessing*)ec->params.webrtc.apm;
     webrtc::AudioFrame play_frame;
-Â Â Â Â const pa_sample_spec *ss = &ec->params.priv.webrtc.sample_spec;
+Â Â Â Â const pa_sample_spec *ss = &ec->params.webrtc.sample_spec;
 
     play_frame.num_channels_ = ss->channels;
     play_frame.sample_rate_hz_ = ss->rate;
     play_frame.interleaved_ = false;
-Â Â Â Â play_frame.samples_per_channel_ = ec->params.priv.webrtc.blocksize / pa_frame_size(ss);
+Â Â Â Â play_frame.samples_per_channel_ = ec->params.webrtc.blocksize / pa_frame_size(ss);
 
     pa_assert(play_frame.samples_per_channel_ <= webrtc::AudioFrame::kMaxDataSizeSamples);
-Â Â Â Â memcpy(play_frame.data_, play, ec->params.priv.webrtc.blocksize);
+Â Â Â Â memcpy(play_frame.data_, play, ec->params.webrtc.blocksize);
 
     apm->ProcessReverseStream(&play_frame);
 }
 
 void pa_webrtc_ec_record(pa_echo_canceller *ec, const uint8_t *rec, uint8_t *out) {
-Â Â Â Â webrtc::AudioProcessing *apm = (webrtc::AudioProcessing*)ec->params.priv.webrtc.apm;
+Â Â Â Â webrtc::AudioProcessing *apm = (webrtc::AudioProcessing*)ec->params.webrtc.apm;
     webrtc::AudioFrame out_frame;
-Â Â Â Â const pa_sample_spec *ss = &ec->params.priv.webrtc.sample_spec;
+Â Â Â Â const pa_sample_spec *ss = &ec->params.webrtc.sample_spec;
     pa_cvolume v;
     int old_volume, new_volume;
 
     out_frame.num_channels_ = ss->channels;
     out_frame.sample_rate_hz_ = ss->rate;
     out_frame.interleaved_ = false;
-Â Â Â Â out_frame.samples_per_channel_ = ec->params.priv.webrtc.blocksize / pa_frame_size(ss);
+Â Â Â Â out_frame.samples_per_channel_ = ec->params.webrtc.blocksize / pa_frame_size(ss);
 
     pa_assert(out_frame.samples_per_channel_ <= webrtc::AudioFrame::kMaxDataSizeSamples);
-Â Â Â Â memcpy(out_frame.data_, rec, ec->params.priv.webrtc.blocksize);
+Â Â Â Â memcpy(out_frame.data_, rec, ec->params.webrtc.blocksize);
 
-Â Â Â Â if (ec->params.priv.webrtc.agc) {
+Â Â Â Â if (ec->params.webrtc.agc) {
         pa_cvolume_init(&v);
         pa_echo_canceller_get_capture_volume(ec, &v);
         old_volume = webrtc_volume_from_pa(pa_cvolume_avg(&v));
@@ -374,13 +374,13 @@ void pa_webrtc_ec_record(pa_echo_canceller *ec, const uint8_t *rec, uint8_t *out
     apm->set_stream_delay_ms(0);
     apm->ProcessStream(&out_frame);
 
-Â Â Â Â if (ec->params.priv.webrtc.agc) {
-Â Â Â Â Â Â Â Â if (PA_UNLIKELY(ec->params.priv.webrtc.first)) {
+Â Â Â Â if (ec->params.webrtc.agc) {
+Â Â Â Â Â Â Â Â 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 */
-Â Â Â Â Â Â Â Â Â Â Â Â ec->params.priv.webrtc.first = false;
+Â Â Â Â Â Â Â Â Â Â Â Â ec->params.webrtc.first = false;
             new_volume = WEBRTC_AGC_START_VOLUME;
         } else {
             new_volume = apm->gain_control()->stream_analog_level();
@@ -392,14 +392,14 @@ void pa_webrtc_ec_record(pa_echo_canceller *ec, const uint8_t *rec, uint8_t *out
         }
     }
 
-Â Â Â Â memcpy(out, out_frame.data_, ec->params.priv.webrtc.blocksize);
+Â Â Â Â memcpy(out, out_frame.data_, ec->params.webrtc.blocksize);
 }
 
 void pa_webrtc_ec_set_drift(pa_echo_canceller *ec, float drift) {
-Â Â Â Â webrtc::AudioProcessing *apm = (webrtc::AudioProcessing*)ec->params.priv.webrtc.apm;
-Â Â Â Â const pa_sample_spec *ss = &ec->params.priv.webrtc.sample_spec;
+Â Â Â Â webrtc::AudioProcessing *apm = (webrtc::AudioProcessing*)ec->params.webrtc.apm;
+Â Â Â Â const pa_sample_spec *ss = &ec->params.webrtc.sample_spec;
 
-Â Â Â Â apm->echo_cancellation()->set_stream_drift_samples(drift * ec->params.priv.webrtc.blocksize / pa_frame_size(ss));
+Â Â Â Â apm->echo_cancellation()->set_stream_drift_samples(drift * ec->params.webrtc.blocksize / pa_frame_size(ss));
 }
 
 void pa_webrtc_ec_run(pa_echo_canceller *ec, const uint8_t *rec, const uint8_t *play, uint8_t *out) {
@@ -408,11 +408,11 @@ 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) {
-Â Â Â Â if (ec->params.priv.webrtc.trace)
+Â Â Â Â if (ec->params.webrtc.trace)
         webrtc::Trace::ReturnTrace();
 
-Â Â Â Â if (ec->params.priv.webrtc.apm) {
-Â Â Â Â Â Â Â Â delete (webrtc::AudioProcessing*)ec->params.priv.webrtc.apm;
-Â Â Â Â Â Â Â Â ec->params.priv.webrtc.apm = NULL;
+Â Â Â Â if (ec->params.webrtc.apm) {
+Â Â Â Â Â Â Â Â delete (webrtc::AudioProcessing*)ec->params.webrtc.apm;
+Â Â Â Â Â Â Â Â ec->params.webrtc.apm = NULL;
     }
 }
-- 
2.5.0