webrtcdsp: indent C++ sources

ext/webrtcdsp/gstwebrtcdsp.cpp

@@ -88,7 +88,7 @@ GST_DEBUG_CATEGORY (webrtc_dsp_debug);
 #define DEFAULT_VOICE_DETECTION_LIKELIHOOD webrtc::VoiceDetection::kLowLikelihood
 
 static GstStaticPadTemplate gst_webrtc_dsp_sink_template =
-GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_STATIC_PAD_TEMPLATE ("sink",
     GST_PAD_SINK,
     GST_PAD_ALWAYS,
     GST_STATIC_CAPS ("audio/x-raw, "
@@ -104,7 +104,7 @@ GST_STATIC_PAD_TEMPLATE ("sink",
     );
 
 static GstStaticPadTemplate gst_webrtc_dsp_src_template =
-GST_STATIC_PAD_TEMPLATE ("src",
+    GST_STATIC_PAD_TEMPLATE ("src",
     GST_PAD_SRC,
     GST_PAD_ALWAYS,
     GST_STATIC_CAPS ("audio/x-raw, "
@@ -119,17 +119,24 @@ GST_STATIC_PAD_TEMPLATE ("src",
         "channels = (int) [1, MAX]")
     );
 
-typedef webrtc::EchoCancellation::SuppressionLevel GstWebrtcEchoSuppressionLevel;
+typedef
+    webrtc::EchoCancellation::SuppressionLevel
+    GstWebrtcEchoSuppressionLevel;
 #define GST_TYPE_WEBRTC_ECHO_SUPPRESSION_LEVEL \
     (gst_webrtc_echo_suppression_level_get_type ())
-static GType
+static
+    GType
 gst_webrtc_echo_suppression_level_get_type (void)
 {
-  static GType suppression_level_type = 0;
-  static const GEnumValue level_types[] = {
+  static
+      GType
+      suppression_level_type = 0;
+  static const
+      GEnumValue
+      level_types[] = {
     {webrtc::EchoCancellation::kLowSuppression, "Low Suppression", "low"},
     {webrtc::EchoCancellation::kModerateSuppression,
-      "Moderate Suppression", "moderate"},
+        "Moderate Suppression", "moderate"},
     {webrtc::EchoCancellation::kHighSuppression, "high Suppression", "high"},
     {0, NULL, NULL}
   };
@@ -141,19 +148,26 @@ gst_webrtc_echo_suppression_level_get_type (void)
   return suppression_level_type;
 }
 
-typedef webrtc::NoiseSuppression::Level GstWebrtcNoiseSuppressionLevel;
+typedef
+    webrtc::NoiseSuppression::Level
+    GstWebrtcNoiseSuppressionLevel;
 #define GST_TYPE_WEBRTC_NOISE_SUPPRESSION_LEVEL \
     (gst_webrtc_noise_suppression_level_get_type ())
-static GType
+static
+    GType
 gst_webrtc_noise_suppression_level_get_type (void)
 {
-  static GType suppression_level_type = 0;
-  static const GEnumValue level_types[] = {
+  static
+      GType
+      suppression_level_type = 0;
+  static const
+      GEnumValue
+      level_types[] = {
     {webrtc::NoiseSuppression::kLow, "Low Suppression", "low"},
     {webrtc::NoiseSuppression::kModerate, "Moderate Suppression", "moderate"},
     {webrtc::NoiseSuppression::kHigh, "High Suppression", "high"},
     {webrtc::NoiseSuppression::kVeryHigh, "Very High Suppression",
-      "very-high"},
+        "very-high"},
     {0, NULL, NULL}
   };
 
@@ -164,15 +178,23 @@ gst_webrtc_noise_suppression_level_get_type (void)
   return suppression_level_type;
 }
 
-typedef webrtc::GainControl::Mode GstWebrtcGainControlMode;
+typedef
+    webrtc::GainControl::Mode
+    GstWebrtcGainControlMode;
 #define GST_TYPE_WEBRTC_GAIN_CONTROL_MODE \
     (gst_webrtc_gain_control_mode_get_type ())
-static GType
+static
+    GType
 gst_webrtc_gain_control_mode_get_type (void)
 {
-  static GType gain_control_mode_type = 0;
-  static const GEnumValue mode_types[] = {
-    {webrtc::GainControl::kAdaptiveDigital, "Adaptive Digital", "adaptive-digital"},
+  static
+      GType
+      gain_control_mode_type = 0;
+  static const
+      GEnumValue
+      mode_types[] = {
+    {webrtc::GainControl::kAdaptiveDigital, "Adaptive Digital",
+        "adaptive-digital"},
     {webrtc::GainControl::kFixedDigital, "Fixed Digital", "fixed-digital"},
     {0, NULL, NULL}
   };
@@ -184,24 +206,34 @@ gst_webrtc_gain_control_mode_get_type (void)
   return gain_control_mode_type;
 }
 
-typedef webrtc::VoiceDetection::Likelihood GstWebrtcVoiceDetectionLikelihood;
+typedef
+    webrtc::VoiceDetection::Likelihood
+    GstWebrtcVoiceDetectionLikelihood;
 #define GST_TYPE_WEBRTC_VOICE_DETECTION_LIKELIHOOD \
     (gst_webrtc_voice_detection_likelihood_get_type ())
-static GType
+static
+    GType
 gst_webrtc_voice_detection_likelihood_get_type (void)
 {
-  static GType likelihood_type = 0;
-  static const GEnumValue likelihood_types[] = {
-    {webrtc::VoiceDetection::kVeryLowLikelihood, "Very Low Likelihood", "very-low"},
+  static
+      GType
+      likelihood_type = 0;
+  static const
+      GEnumValue
+      likelihood_types[] = {
+    {webrtc::VoiceDetection::kVeryLowLikelihood, "Very Low Likelihood",
+        "very-low"},
     {webrtc::VoiceDetection::kLowLikelihood, "Low Likelihood", "low"},
-    {webrtc::VoiceDetection::kModerateLikelihood, "Moderate Likelihood", "moderate"},
+    {webrtc::VoiceDetection::kModerateLikelihood, "Moderate Likelihood",
+        "moderate"},
     {webrtc::VoiceDetection::kHighLikelihood, "High Likelihood", "high"},
     {0, NULL, NULL}
   };
 
   if (!likelihood_type) {
     likelihood_type =
-        g_enum_register_static ("GstWebrtcVoiceDetectionLikelihood", likelihood_types);
+        g_enum_register_static ("GstWebrtcVoiceDetectionLikelihood",
+        likelihood_types);
   }
   return likelihood_type;
 }
@@ -236,42 +268,70 @@ enum
  */
 struct _GstWebrtcDsp
 {
-  GstAudioFilter element;
+  GstAudioFilter
+      element;
 
   /* Protected by the object lock */
-  GstAudioInfo info;
-  gboolean interleaved;
-  guint period_size;
-  guint period_samples;
-  gboolean stream_has_voice;
+  GstAudioInfo
+      info;
+  gboolean
+      interleaved;
+  guint
+      period_size;
+  guint
+      period_samples;
+  gboolean
+      stream_has_voice;
 
   /* Protected by the stream lock */
-  GstAdapter *adapter;
-  GstPlanarAudioAdapter *padapter;
-  webrtc::AudioProcessing * apm;
+  GstAdapter *
+      adapter;
+  GstPlanarAudioAdapter *
+      padapter;
+  webrtc::AudioProcessing *
+      apm;
 
   /* Protected by the object lock */
-  gchar *probe_name;
-  GstWebrtcEchoProbe *probe;
+  gchar *
+      probe_name;
+  GstWebrtcEchoProbe *
+      probe;
 
   /* Properties */
-  gboolean high_pass_filter;
-  gboolean echo_cancel;
-  webrtc::EchoCancellation::SuppressionLevel echo_suppression_level;
-  gboolean noise_suppression;
-  webrtc::NoiseSuppression::Level noise_suppression_level;
-  gboolean gain_control;
-  gboolean experimental_agc;
-  gboolean extended_filter;
-  gboolean delay_agnostic;
-  gint target_level_dbfs;
-  gint compression_gain_db;
-  gint startup_min_volume;
-  gboolean limiter;
-  webrtc::GainControl::Mode gain_control_mode;
-  gboolean voice_detection;
-  gint voice_detection_frame_size_ms;
-  webrtc::VoiceDetection::Likelihood voice_detection_likelihood;
+  gboolean
+      high_pass_filter;
+  gboolean
+      echo_cancel;
+  webrtc::EchoCancellation::SuppressionLevel
+      echo_suppression_level;
+  gboolean
+      noise_suppression;
+  webrtc::NoiseSuppression::Level
+      noise_suppression_level;
+  gboolean
+      gain_control;
+  gboolean
+      experimental_agc;
+  gboolean
+      extended_filter;
+  gboolean
+      delay_agnostic;
+  gint
+      target_level_dbfs;
+  gint
+      compression_gain_db;
+  gint
+      startup_min_volume;
+  gboolean
+      limiter;
+  webrtc::GainControl::Mode
+      gain_control_mode;
+  gboolean
+      voice_detection;
+  gint
+      voice_detection_frame_size_ms;
+  webrtc::VoiceDetection::Likelihood
+      voice_detection_likelihood;
 };
 
 G_DEFINE_TYPE (GstWebrtcDsp, gst_webrtc_dsp, GST_TYPE_AUDIO_FILTER);
@@ -279,7 +339,8 @@ G_DEFINE_TYPE (GstWebrtcDsp, gst_webrtc_dsp, GST_TYPE_AUDIO_FILTER);
 static const gchar *
 webrtc_error_to_string (gint err)
 {
-  const gchar *str = "unkown error";
+  const gchar *
+      str = "unkown error";
 
   switch (err) {
     case webrtc::AudioProcessing::kNoError:
@@ -331,7 +392,8 @@ webrtc_error_to_string (gint err)
 static GstBuffer *
 gst_webrtc_dsp_take_buffer (GstWebrtcDsp * self)
 {
-  GstBuffer *buffer;
+  GstBuffer *
+      buffer;
   GstClockTime timestamp;
   guint64 distance;
   gboolean at_discont;
@@ -343,7 +405,8 @@ gst_webrtc_dsp_take_buffer (GstWebrtcDsp * self)
     timestamp = gst_planar_audio_adapter_prev_pts (self->padapter, &distance);
   }
 
-  timestamp += gst_util_uint64_scale_int (distance, GST_SECOND, self->info.rate);
+  timestamp +=
+      gst_util_uint64_scale_int (distance, GST_SECOND, self->info.rate);
 
   if (self->interleaved) {
     buffer = gst_adapter_take_buffer (self->adapter, self->period_size);
@@ -367,14 +430,17 @@ gst_webrtc_dsp_take_buffer (GstWebrtcDsp * self)
   return buffer;
 }
 
-static GstFlowReturn
+static
+    GstFlowReturn
 gst_webrtc_dsp_analyze_reverse_stream (GstWebrtcDsp * self,
     GstClockTime rec_time)
 {
-  GstWebrtcEchoProbe *probe = NULL;
+  GstWebrtcEchoProbe *
+      probe = NULL;
   webrtc::AudioProcessing * apm;
   webrtc::AudioFrame frame;
-  GstBuffer *buf = NULL;
+  GstBuffer *
+      buf = NULL;
   GstFlowReturn ret = GST_FLOW_OK;
   gint err, delay;
 
@@ -393,7 +459,8 @@ gst_webrtc_dsp_analyze_reverse_stream (GstWebrtcDsp * self,
     rec_time = GST_CLOCK_TIME_NONE;
 
 again:
-  delay = gst_webrtc_echo_probe_read (probe, rec_time, (gpointer) &frame, &buf);
+  delay =
+      gst_webrtc_echo_probe_read (probe, rec_time, (gpointer) & frame, &buf);
   apm->set_stream_delay_ms (delay);
 
   if (delay < 0)
@@ -402,8 +469,8 @@ again:
   if (frame.sample_rate_hz_ != self->info.rate) {
     GST_ELEMENT_ERROR (self, STREAM, FORMAT,
         ("Echo Probe has rate %i , while the DSP is running at rate %i,"
-         " use a caps filter to ensure those are the same.",
-         frame.sample_rate_hz_, self->info.rate), (NULL));
+            " use a caps filter to ensure those are the same.",
+            frame.sample_rate_hz_, self->info.rate), (NULL));
     ret = GST_FLOW_ERROR;
     goto done;
   }
@@ -412,10 +479,11 @@ again:
     webrtc::StreamConfig config (frame.sample_rate_hz_, frame.num_channels_,
         false);
     GstAudioBuffer abuf;
-    float * const * data;
+    float *const *
+        data;
 
     gst_audio_buffer_map (&abuf, &self->info, buf, GST_MAP_READWRITE);
-    data = (float * const *) abuf.planes;
+    data = (float *const *) abuf.planes;
     if ((err = apm->ProcessReverseStream (data, config, config, data)) < 0)
       GST_WARNING_OBJECT (self, "Reverse stream analyses failed: %s.",
           webrtc_error_to_string (err));
@@ -428,7 +496,7 @@ again:
   }
 
   if (self->delay_agnostic)
-      goto again;
+    goto again;
 
 done:
   gst_object_unref (probe);
@@ -438,11 +506,13 @@ done:
 }
 
 static void
-gst_webrtc_vad_post_message (GstWebrtcDsp *self, GstClockTime timestamp,
+gst_webrtc_vad_post_message (GstWebrtcDsp * self, GstClockTime timestamp,
     gboolean stream_has_voice)
 {
-  GstBaseTransform *trans = GST_BASE_TRANSFORM_CAST (self);
-  GstStructure *s;
+  GstBaseTransform *
+      trans = GST_BASE_TRANSFORM_CAST (self);
+  GstStructure *
+      s;
   GstClockTime stream_time;
 
   stream_time = gst_segment_to_stream_time (&trans->segment, GST_FORMAT_TIME,
@@ -459,9 +529,9 @@ gst_webrtc_vad_post_message (GstWebrtcDsp *self, GstClockTime timestamp,
       gst_message_new_element (GST_OBJECT (self), s));
 }
 
-static GstFlowReturn
-gst_webrtc_dsp_process_stream (GstWebrtcDsp * self,
-    GstBuffer * buffer)
+static
+    GstFlowReturn
+gst_webrtc_dsp_process_stream (GstWebrtcDsp * self, GstBuffer * buffer)
 {
   GstAudioBuffer abuf;
   webrtc::AudioProcessing * apm = self->apm;
@@ -484,7 +554,8 @@ gst_webrtc_dsp_process_stream (GstWebrtcDsp * self,
     if (err >= 0)
       memcpy (abuf.planes[0], frame.data_, self->period_size);
   } else {
-    float * const * data = (float * const *) abuf.planes;
+    float *const *
+        data = (float *const *) abuf.planes;
     webrtc::StreamConfig config (self->info.rate, self->info.channels, false);
 
     err = apm->ProcessStream (data, config, config, data);
@@ -498,7 +569,8 @@ gst_webrtc_dsp_process_stream (GstWebrtcDsp * self,
       gboolean stream_has_voice = apm->voice_detection ()->stream_has_voice ();
 
       if (stream_has_voice != self->stream_has_voice)
-        gst_webrtc_vad_post_message (self, GST_BUFFER_PTS (buffer), stream_has_voice);
+        gst_webrtc_vad_post_message (self, GST_BUFFER_PTS (buffer),
+            stream_has_voice);
 
       self->stream_has_voice = stream_has_voice;
     }
@@ -509,19 +581,20 @@ gst_webrtc_dsp_process_stream (GstWebrtcDsp * self,
   return GST_FLOW_OK;
 }
 
-static GstFlowReturn
+static
+    GstFlowReturn
 gst_webrtc_dsp_submit_input_buffer (GstBaseTransform * btrans,
     gboolean is_discont, GstBuffer * buffer)
 {
-  GstWebrtcDsp *self = GST_WEBRTC_DSP (btrans);
+  GstWebrtcDsp *
+      self = GST_WEBRTC_DSP (btrans);
 
   buffer = gst_buffer_make_writable (buffer);
   GST_BUFFER_PTS (buffer) = gst_segment_to_running_time (&btrans->segment,
       GST_FORMAT_TIME, GST_BUFFER_PTS (buffer));
 
   if (is_discont) {
-    GST_DEBUG_OBJECT (self,
-        "Received discont, clearing adapter.");
+    GST_DEBUG_OBJECT (self, "Received discont, clearing adapter.");
     if (self->interleaved)
       gst_adapter_clear (self->adapter);
     else
@@ -536,10 +609,12 @@ gst_webrtc_dsp_submit_input_buffer (GstBaseTransform * btrans,
   return GST_FLOW_OK;
 }
 
-static GstFlowReturn
+static
+    GstFlowReturn
 gst_webrtc_dsp_generate_output (GstBaseTransform * btrans, GstBuffer ** outbuf)
 {
-  GstWebrtcDsp *self = GST_WEBRTC_DSP (btrans);
+  GstWebrtcDsp *
+      self = GST_WEBRTC_DSP (btrans);
   GstFlowReturn ret;
   gboolean not_enough;
 
@@ -563,10 +638,12 @@ gst_webrtc_dsp_generate_output (GstBaseTransform * btrans, GstBuffer ** outbuf)
   return ret;
 }
 
-static gboolean
+static
+    gboolean
 gst_webrtc_dsp_start (GstBaseTransform * btrans)
 {
-  GstWebrtcDsp *self = GST_WEBRTC_DSP (btrans);
+  GstWebrtcDsp *
+      self = GST_WEBRTC_DSP (btrans);
   webrtc::Config config;
 
   GST_OBJECT_LOCK (self);
@@ -574,7 +651,8 @@ gst_webrtc_dsp_start (GstBaseTransform * btrans)
   config.Set < webrtc::ExtendedFilter >
       (new webrtc::ExtendedFilter (self->extended_filter));
   config.Set < webrtc::ExperimentalAgc >
-      (new webrtc::ExperimentalAgc (self->experimental_agc, self->startup_min_volume));
+      (new webrtc::ExperimentalAgc (self->experimental_agc,
+          self->startup_min_volume));
   config.Set < webrtc::DelayAgnostic >
       (new webrtc::DelayAgnostic (self->delay_agnostic));
 
@@ -598,10 +676,12 @@ gst_webrtc_dsp_start (GstBaseTransform * btrans)
   return TRUE;
 }
 
-static gboolean
+static
+    gboolean
 gst_webrtc_dsp_setup (GstAudioFilter * filter, const GstAudioInfo * info)
 {
-  GstWebrtcDsp *self = GST_WEBRTC_DSP (filter);
+  GstWebrtcDsp *
+      self = GST_WEBRTC_DSP (filter);
   webrtc::AudioProcessing * apm;
   webrtc::ProcessingConfig pconfig;
   GstAudioInfo probe_info = *info;
@@ -679,7 +759,8 @@ gst_webrtc_dsp_setup (GstAudioFilter * filter, const GstAudioInfo * info)
   }
 
   if (self->gain_control) {
-    GEnumClass *mode_class = (GEnumClass *)
+    GEnumClass *
+        mode_class = (GEnumClass *)
         g_type_class_ref (GST_TYPE_WEBRTC_GAIN_CONTROL_MODE);
 
     GST_DEBUG_OBJECT (self, "Enabling Digital Gain Control, target level "
@@ -698,20 +779,21 @@ gst_webrtc_dsp_setup (GstAudioFilter * filter, const GstAudioInfo * info)
   }
 
   if (self->voice_detection) {
-    GEnumClass *likelihood_class = (GEnumClass *)
+    GEnumClass *
+        likelihood_class = (GEnumClass *)
         g_type_class_ref (GST_TYPE_WEBRTC_VOICE_DETECTION_LIKELIHOOD);
     GST_DEBUG_OBJECT (self, "Enabling Voice Activity Detection, frame size "
-      "%d milliseconds, likelihood: %s", self->voice_detection_frame_size_ms,
-      g_enum_get_value (likelihood_class,
-          self->voice_detection_likelihood)->value_name);
+        "%d milliseconds, likelihood: %s", self->voice_detection_frame_size_ms,
+        g_enum_get_value (likelihood_class,
+            self->voice_detection_likelihood)->value_name);
     g_type_class_unref (likelihood_class);
 
     self->stream_has_voice = FALSE;
 
     apm->voice_detection ()->Enable (true);
     apm->voice_detection ()->set_likelihood (self->voice_detection_likelihood);
-    apm->voice_detection ()->set_frame_size_ms (
-        self->voice_detection_frame_size_ms);
+    apm->voice_detection ()->
+        set_frame_size_ms (self->voice_detection_frame_size_ms);
   }
 
   GST_OBJECT_UNLOCK (self);
@@ -744,10 +826,12 @@ initialize_failed:
   return FALSE;
 }
 
-static gboolean
+static
+    gboolean
 gst_webrtc_dsp_stop (GstBaseTransform * btrans)
 {
-  GstWebrtcDsp *self = GST_WEBRTC_DSP (btrans);
+  GstWebrtcDsp *
+      self = GST_WEBRTC_DSP (btrans);
 
   GST_OBJECT_LOCK (self);
 
@@ -771,7 +855,8 @@ static void
 gst_webrtc_dsp_set_property (GObject * object,
     guint prop_id, const GValue * value, GParamSpec * pspec)
 {
-  GstWebrtcDsp *self = GST_WEBRTC_DSP (object);
+  GstWebrtcDsp *
+      self = GST_WEBRTC_DSP (object);
 
   GST_OBJECT_LOCK (self);
   switch (prop_id) {
@@ -845,7 +930,8 @@ static void
 gst_webrtc_dsp_get_property (GObject * object,
     guint prop_id, GValue * value, GParamSpec * pspec)
 {
-  GstWebrtcDsp *self = GST_WEBRTC_DSP (object);
+  GstWebrtcDsp *
+      self = GST_WEBRTC_DSP (object);
 
   GST_OBJECT_LOCK (self);
   switch (prop_id) {
@@ -914,7 +1000,8 @@ gst_webrtc_dsp_get_property (GObject * object,
 static void
 gst_webrtc_dsp_finalize (GObject * object)
 {
-  GstWebrtcDsp *self = GST_WEBRTC_DSP (object);
+  GstWebrtcDsp *
+      self = GST_WEBRTC_DSP (object);
 
   gst_object_unref (self->adapter);
   gst_object_unref (self->padapter);
@@ -934,10 +1021,14 @@ gst_webrtc_dsp_init (GstWebrtcDsp * self)
 static void
 gst_webrtc_dsp_class_init (GstWebrtcDspClass * klass)
 {
-  GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
-  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
-  GstBaseTransformClass *btrans_class = GST_BASE_TRANSFORM_CLASS (klass);
-  GstAudioFilterClass *audiofilter_class = GST_AUDIO_FILTER_CLASS (klass);
+  GObjectClass *
+      gobject_class = G_OBJECT_CLASS (klass);
+  GstElementClass *
+      element_class = GST_ELEMENT_CLASS (klass);
+  GstBaseTransformClass *
+      btrans_class = GST_BASE_TRANSFORM_CLASS (klass);
+  GstAudioFilterClass *
+      audiofilter_class = GST_AUDIO_FILTER_CLASS (klass);
 
   gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_webrtc_dsp_finalize);
   gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_webrtc_dsp_set_property);
@@ -1053,7 +1144,7 @@ gst_webrtc_dsp_class_init (GstWebrtcDspClass * klass)
       PROP_COMPRESSION_GAIN_DB,
       g_param_spec_int ("compression-gain-db", "Compression Gain dB",
           "Sets the maximum |gain| the digital compression stage may apply, "
-					"in dB.",
+          "in dB.",
           0, 90, DEFAULT_COMPRESSION_GAIN_DB, (GParamFlags) (G_PARAM_READWRITE |
               G_PARAM_STATIC_STRINGS | G_PARAM_CONSTRUCT)));
 
@@ -1063,8 +1154,9 @@ gst_webrtc_dsp_class_init (GstWebrtcDspClass * klass)
           "At startup the experimental AGC moves the microphone volume up to "
           "|startup_min_volume| if the current microphone volume is set too "
           "low. No effect if experimental-agc isn't enabled.",
-          12, 255, DEFAULT_STARTUP_MIN_VOLUME, (GParamFlags) (G_PARAM_READWRITE |
-              G_PARAM_STATIC_STRINGS | G_PARAM_CONSTRUCT)));
+          12, 255, DEFAULT_STARTUP_MIN_VOLUME,
+          (GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
+              G_PARAM_CONSTRUCT)));
 
   g_object_class_install_property (gobject_class,
       PROP_LIMITER,
@@ -1115,7 +1207,8 @@ gst_webrtc_dsp_class_init (GstWebrtcDspClass * klass)
 
 }
 
-static gboolean
+static
+    gboolean
 plugin_init (GstPlugin * plugin)
 {
   GST_DEBUG_CATEGORY_INIT

ext/webrtcdsp/gstwebrtcechoprobe.cpp

@@ -42,7 +42,7 @@ GST_DEBUG_CATEGORY_EXTERN (webrtc_dsp_debug);
 #define MAX_ADAPTER_SIZE (1*1024*1024)
 
 static GstStaticPadTemplate gst_webrtc_echo_probe_sink_template =
-GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_STATIC_PAD_TEMPLATE ("sink",
     GST_PAD_SINK,
     GST_PAD_ALWAYS,
     GST_STATIC_CAPS ("audio/x-raw, "
@@ -58,7 +58,7 @@ GST_STATIC_PAD_TEMPLATE ("sink",
     );
 
 static GstStaticPadTemplate gst_webrtc_echo_probe_src_template =
-GST_STATIC_PAD_TEMPLATE ("src",
+    GST_STATIC_PAD_TEMPLATE ("src",
     GST_PAD_SRC,
     GST_PAD_ALWAYS,
     GST_STATIC_CAPS ("audio/x-raw, "
@@ -426,13 +426,12 @@ copy:
             NULL);
       } else {
         ret = gst_planar_audio_adapter_take_buffer (self->padapter, size,
-          GST_MAP_READWRITE);
+            GST_MAP_READWRITE);
       }
     } else {
       ret = gst_buffer_new_allocate (NULL, self->period_size, NULL);
       gst_buffer_memset (ret, 0, 0, self->period_size);
-      gst_buffer_add_audio_meta (ret, &self->info, self->period_samples,
-          NULL);
+      gst_buffer_add_audio_meta (ret, &self->info, self->period_samples, NULL);
     }
 
     *buf = ret;