1 files changed, 153 insertions, 156 deletions

diff --git a/plugins/audio_webrtc/audio_webrtc_consumer.cxx b/plugins/audio_webrtc/audio_webrtc_consumer.cxx
index e55097b..8762a2d 100755
--- a/plugins/audio_webrtc/audio_webrtc_consumer.cxx
+++ b/plugins/audio_webrtc/audio_webrtc_consumer.cxx
@@ -1,17 +1,17 @@
 /* Copyright (C) 2012 Doubango Telecom <http://www.doubango.org>
-*	
+*
 * This file is part of Open Source Doubango Framework.
 *
 * DOUBANGO is free software: you can redistribute it and/or modify
 * it under the terms of the GNU General Public License as published by
 * the Free Software Foundation, either version 3 of the License, or
 * (at your option) any later version.
-*	
+*
 * DOUBANGO 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 General Public License
 * along with DOUBANGO.
 */
@@ -24,153 +24,152 @@
 #include "tsk_memory.h"
 #include "tsk_debug.h"
 
-typedef struct audio_consumer_webrtc_s
-{
-	TDAV_DECLARE_CONSUMER_AUDIO;
-	audio_webrtc_instance_handle_t* audioInstHandle;
-	struct{
-		void* ptr;
-		bool isFull;
-		int size;
-		int index;
-	} buffer;
+typedef struct audio_consumer_webrtc_s {
+    TDAV_DECLARE_CONSUMER_AUDIO;
+    audio_webrtc_instance_handle_t* audioInstHandle;
+    struct {
+        void* ptr;
+        bool isFull;
+        int size;
+        int index;
+    } buffer;
 }
 audio_consumer_webrtc_t;
 
 int audio_consumer_webrtc_get_data_10ms(const audio_consumer_webrtc_t* _self, void* audioSamples, int nSamples, int nBytesPerSample, int nChannels, int samplesPerSec, uint32_t &nSamplesOut)
 {
-	nSamplesOut = 0;
-	if(!_self || !audioSamples || !nSamples){
-		DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Invalid parameter");
-		return -1;
-	}
-	if((nSamples != (samplesPerSec / 100))){
-		DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Not producing 10ms samples (nSamples=%d, samplesPerSec=%d)", nSamples, samplesPerSec);
-		return -2;
-	}
-	if((nBytesPerSample != (TMEDIA_CONSUMER(_self)->audio.bits_per_sample >> 3))){
-		DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("%d not valid bytes/samples", nBytesPerSample);
-		return -3;
-	}
-	if((nChannels != TMEDIA_CONSUMER(_self)->audio.out.channels)){
-		DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("%d not the expected number of channels", nChannels);
-		return -4;
-	}
-
-	audio_consumer_webrtc_t* self = const_cast<audio_consumer_webrtc_t*>(_self);
-
-	if(self->buffer.index == self->buffer.size){
-		tdav_consumer_audio_tick(TDAV_CONSUMER_AUDIO(self));
-		self->buffer.index = 0;
-		if((tdav_consumer_audio_get(TDAV_CONSUMER_AUDIO(self), self->buffer.ptr, self->buffer.size)) != self->buffer.size){
-			nSamplesOut = 0;
-			return 0;
-		}
-	}
-	
-	int nSamplesInBits = (nSamples * nBytesPerSample);
-	if(_self->buffer.index + nSamplesInBits <= _self->buffer.size){
-		memcpy(audioSamples, (((uint8_t*)self->buffer.ptr) + self->buffer.index), nSamplesInBits);
-	}
-	self->buffer.index += nSamplesInBits;
-	TSK_CLAMP(0, self->buffer.index, self->buffer.size);
-	nSamplesOut = nSamples;
-	
-	return 0;
+    nSamplesOut = 0;
+    if(!_self || !audioSamples || !nSamples) {
+        DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Invalid parameter");
+        return -1;
+    }
+    if((nSamples != (samplesPerSec / 100))) {
+        DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Not producing 10ms samples (nSamples=%d, samplesPerSec=%d)", nSamples, samplesPerSec);
+        return -2;
+    }
+    if((nBytesPerSample != (TMEDIA_CONSUMER(_self)->audio.bits_per_sample >> 3))) {
+        DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("%d not valid bytes/samples", nBytesPerSample);
+        return -3;
+    }
+    if((nChannels != TMEDIA_CONSUMER(_self)->audio.out.channels)) {
+        DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("%d not the expected number of channels", nChannels);
+        return -4;
+    }
+
+    audio_consumer_webrtc_t* self = const_cast<audio_consumer_webrtc_t*>(_self);
+
+    if(self->buffer.index == self->buffer.size) {
+        tdav_consumer_audio_tick(TDAV_CONSUMER_AUDIO(self));
+        self->buffer.index = 0;
+        if((tdav_consumer_audio_get(TDAV_CONSUMER_AUDIO(self), self->buffer.ptr, self->buffer.size)) != self->buffer.size) {
+            nSamplesOut = 0;
+            return 0;
+        }
+    }
+
+    int nSamplesInBits = (nSamples * nBytesPerSample);
+    if(_self->buffer.index + nSamplesInBits <= _self->buffer.size) {
+        memcpy(audioSamples, (((uint8_t*)self->buffer.ptr) + self->buffer.index), nSamplesInBits);
+    }
+    self->buffer.index += nSamplesInBits;
+    TSK_CLAMP(0, self->buffer.index, self->buffer.size);
+    nSamplesOut = nSamples;
+
+    return 0;
 }
 
 
 /* ============ Media Consumer Interface ================= */
 static int audio_consumer_webrtc_set(tmedia_consumer_t* self, const tmedia_param_t* param)
 {
-	audio_consumer_webrtc_t* webrtc = (audio_consumer_webrtc_t*)self;
-	int ret = tdav_consumer_audio_set(TDAV_CONSUMER_AUDIO(self), param);
+    audio_consumer_webrtc_t* webrtc = (audio_consumer_webrtc_t*)self;
+    int ret = tdav_consumer_audio_set(TDAV_CONSUMER_AUDIO(self), param);
 
-	if(ret == 0){
-		if(tsk_striequals(param->key, "volume")){
-			
-		}
-	}
+    if(ret == 0) {
+        if(tsk_striequals(param->key, "volume")) {
 
-	return ret;
+        }
+    }
+
+    return ret;
 }
 
 static int audio_consumer_webrtc_prepare(tmedia_consumer_t* _self, const tmedia_codec_t* codec)
 {
-	audio_consumer_webrtc_t* self = (audio_consumer_webrtc_t*)_self;
-	if(!self){
-		DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Invalid parameter");
-		return -1;
-	}
-
-	// create audio instance
-	if(!(self->audioInstHandle = audio_webrtc_instance_create(TMEDIA_CONSUMER(self)->session_id))){
-		DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Failed to create audio instance handle");
-		return -1;
-	}
-
-	// initialize input parameters from the codec information
-	TMEDIA_CONSUMER(self)->audio.ptime = codec->plugin->audio.ptime;
-	TMEDIA_CONSUMER(self)->audio.in.channels = codec->plugin->audio.channels;
-	TMEDIA_CONSUMER(self)->audio.in.rate = codec->plugin->rate;
-	
-	// prepare playout device and update output parameters
-	int ret = audio_webrtc_instance_prepare_consumer(self->audioInstHandle, &_self);
-
-	// now that the producer is prepared we can initialize internal buffer using device caps
-	if(ret == 0){
-		// allocate buffer
-		int xsize = ((TMEDIA_CONSUMER(self)->audio.ptime * TMEDIA_CONSUMER(self)->audio.out.rate) / 1000) * (TMEDIA_CONSUMER(self)->audio.bits_per_sample >> 3);
-		if(!(self->buffer.ptr = tsk_realloc(self->buffer.ptr, xsize))){
-			DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Failed to allocate buffer with size = %d", xsize);
-			self->buffer.size = 0;
-			return -1;
-		}
-		memset(self->buffer.ptr, 0, xsize);
-		self->buffer.size = xsize;
-		self->buffer.index = 0;
-		self->buffer.isFull = false;
-	}
-	return ret;
+    audio_consumer_webrtc_t* self = (audio_consumer_webrtc_t*)_self;
+    if(!self) {
+        DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Invalid parameter");
+        return -1;
+    }
+
+    // create audio instance
+    if(!(self->audioInstHandle = audio_webrtc_instance_create(TMEDIA_CONSUMER(self)->session_id))) {
+        DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Failed to create audio instance handle");
+        return -1;
+    }
+
+    // initialize input parameters from the codec information
+    TMEDIA_CONSUMER(self)->audio.ptime = codec->plugin->audio.ptime;
+    TMEDIA_CONSUMER(self)->audio.in.channels = codec->plugin->audio.channels;
+    TMEDIA_CONSUMER(self)->audio.in.rate = codec->plugin->rate;
+
+    // prepare playout device and update output parameters
+    int ret = audio_webrtc_instance_prepare_consumer(self->audioInstHandle, &_self);
+
+    // now that the producer is prepared we can initialize internal buffer using device caps
+    if(ret == 0) {
+        // allocate buffer
+        int xsize = ((TMEDIA_CONSUMER(self)->audio.ptime * TMEDIA_CONSUMER(self)->audio.out.rate) / 1000) * (TMEDIA_CONSUMER(self)->audio.bits_per_sample >> 3);
+        if(!(self->buffer.ptr = tsk_realloc(self->buffer.ptr, xsize))) {
+            DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Failed to allocate buffer with size = %d", xsize);
+            self->buffer.size = 0;
+            return -1;
+        }
+        memset(self->buffer.ptr, 0, xsize);
+        self->buffer.size = xsize;
+        self->buffer.index = 0;
+        self->buffer.isFull = false;
+    }
+    return ret;
 }
 
 static int audio_consumer_webrtc_start(tmedia_consumer_t* _self)
 {
-	audio_consumer_webrtc_t* self = (audio_consumer_webrtc_t*)_self;
-	if(!self){
-		DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Invalid parameter");
-		return -1;
-	}
+    audio_consumer_webrtc_t* self = (audio_consumer_webrtc_t*)_self;
+    if(!self) {
+        DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Invalid parameter");
+        return -1;
+    }
 
-	return audio_webrtc_instance_start_consumer(self->audioInstHandle);
+    return audio_webrtc_instance_start_consumer(self->audioInstHandle);
 }
 
 
 static int audio_consumer_webrtc_consume(tmedia_consumer_t* _self, const void* buffer, tsk_size_t size, const tsk_object_t* proto_hdr)
 {
-	audio_consumer_webrtc_t* self = (audio_consumer_webrtc_t*)_self;
-	if(!self || !buffer || !size){
-		DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("1Invalid parameter");
-		return -1;
-	}	
-	/* buffer is already decoded */
-	return tdav_consumer_audio_put(TDAV_CONSUMER_AUDIO(self), buffer, size, proto_hdr);
+    audio_consumer_webrtc_t* self = (audio_consumer_webrtc_t*)_self;
+    if(!self || !buffer || !size) {
+        DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("1Invalid parameter");
+        return -1;
+    }
+    /* buffer is already decoded */
+    return tdav_consumer_audio_put(TDAV_CONSUMER_AUDIO(self), buffer, size, proto_hdr);
 }
 
 static int audio_consumer_webrtc_pause(tmedia_consumer_t* self)
 {
-	return 0;
+    return 0;
 }
 
 static int audio_consumer_webrtc_stop(tmedia_consumer_t* _self)
 {
-	audio_consumer_webrtc_t* self = (audio_consumer_webrtc_t*)_self;
-	if(!self){
-		DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Invalid parameter");
-		return -1;
-	}
+    audio_consumer_webrtc_t* self = (audio_consumer_webrtc_t*)_self;
+    if(!self) {
+        DOUBANGO_AUDIO_WEBRTC_DEBUG_ERROR("Invalid parameter");
+        return -1;
+    }
 
-	return audio_webrtc_instance_stop_consumer(self->audioInstHandle);
+    return audio_webrtc_instance_stop_consumer(self->audioInstHandle);
 }
 
 
@@ -180,54 +179,52 @@ static int audio_consumer_webrtc_stop(tmedia_consumer_t* _self)
 /* constructor */
 static tsk_object_t* audio_consumer_webrtc_ctor(tsk_object_t *_self, va_list * app)
 {
-	audio_consumer_webrtc_t *self = (audio_consumer_webrtc_t *)_self;
-	if(self){
-		/* init base */
-		tdav_consumer_audio_init(TDAV_CONSUMER_AUDIO(self));
-		/* init self */
-		
-	}
-	return self;
+    audio_consumer_webrtc_t *self = (audio_consumer_webrtc_t *)_self;
+    if(self) {
+        /* init base */
+        tdav_consumer_audio_init(TDAV_CONSUMER_AUDIO(self));
+        /* init self */
+
+    }
+    return self;
 }
 /* destructor */
 static tsk_object_t* audio_consumer_webrtc_dtor(tsk_object_t *_self)
-{ 
-	audio_consumer_webrtc_t *self = (audio_consumer_webrtc_t *)_self;
-	if(self){
-		/* stop */
-		audio_consumer_webrtc_stop(TMEDIA_CONSUMER(self));
-		/* deinit self */
-		if(self->audioInstHandle){
-			audio_webrtc_instance_destroy(&self->audioInstHandle);
-		}
-		TSK_FREE(self->buffer.ptr);
-		/* deinit base */
-		tdav_consumer_audio_deinit(TDAV_CONSUMER_AUDIO(self));
-	}
-
-	return self;
+{
+    audio_consumer_webrtc_t *self = (audio_consumer_webrtc_t *)_self;
+    if(self) {
+        /* stop */
+        audio_consumer_webrtc_stop(TMEDIA_CONSUMER(self));
+        /* deinit self */
+        if(self->audioInstHandle) {
+            audio_webrtc_instance_destroy(&self->audioInstHandle);
+        }
+        TSK_FREE(self->buffer.ptr);
+        /* deinit base */
+        tdav_consumer_audio_deinit(TDAV_CONSUMER_AUDIO(self));
+    }
+
+    return self;
 }
 /* object definition */
-static const tsk_object_def_t audio_consumer_webrtc_def_s = 
-{
-	sizeof(audio_consumer_webrtc_t),
-	audio_consumer_webrtc_ctor, 
-	audio_consumer_webrtc_dtor,
-	tdav_consumer_audio_cmp, 
+static const tsk_object_def_t audio_consumer_webrtc_def_s = {
+    sizeof(audio_consumer_webrtc_t),
+    audio_consumer_webrtc_ctor,
+    audio_consumer_webrtc_dtor,
+    tdav_consumer_audio_cmp,
 };
 /* plugin definition*/
-static const tmedia_consumer_plugin_def_t audio_consumer_webrtc_plugin_def_s = 
-{
-	&audio_consumer_webrtc_def_s,
-	
-	tmedia_audio,
-	"WebRTC audio consumer",
-	
-	audio_consumer_webrtc_set,
-	audio_consumer_webrtc_prepare,
-	audio_consumer_webrtc_start,
-	audio_consumer_webrtc_consume,
-	audio_consumer_webrtc_pause,
-	audio_consumer_webrtc_stop
+static const tmedia_consumer_plugin_def_t audio_consumer_webrtc_plugin_def_s = {
+    &audio_consumer_webrtc_def_s,
+
+    tmedia_audio,
+    "WebRTC audio consumer",
+
+    audio_consumer_webrtc_set,
+    audio_consumer_webrtc_prepare,
+    audio_consumer_webrtc_start,
+    audio_consumer_webrtc_consume,
+    audio_consumer_webrtc_pause,
+    audio_consumer_webrtc_stop
 };
 const tmedia_consumer_plugin_def_t *audio_consumer_webrtc_plugin_def_t = &audio_consumer_webrtc_plugin_def_s;