changed OSS buffer scheme, to work with frames(like ALSA) and reduce ifdefs

git-svn-id: https://recordmydesktop.svn.sourceforge.net/svnroot/recordmydesktop/trunk@272 f606c939-3180-4ac9-a4b8-4b8779d57d0a

5 files changed, 15 insertions, 25 deletions

diff --git a/recordmydesktop/src/cache_audio.c b/recordmydesktop/src/cache_audio.c
index edcabb3..bff4df2 100644
--- a/recordmydesktop/src/cache_audio.c
+++ b/recordmydesktop/src/cache_audio.c
@@ -33,6 +33,8 @@ void *CacheSoundBuffer(ProgData *pdata){
 #ifdef HAVE_LIBASOUND
     int framesize=((snd_pcm_format_width(SND_PCM_FORMAT_S16_LE))/8)*
                   pdata->args.channels;
+#else
+    int framesize=pdata->args.channels<<1;//Always signed 16 bit data
 #endif
     while((pdata->running)){
         SndBuffer *buff;
@@ -58,15 +60,8 @@ void *CacheSoundBuffer(ProgData *pdata){
         //advance the list
         pdata->sound_buffer=pdata->sound_buffer->next;
         pthread_mutex_unlock(&pdata->sound_buffer_mutex);
-#ifdef HAVE_LIBASOUND
         fwrite(buff->data,1,pdata->periodsize*framesize,
                pdata->cache_data->afp);
-#else
-        fwrite(buff->data,1,pdata->args.buffsize,
-               pdata->cache_data->afp);
-#endif
-
-
         pdata->avd-=pdata->periodtime;
 
         free(buff->data);
diff --git a/recordmydesktop/src/capture_sound.c b/recordmydesktop/src/capture_sound.c
index 4314984..3b69764 100644
--- a/recordmydesktop/src/capture_sound.c
+++ b/recordmydesktop/src/capture_sound.c
@@ -94,7 +94,8 @@ void *CaptureSound(ProgData *pdata){
 #ifdef HAVE_LIBASOUND
         newbuf->data=(signed char *)malloc(frames*framesize);
 #else
-        newbuf->data=(signed char *)malloc(pdata->args.buffsize);
+        newbuf->data=(signed char *)malloc(((pdata->args.buffsize<<1)*
+                                            pdata->args.channels));
 #endif
         newbuf->next=NULL;
 
@@ -124,7 +125,8 @@ void *CaptureSound(ProgData *pdata){
         do{
             int temp_sret=read(pdata->sound_handle,
                                &newbuf->data[sret],
-                               pdata->args.buffsize-sret);
+                               ((pdata->args.buffsize<<1)*
+                                pdata->args.channels)-sret);
             if(temp_sret<0){
                 fprintf(stderr,"An error occured while reading from soundcard"
                                "%s\n"
@@ -133,7 +135,8 @@ void *CaptureSound(ProgData *pdata){
             }
             else
                 sret+=temp_sret;
-        }while(sret<pdata->args.buffsize);
+        }while(sret<((pdata->args.buffsize<<1)*
+                     pdata->args.channels));
 #endif
         //queue the new buffer
         pthread_mutex_lock(&pdata->sound_buffer_mutex);
diff --git a/recordmydesktop/src/encode_sound_buffer.c b/recordmydesktop/src/encode_sound_buffer.c
index 0e8fa2c..c8a8781 100644
--- a/recordmydesktop/src/encode_sound_buffer.c
+++ b/recordmydesktop/src/encode_sound_buffer.c
@@ -28,11 +28,8 @@
 #include <recordmydesktop.h>
 
 void *EncodeSoundBuffer(ProgData *pdata){
-#ifdef HAVE_LIBASOUND
     int sampread=pdata->periodsize;
-#else
-    int sampread=(pdata->args.buffsize>>1)/pdata->args.channels;
-#endif
+
     pdata->v_encoding_clean=0;
     while((pdata->running)){
         float **vorbis_buffer;
@@ -103,12 +100,8 @@ void *EncodeSoundBuffer(ProgData *pdata){
 void SyncEncodeSoundBuffer(ProgData *pdata,signed char *buff){
     float **vorbis_buffer;
     int count=0,i,j;
-#ifdef HAVE_LIBASOUND
     int sampread=(buff!=NULL)?pdata->periodsize:0;
-#else
-    int sampread=(buff!=NULL)?((pdata->args.buffsize>>1)/
-                               pdata->args.channels):0;
-#endif
+
     vorbis_buffer=vorbis_analysis_buffer(&pdata->enc_data->m_vo_dsp,sampread);
     for(i=0;i<sampread;i++){
         for(j=0;j<pdata->args.channels;j++){
diff --git a/recordmydesktop/src/initialize_data.c b/recordmydesktop/src/initialize_data.c
index 4df3a6f..d8f52c0 100644
--- a/recordmydesktop/src/initialize_data.c
+++ b/recordmydesktop/src/initialize_data.c
@@ -139,6 +139,8 @@ int InitializeData(ProgData *pdata,
                                     pdata->args.frequency);
         pdata->periodtime=(1000000*pdata->args.buffsize)/
                           ((pdata->args.channels<<1)*pdata->args.frequency);
+        //when using OSS periodsize serves as an alias of buffsize
+        pdata->periodsize=pdata->args.buffsize;
         if(pdata->sound_handle<0){
 #endif
             fprintf(stderr,"Error while opening/configuring soundcard %s\n"
diff --git a/recordmydesktop/src/load_cache.c b/recordmydesktop/src/load_cache.c
index 00ddf96..f75c375 100644
--- a/recordmydesktop/src/load_cache.c
+++ b/recordmydesktop/src/load_cache.c
@@ -135,10 +135,11 @@ void *LoadCache(ProgData *pdata){
 #ifdef HAVE_LIBASOUND
     int framesize=((snd_pcm_format_width(SND_PCM_FORMAT_S16_LE))/8)*
                   pdata->args.channels;//audio frame size
-    signed char *sound_data=(signed char *)malloc(pdata->periodsize*framesize);
 #else
-    signed char *sound_data=(signed char *)malloc(pdata->args.buffsize);
+    int framesize=pdata->args.channels<<1;//Always signed 16 bit data
 #endif
+    signed char *sound_data=(signed char *)malloc(pdata->periodsize*framesize);
+
     u_int32_t YBlocks[(yuv->y_width*yuv->y_height)/Y_UNIT_BYTES],
               UBlocks[(yuv->uv_width*yuv->uv_height)/UV_UNIT_BYTES],
               VBlocks[(yuv->uv_width*yuv->uv_height)/UV_UNIT_BYTES];
@@ -254,11 +255,7 @@ void *LoadCache(ProgData *pdata){
         //audio load and encoding
         else{
             if(!audio_end){
-#ifdef HAVE_LIBASOUND
                 int nbytes=fread(sound_data,1,pdata->periodsize*framesize,afp);
-#else
-                int nbytes=fread(sound_data,1,pdata->args.buffsize,afp);
-#endif
                 if(nbytes<=0)
                     audio_end=1;
                 else