options: move audio_output_channels, audio_output_format to struct

7 files changed, 22 insertions, 15 deletions

diff --git a/libmpcodecs/ad_dmo.c b/libmpcodecs/ad_dmo.c
index 1501c21078..714651a7e4 100644
--- a/libmpcodecs/ad_dmo.c
+++ b/libmpcodecs/ad_dmo.c
@@ -21,6 +21,7 @@
 #include <unistd.h>
 
 #include "config.h"
+#include "options.h"
 #include "mp_msg.h"
 
 #include "ad_internal.h"
@@ -46,9 +47,10 @@ static int init(sh_audio_t *sh)
 
 static int preinit(sh_audio_t *sh_audio)
 {
+  struct MPOpts *opts = sh_audio->opts;
   DMO_AudioDecoder* ds_adec;
-  int chans=(audio_output_channels==sh_audio->wf->nChannels) ?
-      audio_output_channels : (sh_audio->wf->nChannels>=2 ? 2 : 1);
+  int chans=(opts->audio_output_channels==sh_audio->wf->nChannels) ?
+      opts->audio_output_channels : (sh_audio->wf->nChannels>=2 ? 2 : 1);
   if(!(ds_adec=DMO_AudioDecoder_Open(sh_audio->codec->dll,&sh_audio->codec->guid,sh_audio->wf,chans)))
   {
     mp_tmsg(MSGT_DECAUDIO,MSGL_ERR,"ERROR: Could not open required DirectShow codec %s.\n",sh_audio->codec->dll);
diff --git a/libmpcodecs/ad_faad.c b/libmpcodecs/ad_faad.c
index 2dc6c1f580..96abaa5a6d 100644
--- a/libmpcodecs/ad_faad.c
+++ b/libmpcodecs/ad_faad.c
@@ -25,6 +25,7 @@
 #include <unistd.h>
 
 #include "config.h"
+#include "options.h"
 #include "ad_internal.h"
 #include "libaf/reorder_ch.h"
 
@@ -83,6 +84,7 @@ static int aac_probe(unsigned char *buffer, int len)
 
 static int init(sh_audio_t *sh)
 {
+  struct MPOpts *opts = sh->opts;
   unsigned long faac_samplerate;
   unsigned char faac_channels;
   int faac_init, pos = 0;
@@ -105,7 +107,8 @@ static int init(sh_audio_t *sh)
     /* XXX: FAAD support FLOAT output, how do we handle
       * that (FAAD_FMT_FLOAT)? ::atmos
       */
-    if (audio_output_channels <= 2) faac_conf->downMatrix = 1;
+    if (opts->audio_output_channels <= 2)
+        faac_conf->downMatrix = 1;
       switch(sh->samplesize){
 	case 1: // 8Bit
 	  mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: 8Bit samplesize not supported by FAAD, assuming 16Bit!\n");
@@ -177,7 +180,7 @@ static int init(sh_audio_t *sh)
 
   } else { // We have ES DS in codecdata
     faacDecConfigurationPtr faac_conf = faacDecGetCurrentConfiguration(faac_hdec);
-    if (audio_output_channels <= 2) {
+    if (opts->audio_output_channels <= 2) {
         faac_conf->downMatrix = 1;
         faacDecSetConfiguration(faac_hdec, faac_conf);
     }
@@ -199,7 +202,8 @@ static int init(sh_audio_t *sh)
     mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Negotiated samplerate: %ldHz  channels: %d\n", faac_samplerate, faac_channels);
     // 8 channels is aac channel order #7.
     sh->channels = faac_channels == 7 ? 8 : faac_channels;
-    if (audio_output_channels <= 2) sh->channels = faac_channels > 1 ? 2 : 1;
+    if (opts->audio_output_channels <= 2)
+        sh->channels = faac_channels > 1 ? 2 : 1;
     sh->samplerate = faac_samplerate;
     sh->samplesize=2;
     //sh->o_bps = sh->samplesize*faac_channels*faac_samplerate;
diff --git a/libmpcodecs/ad_ffmpeg.c b/libmpcodecs/ad_ffmpeg.c
index 7c398af5b0..7d945794b8 100644
--- a/libmpcodecs/ad_ffmpeg.c
+++ b/libmpcodecs/ad_ffmpeg.c
@@ -120,7 +120,7 @@ static int init(sh_audio_t *sh_audio)
 	lavc_context->block_align = sh_audio->wf->nBlockAlign;
 	lavc_context->bits_per_coded_sample = sh_audio->wf->wBitsPerSample;
     }
-    lavc_context->request_channels = audio_output_channels;
+    lavc_context->request_channels = opts->audio_output_channels;
     lavc_context->codec_tag = sh_audio->format; //FOURCC
     lavc_context->codec_type = CODEC_TYPE_AUDIO;
     lavc_context->codec_id = lavc_codec->id; // not sure if required, imho not --A'rpi
diff --git a/libmpcodecs/ad_internal.h b/libmpcodecs/ad_internal.h
index 2a5dbc43ea..a04d3f6165 100644
--- a/libmpcodecs/ad_internal.h
+++ b/libmpcodecs/ad_internal.h
@@ -28,7 +28,6 @@
 
 #include "ad.h"
 
-extern int audio_output_channels;
 static int init(sh_audio_t *sh);
 static int preinit(sh_audio_t *sh);
 static void uninit(sh_audio_t *sh);
diff --git a/libmpcodecs/ad_liba52.c b/libmpcodecs/ad_liba52.c
index 0cc282776d..abd6a7ac54 100644
--- a/libmpcodecs/ad_liba52.c
+++ b/libmpcodecs/ad_liba52.c
@@ -141,10 +141,11 @@ static sample_t dynrng_call (sample_t c, void *data)
 
 static int preinit(sh_audio_t *sh)
 {
+  struct MPOpts *opts = sh->opts;
   /* Dolby AC3 audio: */
   /* however many channels, 2 bytes in a word, 256 samples in a block, 6 blocks in a frame */
   if (sh->samplesize < 4) sh->samplesize = 4;
-  sh->audio_out_minsize=audio_output_channels*sh->samplesize*256*6;
+  sh->audio_out_minsize=opts->audio_output_channels*sh->samplesize*256*6;
   sh->audio_in_minsize=3840;
   a52_level = 1.0;
   return 1;
@@ -219,7 +220,7 @@ static int init(sh_audio_t *sh_audio)
 
   /* 'a52 cannot upmix' hotfix:*/
   a52_printinfo(sh_audio);
-  sh_audio->channels=audio_output_channels;
+  sh_audio->channels=opts->audio_output_channels;
 while(sh_audio->channels>0){
   switch(sh_audio->channels){
 	    case 1: a52_flags=A52_MONO; break;
diff --git a/libmpcodecs/ad_libdca.c b/libmpcodecs/ad_libdca.c
index c167c92490..a550cbb8d7 100644
--- a/libmpcodecs/ad_libdca.c
+++ b/libmpcodecs/ad_libdca.c
@@ -25,8 +25,9 @@
 #include <stdlib.h>
 #include <unistd.h>
 #include <assert.h>
-#include "config.h"
 
+#include "config.h"
+#include "options.h"
 #include "mp_msg.h"
 #include "ad_internal.h"
 
@@ -280,8 +281,10 @@ end:
 
 static int preinit(sh_audio_t *sh)
 {
+    struct MPOpts *opts = sh->opts;
+
     /* 256 = samples per block, 16 = max number of blocks */
-    sh->audio_out_minsize = audio_output_channels * sizeof(int16_t) * 256 * 16;
+    sh->audio_out_minsize = opts->audio_output_channels * sizeof(int16_t) * 256 * 16;
     sh->audio_in_minsize = DTSBUFFER_SIZE;
     sh->samplesize=2;
 
@@ -308,8 +311,8 @@ static int init(sh_audio_t *sh)
     }
     channels_info(flags);
 
-    assert(audio_output_channels >= 1 && audio_output_channels <= 6);
-    sh->channels = audio_output_channels;
+    assert(opts->audio_output_channels >= 1 && opts->audio_output_channels <= 6);
+    sh->channels = opts->audio_output_channels;
 
     decoded_bytes = decode_audio(sh, sh->a_buffer, 1, sh->a_buffer_size);
     if(decoded_bytes > 0)
diff --git a/libmpcodecs/dec_audio.c b/libmpcodecs/dec_audio.c
index ade2429dc0..572e06a8a1 100644
--- a/libmpcodecs/dec_audio.c
+++ b/libmpcodecs/dec_audio.c
@@ -44,8 +44,6 @@
 int fakemono = 0;
 #endif
 
-/* used for ac3surround decoder - set using -channels option */
-int audio_output_channels = 2;
 af_cfg_t af_cfg = { 1, NULL };	// Configuration for audio filters
 
 void afm_help(void)