options: move audio_output_channels, audio_output_format to struct

12 files changed, 30 insertions, 20 deletions

diff --git a/cfg-common.h b/cfg-common.h
index 1e9871f99b..0ae68d0531 100644
--- a/cfg-common.h
+++ b/cfg-common.h
@@ -37,7 +37,6 @@ extern char * codecs_file;
 extern int field_dominance;
 
 /* from dec_audio, currently used for ac3surround decoder only */
-extern int audio_output_channels;
 extern int fakemono;
 
 /* defined in network.c */
@@ -569,8 +568,8 @@ const m_option_t common_opts[] = {
     // force video/audio rate:
     {"fps", &force_fps, CONF_TYPE_DOUBLE, CONF_MIN, 0, 0, NULL},
     {"srate", &force_srate, CONF_TYPE_INT, CONF_RANGE, 1000, 8*48000, NULL},
-    {"channels", &audio_output_channels, CONF_TYPE_INT, CONF_RANGE, 1, 8, NULL},
-    {"format", &audio_output_format, CONF_TYPE_AFMT, 0, 0, 0, NULL},
+    OPT_INTRANGE("channels", audio_output_channels, 0, 1, 8),
+    OPT_AUDIOFORMAT("format", audio_output_format, 0),
     OPT_FLOATRANGE("speed", playback_speed, 0, 0.01, 100.0),
 
     // set a-v distance
diff --git a/defaultopts.c b/defaultopts.c
index 2cfeda063d..f263437d74 100644
--- a/defaultopts.c
+++ b/defaultopts.c
@@ -29,6 +29,8 @@ void set_default_mplayer_options(struct MPOpts *opts)
         .audio_id = -1,
         .video_id = -1,
         .sub_id = -1,
+        .audio_output_channels = 2,
+        .audio_output_format = -1,  // AF_FORMAT_UNKNOWN
         .playback_speed = 1.,
         .drc_level = 1.,
         .movie_aspect = -1.,
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)
diff --git a/m_option.h b/m_option.h
index 41c8eae64f..a4a262f44f 100644
--- a/m_option.h
+++ b/m_option.h
@@ -530,5 +530,6 @@ m_option_free(const m_option_t* opt,void* dst) {
 #define OPT_FLOATRANGE(optname, varname, flags, min, max) {optname, NULL, &m_option_type_float, (flags)|CONF_RANGE, min, max, NULL, 1, offsetof(struct MPOpts, varname)}
 #define OPT_STRING(optname, varname, flags) {optname, NULL, &m_option_type_string, flags, 0, 0, NULL, 1, offsetof(struct MPOpts, varname)}
 #define OPT_SETTINGSLIST(optname, varname, flags, objlist) {optname, NULL, &m_option_type_obj_settings_list, flags, 0, 0, objlist, 1, offsetof(struct MPOpts, varname)}
+#define OPT_AUDIOFORMAT(optname, varname, flags) {optname, NULL, &m_option_type_afmt, flags, 0, 0, NULL, 1, offsetof(struct MPOpts, varname)}
 
 #endif /* MPLAYER_M_OPTION_H */
diff --git a/mplayer.c b/mplayer.c
index 0e5b21ba49..1fda090641 100644
--- a/mplayer.c
+++ b/mplayer.c
@@ -337,7 +337,6 @@ static int softsleep=0;
 
        double force_fps=0;
 static int force_srate=0;
-static int audio_output_format=-1; // AF_FORMAT_UNKNOWN
        int frame_dropping=0; // option  0=no drop  1= drop vo  2= drop decode
 static int play_n_frames=-1;
 static int play_n_frames_mf=-1;
@@ -1774,7 +1773,7 @@ void reinit_audio_chain(struct MPContext *mpctx)
         current_module="af_preinit";
         ao_data.samplerate=force_srate;
         ao_data.channels=0;
-        ao_data.format=audio_output_format;
+        ao_data.format = opts->audio_output_format;
         // first init to detect best values
         if(!init_audio_filters(mpctx->sh_audio,   // preliminary init
                                // input:
diff --git a/options.h b/options.h
index 08e75fa6a8..9995dc395b 100644
--- a/options.h
+++ b/options.h
@@ -41,6 +41,8 @@ typedef struct MPOpts {
     int sub_id;
     char *audio_lang;
     char *sub_lang;
+    int audio_output_channels;
+    int audio_output_format;
     float playback_speed;
     float drc_level;
     struct m_obj_settings *vf_settings;