diff options
Diffstat (limited to 'libmpcodecs')
| -rw-r--r-- | libmpcodecs/ad_dmo.c | 6 | ||||
| -rw-r--r-- | libmpcodecs/ad_faad.c | 10 | ||||
| -rw-r--r-- | libmpcodecs/ad_ffmpeg.c | 2 | ||||
| -rw-r--r-- | libmpcodecs/ad_internal.h | 1 | ||||
| -rw-r--r-- | libmpcodecs/ad_liba52.c | 5 | ||||
| -rw-r--r-- | libmpcodecs/ad_libdca.c | 11 | ||||
| -rw-r--r-- | libmpcodecs/dec_audio.c | 2 |
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) |