Mercurial > mplayer.hg
changeset 13427:9d0b052c4f74
setting samplesize to 2 in decoders where neccessary.
Needed because initialization of sh_audio was moved from dec_audio to
demuxer.c, and some demuxers set samplesize incorrect or to 0.
| author | reimar |
|---|---|
| date | Tue, 21 Sep 2004 20:34:47 +0000 |
| parents | 0f92ad66bbb5 |
| children | d1cdcd974606 |
| files | libmpcodecs/ad_acm.c libmpcodecs/ad_alaw.c libmpcodecs/ad_dk3adpcm.c libmpcodecs/ad_dmo.c libmpcodecs/ad_dshow.c libmpcodecs/ad_faad.c libmpcodecs/ad_ffmpeg.c libmpcodecs/ad_imaadpcm.c libmpcodecs/ad_liba52.c libmpcodecs/ad_libmad.c libmpcodecs/ad_mp3lib.c libmpcodecs/ad_msadpcm.c libmpcodecs/ad_msgsm.c |
| diffstat | 13 files changed, 14 insertions(+), 1 deletions(-) [+] |
line wrap: on
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--- a/libmpcodecs/ad_acm.c Tue Sep 21 19:50:54 2004 +0000 +++ b/libmpcodecs/ad_acm.c Tue Sep 21 20:34:47 2004 +0000 @@ -127,6 +127,7 @@ sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec; sh_audio->channels=priv->o_wf->nChannels; sh_audio->samplerate=priv->o_wf->nSamplesPerSec; + sh_audio->samplesize=2; mp_msg(MSGT_DECVIDEO,MSGL_V,"INFO: Win32/ACM audio codec init OK!\n"); return 1;
--- a/libmpcodecs/ad_alaw.c Tue Sep 21 19:50:54 2004 +0000 +++ b/libmpcodecs/ad_alaw.c Tue Sep 21 20:34:47 2004 +0000 @@ -25,6 +25,7 @@ sh_audio->channels=sh_audio->wf->nChannels; sh_audio->samplerate=sh_audio->wf->nSamplesPerSec; sh_audio->i_bps=sh_audio->channels*sh_audio->samplerate; + sh_audio->samplesize=2; return 1; }
--- a/libmpcodecs/ad_dk3adpcm.c Tue Sep 21 19:50:54 2004 +0000 +++ b/libmpcodecs/ad_dk3adpcm.c Tue Sep 21 20:34:47 2004 +0000 @@ -84,6 +84,7 @@ sh_audio->samplerate = sh_audio->wf->nSamplesPerSec; sh_audio->i_bps = (sh_audio->ds->ss_mul * sh_audio->samplerate) / sh_audio->ds->ss_div; + sh_audio->samplesize=2; return 1; }
--- a/libmpcodecs/ad_dmo.c Tue Sep 21 19:50:54 2004 +0000 +++ b/libmpcodecs/ad_dmo.c Tue Sep 21 20:34:47 2004 +0000 @@ -43,6 +43,7 @@ sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec; sh_audio->channels=chans; sh_audio->samplerate=sh_audio->wf->nSamplesPerSec; + sh_audio->samplesize=2; sh_audio->audio_in_minsize=4*sh_audio->wf->nBlockAlign; if(sh_audio->audio_in_minsize<8192) sh_audio->audio_in_minsize=8192; sh_audio->audio_out_minsize=4*16384;
--- a/libmpcodecs/ad_dshow.c Tue Sep 21 19:50:54 2004 +0000 +++ b/libmpcodecs/ad_dshow.c Tue Sep 21 20:34:47 2004 +0000 @@ -39,6 +39,7 @@ sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec; sh_audio->channels=sh_audio->wf->nChannels; sh_audio->samplerate=sh_audio->wf->nSamplesPerSec; + sh_audio->samplesize=2; sh_audio->audio_in_minsize=2*sh_audio->wf->nBlockAlign; if(sh_audio->audio_in_minsize<8192) sh_audio->audio_in_minsize=8192; sh_audio->audio_out_minsize=16384;
--- a/libmpcodecs/ad_faad.c Tue Sep 21 19:50:54 2004 +0000 +++ b/libmpcodecs/ad_faad.c Tue Sep 21 20:34:47 2004 +0000 @@ -71,11 +71,11 @@ /* XXX: FAAD support FLOAT output, how do we handle * that (FAAD_FMT_FLOAT)? ::atmos */ - if(sh->samplesize) switch(sh->samplesize){ case 1: // 8Bit mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: 8Bit samplesize not supported by FAAD, assuming 16Bit!\n"); default: + sh->samplesize=2; case 2: // 16Bit faac_conf->outputFormat = FAAD_FMT_16BIT; break; @@ -123,6 +123,7 @@ mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Negotiated samplerate: %dHz channels: %d\n", faac_samplerate, faac_channels); sh->channels = faac_channels; sh->samplerate = faac_samplerate; + sh->samplesize=2; //sh->o_bps = sh->samplesize*faac_channels*faac_samplerate; if(!sh->i_bps) { mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: compressed input bitrate missing, assuming 128kbit/s!\n");
--- a/libmpcodecs/ad_ffmpeg.c Tue Sep 21 19:50:54 2004 +0000 +++ b/libmpcodecs/ad_ffmpeg.c Tue Sep 21 20:34:47 2004 +0000 @@ -110,6 +110,7 @@ sh_audio->samplerate=sh_audio->wf->nSamplesPerSec; sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec; #endif + sh_audio->samplesize=2; return 1; }
--- a/libmpcodecs/ad_imaadpcm.c Tue Sep 21 19:50:54 2004 +0000 +++ b/libmpcodecs/ad_imaadpcm.c Tue Sep 21 20:34:47 2004 +0000 @@ -111,6 +111,7 @@ /* decodes 34 byte -> 64 short*/ sh_audio->i_bps = (sh_audio->ds->ss_mul * sh_audio->samplerate) / sh_audio->ds->ss_div; + sh_audio->samplesize=2; return 1; }
--- a/libmpcodecs/ad_liba52.c Tue Sep 21 19:50:54 2004 +0000 +++ b/libmpcodecs/ad_liba52.c Tue Sep 21 20:34:47 2004 +0000 @@ -67,6 +67,7 @@ mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"a52: len=%d flags=0x%X %d Hz %d bit/s\n",length,flags,sample_rate,bit_rate); sh_audio->samplerate=sample_rate; sh_audio->i_bps=bit_rate/8; + sh_audio->samplesize=2; demux_read_data(sh_audio->ds,sh_audio->a_in_buffer+8,length-8); if(sh_audio->format!=0x2000) swab(sh_audio->a_in_buffer+8,sh_audio->a_in_buffer+8,length-8);
--- a/libmpcodecs/ad_libmad.c Tue Sep 21 19:50:54 2004 +0000 +++ b/libmpcodecs/ad_libmad.c Tue Sep 21 20:34:47 2004 +0000 @@ -88,6 +88,7 @@ sh->channels=(this->frame.header.mode == MAD_MODE_SINGLE_CHANNEL) ? 1 : 2; sh->samplerate=this->frame.header.samplerate; sh->i_bps=this->frame.header.bitrate/8; + sh_audio->samplesize=2; return 1; }
--- a/libmpcodecs/ad_mp3lib.c Tue Sep 21 19:50:54 2004 +0000 +++ b/libmpcodecs/ad_mp3lib.c Tue Sep 21 20:34:47 2004 +0000 @@ -50,6 +50,7 @@ sh->a_buffer_len=MP3_DecodeFrame(sh->a_buffer,-1); if(!sh->a_buffer_len) return 0; // unsupported layer/format sh->channels=2; // hack + sh->samplesize=2; sh->samplerate=MP3_samplerate; sh->i_bps=MP3_bitrate*(1000/8); MP3_PrintHeader();
--- a/libmpcodecs/ad_msadpcm.c Tue Sep 21 19:50:54 2004 +0000 +++ b/libmpcodecs/ad_msadpcm.c Tue Sep 21 20:34:47 2004 +0000 @@ -78,6 +78,7 @@ sh_audio->samplerate=sh_audio->wf->nSamplesPerSec; sh_audio->i_bps = sh_audio->wf->nBlockAlign * (sh_audio->channels*sh_audio->samplerate) / sh_audio->ds->ss_div; + sh_audio->samplesize=2; return 1; }
--- a/libmpcodecs/ad_msgsm.c Tue Sep 21 19:50:54 2004 +0000 +++ b/libmpcodecs/ad_msgsm.c Tue Sep 21 20:34:47 2004 +0000 @@ -25,6 +25,7 @@ GSM_Init(); sh_audio->channels=sh_audio->wf->nChannels; sh_audio->samplerate=sh_audio->wf->nSamplesPerSec; + sh_audio->samplesize=2; // decodes 65 byte -> 320 short // 1 sec: sh_audio->channels*sh_audio->samplerate samples // 1 frame: 320 samples