Mercurial > mplayer.hg
changeset 14279:b4b202086260
Fix channels, sample rate and sample size for 3gp files
Patch by Richard van der Hoff [ richardv at mxtelecom dot com ]
author | rtognimp |
---|---|
date | Wed, 29 Dec 2004 23:26:01 +0000 |
parents | 17dd45302212 |
children | 8631a3803289 |
files | libmpdemux/demux_mov.c |
diffstat | 1 files changed, 26 insertions(+), 8 deletions(-) [+] |
line wrap: on
line diff
--- a/libmpdemux/demux_mov.c Wed Dec 29 23:23:46 2004 +0000 +++ b/libmpdemux/demux_mov.c Wed Dec 29 23:26:01 2004 +0000 @@ -112,6 +112,8 @@ int duration; // 0 = variable int width,height; // for video unsigned int fourcc; + unsigned int nchannels; + unsigned int samplebytes; // int tkdata_len; // track data unsigned char* tkdata; @@ -866,6 +868,23 @@ sh_audio_t* sh=new_sh_audio(demuxer,priv->track_db); sh->format=trak->fourcc; + switch( sh->format ) { + case 0x726D6173: /* samr */ + /* amr narrowband */ + trak->samplebytes=sh->samplesize=1; + trak->nchannels=sh->channels=1; + sh->samplerate=8000; + break; + + case 0x62776173: /* sawb */ + /* amr wideband */ + trak->samplebytes=sh->samplesize=1; + trak->nchannels=sh->channels=1; + sh->samplerate=16000; + break; + + default: + // assumptions for below table: short is 16bit, int is 32bit, intfp is 16bit // XXX: 32bit fixed point numbers (intfp) are only 2 Byte! // short values are usually one byte leftpadded by zero @@ -892,8 +911,8 @@ // 32 char[4] atom type (fourc charater code -> esds) // 36 char[] atom data (len=size-8) - sh->samplesize=char2short(trak->stdata,18)/8; - sh->channels=char2short(trak->stdata,16); + trak->samplebytes=sh->samplesize=char2short(trak->stdata,18)/8; + trak->nchannels=sh->channels=char2short(trak->stdata,16); /*printf("MOV: timescale: %d samplerate: %d durmap: %d (%d) -> %d (%d)\n", trak->timescale, char2short(trak->stdata,24), trak->durmap[0].dur, trak->durmap[0].num, trak->timescale/trak->durmap[0].dur, @@ -916,9 +935,9 @@ sh->samplerate = 44100; } } - + } mp_msg(MSGT_DEMUX, MSGL_INFO, "Audio bits: %d chans: %d rate: %d\n", - trak->stdata[19],trak->stdata[17],sh->samplerate); + sh->samplesize*8,sh->channels,sh->samplerate); if(trak->stdata_len >= 44 && trak->stdata[9]>=1){ mp_msg(MSGT_DEMUX,MSGL_V,"Audio header: samp/pack=%d bytes/pack=%d bytes/frame=%d bytes/samp=%d \n", @@ -985,7 +1004,7 @@ // Emulate WAVEFORMATEX struct: sh->wf=malloc(sizeof(WAVEFORMATEX)); memset(sh->wf,0,sizeof(WAVEFORMATEX)); - sh->wf->nChannels=(trak->stdata[16]<<8)+trak->stdata[17]; + sh->wf->nChannels=sh->channels; sh->wf->wBitsPerSample=(trak->stdata[18]<<8)+trak->stdata[19]; // sh->wf->nSamplesPerSec=trak->timescale; sh->wf->nSamplesPerSec=(trak->stdata[24]<<8)+trak->stdata[25]; @@ -1748,9 +1767,8 @@ // with missing stsd v1 header containing compression rate x/=ds->ss_div; x*=ds->ss_mul; // compression ratio fix ! HACK ! } else { - x*=(trak->stdata[16]<<8)+trak->stdata[17]; //channels - x*=(trak->stdata[18]<<8)+trak->stdata[19]; //bits/sample - x/=8; // bits/sample + x*=trak->nchannels; + x*=trak->samplebytes; } } mp_msg(MSGT_DEMUX, MSGL_DBG2, "Audio sample %d bytes pts %5.3f\n",trak->chunks[trak->pos].size*trak->samplesize,pts);