mirror of
https://github.com/mpv-player/mpv.git
synced 2024-09-20 12:02:23 +02:00
allow sh_audio struct to be initialized by demuxer, add parsing of mp4 esds header to mov demuxer, init faad from info from mov header
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@5207 b3059339-0415-0410-9bf9-f77b7e298cf2
This commit is contained in:
atmos4
parent
682288117e
commit
9ed76eba4b
29
dec_audio.c
29
dec_audio.c
@ -328,15 +328,17 @@ int mplayer_audio_read(char *buf,int size){
|
||||
int init_audio(sh_audio_t *sh_audio){
|
||||
int driver=sh_audio->codec->driver;
|
||||
|
||||
sh_audio->samplesize=2;
|
||||
if(!sh_audio->samplesize)
|
||||
sh_audio->samplesize=2;
|
||||
if(!sh_audio->sample_format)
|
||||
#ifdef WORDS_BIGENDIAN
|
||||
sh_audio->sample_format=AFMT_S16_BE;
|
||||
sh_audio->sample_format=AFMT_S16_BE;
|
||||
#else
|
||||
sh_audio->sample_format=AFMT_S16_LE;
|
||||
sh_audio->sample_format=AFMT_S16_LE;
|
||||
#endif
|
||||
sh_audio->samplerate=0;
|
||||
//sh_audio->samplerate=0;
|
||||
//sh_audio->pcm_bswap=0;
|
||||
sh_audio->o_bps=0;
|
||||
//sh_audio->o_bps=0;
|
||||
|
||||
sh_audio->a_buffer_size=0;
|
||||
sh_audio->a_buffer=NULL;
|
||||
@ -840,16 +842,15 @@ case AFM_AAC: {
|
||||
faacDecConfigurationPtr faac_conf;
|
||||
faac_hdec = faacDecOpen();
|
||||
|
||||
#if 0
|
||||
#if 1
|
||||
/* Set the default object type and samplerate */
|
||||
/* This is useful for RAW AAC files */
|
||||
faac_conf = faacDecGetCurrentConfiguration(faac_hdec);
|
||||
if(sh_audio->samplerate)
|
||||
faac_conf->defSampleRate = sh_audio->samplerate;
|
||||
/* XXX: is outputFormat samplesize of compressed data or samplesize of
|
||||
* decoded data, maybe upsampled? Also, FAAD support FLOAT output,
|
||||
* how do we handle that (FAAD_FMT_FLOAT)? ::atmos
|
||||
*/
|
||||
/* XXX: FAAD support FLOAT output, how do we handle
|
||||
* that (FAAD_FMT_FLOAT)? ::atmos
|
||||
*/
|
||||
if(sh_audio->samplesize)
|
||||
switch(sh_audio->samplesize){
|
||||
case 1: // 8Bit
|
||||
@ -865,7 +866,7 @@ case AFM_AAC: {
|
||||
faac_conf->outputFormat = FAAD_FMT_32BIT;
|
||||
break;
|
||||
}
|
||||
faac_conf->defObjectType = LTP; // => MAIN, LC, SSR, LTP available.
|
||||
//faac_conf->defObjectType = LTP; // => MAIN, LC, SSR, LTP available.
|
||||
|
||||
faacDecSetConfiguration(faac_hdec, faac_conf);
|
||||
#endif
|
||||
@ -887,7 +888,11 @@ case AFM_AAC: {
|
||||
mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Negotiated samplerate: %dHz channels: %d\n", faac_samplerate, faac_channels);
|
||||
sh_audio->channels = faac_channels;
|
||||
sh_audio->samplerate = faac_samplerate;
|
||||
sh_audio->i_bps = 128*1000/8; // XXX: HACK!!! There's currently no way to get bitrate from libfaad2! ::atmos
|
||||
if(!sh_audio->i_bps) {
|
||||
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: compressed input bitrate missing, assuming 128kbit/s!\n");
|
||||
sh_audio->i_bps = 128*1000/8; // XXX: HACK!!! There's currently no way to get bitrate from libfaad2! ::atmos
|
||||
} else
|
||||
mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: got %dkbit/s rate from MP4 header!\n",sh_audio->i_bps*8/1000);
|
||||
}
|
||||
|
||||
} break;
|
||||
|
||||
@ -562,9 +562,37 @@ static void lschunks(demuxer_t* demuxer,int level,off_t endpos,mov_track_t* trak
|
||||
mov_build_index(trak);
|
||||
switch(trak->type){
|
||||
case MOV_TRAK_AUDIO: {
|
||||
#if 0
|
||||
struct {
|
||||
int16_t version; // 0 or 1 (version 1 is qt3.0+)
|
||||
int16_t revision; // 0
|
||||
int32_t vendor_id; // 0
|
||||
int16_t channels; // 1 or 2 (Mono/Stereo)
|
||||
int16_t samplesize; // 8 or 16 (8Bit/16Bit)
|
||||
int16_t compression_id; // if version 0 then 0
|
||||
// if version 1 and vbr then -2 else 0
|
||||
int16_t packet_size; // 0
|
||||
uint32_t sample_rate; // samplerate (Hz)
|
||||
// qt3.0+ (version == 1)
|
||||
uint32_t samples_per_packet; // 0 or num uncompressed samples in a packet
|
||||
// if 0 below three values are also 0
|
||||
uint32_t bytes_per_packet; // 0 or num compressed bytes for one channel
|
||||
uint32_t bytes_per_frame; // 0 or num compressed bytes for all channels
|
||||
// (channels * bytes_per_packet)
|
||||
uint32_t bytes_per_sample; // 0 or size of uncompressed sample
|
||||
// if samples_per_packet and bytes_per_packet are constant (CBR)
|
||||
// then bytes_per_frame and bytes_per_sample must be 0 (else is VBR)
|
||||
// ---
|
||||
// optional additional atom-based fields
|
||||
// ([int32_t size,int32_t type,some data ],repeat)
|
||||
} my_stdata;
|
||||
#endif
|
||||
sh_audio_t* sh=new_sh_audio(demuxer,priv->track_db);
|
||||
sh->format=trak->fourcc;
|
||||
|
||||
// assumptions for below table: short is 16bit, int is 32bit
|
||||
// short values are usually one byte leftpadded by zero
|
||||
// int values are usually two byte leftpadded by zero
|
||||
// stdata[]:
|
||||
// 8 short version
|
||||
// 10 short revision
|
||||
@ -579,8 +607,48 @@ static void lschunks(demuxer_t* demuxer,int level,off_t endpos,mov_track_t* trak
|
||||
// 32 int bytes_per_packet
|
||||
// 36 int bytes_per_frame
|
||||
// 40 int bytes_per_sample
|
||||
// my recoveries from .mp4 files with correct index (counting from 0):
|
||||
// it's always start with padding/real start
|
||||
// 8/ 9 short version
|
||||
// 10/11 short revision
|
||||
// 12/14 int vendor_id
|
||||
// 16/17 short channels
|
||||
// 18/19 short samplesize
|
||||
// 20/21 short compression_id
|
||||
// 22/23 short packet_size (XXX: this overlaps with sample_rate ?)
|
||||
// 22/24 int sample_rate
|
||||
// esds atom:
|
||||
// 28/31 int atom size (bytes of int size, int type and data)
|
||||
// 32/32 int atom type (fourc charater code -> esds)
|
||||
// 62/63 int compressed datarate (Bits)
|
||||
|
||||
mp_msg(MSGT_DEMUX, MSGL_INFO, "Audio bits: %d chans: %d\n",trak->stdata[19],trak->stdata[17]);
|
||||
#define char2short(x,y) ((x[y]<<8)|x[y+1])
|
||||
#define char2int(x,y) ((x[y]<<24)|(x[y+1]<<16)|(x[y+2]<<8)|x[y+3])
|
||||
|
||||
sh->samplesize=char2short(trak->stdata,18)/8;
|
||||
sh->channels=char2short(trak->stdata,16);
|
||||
sh->samplerate=char2short(trak->stdata,24);
|
||||
|
||||
mp_msg(MSGT_DEMUX, MSGL_INFO, "Audio bits: %d chans: %d\n",
|
||||
trak->stdata[19],trak->stdata[17]);
|
||||
mp_msg(MSGT_DEMUX, MSGL_INFO, "Audio sample rate: %d\n",
|
||||
char2short(trak->stdata,24));
|
||||
if((trak->stdata[9]==0) && trak->stdata_len >= 36) { // version 0 with extra atoms
|
||||
int atom_len = char2int(trak->stdata,28);
|
||||
switch(char2int(trak->stdata,32)) { // atom type
|
||||
case MOV_FOURCC('e','s','d','s'):
|
||||
mp_msg(MSGT_DEMUX, MSGL_INFO, "MOV: MP4 esds audio atom found (%d)!\n", atom_len);
|
||||
if(atom_len >= 28)
|
||||
mp_msg(MSGT_DEMUX, MSGL_INFO, "Audio compressed datarate: %dkbit/s\n",
|
||||
char2int(trak->stdata,62)/1000);
|
||||
sh->i_bps=char2int(trak->stdata,62)/8;
|
||||
break;
|
||||
default:
|
||||
mp_msg(MSGT_DEMUX, MSGL_INFO, "MOV: unknown audio atom %c%c%c%c found (%d)!\n",
|
||||
trak->stdata[32],trak->stdata[33],trak->stdata[34],trak->stdata[35],
|
||||
atom_len);
|
||||
}
|
||||
}
|
||||
mp_msg(MSGT_DEMUX, MSGL_INFO, "Fourcc: %.4s\n",&trak->fourcc);
|
||||
#if 0
|
||||
{ FILE* f=fopen("stdata.dat","wb");
|
||||
|
||||
Loading…
Reference in New Issue
Block a user