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cosmetics: ad_ffmpeg.c: reformat

This commit is contained in:
Uoti Urpala 2011-08-21 22:13:49 +03:00
parent 2c5285c151
commit da2b884c70
1 changed files with 137 additions and 131 deletions
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112
libmpcodecs/ad_ffmpeg.c
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@ -20,6 +20,8 @@
#include <stdlib.h> #include <stdlib.h>
#include <unistd.h> #include <unistd.h>
#include <libavcodec/avcodec.h>
#include "config.h" #include "config.h"
#include "mp_msg.h" #include "mp_msg.h"
#include "options.h" #include "options.h"
@ -40,14 +42,9 @@ static const ad_info_t info =
LIBAD_EXTERN(ffmpeg) LIBAD_EXTERN(ffmpeg)
#define assert(x)
#include "libavcodec/avcodec.h"
static int preinit(sh_audio_t *sh) static int preinit(sh_audio_t *sh)
{ {
sh->audio_out_minsize=AVCODEC_MAX_AUDIO_FRAME_SIZE; sh->audio_out_minsize = AVCODEC_MAX_AUDIO_FRAME_SIZE;
return 1; return 1;
} }
@ -56,7 +53,8 @@ static int preinit(sh_audio_t *sh)
* from the codec, since if the codec isn't using the correct values for * from the codec, since if the codec isn't using the correct values for
* those everything breaks anyway. * those everything breaks anyway.
*/ */
static int setup_format(sh_audio_t *sh_audio, const AVCodecContext *lavc_context) static int setup_format(sh_audio_t *sh_audio,
const AVCodecContext *lavc_context)
{ {
int sample_format = sh_audio->sample_format; int sample_format = sh_audio->sample_format;
switch (lavc_context->sample_fmt) { switch (lavc_context->sample_fmt) {
@ -82,10 +80,10 @@ static int setup_format(sh_audio_t *sh_audio, const AVCodecContext *lavc_context
if (lavc_context->channels != sh_audio->channels || if (lavc_context->channels != sh_audio->channels ||
samplerate != sh_audio->samplerate || samplerate != sh_audio->samplerate ||
sample_format != sh_audio->sample_format) { sample_format != sh_audio->sample_format) {
sh_audio->channels=lavc_context->channels; sh_audio->channels = lavc_context->channels;
sh_audio->samplerate=samplerate; sh_audio->samplerate = samplerate;
sh_audio->sample_format = sample_format; sh_audio->sample_format = sample_format;
sh_audio->samplesize=af_fmt2bits(sh_audio->sample_format)/ 8; sh_audio->samplesize = af_fmt2bits(sh_audio->sample_format) / 8;
if (broken_srate) if (broken_srate)
mp_msg(MSGT_DECAUDIO, MSGL_WARN, mp_msg(MSGT_DECAUDIO, MSGL_WARN,
"Ignoring broken container sample rate for AAC with SBR\n"); "Ignoring broken container sample rate for AAC with SBR\n");
@ -100,21 +98,23 @@ static int init(sh_audio_t *sh_audio)
AVCodecContext *lavc_context; AVCodecContext *lavc_context;
AVCodec *lavc_codec; AVCodec *lavc_codec;
mp_msg(MSGT_DECAUDIO,MSGL_V,"FFmpeg's libavcodec audio codec\n"); mp_msg(MSGT_DECAUDIO, MSGL_V, "FFmpeg's libavcodec audio codec\n");
lavc_codec = avcodec_find_decoder_by_name(sh_audio->codec->dll); lavc_codec = avcodec_find_decoder_by_name(sh_audio->codec->dll);
if(!lavc_codec){ if (!lavc_codec) {
mp_tmsg(MSGT_DECAUDIO,MSGL_ERR,"Cannot find codec '%s' in libavcodec...\n",sh_audio->codec->dll); mp_tmsg(MSGT_DECAUDIO, MSGL_ERR,
"Cannot find codec '%s' in libavcodec...\n",
sh_audio->codec->dll);
return 0; return 0;
} }
lavc_context = avcodec_alloc_context(); lavc_context = avcodec_alloc_context();
sh_audio->context=lavc_context; sh_audio->context = lavc_context;
lavc_context->drc_scale = opts->drc_level; lavc_context->drc_scale = opts->drc_level;
lavc_context->sample_rate = sh_audio->samplerate; lavc_context->sample_rate = sh_audio->samplerate;
lavc_context->bit_rate = sh_audio->i_bps * 8; lavc_context->bit_rate = sh_audio->i_bps * 8;
if(sh_audio->wf){ if (sh_audio->wf) {
lavc_context->channels = sh_audio->wf->nChannels; lavc_context->channels = sh_audio->wf->nChannels;
lavc_context->sample_rate = sh_audio->wf->nSamplesPerSec; lavc_context->sample_rate = sh_audio->wf->nSamplesPerSec;
lavc_context->bit_rate = sh_audio->wf->nAvgBytesPerSec * 8; lavc_context->bit_rate = sh_audio->wf->nAvgBytesPerSec * 8;
@ -135,8 +135,8 @@ static int init(sh_audio_t *sh_audio)
} }
// for QDM2 // for QDM2
if (sh_audio->codecdata_len && sh_audio->codecdata && !lavc_context->extradata) if (sh_audio->codecdata_len && sh_audio->codecdata &&
{ !lavc_context->extradata) {
lavc_context->extradata = av_malloc(sh_audio->codecdata_len + lavc_context->extradata = av_malloc(sh_audio->codecdata_len +
FF_INPUT_BUFFER_PADDING_SIZE); FF_INPUT_BUFFER_PADDING_SIZE);
lavc_context->extradata_size = sh_audio->codecdata_len; lavc_context->extradata_size = sh_audio->codecdata_len;
@ -146,21 +146,20 @@ static int init(sh_audio_t *sh_audio)
/* open it */ /* open it */
if (avcodec_open(lavc_context, lavc_codec) < 0) { if (avcodec_open(lavc_context, lavc_codec) < 0) {
mp_tmsg(MSGT_DECAUDIO,MSGL_ERR, "Could not open codec.\n"); mp_tmsg(MSGT_DECAUDIO, MSGL_ERR, "Could not open codec.\n");
return 0; return 0;
} }
mp_msg(MSGT_DECAUDIO,MSGL_V,"INFO: libavcodec \"%s\" init OK!\n", lavc_codec->name); mp_msg(MSGT_DECAUDIO, MSGL_V, "INFO: libavcodec \"%s\" init OK!\n",
lavc_codec->name);
// printf("\nFOURCC: 0x%X\n",sh_audio->format); if (sh_audio->format == 0x3343414D) {
if(sh_audio->format==0x3343414D){
// MACE 3:1 // MACE 3:1
sh_audio->ds->ss_div = 2*3; // 1 samples/packet sh_audio->ds->ss_div = 2 * 3; // 1 samples/packet
sh_audio->ds->ss_mul = 2*sh_audio->wf->nChannels; // 1 byte*ch/packet sh_audio->ds->ss_mul = 2 * sh_audio->wf->nChannels; // 1 byte*ch/packet
} else } else if (sh_audio->format == 0x3643414D) {
if(sh_audio->format==0x3643414D){
// MACE 6:1 // MACE 6:1
sh_audio->ds->ss_div = 2*6; // 1 samples/packet sh_audio->ds->ss_div = 2 * 6; // 1 samples/packet
sh_audio->ds->ss_mul = 2*sh_audio->wf->nChannels; // 1 byte*ch/packet sh_audio->ds->ss_mul = 2 * sh_audio->wf->nChannels; // 1 byte*ch/packet
} }
// Decode at least 1 byte: (to get header filled) // Decode at least 1 byte: (to get header filled)
@ -178,9 +177,9 @@ static int init(sh_audio_t *sh_audio)
} }
} }
sh_audio->i_bps=lavc_context->bit_rate/8; sh_audio->i_bps = lavc_context->bit_rate / 8;
if (sh_audio->wf && sh_audio->wf->nAvgBytesPerSec) if (sh_audio->wf && sh_audio->wf->nAvgBytesPerSec)
sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec; sh_audio->i_bps = sh_audio->wf->nAvgBytesPerSec;
switch (lavc_context->sample_fmt) { switch (lavc_context->sample_fmt) {
case SAMPLE_FMT_U8: case SAMPLE_FMT_U8:
@ -204,10 +203,10 @@ static void uninit(sh_audio_t *sh)
av_freep(&lavc_context); av_freep(&lavc_context);
} }
static int control(sh_audio_t *sh,int cmd,void* arg, ...) static int control(sh_audio_t *sh, int cmd, void *arg, ...)
{ {
AVCodecContext *lavc_context = sh->context; AVCodecContext *lavc_context = sh->context;
switch(cmd){ switch (cmd) {
case ADCTRL_RESYNC_STREAM: case ADCTRL_RESYNC_STREAM:
avcodec_flush_buffers(lavc_context); avcodec_flush_buffers(lavc_context);
ds_clear_parser(sh->ds); ds_clear_parser(sh->ds);
@ -216,16 +215,19 @@ static int control(sh_audio_t *sh,int cmd,void* arg, ...)
return CONTROL_UNKNOWN; return CONTROL_UNKNOWN;
} }
static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen) static int decode_audio(sh_audio_t *sh_audio, unsigned char *buf, int minlen,
int maxlen)
{ {
unsigned char *start=NULL; AVCodecContext *avctx = sh_audio->context;
int y,len=-1;
while(len<minlen){ unsigned char *start = NULL;
int y, len = -1;
while (len < minlen) {
AVPacket pkt; AVPacket pkt;
int len2=maxlen; int len2 = maxlen;
double pts; double pts;
int x=ds_get_packet_pts(sh_audio->ds,&start, &pts); int x = ds_get_packet_pts(sh_audio->ds, &start, &pts);
if(x<=0) { if (x <= 0) {
start = NULL; start = NULL;
x = 0; x = 0;
ds_parse(sh_audio->ds, &start, &x, MP_NOPTS_VALUE, 0); ds_parse(sh_audio->ds, &start, &x, MP_NOPTS_VALUE, 0);
@ -243,32 +245,36 @@ static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int m
sh_audio->pts = pts; sh_audio->pts = pts;
sh_audio->pts_bytes = 0; sh_audio->pts_bytes = 0;
} }
y=avcodec_decode_audio3(sh_audio->context,(int16_t*)buf,&len2,&pkt); y = avcodec_decode_audio3(avctx, (int16_t *)buf, &len2, &pkt);
//printf("return:%d samples_out:%d bitstream_in:%d sample_sum:%d\n", y, len2, x, len); fflush(stdout);
// LATM may need many packets to find mux info // LATM may need many packets to find mux info
if (y == AVERROR(EAGAIN)) if (y == AVERROR(EAGAIN))
continue; continue;
if(y<0){ mp_msg(MSGT_DECAUDIO,MSGL_V,"lavc_audio: error\n");break; } if (y < 0) {
if(!sh_audio->parser && y<x) mp_msg(MSGT_DECAUDIO, MSGL_V, "lavc_audio: error\n");
sh_audio->ds->buffer_pos+=y-x; // put back data (HACK!) break;
if(len2>0){ }
if (((AVCodecContext *)sh_audio->context)->channels >= 5) { if (!sh_audio->parser && y < x)
int samplesize = av_get_bits_per_sample_format(((AVCodecContext *) sh_audio->ds->buffer_pos += y - x; // put back data (HACK!)
sh_audio->context)->sample_fmt) / 8; if (len2 > 0) {
if (avctx->channels >= 5) {
int samplesize = av_get_bits_per_sample_format(
avctx->sample_fmt) / 8;
reorder_channel_nch(buf, AF_CHANNEL_LAYOUT_LAVC_DEFAULT, reorder_channel_nch(buf, AF_CHANNEL_LAYOUT_LAVC_DEFAULT,
AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT, AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
((AVCodecContext *)sh_audio->context)->channels, avctx->channels,
len2 / samplesize, samplesize); len2 / samplesize, samplesize);
} }
//len=len2;break; if (len < 0)
if(len<0) len=len2; else len+=len2; len = len2;
buf+=len2; else
len += len2;
buf += len2;
maxlen -= len2; maxlen -= len2;
sh_audio->pts_bytes += len2; sh_audio->pts_bytes += len2;
} }
mp_dbg(MSGT_DECAUDIO,MSGL_DBG2,"Decoded %d -> %d \n",y,len2); mp_dbg(MSGT_DECAUDIO, MSGL_DBG2, "Decoded %d -> %d \n", y, len2);
if (setup_format(sh_audio, sh_audio->context)) if (setup_format(sh_audio, avctx))
break; break;
} }
return len; return len;