1
0
mirror of https://github.com/mpv-player/mpv synced 2024-12-27 01:22:30 +00:00
mpv/libmpcodecs/ad_ffmpeg.c
Uoti Urpala 0321d683b4 ad_ffmpeg: return failure from init() if initial decode fails
The init() method in ad_ffmpeg tries to decode some audio data after
opening the libavcodec decoder; however the method returned success
even if this part failed. Change it to return failure instead,
indicating that the codec could not be successfully opened.

This improves behavior at least with some AAC files, for which the
libavcodec decoder can be successfully initialized but decoding
packets always fails. Before the audio would be decoded with
libavcodec, producing only a constant stream of errors; after this
commit audio decoder initialization falls back to FAAD (if available)
which works for these samples.
2011-05-03 16:52:57 +03:00

278 lines
9.0 KiB
C

/*
* This file is part of MPlayer.
*
* MPlayer is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* MPlayer is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License along
* with MPlayer; if not, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include "config.h"
#include "mp_msg.h"
#include "options.h"
#include "ad_internal.h"
#include "vd_ffmpeg.h"
#include "libaf/reorder_ch.h"
#include "mpbswap.h"
static const ad_info_t info =
{
"FFmpeg/libavcodec audio decoders",
"ffmpeg",
"Nick Kurshev",
"ffmpeg.sf.net",
""
};
LIBAD_EXTERN(ffmpeg)
#define assert(x)
#include "libavcodec/avcodec.h"
static int preinit(sh_audio_t *sh)
{
sh->audio_out_minsize=AVCODEC_MAX_AUDIO_FRAME_SIZE;
return 1;
}
/* Prefer playing audio with the samplerate given in container data
* if available, but take number the number of channels and sample format
* from the codec, since if the codec isn't using the correct values for
* those everything breaks anyway.
*/
static int setup_format(sh_audio_t *sh_audio, const AVCodecContext *lavc_context)
{
int sample_format = sh_audio->sample_format;
switch (lavc_context->sample_fmt) {
case SAMPLE_FMT_U8: sample_format = AF_FORMAT_U8; break;
case SAMPLE_FMT_S16: sample_format = AF_FORMAT_S16_NE; break;
case SAMPLE_FMT_S32: sample_format = AF_FORMAT_S32_NE; break;
case SAMPLE_FMT_FLT: sample_format = AF_FORMAT_FLOAT_NE; break;
default:
mp_msg(MSGT_DECAUDIO, MSGL_FATAL, "Unsupported sample format\n");
}
bool broken_srate = false;
int samplerate = lavc_context->sample_rate;
int container_samplerate = sh_audio->container_out_samplerate;
if (!container_samplerate && sh_audio->wf)
container_samplerate = sh_audio->wf->nSamplesPerSec;
if (lavc_context->codec_id == CODEC_ID_AAC
&& samplerate == 2 * container_samplerate)
broken_srate = true;
else if (container_samplerate)
samplerate = container_samplerate;
if (lavc_context->channels != sh_audio->channels ||
samplerate != sh_audio->samplerate ||
sample_format != sh_audio->sample_format) {
sh_audio->channels=lavc_context->channels;
sh_audio->samplerate=samplerate;
sh_audio->sample_format = sample_format;
sh_audio->samplesize=af_fmt2bits(sh_audio->sample_format)/ 8;
if (broken_srate)
mp_msg(MSGT_DECAUDIO, MSGL_WARN,
"Ignoring broken container sample rate for AAC with SBR\n");
return 1;
}
return 0;
}
static int init(sh_audio_t *sh_audio)
{
struct MPOpts *opts = sh_audio->opts;
AVCodecContext *lavc_context;
AVCodec *lavc_codec;
mp_msg(MSGT_DECAUDIO,MSGL_V,"FFmpeg's libavcodec audio codec\n");
init_avcodec();
lavc_codec = avcodec_find_decoder_by_name(sh_audio->codec->dll);
if(!lavc_codec){
mp_tmsg(MSGT_DECAUDIO,MSGL_ERR,"Cannot find codec '%s' in libavcodec...\n",sh_audio->codec->dll);
return 0;
}
lavc_context = avcodec_alloc_context();
sh_audio->context=lavc_context;
lavc_context->drc_scale = opts->drc_level;
lavc_context->sample_rate = sh_audio->samplerate;
lavc_context->bit_rate = sh_audio->i_bps * 8;
if(sh_audio->wf){
lavc_context->channels = sh_audio->wf->nChannels;
lavc_context->sample_rate = sh_audio->wf->nSamplesPerSec;
lavc_context->bit_rate = sh_audio->wf->nAvgBytesPerSec * 8;
lavc_context->block_align = sh_audio->wf->nBlockAlign;
lavc_context->bits_per_coded_sample = sh_audio->wf->wBitsPerSample;
}
lavc_context->request_channels = opts->audio_output_channels;
lavc_context->codec_tag = sh_audio->format; //FOURCC
lavc_context->codec_type = AVMEDIA_TYPE_AUDIO;
lavc_context->codec_id = lavc_codec->id; // not sure if required, imho not --A'rpi
/* alloc extra data */
if (sh_audio->wf && sh_audio->wf->cbSize > 0) {
lavc_context->extradata = av_mallocz(sh_audio->wf->cbSize + FF_INPUT_BUFFER_PADDING_SIZE);
lavc_context->extradata_size = sh_audio->wf->cbSize;
memcpy(lavc_context->extradata, sh_audio->wf + 1,
lavc_context->extradata_size);
}
// for QDM2
if (sh_audio->codecdata_len && sh_audio->codecdata && !lavc_context->extradata)
{
lavc_context->extradata = av_malloc(sh_audio->codecdata_len +
FF_INPUT_BUFFER_PADDING_SIZE);
lavc_context->extradata_size = sh_audio->codecdata_len;
memcpy(lavc_context->extradata, (char *)sh_audio->codecdata,
lavc_context->extradata_size);
}
/* open it */
if (avcodec_open(lavc_context, lavc_codec) < 0) {
mp_tmsg(MSGT_DECAUDIO,MSGL_ERR, "Could not open codec.\n");
return 0;
}
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){
// MACE 3:1
sh_audio->ds->ss_div = 2*3; // 1 samples/packet
sh_audio->ds->ss_mul = 2*sh_audio->wf->nChannels; // 1 byte*ch/packet
} else
if(sh_audio->format==0x3643414D){
// MACE 6:1
sh_audio->ds->ss_div = 2*6; // 1 samples/packet
sh_audio->ds->ss_mul = 2*sh_audio->wf->nChannels; // 1 byte*ch/packet
}
// Decode at least 1 byte: (to get header filled)
for (int tries = 0;;) {
int x = decode_audio(sh_audio, sh_audio->a_buffer, 1,
sh_audio->a_buffer_size);
if (x > 0) {
sh_audio->a_buffer_len = x;
break;
}
if (++tries >= 5) {
mp_msg(MSGT_DECAUDIO, MSGL_ERR,
"ad_ffmpeg: initial decode failed\n");
return 0;
}
}
sh_audio->i_bps=lavc_context->bit_rate/8;
if (sh_audio->wf && sh_audio->wf->nAvgBytesPerSec)
sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec;
switch (lavc_context->sample_fmt) {
case SAMPLE_FMT_U8:
case SAMPLE_FMT_S16:
case SAMPLE_FMT_S32:
case SAMPLE_FMT_FLT:
break;
default:
return 0;
}
return 1;
}
static void uninit(sh_audio_t *sh)
{
AVCodecContext *lavc_context = sh->context;
if (avcodec_close(lavc_context) < 0)
mp_tmsg(MSGT_DECVIDEO, MSGL_ERR, "Could not close codec.\n");
av_freep(&lavc_context->extradata);
av_freep(&lavc_context);
}
static int control(sh_audio_t *sh,int cmd,void* arg, ...)
{
AVCodecContext *lavc_context = sh->context;
switch(cmd){
case ADCTRL_RESYNC_STREAM:
avcodec_flush_buffers(lavc_context);
ds_clear_parser(sh->ds);
return CONTROL_TRUE;
}
return CONTROL_UNKNOWN;
}
static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
{
unsigned char *start=NULL;
int y,len=-1;
while(len<minlen){
AVPacket pkt;
int len2=maxlen;
double pts;
int x=ds_get_packet_pts(sh_audio->ds,&start, &pts);
if(x<=0) {
start = NULL;
x = 0;
ds_parse(sh_audio->ds, &start, &x, MP_NOPTS_VALUE, 0);
if (x <= 0)
break; // error
} else {
int in_size = x;
int consumed = ds_parse(sh_audio->ds, &start, &x, pts, 0);
sh_audio->ds->buffer_pos -= in_size - consumed;
}
av_init_packet(&pkt);
pkt.data = start;
pkt.size = x;
if (pts != MP_NOPTS_VALUE) {
sh_audio->pts = pts;
sh_audio->pts_bytes = 0;
}
y=avcodec_decode_audio3(sh_audio->context,(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
if (y == AVERROR(EAGAIN))
continue;
if(y<0){ mp_msg(MSGT_DECAUDIO,MSGL_V,"lavc_audio: error\n");break; }
if(!sh_audio->parser && y<x)
sh_audio->ds->buffer_pos+=y-x; // put back data (HACK!)
if(len2>0){
if (((AVCodecContext *)sh_audio->context)->channels >= 5) {
int samplesize = av_get_bits_per_sample_format(((AVCodecContext *)
sh_audio->context)->sample_fmt) / 8;
reorder_channel_nch(buf, AF_CHANNEL_LAYOUT_LAVC_DEFAULT,
AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
((AVCodecContext *)sh_audio->context)->channels,
len2 / samplesize, samplesize);
}
//len=len2;break;
if(len<0) len=len2; else len+=len2;
buf+=len2;
maxlen -= len2;
sh_audio->pts_bytes += len2;
}
mp_dbg(MSGT_DECAUDIO,MSGL_DBG2,"Decoded %d -> %d \n",y,len2);
if (setup_format(sh_audio, sh_audio->context))
break;
}
return len;
}