Mirrors/mpv
1
0
Fork 0
mirror of https://github.com/mpv-player/mpv.git synced 2025-12-28 05:33:14 +00:00
Code Issues Packages Projects Releases Wiki Activity

Rename directories, move files (step 1 of 2) (does not compile)

Browse Source 
Tis drops the silly lib prefixes, and attempts to organize the tree in
a more logical way. Make the top-level directory less cluttered as
well.

Renames the following directories:
    libaf -> audio/filter
    libao2 -> audio/out
    libvo -> video/out
    libmpdemux -> demux

Split libmpcodecs:
    vf* -> video/filter
    vd*, dec_video.* -> video/decode
    mp_image*, img_format*, ... -> video/
    ad*, dec_audio.* -> audio/decode

libaf/format.* is moved to audio/ - this is similar to how mp_image.*
is located in video/.

Move most top-level .c/.h files to core. (talloc.c/.h is left on top-
level, because it's external.) Park some of the more annoying files
in compat/. Some of these are relicts from the time mplayer used
ffmpeg internals.

sub/ is not split, because it's too much of a mess (subtitle code is
mixed with OSD display and rendering).

Maybe the organization of core is not ideal: it mixes playback core
(like mplayer.c) and utility helpers (like bstr.c/h). Should the need
arise, the playback core will be moved somewhere else, while core
contains all helper and common code.
This commit is contained in:
wm4
2012-11-05 17:02:04 +01:00
parent bd48deba77
commit d4bdd0473d
278 changed files with 0 additions and 0 deletions
Download Patch File Download Diff File Expand all files Collapse all files

50
audio/decode/ad.c Normal file
View File

@@ -0,0 +1,50 @@
/*
* audio decoder interface
*
* 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 <string.h>
#include "config.h"
#include "stream/stream.h"
#include "libmpdemux/demuxer.h"
#include "libmpdemux/stheader.h"
#include "ad.h"
/* Missed vorbis, mad, dshow */
extern const ad_functions_t mpcodecs_ad_mpg123;
extern const ad_functions_t mpcodecs_ad_ffmpeg;
extern const ad_functions_t mpcodecs_ad_pcm;
extern const ad_functions_t mpcodecs_ad_dvdpcm;
extern const ad_functions_t mpcodecs_ad_spdif;
const ad_functions_t * const mpcodecs_ad_drivers[] =
{
#ifdef CONFIG_MPG123
&mpcodecs_ad_mpg123,
#endif
&mpcodecs_ad_ffmpeg,
&mpcodecs_ad_pcm,
&mpcodecs_ad_dvdpcm,
&mpcodecs_ad_spdif,
NULL
};

54
audio/decode/ad.h Normal file
View File

@@ -0,0 +1,54 @@
/*
* 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.
*/
#ifndef MPLAYER_AD_H
#define MPLAYER_AD_H
#include "mpc_info.h"
#include "libmpdemux/stheader.h"
typedef struct mp_codec_info ad_info_t;
/* interface of video decoder drivers */
typedef struct ad_functions
{
const ad_info_t *info;
int (*preinit)(sh_audio_t *sh);
int (*init)(sh_audio_t *sh);
void (*uninit)(sh_audio_t *sh);
int (*control)(sh_audio_t *sh,int cmd,void* arg, ...);
int (*decode_audio)(sh_audio_t *sh, unsigned char *buffer, int minlen,
int maxlen);
} ad_functions_t;
// NULL terminated array of all drivers
extern const ad_functions_t * const mpcodecs_ad_drivers[];
// fallback if ADCTRL_RESYNC not implemented: sh_audio->a_in_buffer_len=0;
#define ADCTRL_RESYNC_STREAM 1 // resync, called after seeking
// fallback if ADCTRL_SKIP not implemented: ds_fill_buffer(sh_audio->ds);
#define ADCTRL_SKIP_FRAME 2 // skip block/frame, called while seeking
// fallback if ADCTRL_QUERY_FORMAT not implemented: sh_audio->sample_format
#define ADCTRL_QUERY_FORMAT 3 // test for availabilty of a format
// fallback: use hw mixer in libao
#define ADCTRL_SET_VOLUME 4 // not used at the moment
#endif /* MPLAYER_AD_H */

162
audio/decode/ad_dvdpcm.c Normal file
View File

@@ -0,0 +1,162 @@
/*
* 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 "ad_internal.h"
static const ad_info_t info =
{
"Uncompressed DVD/VOB LPCM audio decoder",
"dvdpcm",
"Nick Kurshev",
"A'rpi",
""
};
LIBAD_EXTERN(dvdpcm)
static int init(sh_audio_t *sh)
{
/* DVD PCM Audio:*/
sh->i_bps = 0;
if(sh->codecdata_len==3){
// we have LPCM header:
unsigned char h=sh->codecdata[1];
sh->channels=1+(h&7);
switch((h>>4)&3){
case 0: sh->samplerate=48000;break;
case 1: sh->samplerate=96000;break;
case 2: sh->samplerate=44100;break;
case 3: sh->samplerate=32000;break;
}
switch ((h >> 6) & 3) {
case 0:
sh->sample_format = AF_FORMAT_S16_BE;
sh->samplesize = 2;
break;
case 1:
mp_tmsg(MSGT_DECAUDIO, MSGL_INFO, "Samples of this format are needed to improve support. Please contact the developers.\n");
sh->i_bps = sh->channels * sh->samplerate * 5 / 2;
case 2:
sh->sample_format = AF_FORMAT_S24_BE;
sh->samplesize = 3;
break;
default:
sh->sample_format = AF_FORMAT_S16_BE;
sh->samplesize = 2;
}
} else {
// use defaults:
sh->channels=2;
sh->samplerate=48000;
sh->sample_format = AF_FORMAT_S16_BE;
sh->samplesize = 2;
}
if (!sh->i_bps)
sh->i_bps = sh->samplesize * sh->channels * sh->samplerate;
return 1;
}
static int preinit(sh_audio_t *sh)
{
sh->audio_out_minsize=2048;
return 1;
}
static void uninit(sh_audio_t *sh)
{
}
static int control(sh_audio_t *sh,int cmd,void* arg, ...)
{
int skip;
switch(cmd)
{
case ADCTRL_SKIP_FRAME:
skip=sh->i_bps/16;
skip=skip&(~3);
demux_read_data(sh->ds,NULL,skip);
return CONTROL_TRUE;
}
return CONTROL_UNKNOWN;
}
static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
{
int j,len;
if (sh_audio->samplesize == 3) {
if (((sh_audio->codecdata[1] >> 6) & 3) == 1) {
// 20 bit
// not sure if the "& 0xf0" and "<< 4" are the right way around
// can somebody clarify?
for (j = 0; j < minlen; j += 12) {
char tmp[10];
len = demux_read_data(sh_audio->ds, tmp, 10);
if (len < 10) break;
// first sample
buf[j + 0] = tmp[0];
buf[j + 1] = tmp[1];
buf[j + 2] = tmp[8] & 0xf0;
// second sample
buf[j + 3] = tmp[2];
buf[j + 4] = tmp[3];
buf[j + 5] = tmp[8] << 4;
// third sample
buf[j + 6] = tmp[4];
buf[j + 7] = tmp[5];
buf[j + 8] = tmp[9] & 0xf0;
// fourth sample
buf[j + 9] = tmp[6];
buf[j + 10] = tmp[7];
buf[j + 11] = tmp[9] << 4;
}
len = j;
} else {
// 24 bit
for (j = 0; j < minlen; j += 12) {
char tmp[12];
len = demux_read_data(sh_audio->ds, tmp, 12);
if (len < 12) break;
// first sample
buf[j + 0] = tmp[0];
buf[j + 1] = tmp[1];
buf[j + 2] = tmp[8];
// second sample
buf[j + 3] = tmp[2];
buf[j + 4] = tmp[3];
buf[j + 5] = tmp[9];
// third sample
buf[j + 6] = tmp[4];
buf[j + 7] = tmp[5];
buf[j + 8] = tmp[10];
// fourth sample
buf[j + 9] = tmp[6];
buf[j + 10] = tmp[7];
buf[j + 11] = tmp[11];
}
len = j;
}
} else
len=demux_read_data(sh_audio->ds,buf,(minlen+3)&(~3));
return len;
}

46
audio/decode/ad_internal.h Normal file
View File

@@ -0,0 +1,46 @@
/*
* 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.
*/
#ifndef MPLAYER_AD_INTERNAL_H
#define MPLAYER_AD_INTERNAL_H
#include "codec-cfg.h"
#include "libaf/format.h"
#include "stream/stream.h"
#include "libmpdemux/demuxer.h"
#include "libmpdemux/stheader.h"
#include "ad.h"
static int init(sh_audio_t *sh);
static int preinit(sh_audio_t *sh);
static void uninit(sh_audio_t *sh);
static int control(sh_audio_t *sh,int cmd,void* arg, ...);
static int decode_audio(sh_audio_t *sh,unsigned char *buffer,int minlen,int maxlen);
#define LIBAD_EXTERN(x) const ad_functions_t mpcodecs_ad_##x = {\
&info,\
preinit,\
init,\
uninit,\
control,\
decode_audio\
};
#endif /* MPLAYER_AD_INTERNAL_H */

413
audio/decode/ad_lavc.c Normal file
View File

@@ -0,0 +1,413 @@
/*
* 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 <stdbool.h>
#include <assert.h>
#include <libavcodec/avcodec.h>
#include <libavutil/opt.h>
#include "talloc.h"
#include "config.h"
#include "mp_msg.h"
#include "options.h"
#include "ad_internal.h"
#include "libaf/reorder_ch.h"
#include "mpbswap.h"
static const ad_info_t info =
{
"libavcodec audio decoders",
"ffmpeg",
"",
"",
"",
.print_name = "libavcodec",
};
LIBAD_EXTERN(ffmpeg)
struct priv {
AVCodecContext *avctx;
AVFrame *avframe;
char *output;
char *output_packed; // used by deplanarize to store packed audio samples
int output_left;
int unitsize;
int previous_data_left; // input demuxer packet data
};
static int preinit(sh_audio_t *sh)
{
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 (av_get_packed_sample_fmt(lavc_context->sample_fmt)) {
case AV_SAMPLE_FMT_U8: sample_format = AF_FORMAT_U8; break;
case AV_SAMPLE_FMT_S16: sample_format = AF_FORMAT_S16_NE; break;
case AV_SAMPLE_FMT_S32: sample_format = AF_FORMAT_S32_NE; break;
case AV_SAMPLE_FMT_FLT: sample_format = AF_FORMAT_FLOAT_NE; break;
default:
mp_msg(MSGT_DECAUDIO, MSGL_FATAL, "Unsupported sample format\n");
sample_format = AF_FORMAT_UNKNOWN;
}
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;
if (sh_audio->codec->dll) {
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;
}
} else if (!sh_audio->libav_codec_id) {
mp_tmsg(MSGT_DECAUDIO, MSGL_INFO, "No Libav codec ID known. "
"Generic lavc decoder is not applicable.\n");
return 0;
} else {
lavc_codec = avcodec_find_decoder(sh_audio->libav_codec_id);
if (!lavc_codec) {
mp_tmsg(MSGT_DECAUDIO, MSGL_INFO, "Libavcodec has no decoder "
"for this codec\n");
return 0;
}
}
sh_audio->codecname = lavc_codec->long_name;
if (!sh_audio->codecname)
sh_audio->codecname = lavc_codec->name;
struct priv *ctx = talloc_zero(NULL, struct priv);
sh_audio->context = ctx;
lavc_context = avcodec_alloc_context3(lavc_codec);
ctx->avctx = lavc_context;
ctx->avframe = avcodec_alloc_frame();
// Always try to set - option only exists for AC3 at the moment
av_opt_set_double(lavc_context, "drc_scale", opts->drc_level,
AV_OPT_SEARCH_CHILDREN);
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
if (sh_audio->gsh->lavf_codec_tag)
lavc_context->codec_tag = sh_audio->gsh->lavf_codec_tag;
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_open2(lavc_context, lavc_codec, NULL) < 0) {
mp_tmsg(MSGT_DECAUDIO, MSGL_ERR, "Could not open codec.\n");
uninit(sh_audio);
return 0;
}
mp_msg(MSGT_DECAUDIO, MSGL_V, "INFO: libavcodec \"%s\" init OK!\n",
lavc_codec->name);
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");
uninit(sh_audio);
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 (av_get_packed_sample_fmt(lavc_context->sample_fmt)) {
case AV_SAMPLE_FMT_U8:
case AV_SAMPLE_FMT_S16:
case AV_SAMPLE_FMT_S32:
case AV_SAMPLE_FMT_FLT:
break;
default:
uninit(sh_audio);
return 0;
}
return 1;
}
static void uninit(sh_audio_t *sh)
{
sh->codecname = NULL;
struct priv *ctx = sh->context;
if (!ctx)
return;
AVCodecContext *lavc_context = ctx->avctx;
if (lavc_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);
}
avcodec_free_frame(&ctx->avframe);
talloc_free(ctx);
sh->context = NULL;
}
static int control(sh_audio_t *sh, int cmd, void *arg, ...)
{
struct priv *ctx = sh->context;
switch (cmd) {
case ADCTRL_RESYNC_STREAM:
avcodec_flush_buffers(ctx->avctx);
ds_clear_parser(sh->ds);
ctx->previous_data_left = 0;
ctx->output_left = 0;
return CONTROL_TRUE;
}
return CONTROL_UNKNOWN;
}
static av_always_inline void deplanarize(struct sh_audio *sh)
{
struct priv *priv = sh->context;
size_t bps = av_get_bytes_per_sample(priv->avctx->sample_fmt);
size_t nb_samples = priv->avframe->nb_samples;
size_t channels = priv->avctx->channels;
size_t size = bps * nb_samples * channels;
if (talloc_get_size(priv->output_packed) != size)
priv->output_packed =
talloc_realloc_size(priv, priv->output_packed, size);
size_t offset = 0;
unsigned char *output_ptr = priv->output_packed;
unsigned char **src = priv->avframe->data;
for (size_t s = 0; s < nb_samples; s++) {
for (size_t c = 0; c < channels; c++) {
memcpy(output_ptr, src[c] + offset, bps);
output_ptr += bps;
}
offset += bps;
}
priv->output = priv->output_packed;
}
static int decode_new_packet(struct sh_audio *sh)
{
struct priv *priv = sh->context;
AVCodecContext *avctx = priv->avctx;
double pts = MP_NOPTS_VALUE;
int insize;
bool packet_already_used = priv->previous_data_left;
struct demux_packet *mpkt = ds_get_packet2(sh->ds,
priv->previous_data_left);
unsigned char *start;
if (!mpkt) {
assert(!priv->previous_data_left);
start = NULL;
insize = 0;
ds_parse(sh->ds, &start, &insize, pts, 0);
if (insize <= 0)
return -1; // error or EOF
} else {
assert(mpkt->len >= priv->previous_data_left);
if (!priv->previous_data_left) {
priv->previous_data_left = mpkt->len;
pts = mpkt->pts;
}
insize = priv->previous_data_left;
start = mpkt->buffer + mpkt->len - priv->previous_data_left;
int consumed = ds_parse(sh->ds, &start, &insize, pts, 0);
priv->previous_data_left -= consumed;
priv->previous_data_left = FFMAX(priv->previous_data_left, 0);
}
AVPacket pkt;
av_init_packet(&pkt);
pkt.data = start;
pkt.size = insize;
if (mpkt && mpkt->avpacket) {
pkt.side_data = mpkt->avpacket->side_data;
pkt.side_data_elems = mpkt->avpacket->side_data_elems;
}
if (pts != MP_NOPTS_VALUE && !packet_already_used) {
sh->pts = pts;
sh->pts_bytes = 0;
}
int got_frame = 0;
int ret = avcodec_decode_audio4(avctx, priv->avframe, &got_frame, &pkt);
// LATM may need many packets to find mux info
if (ret == AVERROR(EAGAIN))
return 0;
if (ret < 0) {
mp_msg(MSGT_DECAUDIO, MSGL_V, "lavc_audio: error\n");
return -1;
}
// The "insize >= ret" test is sanity check against decoder overreads
if (!sh->parser && insize >= ret)
priv->previous_data_left = insize - ret;
if (!got_frame)
return 0;
uint64_t unitsize = (uint64_t)av_get_bytes_per_sample(avctx->sample_fmt) *
avctx->channels;
if (unitsize > 100000)
abort();
priv->unitsize = unitsize;
uint64_t output_left = unitsize * priv->avframe->nb_samples;
if (output_left > 500000000)
abort();
priv->output_left = output_left;
if (av_sample_fmt_is_planar(avctx->sample_fmt) && avctx->channels > 1) {
deplanarize(sh);
} else {
priv->output = priv->avframe->data[0];
}
mp_dbg(MSGT_DECAUDIO, MSGL_DBG2, "Decoded %d -> %d \n", insize,
priv->output_left);
return 0;
}
static int decode_audio(sh_audio_t *sh_audio, unsigned char *buf, int minlen,
int maxlen)
{
struct priv *priv = sh_audio->context;
AVCodecContext *avctx = priv->avctx;
int len = -1;
while (len < minlen) {
if (!priv->output_left) {
if (decode_new_packet(sh_audio) < 0)
break;
continue;
}
if (setup_format(sh_audio, avctx))
return len;
int size = (minlen - len + priv->unitsize - 1);
size -= size % priv->unitsize;
size = FFMIN(size, priv->output_left);
if (size > maxlen)
abort();
memcpy(buf, priv->output, size);
priv->output += size;
priv->output_left -= size;
if (avctx->channels >= 5) {
int samplesize = av_get_bytes_per_sample(avctx->sample_fmt);
reorder_channel_nch(buf, AF_CHANNEL_LAYOUT_LAVC_DEFAULT,
AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
avctx->channels,
size / samplesize, samplesize);
}
if (len < 0)
len = size;
else
len += size;
buf += size;
maxlen -= size;
sh_audio->pts_bytes += size;
}
return len;
}

489
audio/decode/ad_mpg123.c Normal file
View File

@@ -0,0 +1,489 @@
/*
* MPEG 1.0/2.0/2.5 audio layer I, II, III decoding with libmpg123
*
* Copyright (C) 2010-2012 Thomas Orgis <thomas@orgis.org>
*
* 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 "ad_internal.h"
static const ad_info_t info = {
"MPEG 1.0/2.0/2.5 layers I, II, III",
"mpg123",
"Thomas Orgis",
"mpg123.org",
"High-performance decoder using libmpg123."
};
LIBAD_EXTERN(mpg123)
/* Reducing the ifdeffery to two main variants:
* 1. most compatible to any libmpg123 version
* 2. fastest variant with recent libmpg123 (>=1.14)
* Running variant 2 on older libmpg123 versions may work in
* principle, but is not supported.
* So, please leave the check for MPG123_API_VERSION there, m-kay?
*/
#include <mpg123.h>
/* Enable faster mode of operation with newer libmpg123, avoiding
* unnecessary memcpy() calls. */
#if (defined MPG123_API_VERSION) && (MPG123_API_VERSION >= 33)
#define AD_MPG123_FRAMEWISE
#endif
/* Switch for updating bitrate info of VBR files. Not essential. */
#define AD_MPG123_MEAN_BITRATE
/* Funny thing, that. I assume I shall use it for selecting mpg123 channels.
* Please correct me if I guessed wrong. */
extern int fakemono;
struct ad_mpg123_context {
mpg123_handle *handle;
#ifdef AD_MPG123_MEAN_BITRATE
/* Running mean for bit rate, stream length estimation. */
float mean_rate;
unsigned int mean_count;
/* Time delay for updates. */
short delay;
#endif
/* If the stream is actually VBR. */
char vbr;
};
/* This initializes libmpg123 and prepares the handle, including funky
* parameters. */
static int preinit(sh_audio_t *sh)
{
int err, flag;
struct ad_mpg123_context *con;
/* Assumption: You always call preinit + init + uninit, on every file.
* But you stop at preinit in case it fails.
* If that is not true, one must ensure not to call mpg123_init / exit
* twice in a row. */
if (mpg123_init() != MPG123_OK)
return 0;
sh->context = malloc(sizeof(struct ad_mpg123_context));
con = sh->context;
/* Auto-choice of optimized decoder (first argument NULL). */
con->handle = mpg123_new(NULL, &err);
if (!con->handle)
goto bad_end;
/* Guessing here: Default value triggers forced upmix of mono to stereo. */
flag = fakemono == 0 ? MPG123_FORCE_STEREO :
fakemono == 1 ? MPG123_MONO_LEFT :
fakemono == 2 ? MPG123_MONO_RIGHT : 0;
if (mpg123_param(con->handle, MPG123_ADD_FLAGS, flag, 0.0) != MPG123_OK)
goto bad_end;
/* Basic settings.
* Don't spill messages, enable better resync with non-seekable streams.
* Give both flags individually without error checking to keep going with
* old libmpg123. Generally, it is not fatal if the flags are not
* honored */
mpg123_param(con->handle, MPG123_ADD_FLAGS, MPG123_QUIET, 0.0);
/* Do not bail out on malformed streams at all.
* MPlayer does not handle a decoder throwing the towel on crappy input. */
mpg123_param(con->handle, MPG123_RESYNC_LIMIT, -1, 0.0);
/* Open decisions: Configure libmpg123 to force encoding (or stay open about
* library builds that support only float or int32 output), (de)configure
* gapless decoding (won't work with seeking in MPlayer, though).
* Don't forget to eventually enable ReplayGain/RVA support, too.
* Let's try to run with the default for now. */
/* That would produce floating point output.
* You can get 32 and 24 bit ints, even 8 bit via format matrix. */
/* mpg123_param(con->handle, MPG123_ADD_FLAGS, MPG123_FORCE_FLOAT, 0.); */
/* Example for RVA choice (available since libmpg123 1.0.0):
mpg123_param(con->handle, MPG123_RVA, MPG123_RVA_MIX, 0.0) */
#ifdef AD_MPG123_FRAMEWISE
/* Prevent funky automatic resampling.
* This way, we can be sure that one frame will never produce
* more than 1152 stereo samples. */
mpg123_param(con->handle, MPG123_REMOVE_FLAGS, MPG123_AUTO_RESAMPLE, 0.);
#else
/* Older mpg123 is vulnerable to concatenated streams when gapless cutting
* is enabled (will only play the jingle of a badly constructed radio
* stream). The versions using framewise decoding are fine with that. */
mpg123_param(con->handle, MPG123_REMOVE_FLAGS, MPG123_GAPLESS, 0.);
#endif
return 1;
bad_end:
if (!con->handle)
mp_msg(MSGT_DECAUDIO, MSGL_ERR, "mpg123 preinit error: %s\n",
mpg123_plain_strerror(err));
else
mp_msg(MSGT_DECAUDIO, MSGL_ERR, "mpg123 preinit error: %s\n",
mpg123_strerror(con->handle));
if (con->handle)
mpg123_delete(con->handle);
mpg123_exit();
free(sh->context);
sh->context = NULL;
return 0;
}
/* Compute bitrate from frame size. */
static int compute_bitrate(struct mpg123_frameinfo *i)
{
static const int samples_per_frame[4][4] = {
{-1, 384, 1152, 1152}, /* MPEG 1 */
{-1, 384, 1152, 576}, /* MPEG 2 */
{-1, 384, 1152, 576}, /* MPEG 2.5 */
{-1, -1, -1, -1}, /* Unknown */
};
return (int) ((i->framesize + 4) * 8 * i->rate * 0.001 /
samples_per_frame[i->version][i->layer] + 0.5);
}
/* Opted against the header printout from old mp3lib, too much
* irrelevant info. This is modelled after the mpg123 app's
* standard output line.
* If more verbosity is demanded, one can add more detail and
* also throw in ID3v2 info which libmpg123 collects anyway. */
static void print_header_compact(struct mpg123_frameinfo *i)
{
static const char *smodes[5] = {
"stereo", "joint-stereo", "dual-channel", "mono", "invalid"
};
static const char *layers[4] = {
"Unknown", "I", "II", "III"
};
static const char *versions[4] = {
"1.0", "2.0", "2.5", "x.x"
};
mp_msg(MSGT_DECAUDIO, MSGL_V, "MPEG %s layer %s, ",
versions[i->version], layers[i->layer]);
switch (i->vbr) {
case MPG123_CBR:
if (i->bitrate)
mp_msg(MSGT_DECAUDIO, MSGL_V, "%d kbit/s", i->bitrate);
else
mp_msg(MSGT_DECAUDIO, MSGL_V, "%d kbit/s (free format)",
compute_bitrate(i));
break;
case MPG123_VBR:
mp_msg(MSGT_DECAUDIO, MSGL_V, "VBR");
break;
case MPG123_ABR:
mp_msg(MSGT_DECAUDIO, MSGL_V, "%d kbit/s ABR", i->abr_rate);
break;
default:
mp_msg(MSGT_DECAUDIO, MSGL_V, "???");
}
mp_msg(MSGT_DECAUDIO, MSGL_V, ", %ld Hz %s\n", i->rate,
smodes[i->mode]);
}
/* This tries to extract a requested amount of decoded data.
* Even when you request 0 bytes, it will feed enough input so that
* the decoder _could_ have delivered something.
* Returns byte count >= 0, -1 on error.
*
* Thoughts on exact pts keeping:
* We have to assume that MPEG frames are cut in pieces by packet boundaries.
* Also, it might be possible that the first packet does not contain enough
* data to ensure initial stream sync... or re-sync on erroneous streams.
* So we need something robust to relate the decoded byte count to the correct
* time stamp. This is tricky, though. From the outside, you cannot tell if,
* after having fed two packets until the first output arrives, one should
* start counting from the first packet's pts or the second packet's.
* So, let's just count from the last fed package's pts. If the packets are
* exactly cut to MPEG frames, this will cause one frame mismatch in the
* beginning (when mpg123 peeks ahead for the following header), but will
* be corrected with the third frame already. One might add special code to
* not increment the base pts past the first packet's after a resync before
* the first decoded bytes arrived. */
static int decode_a_bit(sh_audio_t *sh, unsigned char *buf, int count)
{
int ret = MPG123_OK;
int got = 0;
struct ad_mpg123_context *con = sh->context;
/* There will be one MPG123_NEW_FORMAT message on first open.
* This will be handled in init(). */
do {
size_t got_now = 0;
/* Feed the decoder. This will only fire from the second round on. */
if (ret == MPG123_NEED_MORE) {
int incount;
double pts;
unsigned char *inbuf;
/* Feed more input data. */
incount = ds_get_packet_pts(sh->ds, &inbuf, &pts);
if (incount <= 0)
break; /* Apparently that's it. EOF. */
/* Next bytes from that presentation time. */
if (pts != MP_NOPTS_VALUE) {
sh->pts = pts;
sh->pts_bytes = 0;
}
#ifdef AD_MPG123_FRAMEWISE
/* Have to use mpg123_feed() to avoid decoding here. */
ret = mpg123_feed(con->handle, inbuf, incount);
#else
/* Do not use mpg123_feed(), added in later libmpg123 versions. */
ret = mpg123_decode(con->handle, inbuf, incount, NULL, 0, NULL);
#endif
if (ret == MPG123_ERR)
break;
}
/* Theoretically, mpg123 could return MPG123_DONE, so be prepared.
* Should not happen in our usage, but it is a valid return code. */
else if (ret == MPG123_ERR || ret == MPG123_DONE)
break;
/* Try to decode a bit. This is the return value that counts
* for the loop condition. */
#ifdef AD_MPG123_FRAMEWISE
if (!buf) { /* fake call just for feeding to get format */
ret = mpg123_getformat(con->handle, NULL, NULL, NULL);
} else { /* This is the decoding. One frame at a time. */
ret = mpg123_replace_buffer(con->handle, buf, count);
if (ret == MPG123_OK)
ret = mpg123_decode_frame(con->handle, NULL, NULL, &got_now);
}
#else
ret = mpg123_decode(con->handle, NULL, 0, buf + got, count - got,
&got_now);
#endif
got += got_now;
sh->pts_bytes += got_now;
#ifdef AD_MPG123_FRAMEWISE
} while (ret == MPG123_NEED_MORE || (got == 0 && count != 0));
#else
} while (ret == MPG123_NEED_MORE || got < count);
#endif
if (ret == MPG123_ERR) {
mp_msg(MSGT_DECAUDIO, MSGL_ERR, "mpg123 decoding failed: %s\n",
mpg123_strerror(con->handle));
mpg123_close(con->handle);
return -1;
}
return got;
}
/* Close, reopen stream. Feed data until we know the format of the stream.
* 1 on success, 0 on error */
static int reopen_stream(sh_audio_t *sh)
{
struct ad_mpg123_context *con = (struct ad_mpg123_context*) sh->context;
mpg123_close(con->handle);
/* No resetting of the context:
* We do not want to loose the mean bitrate data. */
/* Open and make sure we have fed enough data to get stream properties. */
if (MPG123_OK == mpg123_open_feed(con->handle) &&
/* Feed data until mpg123 is ready (has found stream beginning). */
!decode_a_bit(sh, NULL, 0)) {
return 1;
} else {
mp_msg(MSGT_DECAUDIO, MSGL_ERR,
"mpg123 failed to reopen stream: %s\n",
mpg123_strerror(con->handle));
mpg123_close(con->handle);
return 0;
}
}
/* Now we really start accessing some data and determining file format.
* Paranoia note: The mpg123_close() on errors is not really necessary,
* But it ensures that we don't accidentally continue decoding with a
* bad state (possibly interpreting the format badly or whatnot). */
static int init(sh_audio_t *sh)
{
long rate = 0;
int channels = 0;
int encoding = 0;
mpg123_id3v2 *v2;
struct mpg123_frameinfo finfo;
struct ad_mpg123_context *con = sh->context;
/* We're open about any output format that libmpg123 will suggest.
* Note that a standard build will always default to 16 bit signed and
* the native sample rate of the file. */
if (MPG123_OK == mpg123_format_all(con->handle) &&
reopen_stream(sh) &&
MPG123_OK == mpg123_getformat(con->handle, &rate, &channels, &encoding) &&
/* Forbid the format to change later on. */
MPG123_OK == mpg123_format_none(con->handle) &&
MPG123_OK == mpg123_format(con->handle, rate, channels, encoding) &&
/* Get MPEG header info. */
MPG123_OK == mpg123_info(con->handle, &finfo) &&
/* Since we queried format, mpg123 should have read past ID3v2 tags.
* We need to decide if printing of UTF-8 encoded text info is wanted. */
MPG123_OK == mpg123_id3(con->handle, NULL, &v2)) {
/* If we are here, we passed all hurdles. Yay! Extract the info. */
print_header_compact(&finfo);
/* Do we want to print out the UTF-8 Id3v2 info?
if (v2)
print_id3v2(v2); */
/* Have kb/s, want B/s
* For VBR, the first frame will be a bad estimate. */
sh->i_bps = (finfo.bitrate ? finfo.bitrate : compute_bitrate(&finfo))
* 1000 / 8;
#ifdef AD_MPG123_MEAN_BITRATE
con->delay = 1;
con->mean_rate = 0.;
con->mean_count = 0;
#endif
con->vbr = (finfo.vbr != MPG123_CBR);
sh->channels = channels;
sh->samplerate = rate;
/* Without external force, mpg123 will always choose signed encoding,
* and non-16-bit only on builds that don't support it.
* Be reminded that it doesn't matter to the MPEG file what encoding
* is produced from it. */
switch (encoding) {
case MPG123_ENC_SIGNED_8:
sh->sample_format = AF_FORMAT_S8;
sh->samplesize = 1;
break;
case MPG123_ENC_SIGNED_16:
sh->sample_format = AF_FORMAT_S16_NE;
sh->samplesize = 2;
break;
/* To stay compatible with the oldest libmpg123 headers, do not rely
* on float and 32 bit encoding symbols being defined.
* Those formats came later */
case 0x1180: /* MPG123_ENC_SIGNED_32 */
sh->sample_format = AF_FORMAT_S32_NE;
sh->samplesize = 4;
break;
case 0x200: /* MPG123_ENC_FLOAT_32 */
sh->sample_format = AF_FORMAT_FLOAT_NE;
sh->samplesize = 4;
break;
default:
mp_msg(MSGT_DECAUDIO, MSGL_ERR,
"Bad encoding from mpg123: %i.\n", encoding);
mpg123_close(con->handle);
return 0;
}
#ifdef AD_MPG123_FRAMEWISE
/* Going to decode directly to MPlayer's memory. It is important
* to have MPG123_AUTO_RESAMPLE disabled for the buffer size
* being an all-time limit. */
sh->audio_out_minsize = 1152 * 2 * sh->samplesize;
#endif
return 1;
} else {
mp_msg(MSGT_DECAUDIO, MSGL_ERR, "mpg123 init error: %s\n",
mpg123_strerror(con->handle));
mpg123_close(con->handle);
return 0;
}
}
static void uninit(sh_audio_t *sh)
{
struct ad_mpg123_context *con = (struct ad_mpg123_context*) sh->context;
mpg123_close(con->handle);
mpg123_delete(con->handle);
free(sh->context);
sh->context = NULL;
mpg123_exit();
}
#ifdef AD_MPG123_MEAN_BITRATE
/* Update mean bitrate. This could be dropped if accurate time display
* on audio file playback is not desired. */
static void update_info(sh_audio_t *sh)
{
struct ad_mpg123_context *con = sh->context;
if (con->vbr && --con->delay < 1) {
struct mpg123_frameinfo finfo;
if (MPG123_OK == mpg123_info(con->handle, &finfo)) {
if (++con->mean_count > ((unsigned int) -1) / 2)
con->mean_count = ((unsigned int) -1) / 4;
/* Might not be numerically optimal, but works fine enough. */
con->mean_rate = ((con->mean_count - 1) * con->mean_rate +
finfo.bitrate) / con->mean_count;
sh->i_bps = (int) (con->mean_rate * 1000 / 8);
con->delay = 10;
}
}
}
#endif
static int decode_audio(sh_audio_t *sh, unsigned char *buf, int minlen,
int maxlen)
{
int bytes;
bytes = decode_a_bit(sh, buf, maxlen);
if (bytes == 0)
return -1; /* EOF */
#ifdef AD_MPG123_MEAN_BITRATE
update_info(sh);
#endif
return bytes;
}
static int control(sh_audio_t *sh, int cmd, void *arg, ...)
{
switch (cmd) {
case ADCTRL_RESYNC_STREAM:
/* Close/reopen the stream for mpg123 to make sure it doesn't
* think that it still knows the exact stream position.
* Otherwise, we would have funny effects from the gapless code.
* Oh, and it helps to minimize artifacts from jumping in the stream. */
if (reopen_stream(sh)) {
#ifdef AD_MPG123_MEAN_BITRATE
update_info(sh);
#endif
return CONTROL_TRUE;
} else {
/* MPlayer ignores this case! It just keeps on decoding.
* So we have to make sure resync never fails ... */
mp_msg(MSGT_DECAUDIO, MSGL_ERR,
"mpg123 cannot reopen stream for resync.\n");
return CONTROL_FALSE;
}
break;
}
return CONTROL_UNKNOWN;
}

220
audio/decode/ad_pcm.c Normal file
View File

@@ -0,0 +1,220 @@
/*
* 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 <stdbool.h>
#include <libavutil/common.h>
#include "talloc.h"
#include "config.h"
#include "ad_internal.h"
#include "libaf/format.h"
#include "libaf/reorder_ch.h"
static const ad_info_t info = {
"Uncompressed PCM audio decoder",
"pcm",
"Nick Kurshev",
"A'rpi",
""
};
struct ad_pcm_context {
unsigned char *buffer;
int buffer_pos;
int buffer_len;
int buffer_size;
};
LIBAD_EXTERN(pcm)
static int init(sh_audio_t * sh_audio)
{
WAVEFORMATEX *h = sh_audio->wf;
if (!h)
return 0;
sh_audio->i_bps = h->nAvgBytesPerSec;
sh_audio->channels = h->nChannels;
sh_audio->samplerate = h->nSamplesPerSec;
sh_audio->samplesize = (h->wBitsPerSample + 7) / 8;
sh_audio->sample_format = AF_FORMAT_S16_LE; // default
switch (sh_audio->format) { /* hardware formats: */
case 0x0:
case 0x1: // Microsoft PCM
case 0xfffe: // Extended
switch (sh_audio->samplesize) {
case 1: sh_audio->sample_format = AF_FORMAT_U8; break;
case 2: sh_audio->sample_format = AF_FORMAT_S16_LE; break;
case 3: sh_audio->sample_format = AF_FORMAT_S24_LE; break;
case 4: sh_audio->sample_format = AF_FORMAT_S32_LE; break;
}
break;
case 0x3: // IEEE float
sh_audio->sample_format = AF_FORMAT_FLOAT_LE;
break;
case 0x6: sh_audio->sample_format = AF_FORMAT_A_LAW; break;
case 0x7: sh_audio->sample_format = AF_FORMAT_MU_LAW; break;
case 0x11: sh_audio->sample_format = AF_FORMAT_IMA_ADPCM; break;
case 0x50: sh_audio->sample_format = AF_FORMAT_MPEG2; break;
/* case 0x2000: sh_audio->sample_format=AFMT_AC3; */
case 0x20776172: // 'raw '
sh_audio->sample_format = AF_FORMAT_S16_BE;
if (sh_audio->samplesize == 1)
sh_audio->sample_format = AF_FORMAT_U8;
break;
case 0x736F7774: // 'twos'
sh_audio->sample_format = AF_FORMAT_S16_BE;
// intended fall-through
case 0x74776F73: // 'sowt'
if (sh_audio->samplesize == 1)
sh_audio->sample_format = AF_FORMAT_S8;
break;
case 0x32336c66: // 'fl32', bigendian float32
case 0x32334C46: // 'FL32', bigendian float32 in aiff
sh_audio->sample_format = AF_FORMAT_FLOAT_BE;
sh_audio->samplesize = 4;
break;
case 0x666c3332: // '23lf', little endian float32, MPlayer internal fourCC
case 0x6D63706C: // 'lpcm'
sh_audio->sample_format = AF_FORMAT_FLOAT_LE;
sh_audio->samplesize = 4;
break;
/* case 0x34366c66: // 'fl64', bigendian float64
sh_audio->sample_format=AF_FORMAT_FLOAT_BE;
sh_audio->samplesize=8;
break;
case 0x666c3634: // '46lf', little endian float64, MPlayer internal fourCC
sh_audio->sample_format=AF_FORMAT_FLOAT_LE;
sh_audio->samplesize=8;
break;*/
case 0x34326e69: // 'in24', bigendian int24
sh_audio->sample_format = AF_FORMAT_S24_BE;
sh_audio->samplesize = 3;
break;
case 0x696e3234: // '42ni', little endian int24, MPlayer internal fourCC
sh_audio->sample_format = AF_FORMAT_S24_LE;
sh_audio->samplesize = 3;
break;
case 0x32336e69: // 'in32', bigendian int32
sh_audio->sample_format = AF_FORMAT_S32_BE;
sh_audio->samplesize = 4;
break;
case 0x696e3332: // '23ni', little endian int32, MPlayer internal fourCC
sh_audio->sample_format = AF_FORMAT_S32_LE;
sh_audio->samplesize = 4;
break;
case MKTAG('M', 'P', 'a', 'f'):
sh_audio->sample_format = h->wFormatTag;
sh_audio->samplesize = (af_fmt2bits(sh_audio->sample_format) + 7) / 8;
break;
default:
if (sh_audio->samplesize != 2)
sh_audio->sample_format = AF_FORMAT_U8;
}
if (!sh_audio->samplesize) // this would cause MPlayer to hang later
sh_audio->samplesize = 2;
sh_audio->context = talloc_zero(NULL, struct ad_pcm_context);
return 1;
}
static int preinit(sh_audio_t *sh)
{
sh->audio_out_minsize = 2048;
return 1;
}
static void uninit(sh_audio_t *sh)
{
talloc_free(sh->context);
}
static int control(sh_audio_t *sh, int cmd, void *arg, ...)
{
struct ad_pcm_context *ctx = sh->context;
int skip;
switch (cmd) {
case ADCTRL_RESYNC_STREAM:
ctx->buffer_len = 0;
return true;
case ADCTRL_SKIP_FRAME:
skip = sh->i_bps / 16;
skip = skip & (~3);
demux_read_data(sh->ds, NULL, skip);
return CONTROL_TRUE;
}
return CONTROL_UNKNOWN;
}
static int decode_audio(sh_audio_t *sh_audio, unsigned char *buf, int minlen,
int maxlen)
{
int unitsize = sh_audio->channels * sh_audio->samplesize;
minlen = (minlen + unitsize - 1) / unitsize * unitsize;
if (minlen > maxlen)
// if someone needs hundreds of channels adjust audio_out_minsize
// based on channels in preinit()
return -1;
int len = 0;
struct ad_pcm_context *ctx = sh_audio->context;
while (len < minlen) {
if (ctx->buffer_len - ctx->buffer_pos <= 0) {
double pts;
unsigned char *ptr;
int plen = ds_get_packet_pts(sh_audio->ds, &ptr, &pts);
if (plen < 0)
break;
if (ctx->buffer_size < plen) {
talloc_free(ctx->buffer);
ctx->buffer = talloc_size(ctx, plen);
ctx->buffer_size = plen;
}
memcpy(ctx->buffer, ptr, plen);
ctx->buffer_len = plen;
ctx->buffer_pos = 0;
if (pts != MP_NOPTS_VALUE) {
sh_audio->pts = pts;
sh_audio->pts_bytes = 0;
}
}
int from_stored = ctx->buffer_len - ctx->buffer_pos;
if (from_stored > minlen - len)
from_stored = minlen - len;
memcpy(buf + len, ctx->buffer + ctx->buffer_pos, from_stored);
ctx->buffer_pos += from_stored;
sh_audio->pts_bytes += from_stored;
len += from_stored;
}
if (len % unitsize) {
mp_msg(MSGT_DECAUDIO, MSGL_WARN, "[ad_pcm] discarding partial sample "
"at end\n");
len -= len % unitsize;
}
if (len == 0)
len = -1; // The loop above only exits at error/EOF
if (len > 0 && sh_audio->channels >= 5) {
reorder_channel_nch(buf, AF_CHANNEL_LAYOUT_WAVEEX_DEFAULT,
AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
sh_audio->channels, len / sh_audio->samplesize,
sh_audio->samplesize);
}
return len;
}

310
audio/decode/ad_spdif.c Normal file
View File

@@ -0,0 +1,310 @@
/*
* 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 <string.h>
#include <libavformat/avformat.h>
#include <libavcodec/avcodec.h>
#include <libavutil/opt.h>
#include "config.h"
#include "mp_msg.h"
#include "ad_internal.h"
static const ad_info_t info = {
"libavformat/spdifenc audio pass-through decoder.",
"spdif",
"Naoya OYAMA",
"Naoya OYAMA",
"For ALL hardware decoders"
};
LIBAD_EXTERN(spdif)
#define FILENAME_SPDIFENC "spdif"
#define OUTBUF_SIZE 65536
struct spdifContext {
AVFormatContext *lavf_ctx;
int iec61937_packet_size;
int out_buffer_len;
int out_buffer_size;
uint8_t *out_buffer;
uint8_t pb_buffer[OUTBUF_SIZE];
};
static int read_packet(void *p, uint8_t *buf, int buf_size)
{
// spdifenc does not use read callback.
return 0;
}
static int write_packet(void *p, uint8_t *buf, int buf_size)
{
int len;
struct spdifContext *ctx = p;
len = FFMIN(buf_size, ctx->out_buffer_size -ctx->out_buffer_len);
memcpy(&ctx->out_buffer[ctx->out_buffer_len], buf, len);
ctx->out_buffer_len += len;
return len;
}
static int64_t seek(void *p, int64_t offset, int whence)
{
// spdifenc does not use seek callback.
return 0;
}
static int preinit(sh_audio_t *sh)
{
sh->samplesize = 2;
return 1;
}
static int init(sh_audio_t *sh)
{
int i, x, in_size, srate, bps, *dtshd_rate;
unsigned char *start;
double pts;
static const struct {
const char *name; enum CodecID id;
} fmt_id_type[] = {
{ "aac" , CODEC_ID_AAC },
{ "ac3" , CODEC_ID_AC3 },
{ "dca" , CODEC_ID_DTS },
{ "eac3", CODEC_ID_EAC3 },
{ "mpa" , CODEC_ID_MP3 },
{ "thd" , CODEC_ID_TRUEHD },
{ NULL , 0 }
};
AVFormatContext *lavf_ctx = NULL;
AVStream *stream = NULL;
const AVOption *opt = NULL;
struct spdifContext *spdif_ctx = NULL;
spdif_ctx = av_mallocz(sizeof(*spdif_ctx));
if (!spdif_ctx)
goto fail;
spdif_ctx->lavf_ctx = avformat_alloc_context();
if (!spdif_ctx->lavf_ctx)
goto fail;
sh->context = spdif_ctx;
lavf_ctx = spdif_ctx->lavf_ctx;
lavf_ctx->oformat = av_guess_format(FILENAME_SPDIFENC, NULL, NULL);
if (!lavf_ctx->oformat)
goto fail;
lavf_ctx->priv_data = av_mallocz(lavf_ctx->oformat->priv_data_size);
if (!lavf_ctx->priv_data)
goto fail;
lavf_ctx->pb = avio_alloc_context(spdif_ctx->pb_buffer, OUTBUF_SIZE, 1, spdif_ctx,
read_packet, write_packet, seek);
if (!lavf_ctx->pb)
goto fail;
stream = avformat_new_stream(lavf_ctx, 0);
if (!stream)
goto fail;
lavf_ctx->duration = AV_NOPTS_VALUE;
lavf_ctx->start_time = AV_NOPTS_VALUE;
for (i = 0; fmt_id_type[i].name; i++) {
if (!strcmp(sh->codec->dll, fmt_id_type[i].name)) {
lavf_ctx->streams[0]->codec->codec_id = fmt_id_type[i].id;
break;
}
}
lavf_ctx->raw_packet_buffer_remaining_size = RAW_PACKET_BUFFER_SIZE;
if (AVERROR_PATCHWELCOME == lavf_ctx->oformat->write_header(lavf_ctx)) {
mp_msg(MSGT_DECAUDIO,MSGL_INFO,
"This codec is not supported by spdifenc.\n");
goto fail;
}
// get sample_rate & bitrate from parser
bps = srate = 0;
x = ds_get_packet_pts(sh->ds, &start, &pts);
in_size = x;
if (x <= 0) {
pts = MP_NOPTS_VALUE;
x = 0;
}
ds_parse(sh->ds, &start, &x, pts, 0);
if (x == 0) { // not enough buffer
srate = 48000; //fake value
bps = 768000/8; //fake value
} else if (sh->avctx) {
if (sh->avctx->sample_rate < 44100) {
mp_msg(MSGT_DECAUDIO,MSGL_INFO,
"This stream sample_rate[%d Hz] may be broken. "
"Force reset 48000Hz.\n",
sh->avctx->sample_rate);
srate = 48000; //fake value
} else
srate = sh->avctx->sample_rate;
bps = sh->avctx->bit_rate/8;
}
sh->ds->buffer_pos -= in_size;
switch (lavf_ctx->streams[0]->codec->codec_id) {
case CODEC_ID_AAC:
spdif_ctx->iec61937_packet_size = 16384;
sh->sample_format = AF_FORMAT_IEC61937_LE;
sh->samplerate = srate;
sh->channels = 2;
sh->i_bps = bps;
break;
case CODEC_ID_AC3:
spdif_ctx->iec61937_packet_size = 6144;
sh->sample_format = AF_FORMAT_IEC61937_LE;
sh->samplerate = srate;
sh->channels = 2;
sh->i_bps = bps;
break;
case CODEC_ID_DTS: // FORCE USE DTS-HD
opt = av_opt_find(&lavf_ctx->oformat->priv_class,
"dtshd_rate", NULL, 0, 0);
if (!opt)
goto fail;
dtshd_rate = (int*)(((uint8_t*)lavf_ctx->priv_data) +
opt->offset);
*dtshd_rate = 192000*4;
spdif_ctx->iec61937_packet_size = 32768;
sh->sample_format = AF_FORMAT_IEC61937_LE;
sh->samplerate = 192000; // DTS core require 48000
sh->channels = 2*4;
sh->i_bps = bps;
break;
case CODEC_ID_EAC3:
spdif_ctx->iec61937_packet_size = 24576;
sh->sample_format = AF_FORMAT_IEC61937_LE;
sh->samplerate = 192000;
sh->channels = 2;
sh->i_bps = bps;
break;
case CODEC_ID_MP3:
spdif_ctx->iec61937_packet_size = 4608;
sh->sample_format = AF_FORMAT_MPEG2;
sh->samplerate = srate;
sh->channels = 2;
sh->i_bps = bps;
break;
case CODEC_ID_TRUEHD:
spdif_ctx->iec61937_packet_size = 61440;
sh->sample_format = AF_FORMAT_IEC61937_LE;
sh->samplerate = 192000;
sh->channels = 8;
sh->i_bps = bps;
break;
default:
break;
}
return 1;
fail:
uninit(sh);
return 0;
}
static int decode_audio(sh_audio_t *sh, unsigned char *buf,
int minlen, int maxlen)
{
struct spdifContext *spdif_ctx = sh->context;
AVFormatContext *lavf_ctx = spdif_ctx->lavf_ctx;
AVPacket pkt;
double pts;
int ret, in_size, consumed, x;
unsigned char *start = NULL;
consumed = spdif_ctx->out_buffer_len = 0;
spdif_ctx->out_buffer_size = maxlen;
spdif_ctx->out_buffer = buf;
while (spdif_ctx->out_buffer_len + spdif_ctx->iec61937_packet_size < maxlen
&& spdif_ctx->out_buffer_len < minlen) {
if (sh->ds->eof)
break;
x = ds_get_packet_pts(sh->ds, &start, &pts);
if (x <= 0) {
x = 0;
ds_parse(sh->ds, &start, &x, MP_NOPTS_VALUE, 0);
if (x == 0)
continue; // END_NOT_FOUND
in_size = x;
} else {
in_size = x;
consumed = ds_parse(sh->ds, &start, &x, pts, 0);
if (x == 0) {
mp_msg(MSGT_DECAUDIO,MSGL_V,
"start[%p] in_size[%d] consumed[%d] x[%d].\n",
start, in_size, consumed, x);
continue; // END_NOT_FOUND
}
sh->ds->buffer_pos -= in_size - consumed;
}
av_init_packet(&pkt);
pkt.data = start;
pkt.size = x;
mp_msg(MSGT_DECAUDIO,MSGL_V,
"start[%p] pkt.size[%d] in_size[%d] consumed[%d] x[%d].\n",
start, pkt.size, in_size, consumed, x);
if (pts != MP_NOPTS_VALUE) {
sh->pts = pts;
sh->pts_bytes = 0;
}
ret = lavf_ctx->oformat->write_packet(lavf_ctx, &pkt);
if (ret < 0)
break;
}
sh->pts_bytes += spdif_ctx->out_buffer_len;
return spdif_ctx->out_buffer_len;
}
static int control(sh_audio_t *sh, int cmd, void* arg, ...)
{
unsigned char *start;
double pts;
switch (cmd) {
case ADCTRL_RESYNC_STREAM:
case ADCTRL_SKIP_FRAME:
ds_get_packet_pts(sh->ds, &start, &pts);
return CONTROL_TRUE;
}
return CONTROL_UNKNOWN;
}
static void uninit(sh_audio_t *sh)
{
struct spdifContext *spdif_ctx = sh->context;
AVFormatContext *lavf_ctx = spdif_ctx->lavf_ctx;
if (lavf_ctx) {
if (lavf_ctx->oformat)
lavf_ctx->oformat->write_trailer(lavf_ctx);
av_freep(&lavf_ctx->pb);
if (lavf_ctx->streams) {
av_freep(&lavf_ctx->streams[0]->codec);
av_freep(&lavf_ctx->streams[0]->info);
av_freep(&lavf_ctx->streams[0]);
}
av_freep(&lavf_ctx->streams);
av_freep(&lavf_ctx->priv_data);
}
av_freep(&lavf_ctx);
av_freep(&spdif_ctx);
}

462
audio/decode/dec_audio.c Normal file
View File

@@ -0,0 +1,462 @@
/*
* 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 <assert.h>
#include "config.h"
#include "mp_msg.h"
#include "bstr.h"
#include "stream/stream.h"
#include "libmpdemux/demuxer.h"
#include "codec-cfg.h"
#include "libmpdemux/stheader.h"
#include "dec_audio.h"
#include "ad.h"
#include "libaf/format.h"
#include "libaf/af.h"
int fakemono = 0;
struct af_cfg af_cfg = { 1, NULL }; // Configuration for audio filters
void afm_help(void)
{
int i;
mp_tmsg(MSGT_DECAUDIO, MSGL_INFO, "Available (compiled-in) audio codec families/drivers:\n");
mp_msg(MSGT_IDENTIFY, MSGL_INFO, "ID_AUDIO_DRIVERS\n");
mp_msg(MSGT_DECAUDIO, MSGL_INFO, " afm: info: (comment)\n");
for (i = 0; mpcodecs_ad_drivers[i] != NULL; i++)
if (mpcodecs_ad_drivers[i]->info->comment
&& mpcodecs_ad_drivers[i]->info->comment[0])
mp_msg(MSGT_DECAUDIO, MSGL_INFO, "%9s %s (%s)\n",
mpcodecs_ad_drivers[i]->info->short_name,
mpcodecs_ad_drivers[i]->info->name,
mpcodecs_ad_drivers[i]->info->comment);
else
mp_msg(MSGT_DECAUDIO, MSGL_INFO, "%9s %s\n",
mpcodecs_ad_drivers[i]->info->short_name,
mpcodecs_ad_drivers[i]->info->name);
}
static int init_audio_codec(sh_audio_t *sh_audio)
{
assert(!sh_audio->initialized);
resync_audio_stream(sh_audio);
if ((af_cfg.force & AF_INIT_FORMAT_MASK) == AF_INIT_FLOAT) {
int fmt = AF_FORMAT_FLOAT_NE;
if (sh_audio->ad_driver->control(sh_audio, ADCTRL_QUERY_FORMAT,
&fmt) == CONTROL_TRUE) {
sh_audio->sample_format = fmt;
sh_audio->samplesize = 4;
}
}
sh_audio->audio_out_minsize = 8192; // default, preinit() may change it
if (!sh_audio->ad_driver->preinit(sh_audio)) {
mp_tmsg(MSGT_DECAUDIO, MSGL_ERR, "ADecoder preinit failed :(\n");
return 0;
}
/* allocate audio in buffer: */
if (sh_audio->audio_in_minsize > 0) {
sh_audio->a_in_buffer_size = sh_audio->audio_in_minsize;
mp_tmsg(MSGT_DECAUDIO, MSGL_V, "dec_audio: Allocating %d bytes for input buffer.\n",
sh_audio->a_in_buffer_size);
sh_audio->a_in_buffer = av_mallocz(sh_audio->a_in_buffer_size);
}
const int base_size = 65536;
// At least 64 KiB plus rounding up to next decodable unit size
sh_audio->a_buffer_size = base_size + sh_audio->audio_out_minsize;
mp_tmsg(MSGT_DECAUDIO, MSGL_V, "dec_audio: Allocating %d + %d = %d bytes for output buffer.\n",
sh_audio->audio_out_minsize, base_size, sh_audio->a_buffer_size);
sh_audio->a_buffer = av_mallocz(sh_audio->a_buffer_size);
if (!sh_audio->a_buffer)
abort();
sh_audio->a_buffer_len = 0;
if (!sh_audio->ad_driver->init(sh_audio)) {
mp_tmsg(MSGT_DECAUDIO, MSGL_V, "ADecoder init failed :(\n");
uninit_audio(sh_audio); // free buffers
return 0;
}
sh_audio->initialized = 1;
if (!sh_audio->channels || !sh_audio->samplerate) {
mp_tmsg(MSGT_DECAUDIO, MSGL_ERR, "Audio decoder did not specify "
"audio format!\n");
uninit_audio(sh_audio); // free buffers
return 0;
}
if (!sh_audio->o_bps)
sh_audio->o_bps = sh_audio->channels * sh_audio->samplerate
* sh_audio->samplesize;
return 1;
}
static int init_audio(sh_audio_t *sh_audio, char *codecname, char *afm,
int status, stringset_t *selected)
{
int force = 0;
if (codecname && codecname[0] == '+') {
codecname = &codecname[1];
force = 1;
}
sh_audio->codec = NULL;
while (1) {
const ad_functions_t *mpadec;
sh_audio->ad_driver = 0;
if (!(sh_audio->codec = find_audio_codec(sh_audio->format,
NULL,
sh_audio->codec, force)))
break;
// ok we found one codec
if (stringset_test(selected, sh_audio->codec->name))
continue; // already tried & failed
if (codecname && strcmp(sh_audio->codec->name, codecname))
continue; // -ac
if (afm && strcmp(sh_audio->codec->drv, afm))
continue; // afm doesn't match
if (!force && sh_audio->codec->status < status)
continue; // too unstable
stringset_add(selected, sh_audio->codec->name); // tagging it
// ok, it matches all rules, let's find the driver!
int i;
for (i = 0; mpcodecs_ad_drivers[i] != NULL; i++)
if (!strcmp(mpcodecs_ad_drivers[i]->info->short_name,
sh_audio->codec->drv))
break;
mpadec = mpcodecs_ad_drivers[i];
if (!mpadec) { // driver not available (==compiled in)
mp_tmsg(MSGT_DECAUDIO, MSGL_ERR,
"Requested audio codec family [%s] (afm=%s) not available.\nEnable it at compilation.\n",
sh_audio->codec->name, sh_audio->codec->drv);
continue;
}
// it's available, let's try to init!
// init()
mp_tmsg(MSGT_DECAUDIO, MSGL_V, "Opening audio decoder: [%s] %s\n",
mpadec->info->short_name, mpadec->info->name);
sh_audio->ad_driver = mpadec;
if (!init_audio_codec(sh_audio)) {
mp_tmsg(MSGT_DECAUDIO, MSGL_WARN, "Audio decoder init failed for "
"codecs.conf entry \"%s\".\n", sh_audio->codec->name);
continue; // try next...
}
// Yeah! We got it!
return 1;
}
return 0;
}
int init_best_audio_codec(sh_audio_t *sh_audio, char **audio_codec_list,
char **audio_fm_list)
{
stringset_t selected;
char *ac_l_default[2] = { "", (char *) NULL };
// hack:
if (!audio_codec_list)
audio_codec_list = ac_l_default;
// Go through the codec.conf and find the best codec...
sh_audio->initialized = 0;
stringset_init(&selected);
while (!sh_audio->initialized && *audio_codec_list) {
char *audio_codec = *(audio_codec_list++);
if (audio_codec[0]) {
if (audio_codec[0] == '-') {
// disable this codec:
stringset_add(&selected, audio_codec + 1);
} else {
// forced codec by name:
mp_tmsg(MSGT_DECAUDIO, MSGL_INFO, "Forced audio codec: %s\n",
audio_codec);
init_audio(sh_audio, audio_codec, NULL, -1, &selected);
}
} else {
int status;
// try in stability order: UNTESTED, WORKING, BUGGY.
// never try CRASHING.
if (audio_fm_list) {
char **fmlist = audio_fm_list;
// try first the preferred codec families:
while (!sh_audio->initialized && *fmlist) {
char *audio_fm = *(fmlist++);
mp_tmsg(MSGT_DECAUDIO, MSGL_INFO, "Trying to force audio codec driver family %s...\n",
audio_fm);
for (status = CODECS_STATUS__MAX;
status >= CODECS_STATUS__MIN; --status)
if (init_audio(sh_audio, NULL, audio_fm, status, &selected))
break;
}
}
if (!sh_audio->initialized)
for (status = CODECS_STATUS__MAX; status >= CODECS_STATUS__MIN;
--status)
if (init_audio(sh_audio, NULL, NULL, status, &selected))
break;
}
}
stringset_free(&selected);
if (!sh_audio->initialized) {
mp_tmsg(MSGT_DECAUDIO, MSGL_ERR, "Cannot find codec for audio format 0x%X.\n",
sh_audio->format);
return 0; // failed
}
mp_tmsg(MSGT_DECAUDIO, MSGL_INFO, "Selected audio codec: %s [%s]\n",
sh_audio->codecname ? sh_audio->codecname : sh_audio->codec->info,
sh_audio->ad_driver->info->print_name ?
sh_audio->ad_driver->info->print_name :
sh_audio->ad_driver->info->short_name);
mp_tmsg(MSGT_DECAUDIO, MSGL_V,
"Audio codecs.conf entry: %s (%s) afm: %s\n",
sh_audio->codec->name, sh_audio->codec->info, sh_audio->codec->drv);
mp_msg(MSGT_DECAUDIO, MSGL_V,
"AUDIO: %d Hz, %d ch, %s, %3.1f kbit/%3.2f%% (ratio: %d->%d)\n",
sh_audio->samplerate, sh_audio->channels,
af_fmt2str_short(sh_audio->sample_format),
sh_audio->i_bps * 8 * 0.001,
((float) sh_audio->i_bps / sh_audio->o_bps) * 100.0,
sh_audio->i_bps, sh_audio->o_bps);
mp_msg(MSGT_IDENTIFY, MSGL_INFO,
"ID_AUDIO_BITRATE=%d\nID_AUDIO_RATE=%d\n" "ID_AUDIO_NCH=%d\n",
sh_audio->i_bps * 8, sh_audio->samplerate, sh_audio->channels);
return 1; // success
}
void uninit_audio(sh_audio_t *sh_audio)
{
if (sh_audio->afilter) {
mp_msg(MSGT_DECAUDIO, MSGL_V, "Uninit audio filters...\n");
af_uninit(sh_audio->afilter);
free(sh_audio->afilter);
sh_audio->afilter = NULL;
}
if (sh_audio->initialized) {
mp_tmsg(MSGT_DECAUDIO, MSGL_V, "Uninit audio: %s\n",
sh_audio->codec->drv);
sh_audio->ad_driver->uninit(sh_audio);
sh_audio->initialized = 0;
}
av_freep(&sh_audio->a_buffer);
av_freep(&sh_audio->a_in_buffer);
}
int init_audio_filters(sh_audio_t *sh_audio, int in_samplerate,
int *out_samplerate, int *out_channels, int *out_format)
{
struct af_stream *afs = sh_audio->afilter;
if (!afs) {
afs = calloc(1, sizeof(struct af_stream));
afs->opts = sh_audio->opts;
}
// input format: same as codec's output format:
afs->input.rate = in_samplerate;
afs->input.nch = sh_audio->channels;
afs->input.format = sh_audio->sample_format;
af_fix_parameters(&(afs->input));
// output format: same as ao driver's input format (if missing, fallback to input)
afs->output.rate = *out_samplerate;
afs->output.nch = *out_channels;
afs->output.format = *out_format;
af_fix_parameters(&(afs->output));
// filter config:
memcpy(&afs->cfg, &af_cfg, sizeof(struct af_cfg));
mp_tmsg(MSGT_DECAUDIO, MSGL_V, "Building audio filter chain for %dHz/%dch/%s -> %dHz/%dch/%s...\n",
afs->input.rate, afs->input.nch,
af_fmt2str_short(afs->input.format), afs->output.rate,
afs->output.nch, af_fmt2str_short(afs->output.format));
// let's autoprobe it!
if (0 != af_init(afs)) {
sh_audio->afilter = NULL;
free(afs);
return 0; // failed :(
}
*out_samplerate = afs->output.rate;
*out_channels = afs->output.nch;
*out_format = afs->output.format;
// ok!
sh_audio->afilter = (void *) afs;
return 1;
}
static void set_min_out_buffer_size(struct bstr *outbuf, int len)
{
size_t oldlen = talloc_get_size(outbuf->start);
if (oldlen < len) {
assert(outbuf->start); // talloc context should be already set
mp_msg(MSGT_DECAUDIO, MSGL_V, "Increasing filtered audio buffer size "
"from %zd to %d\n", oldlen, len);
outbuf->start = talloc_realloc_size(NULL, outbuf->start, len);
}
}
static int filter_n_bytes(sh_audio_t *sh, struct bstr *outbuf, int len)
{
assert(len-1 + sh->audio_out_minsize <= sh->a_buffer_size);
int error = 0;
// Decode more bytes if needed
int old_samplerate = sh->samplerate;
int old_channels = sh->channels;
int old_sample_format = sh->sample_format;
while (sh->a_buffer_len < len) {
unsigned char *buf = sh->a_buffer + sh->a_buffer_len;
int minlen = len - sh->a_buffer_len;
int maxlen = sh->a_buffer_size - sh->a_buffer_len;
int ret = sh->ad_driver->decode_audio(sh, buf, minlen, maxlen);
int format_change = sh->samplerate != old_samplerate
|| sh->channels != old_channels
|| sh->sample_format != old_sample_format;
if (ret <= 0 || format_change) {
error = format_change ? -2 : -1;
// samples from format-changing call get discarded too
len = sh->a_buffer_len;
break;
}
sh->a_buffer_len += ret;
}
// Filter
struct mp_audio filter_input = {
.audio = sh->a_buffer,
.len = len,
.rate = sh->samplerate,
.nch = sh->channels,
.format = sh->sample_format
};
af_fix_parameters(&filter_input);
struct mp_audio *filter_output = af_play(sh->afilter, &filter_input);
if (!filter_output)
return -1;
set_min_out_buffer_size(outbuf, outbuf->len + filter_output->len);
memcpy(outbuf->start + outbuf->len, filter_output->audio,
filter_output->len);
outbuf->len += filter_output->len;
// remove processed data from decoder buffer:
sh->a_buffer_len -= len;
memmove(sh->a_buffer, sh->a_buffer + len, sh->a_buffer_len);
return error;
}
/* Try to get at least minlen decoded+filtered bytes in outbuf
* (total length including possible existing data).
* Return 0 on success, -1 on error/EOF (not distinguished).
* In the former case outbuf->len is always >= minlen on return.
* In case of EOF/error it might or might not be.
* Outbuf.start must be talloc-allocated, and will be reallocated
* if needed to fit all filter output. */
int decode_audio(sh_audio_t *sh_audio, struct bstr *outbuf, int minlen)
{
// Indicates that a filter seems to be buffering large amounts of data
int huge_filter_buffer = 0;
// Decoded audio must be cut at boundaries of this many bytes
int unitsize = sh_audio->channels * sh_audio->samplesize * 16;
/* Filter output size will be about filter_multiplier times input size.
* If some filter buffers audio in big blocks this might only hold
* as average over time. */
double filter_multiplier = af_calc_filter_multiplier(sh_audio->afilter);
/* If the decoder set audio_out_minsize then it can do the equivalent of
* "while (output_len < target_len) output_len += audio_out_minsize;",
* so we must guarantee there is at least audio_out_minsize-1 bytes
* more space in the output buffer than the minimum length we try to
* decode. */
int max_decode_len = sh_audio->a_buffer_size - sh_audio->audio_out_minsize;
if (!unitsize)
return -1;
max_decode_len -= max_decode_len % unitsize;
while (outbuf->len < minlen) {
int declen = (minlen - outbuf->len) / filter_multiplier
+ (unitsize << 5); // some extra for possible filter buffering
if (huge_filter_buffer)
/* Some filter must be doing significant buffering if the estimated
* input length didn't produce enough output from filters.
* Feed the filters 2k bytes at a time until we have enough output.
* Very small amounts could make filtering inefficient while large
* amounts can make MPlayer demux the file unnecessarily far ahead
* to get audio data and buffer video frames in memory while doing
* so. However the performance impact of either is probably not too
* significant as long as the value is not completely insane. */
declen = 2000;
declen -= declen % unitsize;
if (declen > max_decode_len)
declen = max_decode_len;
else
/* if this iteration does not fill buffer, we must have lots
* of buffering in filters */
huge_filter_buffer = 1;
int res = filter_n_bytes(sh_audio, outbuf, declen);
if (res < 0)
return res;
}
return 0;
}
void decode_audio_prepend_bytes(struct bstr *outbuf, int count, int byte)
{
set_min_out_buffer_size(outbuf, outbuf->len + count);
memmove(outbuf->start + count, outbuf->start, outbuf->len);
memset(outbuf->start, byte, count);
outbuf->len += count;
}
void resync_audio_stream(sh_audio_t *sh_audio)
{
sh_audio->a_in_buffer_len = 0; // clear audio input buffer
sh_audio->pts = MP_NOPTS_VALUE;
if (!sh_audio->initialized)
return;
sh_audio->ad_driver->control(sh_audio, ADCTRL_RESYNC_STREAM, NULL);
}
void skip_audio_frame(sh_audio_t *sh_audio)
{
if (!sh_audio->initialized)
return;
if (sh_audio->ad_driver->control(sh_audio, ADCTRL_SKIP_FRAME, NULL) ==
CONTROL_TRUE)
return;
// default skip code:
ds_fill_buffer(sh_audio->ds); // skip block
}

38
audio/decode/dec_audio.h Normal file
View File

@@ -0,0 +1,38 @@
/*
* 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.
*/
#ifndef MPLAYER_DEC_AUDIO_H
#define MPLAYER_DEC_AUDIO_H
#include "libmpdemux/stheader.h"
struct bstr;
// dec_audio.c:
void afm_help(void);
int init_best_audio_codec(sh_audio_t *sh_audio, char** audio_codec_list, char** audio_fm_list);
int decode_audio(sh_audio_t *sh_audio, struct bstr *outbuf, int minlen);
void decode_audio_prepend_bytes(struct bstr *outbuf, int count, int byte);
void resync_audio_stream(sh_audio_t *sh_audio);
void skip_audio_frame(sh_audio_t *sh_audio);
void uninit_audio(sh_audio_t *sh_audio);
int init_audio_filters(sh_audio_t *sh_audio, int in_samplerate,
int *out_samplerate, int *out_channels, int *out_format);
#endif /* MPLAYER_DEC_AUDIO_H */