ad_pcm: add back raw decoder

This was removed in commit 6a26b4a665. Add it back, because it was
needed by demuxer_rawaudio and for PCM audio with demuxers other than
demux_lavf. (In practice, this broke rawaudio and PCM-in-Matroska only.)

Unlike with raw video, there is no single raw audio "decoder" in
libavcodec. Instead of trying to mess raw audio input into ad_ffmpeg
using a table to map audio formats to the respective libavcodec
decoders, it seems advantageous to simply add back ad_pcm.
This commit is contained in:
wm4 2012-09-02 19:30:34 +02:00
parent 53bfaecd40
commit ee65b39cbe
4 changed files with 247 additions and 0 deletions

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@ -141,6 +141,7 @@ SRCS_COMMON = asxparser.c \
libaf/window.c \
libmpcodecs/ad.c \
libmpcodecs/ad_ffmpeg.c \
libmpcodecs/ad_pcm.c \
libmpcodecs/ad_spdif.c \
libmpcodecs/dec_audio.c \
libmpcodecs/dec_video.c \

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@ -1753,6 +1753,36 @@ audiocodec ffnellymoser
driver ffmpeg
dll "nellymoser"
audiocodec pcm
info "Uncompressed PCM"
status working
format 0x0
format 0x1
format 0x3 ; IEEE float
format 0xfffe ; Extended
fourcc "raw " ; (MOV files)
fourcc twos ; (MOV files)
fourcc sowt ; (MOV files)
fourcc fl32 ; (MOV files)
fourcc 23lf ; (MOV files)
; fourcc fl64 ; (MOV files)
; fourcc 46lf ; (MOV files)
fourcc NONE ; (MOV files from Kodak CX6320)
fourcc in24 ; (MOV files)
fourcc 42ni ; (MOV files)
fourcc in32 ; (MOV files)
fourcc 23ni ; (MOV files)
fourcc lpcm ; (MOV files)
fourcc FL32 ; (aiff files)
;;;; these are for hardware support only: (alaw,ulaw,ima-adpcm,mpeg,ac3)
; format 0x6
; format 0x7
; format 0x11
; format 0x50
; format 0x2000
;;;;
driver pcm
audiocodec ffpcmdaud
info "D-Cinema audio (FFmpeg)"
status untested

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@ -33,6 +33,7 @@
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_spdif;
const ad_functions_t * const mpcodecs_ad_drivers[] =
@ -41,6 +42,7 @@ const ad_functions_t * const mpcodecs_ad_drivers[] =
&mpcodecs_ad_mpg123,
#endif
&mpcodecs_ad_ffmpeg,
&mpcodecs_ad_pcm,
&mpcodecs_ad_spdif,
NULL
};

214
libmpcodecs/ad_pcm.c Normal file
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@ -0,0 +1,214 @@
/*
* 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 "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;
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;
}