On Sat, Apr 12, 2008 at 7:21 PM, Michael Niedermayer <[EMAIL PROTECTED]> wrote:
> On Sat, Apr 12, 2008 at 06:12:15PM +0200, Robert Marston wrote:
> > On Sat, Apr 12, 2008 at 3:00 PM, Michael Niedermayer <[EMAIL PROTECTED]>
> wrote:
> > >
> > > On Sat, Apr 12, 2008 at 01:39:45PM +0200, Robert Marston wrote:
> > > >
> > > > Thanks for pointing that out, I will be the first to admit that my c
> > > > knowledge is probably not up to standard when it comes to FFMPEG and
> > > > as such would required much feedback from my mentor. I see the GSoC as
> > > > good learning opportunity for the myself and a chance to bolster open
> > > > source development and potential to increase the code base of the
> > > > mentor organizations project.
> > > >
> > > > Would casting the *(src + channel) to a int32_t stop the above from
> happening?
> > >
> > > i would put the cast after <<0x1C but before >>shift[channel]
> > >
> >
> > As a matter of interest is that to avoid loosing higher order bits in
> > the <<0x1C shift?
>
> no
>
Sorry that a was a stupid question, the cast would have dropped the
bits anyway. What is the reason for putting the cast after the 0x1C
shift?
> >>
> > > >
> > > > My logic on this is that there are 2 samples per byte and 14 bytes per
> > > > channel. = 28 x num_channels
> > > > There are is 1 sample every 1 / sample rate seconds and 90 K ticks per
> > > > second if we use a 90 KHz clock
> > > >
> > > > Therefore the pts, which in my understanding of the time base being
> > > > the number of ticks of the 90 KHz clock, will be advanced by 28 x
> > > > num_channels / sample rate * 90 K for each block read from the file.
> > > >
> > > > Have I misunderstood something here?
> > >
> > > yes, there is no 90khz clock. The "clock" is what the code calls
> time_base
> > > and what you set with av_set_pts_info()
> > >
> > > [...]
> >
> >
> > Ok that makes a little more sense. I was using a 90 KHz clock since I
> > had seen it in some of the other game format demuxers and presumed
> > that it was needed. In this case do I only advance the pts by the
> > number of samples sent in each packet?
>
> This depends on how you define sample, so possibly yes. What the pts
> is increased by is the duration of the packet.
>
>
> [...]
Attached is my latest attempt at the patch.
Thanks
Robert
diff -urNt /home/rob/ffmpeg/libavcodec/adpcm.c ffmpeg/libavcodec/adpcm.c
--- /home/rob/ffmpeg/libavcodec/adpcm.c 2008-04-08 12:41:05.000000000 +0200
+++ ffmpeg/libavcodec/adpcm.c 2008-04-12 19:37:17.000000000 +0200
@@ -34,6 +34,7 @@
* EA IMA EACS decoder by Peter Ross ([EMAIL PROTECTED])
* EA IMA SEAD decoder by Peter Ross ([EMAIL PROTECTED])
* EA ADPCM XAS decoder by Peter Ross ([EMAIL PROTECTED])
+ * MAXIS EA ADPCM decoder by Robert Marston ([EMAIL PROTECTED])
* THP ADPCM decoder by Marco Gerards ([EMAIL PROTECTED])
*
* Features and limitations:
@@ -666,7 +667,7 @@
static av_cold int adpcm_decode_init(AVCodecContext * avctx)
{
ADPCMContext *c = avctx->priv_data;
- unsigned int max_channels = 2;
+ unsigned int max_channels = 2, channel;
switch(avctx->codec->id) {
case CODEC_ID_ADPCM_EA_R1:
@@ -900,6 +901,7 @@
int32_t coeff1l, coeff2l, coeff1r, coeff2r;
uint8_t shift_left, shift_right;
int count1, count2;
+ int coeff[2][2], shift[2];//used in EA MAXIS ADPCM
if (!buf_size)
return 0;
@@ -1235,6 +1237,29 @@
}
}
break;
+ case CODEC_ID_ADPCM_EA_MAXIS_XA:
+ for(channel = 0; channel < avctx->channels; channel++) {
+ for (i=0; i<2; i++)
+ coeff[channel][i] = ea_adpcm_table[(*src >> 4) + 4*i];
+ shift[channel] = (*src & 0x0F) + 8;
+ src++;
+ }
+ for (count1 = 0; count1 < ((buf_size - avctx->channels) /
avctx->channels) ; count1++) {
+ for(i = 4; i >= 0; i-=4) { /* Pairwise samples LL RR (st) or LL LL
(mono) */
+ int32_t sample;
+ for(channel = 0; channel < avctx->channels; channel++) {
+ sample = (int32_t)(((*(src+channel) >> i) & 0x0F) << 0x1C)
>> shift[channel];
+ sample = (sample +
+ (c->status[channel].sample1 * coeff[channel][0]) +
+ (c->status[channel].sample2 * coeff[channel][1])
+ 0x80) >> 8;
+ c->status[channel].sample2 = c->status[channel].sample1;
+ c->status[channel].sample1 = av_clip_int16(sample);
+ *samples++ = c->status[channel].sample1;
+ }
+ }
+ src+=avctx->channels;
+ }
+ break;
case CODEC_ID_ADPCM_EA_R1:
case CODEC_ID_ADPCM_EA_R2:
case CODEC_ID_ADPCM_EA_R3: {
@@ -1613,6 +1638,7 @@
ADPCM_DECODER(CODEC_ID_ADPCM_4XM, adpcm_4xm);
ADPCM_DECODER(CODEC_ID_ADPCM_CT, adpcm_ct);
ADPCM_DECODER(CODEC_ID_ADPCM_EA, adpcm_ea);
+ADPCM_DECODER(CODEC_ID_ADPCM_EA_MAXIS_XA, adpcm_ea_maxis_xa);
ADPCM_DECODER(CODEC_ID_ADPCM_EA_R1, adpcm_ea_r1);
ADPCM_DECODER(CODEC_ID_ADPCM_EA_R2, adpcm_ea_r2);
ADPCM_DECODER(CODEC_ID_ADPCM_EA_R3, adpcm_ea_r3);
diff -urNt /home/rob/ffmpeg/libavcodec/allcodecs.c ffmpeg/libavcodec/allcodecs.c
--- /home/rob/ffmpeg/libavcodec/allcodecs.c 2008-04-08 12:41:05.000000000
+0200
+++ ffmpeg/libavcodec/allcodecs.c 2008-04-08 15:26:48.000000000 +0200
@@ -244,6 +244,7 @@
REGISTER_ENCDEC (ADPCM_ADX, adpcm_adx);
REGISTER_DECODER (ADPCM_CT, adpcm_ct);
REGISTER_DECODER (ADPCM_EA, adpcm_ea);
+ REGISTER_DECODER (ADPCM_EA_MAXIS_XA, adpcm_ea_maxis_xa);
REGISTER_DECODER (ADPCM_EA_R1, adpcm_ea_r1);
REGISTER_DECODER (ADPCM_EA_R2, adpcm_ea_r2);
REGISTER_DECODER (ADPCM_EA_R3, adpcm_ea_r3);
diff -urNt /home/rob/ffmpeg/libavcodec/avcodec.h ffmpeg/libavcodec/avcodec.h
--- /home/rob/ffmpeg/libavcodec/avcodec.h 2008-04-08 12:41:05.000000000
+0200
+++ ffmpeg/libavcodec/avcodec.h 2008-04-08 15:24:43.000000000 +0200
@@ -231,6 +231,7 @@
CODEC_ID_ADPCM_IMA_EA_SEAD,
CODEC_ID_ADPCM_IMA_EA_EACS,
CODEC_ID_ADPCM_EA_XAS,
+ CODEC_ID_ADPCM_EA_MAXIS_XA,
/* AMR */
CODEC_ID_AMR_NB= 0x12000,
diff -urNt /home/rob/ffmpeg/libavformat/allformats.c
ffmpeg/libavformat/allformats.c
--- /home/rob/ffmpeg/libavformat/allformats.c 2008-04-08 13:15:39.000000000
+0200
+++ ffmpeg/libavformat/allformats.c 2008-04-08 15:28:45.000000000 +0200
@@ -169,6 +169,7 @@
REGISTER_DEMUXER (WSAUD, wsaud);
REGISTER_DEMUXER (WSVQA, wsvqa);
REGISTER_DEMUXER (WV, wv);
+ REGISTER_DEMUXER (XA, xa);
REGISTER_MUXDEMUX (YUV4MPEGPIPE, yuv4mpegpipe);
/* external libraries */
diff -urNt /home/rob/ffmpeg/libavformat/Makefile ffmpeg/libavformat/Makefile
--- /home/rob/ffmpeg/libavformat/Makefile 2008-04-08 13:15:39.000000000
+0200
+++ ffmpeg/libavformat/Makefile 2008-04-08 15:27:39.000000000 +0200
@@ -179,6 +179,7 @@
OBJS-$(CONFIG_WSAUD_DEMUXER) += westwood.o
OBJS-$(CONFIG_WSVQA_DEMUXER) += westwood.o
OBJS-$(CONFIG_WV_DEMUXER) += wv.o
+OBJS-$(CONFIG_XA_DEMUXER) += xa.o
OBJS-$(CONFIG_YUV4MPEGPIPE_MUXER) += yuv4mpeg.o
OBJS-$(CONFIG_YUV4MPEGPIPE_DEMUXER) += yuv4mpeg.o
diff -urNt /home/rob/ffmpeg/libavformat/xa.c ffmpeg/libavformat/xa.c
--- /home/rob/ffmpeg/libavformat/xa.c 1970-01-01 02:00:00.000000000 +0200
+++ ffmpeg/libavformat/xa.c 2008-04-12 17:54:02.000000000 +0200
@@ -0,0 +1,115 @@
+/*
+ * Maxis XA (.xa) File Demuxer
+ * Copyright (c) 2008 Robert Marston
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg 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
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file xa.c
+ * Maxis XA File Demuxer
+ * by Robert Marston ([EMAIL PROTECTED])
+ * for more information on the XA audio format see
+ * http://wiki.multimedia.cx/index.php?title=Maxis_XA
+ */
+
+#include "avformat.h"
+
+#define XA00_TAG MKTAG('X', 'A', 0, 0)
+#define XAI0_TAG MKTAG('X', 'A', 'I', 0)
+#define XAJ0_TAG MKTAG('X', 'A', 'J', 0)
+
+typedef struct MaxisXADemuxContext {
+ uint32_t out_size;
+ uint32_t sent_bytes;
+ uint32_t audio_frame_counter;
+} MaxisXADemuxContext;
+
+static int xa_probe(AVProbeData *p)
+{
+ switch(AV_RL32(&p->buf[0])) {
+ case XA00_TAG:
+ case XAI0_TAG:
+ case XAJ0_TAG:
+ return AVPROBE_SCORE_MAX;
+ }
+ return 0;
+}
+
+static int xa_read_header(AVFormatContext *s,
+ AVFormatParameters *ap)
+{
+ MaxisXADemuxContext *xa = s->priv_data;
+ ByteIOContext *pb = s->pb;
+ AVStream *st;
+
+ /*Set up the XA Audio Decoder*/
+ st = av_new_stream(s, 0);
+ if (!st)
+ return AVERROR(ENOMEM);
+
+ st->codec->codec_type = CODEC_TYPE_AUDIO;
+ st->codec->codec_id = CODEC_ID_ADPCM_EA_MAXIS_XA;
+ url_fskip(pb, 4); /* Skip the XA ID */
+ xa->out_size = get_le32(pb);
+ url_fskip(pb, 2); /* Skip the tag */
+ st->codec->channels = get_le16(pb);
+ st->codec->sample_rate = get_le32(pb);
+ /* Value in file is average byte rate*/
+ st->codec->bit_rate = get_le32(pb) * 8;
+ st->codec->block_align = get_le16(pb);
+ st->codec->bits_per_sample = get_le16(pb);
+
+ av_set_pts_info(st, 64, 1, st->codec->sample_rate);
+ xa->audio_frame_counter = 0;
+
+ return 0;
+}
+
+static int xa_read_packet(AVFormatContext *s,
+ AVPacket *pkt)
+{
+ MaxisXADemuxContext *xa = s->priv_data;
+ AVStream *st = s->streams[0];
+ ByteIOContext *pb = s->pb;
+ unsigned int packet_size;
+ int ret = 0;
+
+ if(xa->sent_bytes > xa->out_size)
+ return AVERROR(EIO);
+ /* 1 byte header and 14 bytes worth of samples * number channels per block
*/
+ packet_size = 15*st->codec->channels;
+
+ ret = av_get_packet(pb, pkt, packet_size);
+ pkt->stream_index = st->index;
+
+ xa->sent_bytes += packet_size;
+ pkt->pts = xa->audio_frame_counter;
+ /* 14 bytes Samples per channel with 2 samples per byte */
+ xa->audio_frame_counter += (28 * st->codec->channels);
+
+ return ret;
+}
+
+AVInputFormat xa_demuxer = {
+ "xa",
+ "Maxis XA File Format",
+ sizeof(MaxisXADemuxContext),
+ xa_probe,
+ xa_read_header,
+ xa_read_packet,
+};
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