On Wed, Apr 9, 2008 at 7:50 PM, Michael Niedermayer <[EMAIL PROTECTED]> wrote:
> On Wed, Apr 09, 2008 at 04:28:50PM +0200, Robert Marston wrote:
>  > Attached patch has the following changes to it.
>  [...]
>
> > >
>  > >  [...]
>  > >  >  typedef struct ADPCMContext {
>  > >  >      int channel; /* for stereo MOVs, decode left, then decode right, 
> then tell it's decoded */
>  > >  >      ADPCMChannelStatus status[6];
>  > >  > +    int32_t mxa_current_left_sample; /*Needed to keep track of left 
> and right samples for Maxis EA XA */
>  > >  > +    int32_t mxa_previous_left_sample;
>  > >  > +    int32_t mxa_current_right_sample;
>  > >  > +    int32_t mxa_previous_right_sample;
>  > >  >  } ADPCMContext;
>  > >  >
>  > >  >  /* XXX: implement encoding */
>  > >
>  > Unless there is a way of getting the samples from the previous decoded
>  > frame then this is needed
>
>  > since other ADPCM formats have the previous
>  > and current samples encoded in the stream
>
>  No some do not.
>
>
Correct my code to use the variables the other formats use.
>  [...]
>
> > >
>  > >  > +    else {
>  > >
>  > > > +        pkt->stream_index = xa->audio_stream_index;
>  > >  > +        pkt->pts = 90000;
>  > >  > +        pkt->pts *= xa->audio_frame_counter;
>  > >  > +        pkt->pts /= xa->sampleRate;
>  > >  > +        xa->audio_frame_counter += (14 * xa->channels);  /* 14 
> Samples per channel  */
>  > >
>  > >  to quote the documentation
>  > >  "int64_t pts;                            ///< presentation time stamp 
> in time_base units"
>  > >
>  > >  This is not what you set it to
>  > >
>  > Removed this as I am unsure of the time base.
>
>  So i assume you are sure that its unneeded to set pts here? If so please
>  explain why it is unneeded. If not please set pts to a correct value.
>
Ignore the previous email with the reply to this, I think this is now correct?
>
>  [...]
>  > @@ -667,8 +669,12 @@
>  >  {
>  >      ADPCMContext *c = avctx->priv_data;
>  >      unsigned int max_channels = 2;
>  > -
>
>  cosmetic change
>
Mistake, corrected.
>
>  [...]
>  > @@ -1235,6 +1242,30 @@
>
> >              }
>  >          }
>  >          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) & 
> 0x0F)+(4*i)];
>
>  one of the operations in there does nothing
>
I know the shift may not be needed, though the other decoders add it
and the maxis xa wiki page warns that some compilers need it.
>
>  > +            shift[channel] = (*src & 0x0F) + 8;
>  > +            src++;
>  > +        }
>
>  indention is inconsistant
>
>
Corrected.
>
>  > +        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;
>
>  random indention
>
Ditto.
>
>  > +            int prev_next = 0;
>
>  contradictionary variable name
>
No longer used.
>
>  > +                for(channel = 0; channel < avctx->channels; channel++, 
> prev_next+=2) {
>  > +                    sample = ((((*(src+channel) >> i) & 0x0F) << 0x1C) >> 
> shift[channel]);
>  > +                    sample = (sample +
>
>  > +                        (c->curr_prev_samples[prev_next+1] * 
> coeff[channel][0]) +
>  > +                        (c->curr_prev_samples[prev_next] * 
> coeff[channel][1]) + 0x80) >> 8;
>
>
>  this can be vertically aligned
>
Aligned a bit better.
>
>  > +                    c->curr_prev_samples[prev_next] = 
> c->curr_prev_samples[prev_next+1];
>  > +                    c->curr_prev_samples[prev_next+1] = 
> av_clip_int16(sample);
>
>  > +                    *samples++ = (unsigned 
> short)c->curr_prev_samples[prev_next+1];
>
>  unneeded cast
>
Removed.
>
>  [...]
>  > +typedef struct MaxisXADemuxContext {
>
>  > +    uint32_t out_size;
>  > +    uint32_t  sent_bytes;
>
>  inconsistant whitespace
>
Corrected.
>  [...]
>
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   2008-04-10 11:43:25.000000000 +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:
@@ -666,9 +667,15 @@
 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_MAXIS_XA:
+        for(channel = 0; channel < avctx->channels; channel++) {
+                c->status[channel].sample1 = 0;
+                c->status[channel].sample2 = 0;
+        }
+        break;
     case CODEC_ID_ADPCM_EA_R1:
     case CODEC_ID_ADPCM_EA_R2:
     case CODEC_ID_ADPCM_EA_R3:
@@ -900,6 +907,7 @@
     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 +1243,29 @@
             }
         }
         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 = ((((*(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 +1644,7 @@
 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     2008-04-10 10:57:41.000000000 +0200
@@ -0,0 +1,116 @@
+/*
+ * 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 visi
+ *   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);
+
+    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 = 90000;
+    pkt->pts *= xa->audio_frame_counter;
+    pkt->pts /= st->codec->sample_rate;
+    /* 14 Samples per channel  */
+    xa->audio_frame_counter += 14;
+
+    return ret;
+}
+
+AVInputFormat xa_demuxer = {
+    "xa",
+    "Maxis XA File Format",
+    sizeof(MaxisXADemuxContext),
+    xa_probe,
+    xa_read_header,
+    xa_read_packet,
+};
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