Attached patch has the following changes to it.
On Tue, Apr 8, 2008 at 8:11 PM, Michael Niedermayer <[EMAIL PROTECTED]> wrote:
> On Tue, Apr 08, 2008 at 04:43:25PM +0200, Robert Marston wrote:
> [...]
>
> > >
> > > > int tag;
> > > > int channels;
> > > > uint32_t sampleRate;
> > >
> > > redundant
> > >
> >
> > Removed tag.
>
> The others are redundant as well
>
Removed.
>
>
> >
> >
> > >
> > >
> > > [...]
> > > > static int xa_read_header(AVFormatContext *s,
> > > > AVFormatParameters *ap)
> > > > {
> > > > MaxisXADemuxContext *xa = s->priv_data;
> > > > AVStream *st;
> > > >
> > > > /* Extract Header Information from xa file */
> > > > if (!xa_extract_header_info(s))
> > > > return AVERROR(EIO);
> > > >
> > > > /*Set up the XA Audio Decoder*/
> > > > st = av_new_stream(s, 0);
> > > > if (!st)
> > > > return AVERROR(ENOMEM);
> > >
> > > > av_set_pts_info(st, 33, 1, xa->sampleRate);
> > >
> > > i doubt 33 is correct
> >
> >
> > I am not entirely sure what the wrap bits is and what its value should be,
> I
> > copied that from the original ADPCM EA in electronicarts.c
>
> Its the number of bits in the pts, mpeg stores them in 33 bits people
> copied that all over the place without undestanding it ...
>
Changed to 64.
>
> [...]
>
> > >
> > >
> > > > if (ret != packet_size) {
> > > > av_log(s, AV_LOG_WARNING, "Size mismatch");
> > > > return AVERROR(EIO);
> > > > }
> > >
> > > memleak
> > >
> >
> > Added a av_free for pkt here. The test files seem to have an extra few
> > samples but according to the description the samples are contained in
> blocks
> > of 14 bytes for each channel and so I am not sure why there are these extra
> > samples in the file.
>
> Probably because the source did not contain a multiple of 28 samples, it
> would be nice if tha last incomplete packet would be decoded as well.
> Assuming iam guessing correct that it is actual audio at all instead of
> junk.
>
Changed to run check that haven't gone past size specified in header of file.
>
> [...]
> > 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 and hence they are just read
out. XA files don't have these and so the decoder needs to keep track
of them as far as I understand it.
>
>
> [...]
> > @@ -1235,6 +1246,53 @@
>
> > }
> > }
> > break;
> > + case CODEC_ID_ADPCM_EA_MAXIS_XA:
> > + channel = avctx->channels;
> > +
> > + if(channel > 2) {
> > + av_log(avctx, AV_LOG_ERROR, "Only 1 or 2 channels supported");
> > + return -1;
> > + }
> > +
>
> > + coeff1l = ea_adpcm_table[(*src >> 4) & 0x0F];
> > + coeff2l = ea_adpcm_table[((*src >> 4) & 0x0F) + 4];
> > + shift_left = (*src & 0x0F) + 8;
> > + src++;
> > +
> > + if(st) {
> > + coeff1r = ea_adpcm_table[(*src >> 4) & 0x0F];
> > + coeff2r = ea_adpcm_table[((*src >> 4) & 0x0F) + 4];
> > + shift_right = (*src & 0x0F) + 8;
> > + src++;
> > + }
>
> duplicate
>
Removed duplicated code
>
>
> > +
> > + for (count1 = 0; count1 < 14 ; count1++) {
> > + for(count2 = 4; count2 >= 0; count2-=4) { /* Pairwise samples
> LL RR (st) or LL LL (mono) */
>
> > + next_left_sample = ((((*src >> count2) & 0x0F) << 0x1C)
> >> shift_left);
>
> > + next_left_sample = (next_left_sample +
> > + (c->mxa_current_left_sample * coeff1l) +
> > + (c->mxa_previous_left_sample * coeff2l) + 0x80) >> 8;
> > + c->mxa_previous_left_sample = c->mxa_current_left_sample;
> > + c->mxa_current_left_sample =
> av_clip_int16(next_left_sample);
> > + *samples++ = (unsigned short)c->mxa_current_left_sample;
> > +
> > + if(st) {
> > + src++;
> > + next_right_sample = ((((*src >> count2) & 0x0F) <<
> 0x1C) >> shift_right);
>
> > + next_right_sample = (next_right_sample +
> > + (c->mxa_current_right_sample * coeff1r) +
> > + (c->mxa_previous_right_sample * coeff2r) + 0x80)
> >> 8;
> > + c->mxa_previous_right_sample =
> c->mxa_current_right_sample;
> > + c->mxa_current_right_sample =
> av_clip_int16(next_right_sample);
> > + *samples++ = (unsigned
> short)c->mxa_current_right_sample;
> > + src--;
> > + }
>
> duplicate
>
ditto.
>
> [...]
> > +typedef struct MaxisXADemuxContext {
> > + int channels;
> > + uint32_t sampleRate;
> > + int bits;
>
> > + int audio_stream_index;
>
> unneeded
>
Removed.
>
> [...]
> > +static int xa_extract_header_info(AVFormatContext *s)
> > +{
> > + MaxisXADemuxContext *xa = s->priv_data;
> > + ByteIOContext *pb = s->pb;
> > +
> > + url_fskip(pb, 10); /* Skip the XA ID, data size and tag parameters */
> > + xa->channels = get_le16(pb);
> > + xa->sampleRate = get_le32(pb);
> > + url_fskip(pb, 6); /* Skip the avgByteRate and Align parameters, since
> can be calculated */
> > + xa->bits = get_le16(pb);
> > +
> > + av_log(s, AV_LOG_DEBUG, "Channels = %d Sample Rate = %d, Bits =
> %d\n", xa->channels, xa->sampleRate, xa->bits);
> > +
> > + return 1;
> > +}
>
> spliting this out in its own function makes no sense
>
>
Removed.
> > +static int xa_read_header(AVFormatContext *s,
>
> > + AVFormatParameters *ap)
> > +{
> > + MaxisXADemuxContext *xa = s->priv_data;
> > + AVStream *st;
> > +
> > + /* Extract Header Information from xa file */
> > + if (!xa_extract_header_info(s))
> > + return AVERROR(EIO);
> > +
> > + /*Set up the XA Audio Decoder*/
> > + st = av_new_stream(s, 0);
> > + if (!st)
> > + return AVERROR(ENOMEM);
> > + av_set_pts_info(st, 33, 1, xa->sampleRate);
> > + st->codec->codec_type = CODEC_TYPE_AUDIO;
>
> > + st->codec->codec_id = CODEC_ID_ADPCM_EA_MAXIS_XA;
> > + st->codec->bits_per_sample = xa->bits;
>
> > + st->codec->block_align = (xa->bits/8) * xa->channels;
>
> This is wrong.
>
>
Changed values to be read from file. Unless there is something else
that was wrong with this?
> [...]
> > + packet_size = 15*xa->channels;/* 1 byte header and 14 bytes worth of
> samples * number channels per block */
> > + if (av_get_packet(pb, pkt, packet_size) != packet_size) {
> > + av_log(s, AV_LOG_WARNING, "Size mismatch when reading data from
> file\n");
> > + av_free_packet(pkt);
> > + return AVERROR(EIO);
> > + }
>
> This is wrong as well
>
Changed the way the data was read so this shouldn't apply to the current method.
>
> > + 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.
Thanks for your help and patience Micheal.
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 2008-04-09 14:44:44.000000000 +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:
@@ -149,6 +150,7 @@
typedef struct ADPCMContext {
int channel; /* for stereo MOVs, decode left, then decode right, then tell
it's decoded */
ADPCMChannelStatus status[6];
+ int32_t curr_prev_samples[4];
} ADPCMContext;
/* XXX: implement encoding */
@@ -667,8 +669,12 @@
{
ADPCMContext *c = avctx->priv_data;
unsigned int max_channels = 2;
-
+ int i;
switch(avctx->codec->id) {
+ case CODEC_ID_ADPCM_EA_MAXIS_XA:
+ for(i = 0; i < 4; i++)
+ c->curr_prev_samples[i] = 0;
+ break;
case CODEC_ID_ADPCM_EA_R1:
case CODEC_ID_ADPCM_EA_R2:
case CODEC_ID_ADPCM_EA_R3:
@@ -900,6 +906,7 @@
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 +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)];
+ 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;
+ int prev_next = 0;
+ 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;
+ 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];
+ }
+ }
+ 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 2008-04-09 15:14:45.000000000 +0200
@@ -0,0 +1,108 @@
+/*
+ * 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 Audio Demuxer
+ * by Robert Marston ([EMAIL PROTECTED])
+ * for more information on the XA audio format visit
+ * 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;
+} 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);
+ st->codec->bit_rate = get_le32(pb) * 8;/* Value in file is average
byte rate*/
+ 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);
+ packet_size = 15*st->codec->channels;/* 1 byte header and 14 bytes worth
of samples * number channels per block */
+
+ ret = av_get_packet(pb, pkt, packet_size);
+ pkt->stream_index = st->index;
+
+ xa->sent_bytes += packet_size;
+
+ return ret;
+}
+
+AVInputFormat xa_demuxer = {
+ "xa",
+ "Maxis XA Audio Format",
+ sizeof(MaxisXADemuxContext),
+ xa_probe,
+ xa_read_header,
+ xa_read_packet,
+};
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