[Ffmpeg-devel] [PATCH] THP PCM decoder (GSoC Qualification)

Marco Gerards mgerards
Wed Apr 4 02:44:18 CEST 2007


Michael Niedermayer <michaelni at gmx.at> writes:

Hi,

> Hi
>
> On Tue, Apr 03, 2007 at 08:59:31PM +0200, Marco Gerards wrote:
>> Michael Niedermayer <michaelni at gmx.at> writes:
>> 
>> > Hi
>> >
>> > On Tue, Apr 03, 2007 at 03:21:56PM +0200, Marco Gerards wrote:
>> > [...]
>> >> In this new patch there are still two issues.  For some reason the
>> >> quality of the sound was not that good.  It wasn't as bad as someone
>> >> reported, for example he said that there were issues with mono sound.
>> >> I can not reproduce this.
>> >> 
>> >> The problem is in stereo sound.  I have disabled stereo in this patch
>> >> and the sound is just fine now.  To be honest, I am not sure what the
>> >> problem is.  I have tried all kinds of things without much success.
>> >> Perhaps it is better to commit this first.  After that I can fix this
>> >> with another patch.
>> >
>> > hmm did you properly interleave right and left channel samples? that is
>> > does it work if you decode just the other channel?
>> > could it be that the encoding is R+L,R-L ? instead of R,L
>> 
>> I tried about anything.  This is the first time I am working with
>> audio, so I perhaps made a silly mistake.
>> 
>> What I tried was interleaving the data send to the output buffer.  The
>> PCM data in the THP files is stored in packets.  First x samples for
>> the first channel are stored, after that x samples for the second
>> channel.
>> 
>> When I decode the other channel it seems to work as well.  So I assume
>> the problem is in how I fill the output buffer and how I tell ffmpeg
>> the format (stereo, interleaved samples) of this data.  How does one
>> in general deal with stereo from a decoder?
>
> stereo in ffmpeg is always
> channel0-sample0, channel1-sample0, channel0-sample1, channel1-sample1, channel0-sample2, channel1-sample2, ...

Well, I have this now and it works... kinda.  When I am playing either
one of the two channels as mono sound, it works.  When I am using both
channels, there is some kind of noise/cracks.  The noise is not there
all the time and it is not very loud, but I am mentioning this because
there must still be something that I do wrong...

>> 
>> [...]
>> 
>> >> Index: libavcodec/allcodecs.c
>> >> ===================================================================
>> >> --- libavcodec/allcodecs.c	(revision 8605)
>> >> +++ libavcodec/allcodecs.c	(working copy)
>> >> @@ -242,6 +242,7 @@
>> >>      REGISTER_ENCDEC (ADPCM_SBPRO_3, adpcm_sbpro_3);
>> >>      REGISTER_ENCDEC (ADPCM_SBPRO_4, adpcm_sbpro_4);
>> >>      REGISTER_ENCDEC (ADPCM_SWF, adpcm_swf);
>> >> +    REGISTER_ENCDEC (ADPCM_THP, adpcm_thp);
>> >
>> > ENCDEC ?
>> 
>> 
>> Baptiste said "use CODEC and dummy return -1 to stay in conformance
>> with others (see IMA_QT)".  I assumed it also applied to allcodecs.c.
>> Should I change this to REGISTER_DECODER?
>
> yes and flame baptiste ;)

Sorry, I didn't mean to blame anyone for my mistakes.  Neither do I
want to insult anyone.  I just misunderstood him, which is totally my
fault.

>>
>> >>      REGISTER_ENCDEC (ADPCM_XA, adpcm_xa);
>> >>      REGISTER_ENCDEC (ADPCM_YAMAHA, adpcm_yamaha);
>> >>  
>> >> Index: libavcodec/Makefile
>> >> ===================================================================
>> >> --- libavcodec/Makefile	(revision 8605)
>> >> +++ libavcodec/Makefile	(working copy)
>> >> @@ -246,6 +246,8 @@
>> >>  OBJS-$(CONFIG_ADPCM_SBPRO_4_ENCODER)   += adpcm.o
>> >>  OBJS-$(CONFIG_ADPCM_SWF_DECODER)       += adpcm.o
>> >>  OBJS-$(CONFIG_ADPCM_SWF_ENCODER)       += adpcm.o
>> >> +OBJS-$(CONFIG_ADPCM_THP_DECODER)       += adpcm.o
>> >> +OBJS-$(CONFIG_ADPCM_THP_ENCODER)       += adpcm.o
>> >>  OBJS-$(CONFIG_ADPCM_XA_DECODER)        += adpcm.o
>> >
>> > unless you write an ADPCM_THP encoder there shouldnt be
>> > a CONFIG_ADPCM_THP_ENCODER ...
>> 
>> Same as above.  Should I remove this?
>
> yes

Done.

Hopefully I got everything right this time, except stereo...  I hope
you guys aren't losing your patience with me... ;-)

--
Marco

Index: libavcodec/Makefile
===================================================================
--- libavcodec/Makefile	(revision 8622)
+++ libavcodec/Makefile	(working copy)
@@ -247,6 +247,7 @@
 OBJS-$(CONFIG_ADPCM_SBPRO_4_ENCODER)   += adpcm.o
 OBJS-$(CONFIG_ADPCM_SWF_DECODER)       += adpcm.o
 OBJS-$(CONFIG_ADPCM_SWF_ENCODER)       += adpcm.o
+OBJS-$(CONFIG_ADPCM_THP_DECODER)       += adpcm.o
 OBJS-$(CONFIG_ADPCM_XA_DECODER)        += adpcm.o
 OBJS-$(CONFIG_ADPCM_XA_ENCODER)        += adpcm.o
 OBJS-$(CONFIG_ADPCM_YAMAHA_DECODER)    += adpcm.o
Index: libavcodec/allcodecs.c
===================================================================
--- libavcodec/allcodecs.c	(revision 8622)
+++ libavcodec/allcodecs.c	(working copy)
@@ -242,6 +242,7 @@
     REGISTER_ENCDEC (ADPCM_SBPRO_3, adpcm_sbpro_3);
     REGISTER_ENCDEC (ADPCM_SBPRO_4, adpcm_sbpro_4);
     REGISTER_ENCDEC (ADPCM_SWF, adpcm_swf);
+    REGISTER_DECODER(ADPCM_THP, adpcm_thp);
     REGISTER_ENCDEC (ADPCM_XA, adpcm_xa);
     REGISTER_ENCDEC (ADPCM_YAMAHA, adpcm_yamaha);
 
Index: libavcodec/avcodec.h
===================================================================
--- libavcodec/avcodec.h	(revision 8622)
+++ libavcodec/avcodec.h	(working copy)
@@ -198,6 +198,7 @@
     CODEC_ID_ADPCM_SBPRO_4,
     CODEC_ID_ADPCM_SBPRO_3,
     CODEC_ID_ADPCM_SBPRO_2,
+    CODEC_ID_ADPCM_THP,
 
     /* AMR */
     CODEC_ID_AMR_NB= 0x12000,
@@ -2409,6 +2410,7 @@
 PCM_CODEC(CODEC_ID_ADPCM_SBPRO_4, adpcm_sbpro_4);
 PCM_CODEC(CODEC_ID_ADPCM_SMJPEG,  adpcm_ima_smjpeg);
 PCM_CODEC(CODEC_ID_ADPCM_SWF,     adpcm_swf);
+PCM_CODEC(CODEC_ID_ADPCM_THP,     adpcm_thp);
 PCM_CODEC(CODEC_ID_ADPCM_XA,      adpcm_xa);
 PCM_CODEC(CODEC_ID_ADPCM_YAMAHA,  adpcm_yamaha);
 
Index: libavcodec/adpcm.c
===================================================================
--- libavcodec/adpcm.c	(revision 8622)
+++ libavcodec/adpcm.c	(working copy)
@@ -1308,6 +1308,66 @@
             src++;
         }
         break;
+    case CODEC_ID_ADPCM_THP:
+      {
+        GetBitContext gb;
+        long table[16][2];
+        int samplecnt;
+        int prev1[2], prev2[2];
+        int ch;
+
+        if (buf_size < 80) {
+            av_log(avctx, AV_LOG_ERROR, "frame too small\n");
+            return -1;
+        }
+
+        init_get_bits(&gb, src, buf_size * 8);
+        src += buf_size;
+
+                    get_bits(&gb, 32); /* Channel size */
+        samplecnt = get_bits(&gb, 32);
+
+        for (ch = 0; ch < 2; ch++)
+            for (i = 0; i < 16; i++)
+                table[i][ch] = get_sbits(&gb, 16);
+
+        /* Initialize the previous sample.  */
+        for (ch = 0; ch < 2; ch++) {
+            prev1[ch] = get_sbits(&gb, 16);
+            prev2[ch] = get_sbits(&gb, 16);
+        }
+
+        if (samples + samplecnt >= samples_end) {
+            av_log(avctx, AV_LOG_ERROR, "allocated output buffer is too small\n");
+            return -1;
+        }
+
+        for (ch = 0; ch <= st; ch++) {
+            samples = (unsigned short *) data + ch;
+
+            /* Read in every sample for this channel.  */
+            for (i = 0; i < samplecnt / 14; i++) {
+                uint8_t index = get_bits (&gb, 4) & 7;
+                unsigned int exp = 1 << get_bits (&gb, 4);
+                signed long factor1 = table[index * 2][ch];
+                signed long factor2 = table[index * 2 + 1][ch];
+
+                /* Decode 14 samples.  */
+                for (n = 0; n < 14; n++) {
+                    int sampledat = get_sbits (&gb, 4);
+
+                    *samples = ((prev1[ch]*factor1 
+                                + prev2[ch]*factor2) >> 11) + (sampledat * exp);
+                    prev2[ch] = prev1[ch];
+                    prev1[ch] = *samples++;
+                    samples += st;
+                }
+            }
+        }
+
+        break;
+      }
+
     default:
         return -1;
     }
@@ -1368,5 +1428,6 @@
 ADPCM_CODEC(CODEC_ID_ADPCM_SBPRO_4, adpcm_sbpro_4);
 ADPCM_CODEC(CODEC_ID_ADPCM_SBPRO_3, adpcm_sbpro_3);
 ADPCM_CODEC(CODEC_ID_ADPCM_SBPRO_2, adpcm_sbpro_2);
+ADPCM_CODEC(CODEC_ID_ADPCM_THP, adpcm_thp);
 
 #undef ADPCM_CODEC
Index: doc/ffmpeg-doc.texi
===================================================================
--- doc/ffmpeg-doc.texi	(revision 8622)
+++ doc/ffmpeg-doc.texi	(working copy)
@@ -902,7 +902,7 @@
 @tab This format is used in non-Windows version of Feeble Files game and
 different game cutscenes repacked for use with ScummVM.
 @item THP @tab    @tab X
- at tab Used on the Nintendo GameCube (video only)
+ at tab Used on the Nintendo GameCube
 @end multitable
 
 @code{X} means that encoding (resp. decoding) is supported.
Index: libavformat/thp.c
===================================================================
--- libavformat/thp.c	(revision 8622)
+++ libavformat/thp.c	(working copy)
@@ -35,10 +35,12 @@
     int              next_frame;
     int              next_framesz;
     int              video_stream_index;
+    int              audio_stream_index;
     int              compcount;
     unsigned char    components[16];
     AVStream*        vst;
     int              has_audio;
+    int              audiosize;
 } ThpDemuxContext;
 
 
@@ -116,7 +118,23 @@
              get_be32(pb); /* Unknown.  */
         }
       else if (thp->components[i] == 1) {
-          /* XXX: Required for audio playback.  */
+          if (thp->has_audio != 0)
+             break;
+
+          /* Audio component.  */
+          st = av_new_stream(s, 0);
+          if (!st)
+              return AVERROR_NOMEM;
+
+          st->codec->codec_type = CODEC_TYPE_AUDIO;
+          st->codec->codec_id = CODEC_ID_ADPCM_THP;
+          st->codec->codec_tag = 0;  /* no fourcc */
+          st->codec->channels    = get_be32(pb); /* numChannels.  */
+          st->codec->sample_rate = get_be32(pb); /* Frequency.  */
+
+          av_set_pts_info(st, 64, 1, st->codec->sample_rate);
+
+          thp->audio_stream_index = st->index;
           thp->has_audio = 1;
       }
     }
@@ -132,6 +150,8 @@
     int size;
     int ret;
 
+    if (thp->audiosize == 0) {
+
     /* Terminate when last frame is reached.  */
     if (thp->frame >= thp->framecnt)
        return AVERROR_IO;
@@ -145,8 +165,12 @@
                         get_be32(pb); /* Previous total size.  */
     size              = get_be32(pb); /* Total size of this frame.  */
 
+    /* Store the audiosize so the next time this function is called,
+       the audio can be read.  */
     if (thp->has_audio)
-                        get_be32(pb); /* Audio size.  */
+       thp->audiosize = get_be32(pb); /* Audio size.  */
+    else
+       thp->frame++;
 
     ret = av_get_packet(pb, pkt, size);
     if (ret != size) {
@@ -155,8 +179,19 @@
     }
 
     pkt->stream_index = thp->video_stream_index;
-    thp->frame++;
+    }
+    else {
+       ret = av_get_packet(pb, pkt, thp->audiosize);
+       if (ret != thp->audiosize) {
+          av_free_packet(pkt);
+          return AVERROR_IO;
+       }
 
+      pkt->stream_index = thp->audio_stream_index;
+      thp->audiosize = 0;
+      thp->frame++;
+    }
+
     return 0;
 }
 





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