Add support for AC3 audio in RM container.

git-svn-id: svn://svn.rockbox.org/rockbox/trunk@22155 a1c6a512-1295-4272-9138-f99709370657

8 files changed, 231 insertions, 8 deletions

diff --git a/apps/codecs/SOURCES b/apps/codecs/SOURCES
index dc6819d272..92e4d2d254 100644
--- a/apps/codecs/SOURCES
+++ b/apps/codecs/SOURCES
@@ -11,6 +11,7 @@ alac.c
 #endif
 cook.c
 raac.c
+dnet.c
 mpc.c
 wma.c
 sid.c
diff --git a/apps/codecs/codecs.make b/apps/codecs/codecs.make
index b327bd7c69..41f5bdccd1 100644
--- a/apps/codecs/codecs.make
+++ b/apps/codecs/codecs.make
@@ -77,6 +77,7 @@ $(CODECDIR)/wavpack_enc.codec: $(CODECDIR)/libwavpack.a
 $(CODECDIR)/asap.codec : $(CODECDIR)/libasap.a
 $(CODECDIR)/cook.codec : $(CODECDIR)/libcook.a $(CODECDIR)/librm.a
 $(CODECDIR)/raac.codec : $(CODECDIR)/libfaad.a $(CODECDIR)/librm.a
+$(CODECDIR)/dnet.codec : $(CODECDIR)/liba52.a $(CODECDIR)/librm.a
 
 $(CODECS): $(CODECLIB) # this must be last in codec dependency list
 
diff --git a/apps/codecs/dnet.c b/apps/codecs/dnet.c
new file mode 100644
index 0000000000..12352ed903
--- /dev/null
+++ b/apps/codecs/dnet.c
@@ -0,0 +1,190 @@
+/***************************************************************************
+ *             __________               __   ___.
+ *   Open      \______   \ ____   ____ |  | _\_ |__   _______  ___
+ *   Source     |       _//  _ \_/ ___\|  |/ /| __ \ /  _ \  \/  /
+ *   Jukebox    |    |   (  <_> )  \___|    < | \_\ (  <_> > <  <
+ *   Firmware   |____|_  /\____/ \___  >__|_ \|___  /\____/__/\_ \
+ *                     \/            \/     \/    \/            \/
+ * $Id:$
+ *
+ * Copyright (C) 2009 Mohamed Tarek
+ *
+ * This program 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.
+ *
+ * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
+ * KIND, either express or implied.
+ *
+ ****************************************************************************/
+
+#include "codeclib.h"
+#include <codecs/librm/rm.h>
+#include <inttypes.h>  /* Needed by a52.h */
+#include <codecs/liba52/config-a52.h>
+#include <codecs/liba52/a52.h>
+
+CODEC_HEADER
+
+#define BUFFER_SIZE 4096
+
+#define A52_SAMPLESPERFRAME (6*256)
+
+static a52_state_t *state;
+unsigned long samplesdone;
+unsigned long frequency;
+RMContext rmctx;
+RMPacket pkt;
+
+static void init_rm(RMContext *rmctx)
+{
+    memcpy(rmctx, (void*)(( (intptr_t)ci->id3->id3v2buf + 3 ) &~ 3), sizeof(RMContext));
+}
+
+/* used outside liba52 */
+static uint8_t buf[3840] IBSS_ATTR;
+
+static inline void output_audio(sample_t *samples)
+{
+    ci->yield();
+    ci->pcmbuf_insert(&samples[0], &samples[256], 256);
+}
+
+static void a52_decode_data(uint8_t *start, uint8_t *end)
+{
+    static uint8_t *bufptr = buf;
+    static uint8_t *bufpos = buf + 7;
+    /*
+     * sample_rate and flags are static because this routine could
+     * exit between the a52_syncinfo() and the ao_setup(), and we want
+     * to have the same values when we get back !
+     */
+    static int sample_rate;
+    static int flags;
+    int bit_rate;
+    int len;
+
+    while (1) {
+        len = end - start;
+        if (!len)
+            break;
+        if (len > bufpos - bufptr)
+            len = bufpos - bufptr;
+        memcpy(bufptr, start, len);
+        bufptr += len;
+        start += len;
+        if (bufptr == bufpos) {
+            if (bufpos == buf + 7) {
+                int length;
+
+                length = a52_syncinfo(buf, &flags, &sample_rate, &bit_rate);
+                if (!length) {
+                    //DEBUGF("skip\n");
+                    for (bufptr = buf; bufptr < buf + 6; bufptr++)
+                        bufptr[0] = bufptr[1];
+                    continue;
+                }
+                bufpos = buf + length;
+            } else {
+                /* Unity gain is 1 << 26, and we want to end up on 28 bits
+                   of precision instead of the default 30.
+                 */
+                level_t level = 1 << 24;
+                sample_t bias = 0;
+                int i;
+
+                /* This is the configuration for the downmixing: */
+                flags = A52_STEREO | A52_ADJUST_LEVEL;
+
+                if (a52_frame(state, buf, &flags, &level, bias))
+                    goto error;
+                a52_dynrng(state, NULL, NULL);
+                frequency = sample_rate;
+
+                /* An A52 frame consists of 6 blocks of 256 samples
+                   So we decode and output them one block at a time */
+                for (i = 0; i < 6; i++) {
+                    if (a52_block(state))
+                        goto error;
+                    output_audio(a52_samples(state));
+                    samplesdone += 256;
+                }
+                ci->set_elapsed(samplesdone/(frequency/1000));
+                bufptr = buf;
+                bufpos = buf + 7;
+                continue;
+            error:
+                //logf("Error decoding A52 stream\n");
+                bufptr = buf;
+                bufpos = buf + 7;
+            }
+        }   
+    }
+}
+
+
+/* this is the codec entry point */
+enum codec_status codec_main(void)
+{
+    size_t n;
+    uint8_t *filebuf;
+    int retval, consumed, packet_offset;    
+
+    /* Generic codec initialisation */
+    ci->configure(DSP_SET_STEREO_MODE, STEREO_NONINTERLEAVED);
+    ci->configure(DSP_SET_SAMPLE_DEPTH, 28);
+
+next_track:
+    if (codec_init()) {
+        retval = CODEC_ERROR;
+        goto exit;
+    }
+
+    while (!ci->taginfo_ready)
+        ci->yield();
+
+    ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
+    codec_set_replaygain(ci->id3);
+
+    /* Intializations */
+    state = a52_init(0);
+    ci->memset(&rmctx,0,sizeof(RMContext)); 
+    ci->memset(&pkt,0,sizeof(RMPacket));
+    init_rm(&rmctx);
+
+    /* Seek to the first packet */
+    ci->advance_buffer(rmctx.data_offset + DATA_HEADER_SIZE );
+
+    /* The main decoding loop */
+    while(pkt.timestamp < rmctx.duration) {
+        ci->yield();
+        if (ci->stop_codec || ci->new_track)
+            break;
+
+        if (ci->seek_time) {
+            packet_offset = ci->seek_time / (((rmctx.block_align + PACKET_HEADER_SIZE)*8*1000)/rmctx.bit_rate);
+            ci->seek_buffer(rmctx.data_offset + DATA_HEADER_SIZE + packet_offset*(rmctx.block_align + PACKET_HEADER_SIZE));
+            samplesdone = A52_SAMPLESPERFRAME * packet_offset;          
+            ci->seek_complete();
+        }
+
+        filebuf = ci->request_buffer(&n, rmctx.block_align + PACKET_HEADER_SIZE);
+        consumed = rm_get_packet(&filebuf, &rmctx, &pkt);
+        if(consumed < 0) {
+            DEBUGF("rm_get_packet failed\n");
+            return CODEC_ERROR;
+        }
+        a52_decode_data(filebuf, filebuf + rmctx.block_align);
+        ci->advance_buffer(pkt.length);
+    }
+
+    retval = CODEC_OK;
+
+    if (ci->request_next_track())
+        goto next_track;
+
+exit:
+    a52_free(state);
+    return retval;
+}
diff --git a/apps/codecs/librm/rm.c b/apps/codecs/librm/rm.c
index c802a0c5a9..b205e7f88d 100644
--- a/apps/codecs/librm/rm.c
+++ b/apps/codecs/librm/rm.c
@@ -27,6 +27,8 @@
 #include "codeclib.h"
 #endif
 
+#define SWAP(a, b) do{uint8_t SWAP_tmp= b; b= a; a= SWAP_tmp;}while(0)
+
 void advance_buffer(uint8_t **buf, int val)
 {
     *buf += val;
@@ -464,7 +466,6 @@ void rm_get_packet_fd(int fd,RMContext *rmctx, RMPacket *pkt)
                "    stream  = %d\n"
                "    timestmp= %d\n",pkt->version,pkt->length,pkt->stream_number,pkt->timestamp);
 
-        //getchar();
         if(pkt->version == 0)
         {
             read_uint8(fd,&packet_group);
@@ -550,7 +551,16 @@ int rm_get_packet(uint8_t **src,RMContext *rmctx, RMPacket *pkt)
                 }                
                 rmctx->audio_pkt_cnt = --rmctx->sub_packet_cnt;
             }
-        }        
+        } 
+
+        else if (rmctx->codec_type == CODEC_AC3) {
+        /* The byte order of the data is reversed from standard AC3 */
+            for(x = 0; x < pkt->length - PACKET_HEADER_SIZE; x+=2) {
+                SWAP((*src)[0], (*src)[1]);
+                *src += 2;                                
+            }
+            *src -= x;
+        }
         rmctx->audio_pkt_cnt++;
     }while(++(rmctx->sub_packet_cnt) < h);
 
diff --git a/apps/codecs/librm/rm.h b/apps/codecs/librm/rm.h
index 12e9b18fa3..86fe5e7f1a 100644
--- a/apps/codecs/librm/rm.h
+++ b/apps/codecs/librm/rm.h
@@ -28,7 +28,12 @@
 #define DATA_HEADER_SIZE 18
 #define PACKET_HEADER_SIZE 12
 
-enum codecs{CODEC_COOK, CODEC_AAC};
+enum codecs {
+    CODEC_COOK, 
+    CODEC_AAC,
+    CODEC_AC3
+};
+
 typedef struct rm_packet
 {
     uint8_t *frames[100]; /* Pointers to ordered audio frames in buffer */
diff --git a/apps/metadata.c b/apps/metadata.c
index f227776c0a..63547646ca 100644
--- a/apps/metadata.c
+++ b/apps/metadata.c
@@ -121,6 +121,9 @@ const struct afmt_entry audio_formats[AFMT_NUM_CODECS] =
     /* AAC in RM/RA */
     [AFMT_RM_AAC] =
         AFMT_ENTRY("RAAC",  "raac",     NULL,        "rm\0ra\0rmvb\0"      ),
+    /* AC3 in RM/RA */
+    [AFMT_RM_AC3] =
+        AFMT_ENTRY("AC3",   "dnet",     NULL,        "rm\0ra\0rmvb\0"      ),
 #endif
 };
 
diff --git a/apps/metadata.h b/apps/metadata.h
index cefc6c3945..c43d2c2260 100644
--- a/apps/metadata.h
+++ b/apps/metadata.h
@@ -63,6 +63,7 @@ enum
     AFMT_SAP,          /* Amiga 8Bit SAP Format */
     AFMT_RM_COOK,      /* Cook in RM/RA */
     AFMT_RM_AAC,       /* AAC in RM/RA */
+    AFMT_RM_AC3,       /* AC3 in RM/RA */
 #endif
 
     /* add new formats at any index above this line to have a sensible order -
diff --git a/apps/metadata/rm.c b/apps/metadata/rm.c
index 4be0de647f..c74acef5dd 100644
--- a/apps/metadata/rm.c
+++ b/apps/metadata/rm.c
@@ -160,24 +160,32 @@ static inline int real_read_audio_stream_info(int fd, RMContext *rmctx)
            skipped += 1;
        }
   
-       read_uint32be(fd, &rmctx->extradata_size);
-       skipped += 4;
-       read(fd, rmctx->codec_extradata, rmctx->extradata_size);
-       skipped += rmctx->extradata_size;
        switch(fourcc) {
            case FOURCC('c','o','o','k'):               
                rmctx->codec_type = CODEC_COOK;
+               read_uint32be(fd, &rmctx->extradata_size);
+               skipped += 4;
+               read(fd, rmctx->codec_extradata, rmctx->extradata_size);
+               skipped += rmctx->extradata_size;
                break;
 
            case FOURCC('r','a','a','c'):
            case FOURCC('r','a','c','p'):
                rmctx->codec_type = CODEC_AAC;
+               read_uint32be(fd, &rmctx->extradata_size);
+               skipped += 4;
+               read(fd, rmctx->codec_extradata, rmctx->extradata_size);
+               skipped += rmctx->extradata_size;
+               break;
+
+           case FOURCC('d','n','e','t'):
+               rmctx->codec_type = CODEC_AC3;
                break;
 
            default: /* Not a supported codec */
                return -1;
        }
-
+       
        DEBUGF("        flavor = %d\n",flavor);
        DEBUGF("        coded_frame_size = %ld\n",coded_framesize);
        DEBUGF("        sub_packet_h = %d\n",rmctx->sub_packet_h);
@@ -407,6 +415,10 @@ bool get_rm_metadata(int fd, struct mp3entry* id3)
         case CODEC_AAC:
             id3->codectype = AFMT_RM_AAC;
             break;
+
+        case CODEC_AC3:
+            id3->codectype = AFMT_RM_AC3;
+            break;
     }
     
     id3->bitrate = rmctx->bit_rate / 1000;