Adding support for playback of aac audio in rm container, with seeking.

git-svn-id: svn://svn.rockbox.org/rockbox/trunk@22031 a1c6a512-1295-4272-9138-f99709370657

7 files changed, 217 insertions, 13 deletions

diff --git a/apps/codecs/SOURCES b/apps/codecs/SOURCES
index 44a8498fa9..dc6819d272 100644
--- a/apps/codecs/SOURCES
+++ b/apps/codecs/SOURCES
@@ -10,6 +10,7 @@ wavpack.c
 alac.c
 #endif
 cook.c
+raac.c
 mpc.c
 wma.c
 sid.c
diff --git a/apps/codecs/codecs.make b/apps/codecs/codecs.make
index fbe3b1c978..b327bd7c69 100644
--- a/apps/codecs/codecs.make
+++ b/apps/codecs/codecs.make
@@ -76,6 +76,7 @@ $(CODECDIR)/wma.codec : $(CODECDIR)/libwma.a
 $(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
 
 $(CODECS): $(CODECLIB) # this must be last in codec dependency list
 
diff --git a/apps/codecs/librm/rm.c b/apps/codecs/librm/rm.c
index c9c40986ba..a48fb51cc6 100644
--- a/apps/codecs/librm/rm.c
+++ b/apps/codecs/librm/rm.c
@@ -529,13 +529,28 @@ int rm_get_packet(uint8_t **src,RMContext *rmctx, RMPacket *pkt)
         
         advance_buffer(src,12);
         consumed += 12;
-        for(x = 0 ; x < w/sps; x++)
-        {
-            place = sps*(h*x+((h+1)/2)*(y&1)+(y>>1)); 
-            pkt->frames[place/sps] = *src;
-            advance_buffer(src,sps);
-            consumed += sps;
-        }
+        if (rmctx->codec_type == cook) {
+            for(x = 0 ; x < w/sps; x++)
+            {
+                place = sps*(h*x+((h+1)/2)*(y&1)+(y>>1)); 
+                pkt->frames[place/sps] = *src;
+                advance_buffer(src,sps);
+                consumed += sps;
+            }
+         }
+        else if (rmctx->codec_type == aac) {
+            rmctx->sub_packet_cnt = (get_uint16be(*src) & 0xf0) >> 4;
+            advance_buffer(src, 2);
+            consumed += 2;
+            if (rmctx->sub_packet_cnt) {
+                for(x = 0; x < rmctx->sub_packet_cnt; x++) {
+                    rmctx->sub_packet_lengths[x] = get_uint16be(*src);
+                    advance_buffer(src, 2);
+                    consumed += 2;
+                }                
+                rmctx->audio_pkt_cnt = --rmctx->sub_packet_cnt;
+            }
+        }        
         rmctx->audio_pkt_cnt++;
     }while(++(rmctx->sub_packet_cnt) < h);
 
diff --git a/apps/codecs/raac.c b/apps/codecs/raac.c
new file mode 100644
index 0000000000..cfc5b4a460
--- /dev/null
+++ b/apps/codecs/raac.c
@@ -0,0 +1,176 @@
+/***************************************************************************
+ *             __________               __   ___.
+ *   Open      \______   \ ____   ____ |  | _\_ |__   _______  ___
+ *   Source     |       _//  _ \_/ ___\|  |/ /| __ \ /  _ \  \/  /
+ *   Jukebox    |    |   (  <_> )  \___|    < | \_\ (  <_> > <  <
+ *   Firmware   |____|_  /\____/ \___  >__|_ \|___  /\____/__/\_ \
+ *                     \/            \/     \/    \/            \/
+ * $Id: aac.c 19743 2009-01-10 21:10:56Z zagor $
+ *
+ * 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 "librm/rm.h"
+#include "libfaad/common.h"
+#include "libfaad/structs.h"
+#include "libfaad/decoder.h"
+#include "libfaad/output.h"
+
+CODEC_HEADER
+#define DATA_HEADER_SIZE 18
+static void init_rm(RMContext *rmctx)
+{
+    memcpy(rmctx, (void*)(( (intptr_t)ci->id3->id3v2buf + 3 ) &~ 3), sizeof(RMContext));
+}
+
+RMContext rmctx;
+RMPacket pkt;
+/* this is the codec entry point */
+enum codec_status codec_main(void)
+{
+    static NeAACDecFrameInfo frame_info;
+    NeAACDecHandle decoder;
+    size_t n;    
+    int32_t *output;
+    unsigned int i;
+    unsigned char* buffer;
+    int err, consumed, pkt_offset, skipped = 0;
+    uint32_t s = 0; /* sample rate */
+    unsigned char c = 0; /* channels */
+    /* Generic codec initialisation */
+    ci->configure(DSP_SET_STEREO_MODE, STEREO_INTERLEAVED);
+    ci->configure(DSP_SET_SAMPLE_DEPTH, 16);
+
+next_track:
+    err = CODEC_OK;
+
+    if (codec_init()) {
+        DEBUGF("FAAD: Codec init error\n");
+        return CODEC_ERROR;
+    }
+
+    while (!*ci->taginfo_ready && !ci->stop_codec)
+        ci->sleep(1);
+
+    ci->memset(&rmctx,0,sizeof(RMContext));
+    ci->memset(&pkt,0,sizeof(RMPacket));
+    init_rm(&rmctx);    
+    ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
+    codec_set_replaygain(ci->id3);
+   
+    /* initialise the sound converter */
+    decoder = NeAACDecOpen();
+    
+    if (!decoder) {
+        DEBUGF("FAAD: Decode open error\n");
+        err = CODEC_ERROR;
+        goto done;
+    }   
+    NeAACDecConfigurationPtr conf = NeAACDecGetCurrentConfiguration(decoder);
+    conf->outputFormat = FAAD_FMT_16BIT;    
+    NeAACDecSetConfiguration(decoder, conf);    
+    
+    decoder->config.defObjectType = rmctx.codec_extradata[0];
+    decoder->config.defSampleRate = rmctx.sample_rate;      
+    err = NeAACDecInit(decoder, NULL, 0, &s, &c);    
+   
+    if (err) {
+        DEBUGF("FAAD: DecInit: %d, %d\n", err, decoder->object_type);
+        err = CODEC_ERROR;
+        goto done;
+    }   
+    ci->id3->frequency = s;    
+    ci->set_elapsed(0);
+    ci->advance_buffer(rmctx.data_offset + DATA_HEADER_SIZE);
+
+    /* The main decoding loop */
+seek_start:
+    while (1) {
+        ci->yield();
+        if (ci->stop_codec || ci->new_track) {
+            break;
+        }
+
+        if (ci->seek_time) {            
+            
+            /* Do not allow seeking beyond the file's length */
+            if ((unsigned) ci->seek_time > ci->id3->length) {
+                ci->seek_complete();
+                goto done;
+            }       
+
+            ci->seek_buffer(rmctx.data_offset + DATA_HEADER_SIZE);            
+
+            /* Seek to the start of the track */
+            if (ci->seek_time == 1) {
+                ci->set_elapsed(0);
+                ci->seek_complete();
+                goto seek_start;           
+            }
+            
+            skipped = 0;                                                                                       
+            while(1) {                
+                buffer = ci->request_buffer(&n,rmctx.audio_framesize + 1000);               
+                pkt_offset = skipped - pkt.length;
+                consumed = rm_get_packet(&buffer, &rmctx, &pkt);
+                if(consumed < 0) {
+                     DEBUGF("rm_get_packet failed\n");
+                     return CODEC_ERROR;
+                }
+                skipped += pkt.length;
+                if(pkt.timestamp > (unsigned)ci->seek_time) break;                
+                ci->advance_buffer(pkt.length);
+            }           
+            ci->seek_buffer(pkt_offset + rmctx.data_offset + DATA_HEADER_SIZE);
+            buffer = ci->request_buffer(&n,rmctx.audio_framesize + 1000);
+            ci->seek_complete();            
+        }
+
+        /* Request the required number of bytes from the input buffer */ 
+        buffer=ci->request_buffer(&n,rmctx.audio_framesize + 1000);        
+        consumed = rm_get_packet(&buffer, &rmctx, &pkt);
+        if(consumed < 0) {
+             DEBUGF("rm_get_packet failed\n");
+             return CODEC_ERROR;
+        }
+
+        if (pkt.timestamp >= ci->id3->length)
+            goto done;
+        /* Decode one block - returned samples will be host-endian */                           
+        for(i = 0; i < rmctx.sub_packet_cnt; i++) {
+            output = (int32_t *)NeAACDecDecode(decoder, &frame_info, buffer, rmctx.sub_packet_lengths[i]);                                           
+            buffer += rmctx.sub_packet_lengths[i];                      
+            if (frame_info.error > 0) {
+                DEBUGF("FAAD: decode error '%s'\n", NeAACDecGetErrorMessage(frame_info.error));
+                err = CODEC_ERROR;
+                goto exit;
+            }
+            output = (int32_t *) output_to_PCM(decoder, decoder->time_out, output,
+                rmctx.nb_channels, decoder->frameLength, decoder->config.outputFormat);
+            ci->pcmbuf_insert(output, NULL, frame_info.samples/rmctx.nb_channels);
+            ci->set_elapsed(pkt.timestamp);            
+        }                                       
+        
+        ci->advance_buffer(pkt.length);              
+    }
+
+    err = CODEC_OK;
+
+done:
+    if (ci->request_next_track())
+        goto next_track;
+
+exit:
+    return err;
+}
+ 
diff --git a/apps/metadata.c b/apps/metadata.c
index a0409a83ac..80f2c126e2 100644
--- a/apps/metadata.c
+++ b/apps/metadata.c
@@ -118,6 +118,9 @@ const struct afmt_entry audio_formats[AFMT_NUM_CODECS] =
     /* Cook in RM/RA */
     [AFMT_COOK] =
         AFMT_ENTRY("Cook",  "cook",   NULL,          "rm\0ra\0"      ),
+    /* AAC in RM/RA */
+    [AFMT_RAAC] =
+        AFMT_ENTRY("RAAC",  "raac",     NULL,          NULL      ),
 #endif
 };
 
diff --git a/apps/metadata.h b/apps/metadata.h
index f3b50c947d..0125b29e5e 100644
--- a/apps/metadata.h
+++ b/apps/metadata.h
@@ -62,6 +62,7 @@ enum
     AFMT_MOD,          /* Amiga MOD File Format */
     AFMT_SAP,          /* Amiga 8Bit SAP Format */
     AFMT_COOK,         /* Cook in RM/RA */
+    AFMT_RAAC,         /* AAC 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 bfaa803be7..95889c35c3 100644
--- a/apps/metadata/rm.c
+++ b/apps/metadata/rm.c
@@ -169,6 +169,11 @@ static inline int real_read_audio_stream_info(int fd, RMContext *rmctx)
                rmctx->codec_type = cook;
                break;
 
+           case FOURCC('r','a','a','c'):
+           case FOURCC('r','a','c','p'):
+               rmctx->codec_type = aac;
+               break;
+
            default: /* Not a supported codec */
                return -1;
        }
@@ -183,7 +188,10 @@ static inline int real_read_audio_stream_info(int fd, RMContext *rmctx)
        DEBUGF("        fourcc = %s\n",fourcc2str(fourcc));
        DEBUGF("        codec_extra_data_length = %ld\n",rmctx->extradata_size);
        DEBUGF("        codec_extradata :\n");
-       print_cook_extradata(rmctx);
+       if(rmctx->codec_type == cook) {
+           DEBUGF("        cook_extradata :\n");
+           print_cook_extradata(rmctx);
+       }
 
     }
 
@@ -391,12 +399,11 @@ bool get_rm_metadata(int fd, struct mp3entry* id3)
     id3->artist = id3->id3v1buf[1];
     id3->comment = id3->id3v1buf[3];
 
-    /*switch(rmctx->codec_type)
+    switch(rmctx->codec_type)
     {
-        case cook:
-            id3->codectype = AFMT_COOK;
-            break;
-    }*/
+        case aac:
+            id3->codectype = AFMT_RAAC;            
+    }
     
     id3->bitrate = rmctx->bit_rate / 1000;
     id3->frequency = rmctx->sample_rate;