Submit FS#11918: Add 2 more codec types to be able to differentiate between AAC / AAC-HE and MPC SV7 / SV8. Additionally handle ATARI soundfiles in get_codec_base_type() as intended.

git-svn-id: svn://svn.rockbox.org/rockbox/trunk@29199 a1c6a512-1295-4272-9138-f99709370657

apps/buffering.c

@@ -25,6 +25,8 @@
 #include <stdlib.h>
 #include <ctype.h>
 #include <inttypes.h>
+#define assert(a)
+
 #include "buffering.h"
 
 #include "storage.h"
@@ -611,6 +613,16 @@ static inline bool buffer_is_low(void)
     return data_counters.useful < (conf_watermark / 2);
 }
 
+static uintptr_t beyond_handle(struct memory_handle *h)
+{
+    /*
+     * the last handle on the chain must leave at least one byte
+     * between itself and the first handle, to avoid overflowing the
+     * ring by advancing buf_widx up to buf_ridx
+     */
+    return h->next != 0 ? ringbuf_offset(h->next) : ringbuf_sub(buf_ridx, 1);
+}
+
 /* Buffer data for the given handle.
    Return whether or not the buffering should continue explicitly.  */
 static bool buffer_handle(int handle_id)
@@ -669,10 +681,10 @@ static bool buffer_handle(int handle_id)
                              buffer_len - h->widx);
 
         ssize_t overlap;
-        uintptr_t next_handle = ringbuf_offset(h->next);
+        uintptr_t next_handle = beyond_handle(h);
 
         /* stop copying if it would overwrite the reading position */
-        if (ringbuf_add_cross(h->widx, copy_n, buf_ridx) >= 0)
+        if (h->widx == next_handle || ringbuf_add_cross(h->widx, copy_n, buf_ridx) >= 0)
             return false;
 
         /* FIXME: This would overwrite the next handle
@@ -789,8 +801,7 @@ static void rebuffer_handle(int handle_id, size_t newpos)
     LOGFQUEUE("buffering >| Q_RESET_HANDLE %d", handle_id);
     queue_send(&buffering_queue, Q_RESET_HANDLE, handle_id);
 
-    uintptr_t next = ringbuf_offset(h->next);
+    if (ringbuf_sub(beyond_handle(h), h->data) < h->filesize - newpos)
-    if (ringbuf_sub(next, h->data) < h->filesize - newpos)
     {
         /* There isn't enough space to rebuffer all of the track from its new
            offset, so we ask the user to free some */

apps/codec_thread.c

@@ -102,16 +102,45 @@ static bool codec_load_next_track(void);
 
 /** misc external functions */
 
+/* Used to check whether a new codec must be loaded. See array audio_formats[]
+ * in metadata.c */
 int get_codec_base_type(int type)
 {
+    int base_type = type;
     switch (type) {
         case AFMT_MPA_L1:
         case AFMT_MPA_L2:
         case AFMT_MPA_L3:
-            return AFMT_MPA_L3;
+            base_type = AFMT_MPA_L3;
+            break;
+        case AFMT_MPC_SV7:
+        case AFMT_MPC_SV8:
+            base_type = AFMT_MPC_SV7;
+            break;
+        case AFMT_MP4_AAC:
+        case AFMT_MP4_AAC_HE:
+            base_type = AFMT_MP4_AAC;
+            break;
+        case AFMT_SAP:
+        case AFMT_CMC:
+        case AFMT_CM3:
+        case AFMT_CMR:
+        case AFMT_CMS:
+        case AFMT_DMC:
+        case AFMT_DLT:
+        case AFMT_MPT:
+        case AFMT_MPD:
+        case AFMT_RMT:
+        case AFMT_TMC:
+        case AFMT_TM8:
+        case AFMT_TM2:
+            base_type = AFMT_SAP;
+            break;
+        default:
+            break;
     }
 
-    return type;
+    return base_type;
 }
 
 const char *get_codec_filename(int cod_spec)

apps/codecs/libm4a/demux.c

@@ -257,7 +257,8 @@ static bool read_chunk_stsd(qtmovie_t *qtmovie, size_t chunk_len)
               stream_skip(qtmovie->stream, entry_remaining);
 
         } else if (qtmovie->res->format==MAKEFOURCC('m','p','4','a')) {
-          if (qtmovie->stream->ci->id3->codectype!=AFMT_MP4_AAC) {
+          if (qtmovie->stream->ci->id3->codectype!=AFMT_MP4_AAC &&
+              qtmovie->stream->ci->id3->codectype!=AFMT_MP4_AAC_HE) {
                return false;
           }
 

apps/metadata.c

@@ -86,9 +86,12 @@ const struct afmt_entry audio_formats[AFMT_NUM_CODECS] =
     /* FLAC */
     [AFMT_FLAC] =
         AFMT_ENTRY("FLAC",  "flac", NULL,       get_flac_metadata,  "flac\0"),
-    /* Musepack */
+    /* Musepack SV7 */
-    [AFMT_MPC] =
+    [AFMT_MPC_SV7] =
-        AFMT_ENTRY("MPC",   "mpc",  NULL,       get_musepack_metadata,"mpc\0"),
+        AFMT_ENTRY("MPCv7", "mpc",  NULL,       get_musepack_metadata,"mpc\0"),
+    /* Musepack SV8 */
+    [AFMT_MPC_SV8] =
+        AFMT_ENTRY("MPCv8", "mpc",  NULL,       get_musepack_metadata,"mpc\0"),
     /* A/52 (aka AC3) audio */                  
     [AFMT_A52] =
         AFMT_ENTRY("AC3",   "a52",  NULL,       get_a52_metadata,   "a52\0ac3\0"),
@@ -101,6 +104,9 @@ const struct afmt_entry audio_formats[AFMT_NUM_CODECS] =
     /* Advanced Audio Coding in M4A container */
     [AFMT_MP4_AAC] =
         AFMT_ENTRY("AAC",   "aac",  NULL,       get_mp4_metadata,   "mp4\0"),
+    /* Advanced Audio Coding High Efficiency in M4A container */
+    [AFMT_MP4_AAC_HE] =
+        AFMT_ENTRY("AAC-HE","aac",  NULL,       get_mp4_metadata,   "mp4\0"),
     /* Shorten */
     [AFMT_SHN] =
         AFMT_ENTRY("SHN","shorten", NULL,       get_shn_metadata,   "shn\0"),

apps/metadata.h

@@ -46,11 +46,13 @@ enum
     AFMT_PCM_WAV,      /* Uncompressed PCM in a WAV file */
     AFMT_OGG_VORBIS,   /* Ogg Vorbis */
     AFMT_FLAC,         /* FLAC */
-    AFMT_MPC,          /* Musepack */
+    AFMT_MPC_SV7,      /* Musepack SV7 */
+    AFMT_MPC_SV8,      /* Musepack SV8 */
     AFMT_A52,          /* A/52 (aka AC3) audio */
     AFMT_WAVPACK,      /* WavPack */
     AFMT_MP4_ALAC,     /* Apple Lossless Audio Codec */
     AFMT_MP4_AAC,      /* Advanced Audio Coding (AAC) in M4A container */
+    AFMT_MP4_AAC_HE,   /* Advanced Audio Coding (AAC-HE) in M4A container */
     AFMT_SHN,          /* Shorten */
     AFMT_SID,          /* SID File Format */
     AFMT_ADX,          /* ADX File Format */

apps/metadata/mp4.c

@@ -300,8 +300,7 @@ static bool read_mp4_esds(int fd, struct mp3entry* id3, uint32_t* size)
         /* Skip 13 bits from above, plus 3 bits, then read 11 bits */
         else if ((length >= 4) && (((bits >> 5) & 0x7ff) == 0x2b7))
         {
-            /* extensionAudioObjectType */
+            /* We found an extensionAudioObjectType */
-            DEBUGF("MP4: extensionAudioType\n");
             type = bits & 0x1f;         /* Object type - 5 bits*/
             bits = get_long_be(&buf[4]);
             
@@ -680,7 +679,6 @@ static bool read_mp4_container(int fd, struct mp3entry* id3,
             {
                 uint32_t frequency;
 
-                id3->codectype = (type == MP4_mp4a) ? AFMT_MP4_AAC : AFMT_MP4_ALAC;
                 lseek(fd, 22, SEEK_CUR);
                 read_uint32be(fd, &frequency);
                 size -= 26;
@@ -700,11 +698,13 @@ static bool read_mp4_container(int fd, struct mp3entry* id3,
                     read_mp4_atom(fd, &subsize, &subtype, size);
                     size -= 10;
                     
+                    id3->codectype = AFMT_MP4_AAC;
                     if (subtype == MP4_esds)
                     {
                         sbr_used = read_mp4_esds(fd, id3, &size);
                         if (sbr_used)
                         {
+                            id3->codectype = AFMT_MP4_AAC_HE;
                             if (SBR_upsampling_used)
                                 DEBUGF("MP4: AAC-HE, SBR upsampling\n");
                             else
@@ -712,6 +712,11 @@ static bool read_mp4_container(int fd, struct mp3entry* id3,
                         }
                     }
                 }
+                else
+                {
+                    id3->codectype = AFMT_MP4_ALAC;
+                }
+
             }
             break;
 

apps/metadata/mpc.c

@@ -125,6 +125,8 @@ bool get_musepack_metadata(int fd, struct mp3entry *id3)
            
             bufused = set_replaygain_sv7(id3, false, header[3], bufused);
             bufused = set_replaygain_sv7(id3, true , header[4], bufused);
+            
+            id3->codectype = AFMT_MPC_SV7;
         } else {
             return false; /* only SV7 is allowed within a "MP+" signature */
         }
@@ -191,6 +193,8 @@ bool get_musepack_metadata(int fd, struct mp3entry *id3)
                     bufused += set_replaygain_sv8(id3, true , gain, peak, bufused);
                 }
             }
+            
+            id3->codectype = AFMT_MPC_SV8;
         } else {
             /* No sv8 stream header found */
             return false;