New FLAC decoder from the ffmpeg project

git-svn-id: svn://svn.rockbox.org/rockbox/trunk@7656 a1c6a512-1295-4272-9138-f99709370657

1 files changed, 292 insertions, 0 deletions

diff --git a/apps/codecs/libffmpegFLAC/main.c b/apps/codecs/libffmpegFLAC/main.c
new file mode 100644
index 0000000..eea97f4
--- /dev/null
+++ b/apps/codecs/libffmpegFLAC/main.c
@@ -0,0 +1,292 @@
+/***************************************************************************
+ *             __________               __   ___.
+ *   Open      \______   \ ____   ____ |  | _\_ |__   _______  ___
+ *   Source     |       _//  _ \_/ ___\|  |/ /| __ \ /  _ \  \/  /
+ *   Jukebox    |    |   (  <_> )  \___|    < | \_\ (  <_> > <  <
+ *   Firmware   |____|_  /\____/ \___  >__|_ \|___  /\____/__/\_ \
+ *                     \/            \/     \/    \/            \/
+ * $Id$
+ *
+ * Copyright (C) 2005 Dave Chapman
+ *
+ * All files in this archive are subject to the GNU General Public License.
+ * See the file COPYING in the source tree root for full license agreement.
+ *
+ * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
+ * KIND, either express or implied.
+ *
+ ****************************************************************************/
+
+/* A test program for the Rockbox version of the ffmpeg FLAC decoder.
+
+   Compile using Makefile.test - run it as "./test file.flac" to decode the
+   FLAC file to the file "test.wav" in the current directory
+
+   This test program should support 16-bit and 24-bit mono and stereo files.
+
+   The resulting "test.wav" should have the same md5sum as a WAV file created
+   by the official FLAC decoder (it produces the same 44-byte canonical WAV 
+   header.
+*/
+
+#include <stdio.h>
+#include <string.h>
+#include <inttypes.h>
+#include <stdbool.h>
+#include <fcntl.h>
+#include <unistd.h>
+#include <sys/types.h>
+#include <sys/stat.h>
+
+#include "decoder.h"
+
+static unsigned char wav_header[44]={
+    'R','I','F','F',//  0 - ChunkID
+    0,0,0,0,        //  4 - ChunkSize (filesize-8)
+    'W','A','V','E',//  8 - Format
+    'f','m','t',' ',// 12 - SubChunkID
+    16,0,0,0,       // 16 - SubChunk1ID  // 16 for PCM
+    1,0,            // 20 - AudioFormat (1=Uncompressed)
+    2,0,            // 22 - NumChannels
+    0,0,0,0,        // 24 - SampleRate in Hz
+    0,0,0,0,        // 28 - Byte Rate (SampleRate*NumChannels*(BitsPerSample/8)
+    4,0,            // 32 - BlockAlign (== NumChannels * BitsPerSample/8)
+    16,0,           // 34 - BitsPerSample
+    'd','a','t','a',// 36 - Subchunk2ID
+    0,0,0,0         // 40 - Subchunk2Size
+};
+
+int open_wav(char* filename) {
+    int fd;
+
+    fd=open(filename,O_CREAT|O_WRONLY|O_TRUNC,S_IRUSR|S_IWUSR);
+    if (fd >= 0) {
+        write(fd,wav_header,sizeof(wav_header));
+    }
+    return(fd);
+}
+
+void close_wav(int fd, FLACContext* fc) {
+    int x;
+    int filesize;
+    int bytespersample;
+
+    bytespersample=fc->bps/8;
+
+    filesize=fc->totalsamples*bytespersample*fc->channels+44;
+
+    // ChunkSize
+    x=filesize-8;
+    wav_header[4]=(x&0xff);
+    wav_header[5]=(x&0xff00)>>8;
+    wav_header[6]=(x&0xff0000)>>16;
+    wav_header[7]=(x&0xff000000)>>24;
+
+    // Number of channels
+    wav_header[22]=fc->channels;
+
+    // Samplerate
+    wav_header[24]=fc->samplerate&0xff;
+    wav_header[25]=(fc->samplerate&0xff00)>>8;
+    wav_header[26]=(fc->samplerate&0xff0000)>>16;
+    wav_header[27]=(fc->samplerate&0xff000000)>>24;
+
+    // ByteRate
+    x=fc->samplerate*(fc->bps/8)*fc->channels;
+    wav_header[28]=(x&0xff);
+    wav_header[29]=(x&0xff00)>>8;
+    wav_header[30]=(x&0xff0000)>>16;
+    wav_header[31]=(x&0xff000000)>>24;
+
+    // BlockAlign
+    wav_header[32]=(fc->bps/8)*fc->channels;
+
+    // Bits per sample
+    wav_header[34]=fc->bps;
+    
+    // Subchunk2Size
+    x=filesize-44;
+    wav_header[40]=(x&0xff);
+    wav_header[41]=(x&0xff00)>>8;
+    wav_header[42]=(x&0xff0000)>>16;
+    wav_header[43]=(x&0xff000000)>>24;
+
+    lseek(fd,0,SEEK_SET);
+    write(fd,wav_header,sizeof(wav_header));
+    close(fd);
+}
+
+static void dump_headers(FLACContext *s)
+{
+    fprintf(stderr,"  Blocksize: %d .. %d\n", s->min_blocksize, 
+                   s->max_blocksize);
+    fprintf(stderr,"  Framesize: %d .. %d\n", s->min_framesize, 
+                   s->max_framesize);
+    fprintf(stderr,"  Samplerate: %d\n", s->samplerate);
+    fprintf(stderr,"  Channels: %d\n", s->channels);
+    fprintf(stderr,"  Bits per sample: %d\n", s->bps);
+    fprintf(stderr,"  Metadata length: %d\n", s->metadatalength);
+    fprintf(stderr,"  Total Samples: %lu\n",s->totalsamples);
+    fprintf(stderr,"  Duration: %d ms\n",s->length);
+    fprintf(stderr,"  Bitrate: %d kbps\n",s->bitrate);
+}
+
+static bool flac_init(int fd, FLACContext* fc)
+{
+    unsigned char buf[255];
+    struct stat statbuf;
+    bool found_streaminfo=false;
+    int endofmetadata=0;
+    int blocklength;
+
+    if (lseek(fd, 0, SEEK_SET) < 0) 
+    {
+        return false;
+    }
+
+    if (read(fd, buf, 4) < 4)
+    {
+        return false;
+    }
+
+    if (memcmp(buf,"fLaC",4) != 0) 
+    {
+        return false;
+    }
+    fc->metadatalength = 4;
+
+    while (!endofmetadata) {
+        if (read(fd, buf, 4) < 4)
+        {
+            return false;
+        }
+
+        endofmetadata=(buf[0]&0x80);
+        blocklength = (buf[1] << 16) | (buf[2] << 8) | buf[3];
+        fc->metadatalength+=blocklength+4;
+
+        if ((buf[0] & 0x7f) == 0)       /* 0 is the STREAMINFO block */
+        {
+            /* FIXME: Don't trust the value of blocklength */
+            if (read(fd, buf, blocklength) < 0)
+            {
+                return false;
+            }
+          
+            fstat(fd,&statbuf);
+            fc->filesize = statbuf.st_size;
+            fc->min_blocksize = (buf[0] << 8) | buf[1];
+            fc->max_blocksize = (buf[2] << 8) | buf[3];
+            fc->min_framesize = (buf[4] << 16) | (buf[5] << 8) | buf[6];
+            fc->max_framesize = (buf[7] << 16) | (buf[8] << 8) | buf[9];
+            fc->samplerate = (buf[10] << 12) | (buf[11] << 4) 
+                             | ((buf[12] & 0xf0) >> 4);
+            fc->channels = ((buf[12]&0x0e)>>1) + 1;
+            fc->bps = (((buf[12]&0x01) << 4) | ((buf[13]&0xf0)>>4) ) + 1;
+
+            /* totalsamples is a 36-bit field, but we assume <= 32 bits are 
+               used */
+            fc->totalsamples = (buf[14] << 24) | (buf[15] << 16) 
+                               | (buf[16] << 8) | buf[17];
+
+            /* Calculate track length (in ms) and estimate the bitrate 
+               (in kbit/s) */
+            fc->length = (fc->totalsamples / fc->samplerate) * 1000;
+
+            found_streaminfo=true;
+        } else if ((buf[0] & 0x7f) == 3) { /* 3 is the SEEKTABLE block */
+            fprintf(stderr,"Seektable length = %d bytes\n",blocklength);
+            if (lseek(fd, blocklength, SEEK_CUR) < 0) {
+                return false;
+          }
+        } else {
+            /* Skip to next metadata block */
+            if (lseek(fd, blocklength, SEEK_CUR) < 0)
+            {
+                return false;
+            }
+        }
+    }
+
+   if (found_streaminfo) {
+       fc->bitrate = ((fc->filesize-fc->metadatalength) * 8) / fc->length;
+       return true;
+   } else {
+       return false;
+   }
+}
+
+/* Dummy function needed to pass to flac_decode_frame() */
+void yield() {
+}
+
+int main(int argc, char* argv[]) {
+    FLACContext fc;
+    int fd,fdout;
+    int n;
+    int i;
+    int bytesleft;
+    int consumed;
+    char buf[MAX_FRAMESIZE]; /* The input buffer */
+    /* The output buffers containing the decoded samples (channels 0 and 1) */
+    int32_t decoded0[MAX_BLOCKSIZE];
+    int32_t decoded1[MAX_BLOCKSIZE];
+
+    /* For testing */
+    int8_t wavbuf[MAX_CHANNELS*MAX_BLOCKSIZE*3];
+    int8_t* p;
+    int scale;
+
+    fd=open(argv[1],O_RDONLY);
+
+    if (fd < 0) {
+        fprintf(stderr,"Can not parse %s\n",argv[1]);
+        return(1);
+    }
+
+    /* Read the metadata and position the file pointer at the start of the 
+       first audio frame */
+    flac_init(fd,&fc);
+
+    dump_headers(&fc);
+
+    fdout=open_wav("test.wav");
+    bytesleft=read(fd,buf,sizeof(buf));
+    while (bytesleft) {
+        if(flac_decode_frame(&fc,decoded0,decoded1,buf,bytesleft,yield) < 0) {
+             fprintf(stderr,"DECODE ERROR, ABORTING\n");
+             break;
+        }
+        consumed=fc.gb.index/8;
+
+        scale=FLAC_OUTPUT_DEPTH-fc.bps;
+        p=wavbuf;
+        for (i=0;i<fc.blocksize;i++) {
+             /* Left sample */
+             decoded0[i]=decoded0[i]>>scale;
+             *(p++)=decoded0[i]&0xff;
+             *(p++)=(decoded0[i]&0xff00)>>8;
+             if (fc.bps==24) *(p++)=(decoded0[i]&0xff0000)>>16;
+
+             if (fc.channels==2) {
+                 /* Right sample */
+                 decoded1[i]=decoded1[i]>>scale;
+                 *(p++)=decoded1[i]&0xff;
+                 *(p++)=(decoded1[i]&0xff00)>>8;
+                 if (fc.bps==24) *(p++)=(decoded1[i]&0xff0000)>>16;
+             }
+        }
+        write(fdout,wavbuf,fc.blocksize*fc.channels*(fc.bps/8));
+
+        memmove(buf,&buf[consumed],bytesleft-consumed);
+        bytesleft-=consumed;
+
+        n=read(fd,&buf[bytesleft],sizeof(buf)-bytesleft);
+        if (n > 0) {
+            bytesleft+=n;
+        }
+    }
+    close_wav(fdout,&fc);
+    close(fd);
+    return(0);
+}