嘗試在Mac上使用Eclipse編譯OpenAL結果未找到架構x86_64的符號錯誤

Question

我對C++相當陌生，但我一直想用音頻製作程序，並且找到了OpenAL。作爲參考，我使用Eclipse和運行OS X 10.10的Mac。我發現了一個簡單的教程程序：

#include <stdio.h> 

#include <stdio.h> 
#include <stdlib.h> 
#include <GLUT/glut.h> 
#include <OpenAL/al.h> 
#include <OpenAL/alc.h> 

#include <stdio.h> 
#include <sys/types.h> 
#include <sys/stat.h> 
#include <unistd.h> 
#include <stdlib.h> 

int main(int argc, char **argv) 
{ 
ALCdevice *dev; 
ALCcontext *ctx; 
struct stat statbuf; 

if(argc < 2) 
{ 
    fprintf(stderr, "Usage: %s <audiofile>\n", argv[0]); 
    return 0; 
} 

/* First the standard open-device, create-context, set-context.. */ 
dev = alcOpenDevice(NULL); 
if(!dev) 
{ 
    fprintf(stderr, "Oops\n"); 
    return 1; 
} 
ctx = alcCreateContext(dev, NULL); 
alcMakeContextCurrent(ctx); 
if(!ctx) 
{ 
    fprintf(stderr, "Oops2\n"); 
    return 1; 
} 

{ 
    /* The number of buffers and bytes-per-buffer for our stream are set 
    * here. The number of buffers should be two or more, and the buffer 
    * size should be a multiple of the frame size (by default, OpenAL's 
    * largest frame size is 4, however extensions that can add more formats 
    * may be larger). Slower systems may need more buffers/larger buffer 
    * sizes. */ 
#define NUM_BUFFERS 3 
#define BUFFER_SIZE 4096 
    /* These are what we'll use for OpenAL playback */ 
    ALuint source, buffers[NUM_BUFFERS]; 
    ALuint frequency; 
    ALenum format; 
    unsigned char *buf; 
    /* These are used for interacting with mplayer */ 
    int pid, files[2]; 
    FILE *f; 

    /* Generate the buffers and sources */ 
    alGenBuffers(NUM_BUFFERS, buffers); 
    alGenSources(1, &source); 
    if(alGetError() != AL_NO_ERROR) 
    { 
     fprintf(stderr, "Error generating :(\n"); 
     return 1; 
    } 

    /* Here's where our magic begins. First, we want to call stat on the 
    * filename since mplayer will just silently exit if it tries to play a 
    * non-existant file **/ 
    if(stat(argv[1], &statbuf) != 0 || !S_ISREG(statbuf.st_mode)) 
    { 
     fprintf(stderr, "%s doesn't seem to be a regular file :(\n", argv[1]); 
     return 1; 
    } 
    /* Open a file pipe. This will create two file-descriptors, one for 
    * reading and another for writing. The data will be passed in memory, 
    * so it won't be bogged by disk access. */ 
    if(pipe(files) != 0) 
    { 
     fprintf(stderr, "Pipe failed :(\n"); 
     return 1; 
    } 

    /* Now we fork. The forked process will inherit the original process's 
    * file descriptors, so each process will have access to the same pipe. 
    * Note that the process memory isn't shared (if you change something in 
    * one process, the other will be unaffected). */ 
    pid = fork(); 
    switch(pid) 
    { 
     case -1: 
      /* If it returns -1, there was an error */ 
      fprintf(stderr, "Fork failed :(\n"); 
      return 1; 
      break; 

     case 0: 
      /* Returning 0 means that we're now in the child process, that 
      * we'll turn into mplayer. First, we can close the read file 
      * descriptor since this process won't be reading from it. */ 
      close(files[0]); 

      /* Here's part of the trick. After closing the stdout file 
      * descriptor, dup2 assigns it the pipe's write file descriptor. 
      * So now, whenever anything writes to stdout, it'll go to the 
      * pipe instead! */ 
      close(STDOUT_FILENO); 
      dup2(files[1], STDOUT_FILENO); 

      /* We can use execlp to run mplayer with the options we need. To 
      * output audio as a standard .wav-formatted file, we use the 
      * pcm audio-out device, and tell it to write to stdout. By 
      * running this, we overwrite the current process memory with 
      * the named commmand, which causes it to start mplayer with the 
      * overridden stdout */ 
      execlp("mplayer", "-nogui", "-really-quiet", "-novideo", 
        "-noconsolecontrols", "-ao", "pcm:file=/dev/stdout", 
        argv[1], (char*)NULL); 
      /* The exec* functions should never return. If it does, 
      * something went wrong, so just _exit. */ 
      _exit(1); 
     default: 
      /* Any other return value means we're in the parent process. 
      * Here, we don't need the write file descriptor, so close it. 
      * Now we can begin using the read file descriptor to read 
      * mplayer's stdout, which will be the file decoded in real- 
      * time! */ 
      close(files[1]); 
      break; 
    } 

    /* fdopen simply creates a FILE* from the given file descriptor. This is 
    * generally easier to work with, but there's no reason you couldn't use 
    * the lower-level io routines on the descriptor if you wanted */ 
    f = fdopen(files[0], "rb"); 

    /* Allocate the buffer, and read the RIFF-WAVE header. We don't actually 
    * need to read it, so just ignore what it writes to the buffer. Because 
    * this is a file pipe, it is unseekable, so we have to read bytes we 
    * want to skip. Also note that because mplayer is writing out the file 
    * in real-time, the chunk size information may not be filled out. */ 
    malloc(BUFFER_SIZE); 
    fread(buf, 1, 12, f); 

    /* This is the first .wav file chunk. Check the chunk header to make 
    * sure it is the format information. The first four bytes is the 
    * indentifier (which we check), and the last four is the chunk size 
    * (which we ignore) */ 
    fread(buf, 1, 8, f); 
    if(buf[0] != 'f' || buf[1] != 'm' || buf[2] != 't' || buf[3] != ' ') 
    { 
     /* If this isn't the format info, it probably means it was an 
     * unsupported audio format for mplayer, or the file didn't contain 
     * an audio track. */ 
     fprintf(stderr, "Not 'fmt ' :(\n"); 
     /* Note that closing the file will leave mplayer's write file 
     * descriptor without a read counterpart. This will cause mplayer to 
     * receive a SIGPIPE signal, which will cause it to abort and exit 
     * automatically for us. Alternatively, you can use the pid returned 
     * from fork() to send it a signal explicitly. */ 
     fclose(f); 
     return 1; 
    } 

    { 
     int channels, bits; 

     /* Read the wave format type, as a 16-bit little-endian integer. 
     * There's no reason this shouldn't be 1. */ 
     fread(buf, 1, 2, f); 
     if(buf[1] != 0 || buf[0] != 1) 
     { 
      fprintf(stderr, "Not PCM :(\n"); 
      fclose(f); 
      return 1; 
     } 

     /* Get the channel count (16-bit little-endian) */ 
     fread(buf, 1, 2, f); 
     channels = buf[1]<<8; 
     channels |= buf[0]; 

     /* Get the sample frequency (32-bit little-endian) */ 
     fread(buf, 1, 4, f); 
     frequency = buf[3]<<24; 
     frequency |= buf[2]<<16; 
     frequency |= buf[1]<<8; 
     frequency |= buf[0]; 

     /* The next 6 bytes hold the block size and bytes-per-second. We 
     * don't need that info, so just read and ignore it. */ 
     fread(buf, 1, 6, f); 

     /* Get the bit depth (16-bit little-endian) */ 
     fread(buf, 1, 2, f); 
     bits = buf[1]<<8; 
     bits |= buf[0]; 

     /* Now convert the given channel count and bit depth into an OpenAL 
     * format. We could use extensions to support more formats (eg. 
     * surround sound, floating-point samples), but that is beyond the 
     * scope of this tutorial */ 
     format = 0; 
     if(bits == 8) 
     { 
      if(channels == 1) 
       format = AL_FORMAT_MONO8; 
      else if(channels == 2) 
       format = AL_FORMAT_STEREO8; 
     } 
     else if(bits == 16) 
     { 
      if(channels == 1) 
       format = AL_FORMAT_MONO16; 
      else if(channels == 2) 
       format = AL_FORMAT_STEREO16; 
     } 
     if(!format) 
     { 
      fprintf(stderr, "Incompatible format (%d, %d) :(\n", channels, bits); 
      fclose(f); 
      return 1; 
     } 
    } 

    /* Next up is the data chunk, which will hold the decoded sample data */ 
    fread(buf, 1, 8, f); 
    if(buf[0] != 'd' || buf[1] != 'a' || buf[2] != 't' || buf[3] != 'a') 
    { 
     fclose(f); 
     fprintf(stderr, "Not 'data' :(\n"); 
     return 1; 
    } 

    /* Now we have everything we need. To read the decoded data, all we have 
    * to do is read from the file handle! Note that the .wav format spec 
    * has multibyte sample foramts stored as little-endian. If you were on 
    * a big-endian machine, you'd have to iterate over the returned data 
    * and flip the bytes for those formats before giving it to OpenAL. Also 
    * be aware that there is no seeking on the file handle. A slightly more 
    * complex setup could be made to send commands back to mplayer to seek 
    * on the stream, however that is beyond the scope of this tutorial. */ 
    { 
     int ret; 

     /* Fill the data buffer with the amount of bytes-per-buffer, and 
     * buffer it into OpenAL. This may read (and return) less than the 
     * requested amount when it hits the end of the "stream" */ 
     ret = fread(buf, 1, BUFFER_SIZE, f); 
     alBufferData(buffers[0], format, buf, ret, frequency); 

     /* Once the data's buffered into OpenAL, we're free to modify our 
     * data buffer, so reuse it to fill the remaining OpenAL buffers. */ 
     ret = fread(buf, 1, BUFFER_SIZE, f); 
     alBufferData(buffers[1], format, buf, ret, frequency); 
     ret = fread(buf, 1, BUFFER_SIZE, f); 
     alBufferData(buffers[2], format, buf, ret, frequency); 
     if(alGetError() != AL_NO_ERROR) 
     { 
      fprintf(stderr, "Error loading :(\n"); 
      return 1; 
     } 

     /* Queue the buffers onto the source, and start playback! */ 
     alSourceQueueBuffers(source, NUM_BUFFERS, buffers); 
     alSourcePlay(source); 
     if(alGetError() != AL_NO_ERROR) 
     { 
      fprintf(stderr, "Error starting :(\n"); 
      return 1; 
     } 

     /* While not at the end of the stream... */ 
     while(!feof(f)) 
     { 
      ALuint buffer; 
      ALint val; 

      /* Check if OpenAL is done with any of the queued buffers */ 
      alGetSourcei(source, AL_BUFFERS_PROCESSED, &val); 
      if(val <= 0) 
       continue; 

      /* For each processed buffer... */ 
      while(val--) 
      { 
       /* Read the next chunk of decoded data from the stream */ 
       ret = fread(buf, 1, BUFFER_SIZE, f); 

       /* Pop the oldest queued buffer from the source, fill it 
       * with the new data, then requeue it */ 
       alSourceUnqueueBuffers(source, 1, &buffer); 
       alBufferData(buffer, format, buf, ret, frequency); 
       alSourceQueueBuffers(source, 1, &buffer); 
       if(alGetError() != AL_NO_ERROR) 
       { 
        fprintf(stderr, "Error buffering :(\n"); 
        return 1; 
       } 
      } 
      /* Make sure the source is still playing, and restart it if 
      * needed. */ 
      alGetSourcei(source, AL_SOURCE_STATE, &val); 
      if(val != AL_PLAYING) 
       alSourcePlay(source); 
     } 
    } 

    /* File's done decoding. We can close the pipe and free the data buffer 
    * now. */ 
    fclose(f); 
    free(buf); 
    { 
     ALint val; 
     /* Although mplayer is done giving us data, OpenAL may still be 
     * playing the remaining buffers. Wait until it stops. */ 
     do { 
      alGetSourcei(source, AL_SOURCE_STATE, &val); 
     } while(val == AL_PLAYING); 
    } 

    /* Done playing. Delete the source and buffers */ 
    alDeleteSources(1, &source); 
    alDeleteBuffers(NUM_BUFFERS, buffers); 
} 

/* All done. Close OpenAL and exit. */ 
alcMakeContextCurrent(NULL); 
alcDestroyContext(ctx); 
alcCloseDevice(dev); 

return 0; 
} 

然而，當我嘗試編譯這個節目，我得到一個錯誤。這裏是編譯：

19:25:37 **** Incremental Build of configuration Release for project AudioPlayer **** 
make all 
Building file: ../src/main.cpp 
Invoking: GCC C++ Compiler 
g++ -I/usr/local/include/SDL2 -O3 -Wall -c -fmessage-length=0 -MMD -MP -MF"src/main.d" -MT"src/main.o" -o "src/main.o" "../src/main.cpp" 
../src/main.cpp:146:15: warning: variable 'buf' is uninitialized when used here [-Wuninitialized] 
     fread(buf, 1, 12, f); 
       ^~~ 
../src/main.cpp:56:27: note: initialize the variable 'buf' to silence this warning 
     unsigned char *buf; 
         ^
          = NULL 
1 warning generated. 
Finished building: ../src/main.cpp 

Building target: AudioPlayer 
Invoking: MacOS X C++ Linker 
g++ -L/usr/local/lib -o "AudioPlayer" ./src/Circle.o ./src/Screen.o ./src/main.o -lSDL2 
Undefined symbols for architecture x86_64: 
    "_alGenBuffers", referenced from: 
     _main in main.o 
    "_alGenSources", referenced from: 
     _main in main.o 
    "_alGetError", referenced from: 
     _main in main.o 
    "_alcCreateContext", referenced from: 
     _main in main.o 
    "_alcMakeContextCurrent", referenced from: 
     _main in main.o 
    "_alcOpenDevice", referenced from: 
     _main in main.o 
ld: symbol(s) not found for architecture x86_64 
clang: error: linker command failed with exit code 1 (use -v to see invocation) 
make: *** [AudioPlayer] Error 1 

19:25:39 Build Finished (took 2s.492ms) 

我很難搞清楚這是什麼意思。我成功編譯並鏈接了SDL2。我不認爲警告會導致這種情況，所以什麼可能導致這個ld: symbol(s) not found for architecture x86_64錯誤？

[編輯]我不認爲我必須在Eclipse設置中鏈接任何東西。 This目前什麼是我/usr/local/lib

Answer 1

我不知道太多的OpenAL和編譯在Mac上的，但如果是說symbol(s) not found，我想你需要添加一個包括對x86_64的庫到G ++的搜索路徑。如果您可以告訴我們/ usr/local/lib中的內容，那將非常有用。

編輯：剛剛看到您的編輯， 我沒有看到任何與OpenAL相關的內容， 確定它已安裝？也許它在另一個 文件夾中，但您必須包含它。庫文件通常在.a或.so中結束 ...我發現了其他可能有用的內容：StackOverflow Situation Similar。嘗試第三條評論。如果可能，請將這些行放在g ++選項中。您可能需要編輯構建過程。你必須編輯Makefile，這將是最簡單的方法。 （我認爲有一個Makefile，我也是新的）