view libao2/ao_macosx.c @ 11619:179138947307

This patch contains bugfixes for the esd audio output driver that I uncovered while trying to send sound to a remote esd server over a wireless (11 mbs, just enough to handle to sound) link. First, the sound was full "ticking" sounds. I found a bug that prevented the "send the remainder of this block" code from ever being called - so large chunks of audio were simply being ignored. Fixing this bug removed the "ticking" from audio streams. Fixing this bug, however, uncovered another problem - when the socket buffer was full, doing a blocking write to finish the buffer would take far too long and would turn video into a chunky mess. I'd imagine this blocking write would be fine for an audio-only stream, but it turns out to hold up the video far too much. The solution in this patch is to write as much data as possible to the socket, and then return as soon as possible, reporting the number of bytes actually written accurately back to mplayer. I've tested it on both local and remote esd servers, and it works well. Patch by Benjamin Osheroff <[e-mail]>

/*
 *
 *  ao_macosx.c
 *
 *      Original Copyright (C) Timothy J. Wood - Aug 2000
 *
 *  This file is part of libao, a cross-platform library.  See
 *  README for a history of this source code.
 *
 *  libao 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, or (at your option)
 *  any later version.
 *
 *  libao is distributed in the hope that it will be useful,
 *  but WITHOUT ANY WARRANTY; without even the implied warranty of
 *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 *  GNU General Public License for more details.
 *
 *  You should have received a copy of the GNU General Public License
 *  along with GNU Make; see the file COPYING.  If not, write to
 *  the Free Software Foundation, 675 Mass Ave, Cambridge, MA 02139, USA.
 */

/*
 * The MacOS X CoreAudio framework doesn't mesh as simply as some
 * simpler frameworks do.  This is due to the fact that CoreAudio pulls
 * audio samples rather than having them pushed at it (which is nice
 * when you are wanting to do good buffering of audio). 
 */

/* Change log:
 * 
 * 14/5-2003: Ported to MPlayer libao2 by Dan Christiansen
 *
 *            AC-3 and MPEG audio passthrough is possible, but I don't have
 *            access to a sound card that supports it.
 */

#include <CoreAudio/AudioHardware.h>
#include <stdio.h>
#include <string.h>
#include <inttypes.h>
#include <pthread.h>

#include "../mp_msg.h"

#include "audio_out.h"
#include "audio_out_internal.h"
#include "afmt.h"

static ao_info_t info =
  {
    "Darwin/Mac OS X native audio output",
    "macosx",
    "Timothy J. Wood & Dan Christiansen",
    ""
  };

LIBAO_EXTERN(macosx)

/* Prefix for all mp_msg() calls */
#define ao_msg(a, b, c...) mp_msg(a, b, "AO: [macosx] " c)

/* This is large, but best (maybe it should be even larger).
 * CoreAudio supposedly has an internal latency in the order of 2ms */
#define NUM_BUFS 128

typedef struct ao_macosx_s
{
  /* CoreAudio */
  AudioDeviceID outputDeviceID;
  AudioStreamBasicDescription outputStreamBasicDescription;

  /* Ring-buffer */
  pthread_mutex_t buffer_mutex; /* mutex covering buffer variables */

  unsigned char *buffer[NUM_BUFS];
  unsigned int buffer_len;
  
  unsigned int buf_read;
  unsigned int buf_write;
  unsigned int buf_read_pos;
  unsigned int buf_write_pos;
  int full_buffers;
  int buffered_bytes;
} ao_macosx_t;

static ao_macosx_t *ao;

/* General purpose Ring-buffering routines */
static int write_buffer(unsigned char* data,int len){
  int len2=0;
  int x;

  while(len>0){
    if(ao->full_buffers==NUM_BUFS) {
      ao_msg(MSGT_AO,MSGL_V, "Buffer overrun\n");
      break;
    }

    x=ao->buffer_len-ao->buf_write_pos;
    if(x>len) x=len;
    memcpy(ao->buffer[ao->buf_write]+ao->buf_write_pos,data+len2,x);

    /* accessing common variables, locking mutex */
    pthread_mutex_lock(&ao->buffer_mutex);
    len2+=x; len-=x;
    ao->buffered_bytes+=x; ao->buf_write_pos+=x;
    if(ao->buf_write_pos>=ao->buffer_len) {
      /* block is full, find next! */
      ao->buf_write=(ao->buf_write+1)%NUM_BUFS;
      ++ao->full_buffers;
      ao->buf_write_pos=0;
    }
    pthread_mutex_unlock(&ao->buffer_mutex);
  }

  return len2;
}

static int read_buffer(unsigned char* data,int len){
  int len2=0;
  int x;

  while(len>0){
    if(ao->full_buffers==0) {
      ao_msg(MSGT_AO,MSGL_V, "Buffer underrun\n");
      break;
    }

    x=ao->buffer_len-ao->buf_read_pos;
    if(x>len) x=len;
    memcpy(data+len2,ao->buffer[ao->buf_read]+ao->buf_read_pos,x);
    len2+=x; len-=x;

    /* accessing common variables, locking mutex */
    pthread_mutex_lock(&ao->buffer_mutex);
    ao->buffered_bytes-=x; ao->buf_read_pos+=x;
    if(ao->buf_read_pos>=ao->buffer_len){
      /* block is empty, find next! */
       ao->buf_read=(ao->buf_read+1)%NUM_BUFS;
       --ao->full_buffers;
       ao->buf_read_pos=0;
    }
    pthread_mutex_unlock(&ao->buffer_mutex);
  }
  
  
  return len2;
}

/* end ring buffer stuff */

/* The function that the CoreAudio thread calls when it wants more data */
static OSStatus audioDeviceIOProc(AudioDeviceID inDevice, const AudioTimeStamp *inNow, const AudioBufferList *inInputData, const AudioTimeStamp *inInputTime, AudioBufferList *outOutputData, const AudioTimeStamp *inOutputTime, void *inClientData)
{
  outOutputData->mBuffers[0].mDataByteSize =
    read_buffer((char *)outOutputData->mBuffers[0].mData, ao->buffer_len);

  return 0;
}


static int control(int cmd,void *arg){
	switch (cmd) {
	case AOCONTROL_SET_DEVICE:
	case AOCONTROL_GET_DEVICE:
	case AOCONTROL_GET_VOLUME:
	case AOCONTROL_SET_VOLUME:
	  /* unimplemented/meaningless */
	  return CONTROL_FALSE;
	case AOCONTROL_QUERY_FORMAT:
	  /* stick with what CoreAudio requests */
	  return CONTROL_FALSE;
	default:
	  return CONTROL_FALSE;
	}
	
}


static int init(int rate,int channels,int format,int flags)
{
  OSStatus status;
  UInt32 propertySize;
  int rc;
  int i;

  ao = (ao_macosx_t *)malloc(sizeof(ao_macosx_t));

  /* initialise mutex */
  pthread_mutex_init(&ao->buffer_mutex, NULL);
  pthread_mutex_unlock(&ao->buffer_mutex);

  /* get default output device */ 
  propertySize = sizeof(ao->outputDeviceID);
  status = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice, &propertySize, &(ao->outputDeviceID));
  if (status) {
        ao_msg(MSGT_AO,MSGL_WARN, 
		"AudioHardwareGetProperty returned %d\n",
		(int)status);
	return CONTROL_FALSE;
    }
    
    if (ao->outputDeviceID == kAudioDeviceUnknown) {
        ao_msg(MSGT_AO,MSGL_WARN, "AudioHardwareGetProperty: ao->outputDeviceID is kAudioDeviceUnknown\n");
	return CONTROL_FALSE;
    }
    
    /* get default output format
     * TODO: get all support formats and iterate through them
     */
    propertySize = sizeof(ao->outputStreamBasicDescription);
    status = AudioDeviceGetProperty(ao->outputDeviceID, 0, false, kAudioDevicePropertyStreamFormat, &propertySize, &ao->outputStreamBasicDescription);
    if (status) {
        ao_msg(MSGT_AO,MSGL_WARN, "AudioDeviceGetProperty returned %d when getting kAudioDevicePropertyStreamFormat\n", (int)status);
	return CONTROL_FALSE;
    }

    ao_msg(MSGT_AO,MSGL_V, "hardware format...\n");
    ao_msg(MSGT_AO,MSGL_V, "%f mSampleRate\n", ao->outputStreamBasicDescription.mSampleRate);
    ao_msg(MSGT_AO,MSGL_V, " %c%c%c%c mFormatID\n", 
	    (int)(ao->outputStreamBasicDescription.mFormatID & 0xff000000) >> 24,
	    (int)(ao->outputStreamBasicDescription.mFormatID & 0x00ff0000) >> 16,
	    (int)(ao->outputStreamBasicDescription.mFormatID & 0x0000ff00) >>  8,
	    (int)(ao->outputStreamBasicDescription.mFormatID & 0x000000ff) >>  0);
    ao_msg(MSGT_AO,MSGL_V, "%5d mBytesPerPacket\n",
	    (int)ao->outputStreamBasicDescription.mBytesPerPacket);
    ao_msg(MSGT_AO,MSGL_V, "%5d mFramesPerPacket\n",
	    (int)ao->outputStreamBasicDescription.mFramesPerPacket);
    ao_msg(MSGT_AO,MSGL_V, "%5d mBytesPerFrame\n",
	    (int)ao->outputStreamBasicDescription.mBytesPerFrame);
    ao_msg(MSGT_AO,MSGL_V, "%5d mChannelsPerFrame\n",
	    (int)ao->outputStreamBasicDescription.mChannelsPerFrame);

    /* get requested buffer length */
    propertySize = sizeof(ao->buffer_len);
    status = AudioDeviceGetProperty(ao->outputDeviceID, 0, false, kAudioDevicePropertyBufferSize, &propertySize, &ao->buffer_len);
    if (status) {
        ao_msg(MSGT_AO,MSGL_WARN, "AudioDeviceGetProperty returned %d when getting kAudioDevicePropertyBufferSize\n", (int)status);
	return CONTROL_FALSE;
    }
    ao_msg(MSGT_AO,MSGL_V, "%5d ao->buffer_len\n", (int)ao->buffer_len);

    /* FIXME:
     *
     * Resampling of 32-bit float audio is broken in MPlayer. Refuse to 
     * handle anything other than the native format until this is fixed
     * or this module is rewritten, whichever comes first.
     */
    if (ao_data.samplerate == ao->outputStreamBasicDescription.mSampleRate) {
      ao_data.samplerate = (int)ao->outputStreamBasicDescription.mSampleRate;
    } else {
      ao_msg(MSGT_AO,MSGL_WARN, "Resampling not supported yet.\n");
      return 0;
    }

    ao_data.channels = ao->outputStreamBasicDescription.mChannelsPerFrame;
    ao_data.outburst = ao_data.buffersize = ao->buffer_len;
    ao_data.bps = 
      ao_data.samplerate * ao->outputStreamBasicDescription.mBytesPerFrame;

    if (ao->outputStreamBasicDescription.mFormatID == kAudioFormatLinearPCM) {
      uint32_t flags = ao->outputStreamBasicDescription.mFormatFlags;
      if (flags & kAudioFormatFlagIsFloat) {
	ao_data.format = AFMT_FLOAT;
      } else {
	ao_msg(MSGT_AO,MSGL_WARN, "Unsupported audio output "
	       "format %d. Please report this to the developer\n",
	       (int)status);
	return CONTROL_FALSE;
      }
      
    } else {
      /* TODO: handle AFMT_AC3, AFMT_MPEG & friends */
      ao_msg(MSGT_AO,MSGL_WARN, "Default Audio Device doesn't "
	     "support Linear PCM!\n");
      return CONTROL_FALSE;
    }
  
    /* Allocate ring-buffer memory */
    for(i=0;i<NUM_BUFS;i++) 
      ao->buffer[i]=(unsigned char *) malloc(ao->buffer_len);


    /* Prepare for playback */

    reset();
    
    /* Set the IO proc that CoreAudio will call when it needs data */
    status = AudioDeviceAddIOProc(ao->outputDeviceID, audioDeviceIOProc, NULL);
    if (status) {
        ao_msg(MSGT_AO,MSGL_WARN, "AudioDeviceAddIOProc returned %d\n", (int)status);
	return CONTROL_FALSE;
    }
 
    /* Start callback */
    status = AudioDeviceStart(ao->outputDeviceID, audioDeviceIOProc);
    if (status) {
      ao_msg(MSGT_AO,MSGL_WARN, "AudioDeviceStart returned %d\n",
	     (int)status);
      return CONTROL_FALSE;
    }
    
    return CONTROL_OK;
}


static int play(void* output_samples,int num_bytes,int flags)
{  
  return write_buffer(output_samples, num_bytes);
}

/* set variables and buffer to initial state */
static void reset()
{
  int i;
  
  pthread_mutex_lock(&ao->buffer_mutex);
  
  /* reset ring-buffer state */
  ao->buf_read=0;
  ao->buf_write=0;
  ao->buf_read_pos=0;
  ao->buf_write_pos=0;
  
  ao->full_buffers=0;
  ao->buffered_bytes=0;
  
  /* zero output buffer */
  for (i = 0; i < NUM_BUFS; i++)
    bzero(ao->buffer[i], ao->buffer_len);

  pthread_mutex_unlock(&ao->buffer_mutex);
       
  return;
}


/* return available space */
static int get_space()
{
  return (NUM_BUFS-ao->full_buffers)*ao_data.buffersize - ao->buf_write_pos;
}


/* return delay until audio is played */
static float get_delay()
{
  return (float)(ao->buffered_bytes)/(float)ao_data.bps;
}


/* unload plugin and deregister from coreaudio */
static void uninit()
{
  int i;
  OSErr status;

  reset();

  status = AudioDeviceRemoveIOProc(ao->outputDeviceID, audioDeviceIOProc);
  if (status)
    ao_msg(MSGT_AO,MSGL_WARN, "AudioDeviceRemoveIOProc "
	   "returned %d\n", (int)status);

  for(i=0;i<NUM_BUFS;i++) free(ao->buffer[i]);
  free(ao);
}


/* stop playing, keep buffers (for pause) */
static void audio_pause()
{
  OSErr status;

  /* stop callback */
  status = AudioDeviceStop(ao->outputDeviceID, audioDeviceIOProc);
  if (status)
    ao_msg(MSGT_AO,MSGL_WARN, "AudioDeviceStop returned %d\n",
	   (int)status);
}


/* resume playing, after audio_pause() */
static void audio_resume()
{
  OSErr status = AudioDeviceStart(ao->outputDeviceID, audioDeviceIOProc);
  if (status)
    ao_msg(MSGT_AO,MSGL_WARN, "AudioDeviceStart returned %d\n",
	   (int)status);
}