mpv/libao2/ao_macosx.c

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/*
*
* 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 <stdlib.h>
#include <inttypes.h>
#include <pthread.h>
#include "config.h"
#include "mp_msg.h"
#include "audio_out.h"
#include "audio_out_internal.h"
#include "libaf/af_format.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 32
typedef struct ao_macosx_s
{
/* CoreAudio */
AudioDeviceID outputDeviceID;
AudioStreamBasicDescription outputStreamBasicDescription;
/* Ring-buffer */
/* does not need explicit synchronization, but needs to allocate
* (num_chunks + 1) * chunk_size memory to store num_chunks * chunk_size
* data */
unsigned char *buffer;
unsigned int buffer_len; ///< must always be (num_chunks + 1) * chunk_size
unsigned int num_chunks;
unsigned int chunk_size;
unsigned int buf_read_pos;
unsigned int buf_write_pos;
} ao_macosx_t;
static ao_macosx_t *ao;
/**
* \brief return number of free bytes in the buffer
* may only be called by mplayer's thread
* \return minimum number of free bytes in buffer, value may change between
* two immediately following calls, and the real number of free bytes
* might actually be larger!
*/
static int buf_free() {
int free = ao->buf_read_pos - ao->buf_write_pos - ao->chunk_size;
if (free < 0) free += ao->buffer_len;
return free;
}
/**
* \brief return number of buffered bytes
* may only be called by playback thread
* \return minimum number of buffered bytes, value may change between
* two immediately following calls, and the real number of buffered bytes
* might actually be larger!
*/
static int buf_used() {
int used = ao->buf_write_pos - ao->buf_read_pos;
if (used < 0) used += ao->buffer_len;
return used;
}
/**
* \brief add data to ringbuffer
*/
static int write_buffer(unsigned char* data, int len){
int first_len = ao->buffer_len - ao->buf_write_pos;
int free = buf_free();
if (len > free) len = free;
if (first_len > len) first_len = len;
// till end of buffer
memcpy (&ao->buffer[ao->buf_write_pos], data, first_len);
if (len > first_len) { // we have to wrap around
// remaining part from beginning of buffer
memcpy (ao->buffer, &data[first_len], len - first_len);
}
ao->buf_write_pos = (ao->buf_write_pos + len) % ao->buffer_len;
return len;
}
/**
* \brief remove data from ringbuffer
*/
static int read_buffer(unsigned char* data,int len){
int first_len = ao->buffer_len - ao->buf_read_pos;
int buffered = buf_used();
if (len > buffered) len = buffered;
if (first_len > len) first_len = len;
// till end of buffer
memcpy (data, &ao->buffer[ao->buf_read_pos], first_len);
if (len > first_len) { // we have to wrap around
// remaining part from beginning of buffer
memcpy (&data[first_len], ao->buffer, len - first_len);
}
ao->buf_read_pos = (ao->buf_read_pos + len) % ao->buffer_len;
return len;
}
/* 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->chunk_size);
return 0;
}
static int control(int cmd,void *arg){
OSStatus status;
UInt32 propertySize;
ao_control_vol_t* vol = (ao_control_vol_t*)arg;
UInt32 stereoChannels[2];
static float volume=0.5;
switch (cmd) {
case AOCONTROL_SET_DEVICE:
case AOCONTROL_GET_DEVICE:
/* unimplemented/meaningless */
return CONTROL_FALSE;
case AOCONTROL_GET_VOLUME:
propertySize=sizeof(stereoChannels);
status = AudioDeviceGetProperty(ao->outputDeviceID, NULL, 0,
kAudioDevicePropertyPreferredChannelsForStereo, &propertySize,
&stereoChannels);
// printf("OSX: stereochannels %d ; %d \n",stereoChannels[0],stereoChannels[1]);
propertySize=sizeof(volume);
status = AudioDeviceGetProperty(ao->outputDeviceID, stereoChannels[0], false, kAudioDevicePropertyVolumeScalar, &propertySize, &volume);
// printf("OSX: get volume=%5.3f status=%d \n",volume,status);
vol->left=(int)(volume*100.0);
status = AudioDeviceGetProperty(ao->outputDeviceID, stereoChannels[1], false, kAudioDevicePropertyVolumeScalar, &propertySize, &volume);
vol->right=(int)(volume*100.0);
return CONTROL_TRUE;
case AOCONTROL_SET_VOLUME:
propertySize=sizeof(stereoChannels);
status = AudioDeviceGetProperty(ao->outputDeviceID, NULL, 0,
kAudioDevicePropertyPreferredChannelsForStereo, &propertySize,
&stereoChannels);
// printf("OSX: stereochannels %d ; %d \n",stereoChannels[0],stereoChannels[1]);
propertySize=sizeof(volume);
volume=vol->left/100.0;
status = AudioDeviceSetProperty(ao->outputDeviceID, 0, stereoChannels[0], 0, kAudioDevicePropertyVolumeScalar, propertySize, &volume);
// printf("OSX: set volume=%5.3f status=%d\n",volume,status);
volume=vol->right/100.0;
status = AudioDeviceSetProperty(ao->outputDeviceID, 0, stereoChannels[1], 0, kAudioDevicePropertyVolumeScalar, propertySize, &volume);
return CONTROL_TRUE;
case AOCONTROL_QUERY_FORMAT:
/* stick with what CoreAudio requests */
return CONTROL_FALSE;
default:
return CONTROL_FALSE;
}
}
static void print_format(const char* str,AudioStreamBasicDescription *f){
uint32_t flags=(uint32_t) f->mFormatFlags;
ao_msg(MSGT_AO,MSGL_V, "%s %7.1fHz %dbit [%c%c%c%c] %s %s %s%s%s%s\n",
str, f->mSampleRate, f->mBitsPerChannel,
(int)(f->mFormatID & 0xff000000) >> 24,
(int)(f->mFormatID & 0x00ff0000) >> 16,
(int)(f->mFormatID & 0x0000ff00) >> 8,
(int)(f->mFormatID & 0x000000ff) >> 0,
(flags&kAudioFormatFlagIsFloat) ? "float" : "int",
(flags&kAudioFormatFlagIsBigEndian) ? "BE" : "LE",
(flags&kAudioFormatFlagIsSignedInteger) ? "S" : "U",
(flags&kAudioFormatFlagIsPacked) ? " packed" : "",
(flags&kAudioFormatFlagIsAlignedHigh) ? " aligned" : "",
(flags&kAudioFormatFlagIsNonInterleaved) ? " ni" : "" );
ao_msg(MSGT_AO,MSGL_DBG2, "%5d mBytesPerPacket\n",
(int)f->mBytesPerPacket);
ao_msg(MSGT_AO,MSGL_DBG2, "%5d mFramesPerPacket\n",
(int)f->mFramesPerPacket);
ao_msg(MSGT_AO,MSGL_DBG2, "%5d mBytesPerFrame\n",
(int)f->mBytesPerFrame);
ao_msg(MSGT_AO,MSGL_DBG2, "%5d mChannelsPerFrame\n",
(int)f->mChannelsPerFrame);
}
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));
/* 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;
}
propertySize = sizeof(ao->outputStreamBasicDescription);
status = AudioDeviceGetProperty(ao->outputDeviceID, 0, false, kAudioDevicePropertyStreamFormat, &propertySize, &ao->outputStreamBasicDescription);
if(!status) print_format("default:",&ao->outputStreamBasicDescription);
#if 1
// dump supported format list:
{ AudioStreamBasicDescription* p;
Boolean ow;
int i;
propertySize=0; //sizeof(p);
// status = AudioDeviceGetPropertyInfo(ao->outputDeviceID, 0, false, kAudioStreamPropertyPhysicalFormats, &propertySize, &ow);
status = AudioDeviceGetPropertyInfo(ao->outputDeviceID, 0, false, kAudioDevicePropertyStreamFormats, &propertySize, &ow);
if (status) {
ao_msg(MSGT_AO,MSGL_WARN, "AudioDeviceGetPropertyInfo returned 0x%X when getting kAudioDevicePropertyStreamFormats\n", (int)status);
}
p=malloc(propertySize);
// status = AudioDeviceGetProperty(ao->outputDeviceID, 0, false, kAudioStreamPropertyPhysicalFormats, &propertySize, p);
status = AudioDeviceGetProperty(ao->outputDeviceID, 0, false, kAudioDevicePropertyStreamFormats, &propertySize, p);
if (status) {
ao_msg(MSGT_AO,MSGL_WARN, "AudioDeviceGetProperty returned 0x%X when getting kAudioDevicePropertyStreamFormats\n", (int)status);
// return CONTROL_FALSE;
}
for(i=0;i<propertySize/sizeof(AudioStreamBasicDescription);i++)
print_format("support:",&p[i]);
// printf("FORMATS: (%d) %p %p %p %p\n",propertySize,p[0],p[1],p[2],p[3]);
free(p);
}
#endif
// fill in our wanted format, and let's see if the driver accepts it or
// offers some similar alternative:
propertySize = sizeof(ao->outputStreamBasicDescription);
memset(&ao->outputStreamBasicDescription,0,propertySize);
ao->outputStreamBasicDescription.mSampleRate=rate;
ao->outputStreamBasicDescription.mFormatID=kAudioFormatLinearPCM;
ao->outputStreamBasicDescription.mChannelsPerFrame=channels;
switch(format&AF_FORMAT_BITS_MASK){
case AF_FORMAT_8BIT: ao->outputStreamBasicDescription.mBitsPerChannel=8; break;
case AF_FORMAT_16BIT: ao->outputStreamBasicDescription.mBitsPerChannel=16; break;
case AF_FORMAT_24BIT: ao->outputStreamBasicDescription.mBitsPerChannel=24; break;
case AF_FORMAT_32BIT: ao->outputStreamBasicDescription.mBitsPerChannel=32; break;
}
if((format&AF_FORMAT_POINT_MASK)==AF_FORMAT_F){
// float
ao->outputStreamBasicDescription.mFormatFlags=kAudioFormatFlagIsFloat|kAudioFormatFlagIsPacked;
} else if((format&AF_FORMAT_SIGN_MASK)==AF_FORMAT_SI){
// signed int
ao->outputStreamBasicDescription.mFormatFlags=kAudioFormatFlagIsSignedInteger|kAudioFormatFlagIsPacked;
} else {
// unsigned int
ao->outputStreamBasicDescription.mFormatFlags=kAudioFormatFlagIsPacked;
}
if((format&AF_FORMAT_END_MASK)==AF_FORMAT_BE)
ao->outputStreamBasicDescription.mFormatFlags|=kAudioFormatFlagIsBigEndian;
ao->outputStreamBasicDescription.mBytesPerPacket=
ao->outputStreamBasicDescription.mBytesPerFrame=channels*(ao->outputStreamBasicDescription.mBitsPerChannel/8);
ao->outputStreamBasicDescription.mFramesPerPacket=1;
print_format("wanted: ",&ao->outputStreamBasicDescription);
// try 1: ask if it accepts our specific requirements?
propertySize = sizeof(ao->outputStreamBasicDescription);
// status = AudioDeviceGetProperty(ao->outputDeviceID, 0, false, kAudioStreamPropertyPhysicalFormatMatch, &propertySize, &ao->outputStreamBasicDescription);
status = AudioDeviceGetProperty(ao->outputDeviceID, 0, false, kAudioDevicePropertyStreamFormatMatch, &propertySize, &ao->outputStreamBasicDescription);
if (status || ao->outputStreamBasicDescription.mSampleRate!=rate
|| ao->outputStreamBasicDescription.mFormatID!=kAudioFormatLinearPCM) {
ao_msg(MSGT_AO,MSGL_V, "AudioDeviceGetProperty returned 0x%X when getting kAudioDevicePropertyStreamFormatMatch\n", (int)status);
// failed (error, bad rate or bad type)
// try 2: set only rate & type, no format details (bits, channels etc)
propertySize = sizeof(ao->outputStreamBasicDescription);
memset(&ao->outputStreamBasicDescription,0,propertySize);
ao->outputStreamBasicDescription.mSampleRate=rate;
ao->outputStreamBasicDescription.mFormatID=kAudioFormatLinearPCM;
// status = AudioDeviceGetProperty(ao->outputDeviceID, 0, false, kAudioStreamPropertyPhysicalFormatMatch, &propertySize, &ao->outputStreamBasicDescription);
status = AudioDeviceGetProperty(ao->outputDeviceID, 0, false, kAudioDevicePropertyStreamFormatMatch, &propertySize, &ao->outputStreamBasicDescription);
if (status || ao->outputStreamBasicDescription.mFormatID!=kAudioFormatLinearPCM) {
ao_msg(MSGT_AO,MSGL_V, "AudioDeviceGetProperty returned 0x%X when getting kAudioDevicePropertyStreamFormatMatch\n", (int)status);
// failed again. (error or bad type)
// giving up... just read the default.
propertySize = sizeof(ao->outputStreamBasicDescription);
// status = AudioDeviceGetProperty(ao->outputDeviceID, 0, false, kAudioStreamPropertyPhysicalFormat, &propertySize, &ao->outputStreamBasicDescription);
status = AudioDeviceGetProperty(ao->outputDeviceID, 0, false, kAudioDevicePropertyStreamFormat, &propertySize, &ao->outputStreamBasicDescription);
if (status) {
// failed to read the default format - WTF?
ao_msg(MSGT_AO,MSGL_WARN, "AudioDeviceGetProperty returned 0x%X when getting kAudioDevicePropertyStreamFormat\n", (int)status);
return CONTROL_FALSE;
}
}
}
// propertySize = sizeof(ao->outputStreamBasicDescription);
// status = AudioDeviceGetProperty(ao->outputDeviceID, 0, false, kAudioDevicePropertyStreamFormatSupported, &propertySize, &ao->outputStreamBasicDescription);
// if (status) {
// ao_msg(MSGT_AO,MSGL_V, "AudioDeviceGetProperty returned 0x%X when getting kAudioDevicePropertyStreamFormatSupported\n", (int)status);
// }
// ok, now try to set the new (default or matched) audio format:
print_format("best: ",&ao->outputStreamBasicDescription);
propertySize = sizeof(ao->outputStreamBasicDescription);
status = AudioDeviceSetProperty(ao->outputDeviceID, 0, 0, false, kAudioDevicePropertyStreamFormat, propertySize, &ao->outputStreamBasicDescription);
if(status)
ao_msg(MSGT_AO,MSGL_WARN, "AudioDeviceSetProperty returned 0x%X when getting kAudioDevicePropertyStreamFormat\n", (int)status);
// see what did we get finally... we'll be forced to use this anyway :(
propertySize = sizeof(ao->outputStreamBasicDescription);
status = AudioDeviceGetProperty(ao->outputDeviceID, 0, false, kAudioDevicePropertyStreamFormat, &propertySize, &ao->outputStreamBasicDescription);
print_format("final: ",&ao->outputStreamBasicDescription);
/* get requested buffer length */
// TODO: set NUM_BUFS dinamically, based on buffer size!
propertySize = sizeof(ao->chunk_size);
status = AudioDeviceGetProperty(ao->outputDeviceID, 0, false, kAudioDevicePropertyBufferSize, &propertySize, &ao->chunk_size);
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 chunk size\n", (int)ao->chunk_size);
ao_data.samplerate = ao->outputStreamBasicDescription.mSampleRate;
ao_data.channels = channels;
ao_data.outburst = ao_data.buffersize = ao->chunk_size;
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 = (flags&kAudioFormatFlagIsBigEndian) ? AF_FORMAT_FLOAT_BE : AF_FORMAT_FLOAT_LE;
} else {
ao_msg(MSGT_AO,MSGL_WARN, "Unsupported audio output "
"format 0x%X. Please report this to the developer\n", format);
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 */
ao->num_chunks = NUM_BUFS;
ao->buffer_len = (ao->num_chunks + 1) * ao->chunk_size;
ao->buffer = (unsigned char *)malloc(ao->buffer_len);
/* Prepare for playback */
/* 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 */
reset();
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()
{
audio_pause();
/* reset ring-buffer state */
ao->buf_read_pos=0;
ao->buf_write_pos=0;
audio_resume();
return;
}
/* return available space */
static int get_space()
{
return buf_free();
}
/* return delay until audio is played */
static float get_delay()
{
int buffered = ao->buffer_len - ao->chunk_size - buf_free(); // could be less
// inaccurate, should also contain the data buffered e.g. by the OS
return (float)(buffered)/(float)ao_data.bps;
}
/* unload plugin and deregister from coreaudio */
static void uninit(int immed)
{
int i;
OSErr status;
reset();
status = AudioDeviceRemoveIOProc(ao->outputDeviceID, audioDeviceIOProc);
if (status)
ao_msg(MSGT_AO,MSGL_WARN, "AudioDeviceRemoveIOProc "
"returned %d\n", (int)status);
free(ao->buffer);
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);
}