RepoMirrors/mpv
1
0
Fork 0
mirror of https://github.com/mpv-player/mpv synced 2024-12-25 16:33:02 +00:00
Code Issues Releases Wiki Activity
92f9a746ae
mpv/libao2/ao_macosx.c
diego 92f79295d2 Use standard license header, merge changelog into license header and TODO. 
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@26994 b3059339-0415-0410-9bf9-f77b7e298cf2
2008-06-06 00:45:35 +00:00

1206 lines
43 KiB
C
Raw Blame History

/*
* Mac OS X audio output driver
*
* original copyright (C) Timothy J. Wood - Aug 2000
* ported to MPlayer libao2 by Dan Christiansen
*
* The S/PDIF part of the code is based on the auhal audio output
* module from VideoLAN:
* Copyright (c) 2006 Derk-Jan Hartman <hartman at videolan dot org>
*
* This file is part of MPlayer.
*
* MPlayer 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 of the License, or
* (at your option) any later version.
*
* MPlayer 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
* along with MPlayer; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 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).
*
* AC-3 and MPEG audio passthrough is possible, but has never been tested
* due to lack of a soundcard that supports it.
*/
#include <CoreServices/CoreServices.h>
#include <AudioUnit/AudioUnit.h>
#include <AudioToolbox/AudioToolbox.h>
#include <stdio.h>
#include <string.h>
#include <stdlib.h>
#include <inttypes.h>
#include <sys/types.h>
#include <unistd.h>
#include "config.h"
#include "mp_msg.h"
#include "audio_out.h"
#include "audio_out_internal.h"
#include "libaf/af_format.h"
#include "osdep/timer.h"
static ao_info_t info =
{
"Darwin/Mac OS X native audio output",
"macosx",
"Timothy J. Wood & Dan Christiansen & Chris Roccati",
""
};
LIBAO_EXTERN(macosx)
/* Prefix for all mp_msg() calls */
#define ao_msg(a, b, c...) mp_msg(a, b, "AO: [macosx] " c)
typedef struct ao_macosx_s
{
AudioDeviceID i_selected_dev; /* Keeps DeviceID of the selected device. */
int b_supports_digital; /* Does the currently selected device support digital mode? */
int b_digital; /* Are we running in digital mode? */
int b_muted; /* Are we muted in digital mode? */
/* AudioUnit */
AudioUnit theOutputUnit;
/* CoreAudio SPDIF mode specific */
pid_t i_hog_pid; /* Keeps the pid of our hog status. */
AudioStreamID i_stream_id; /* The StreamID that has a cac3 streamformat */
int i_stream_index; /* The index of i_stream_id in an AudioBufferList */
AudioStreamBasicDescription stream_format;/* The format we changed the stream to */
AudioStreamBasicDescription sfmt_revert; /* The original format of the stream */
int b_revert; /* Whether we need to revert the stream format */
int b_changed_mixing; /* Whether we need to set the mixing mode back */
int b_stream_format_changed; /* Flag for main thread to reset stream's format to digital and reset buffer */
/* Original common part */
int packetSize;
int paused;
/* 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 = NULL;
/**
* \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(void) {
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(void) {
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
if (data) {
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;
}
OSStatus theRenderProc(void *inRefCon, AudioUnitRenderActionFlags *inActionFlags, const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, UInt32 inNumFrames, AudioBufferList *ioData)
{
int amt=buf_used();
int req=(inNumFrames)*ao->packetSize;
if(amt>req)
amt=req;
if(amt)
read_buffer((unsigned char *)ioData->mBuffers[0].mData, amt);
else audio_pause();
ioData->mBuffers[0].mDataByteSize = amt;
return noErr;
}
static int control(int cmd,void *arg){
ao_control_vol_t *control_vol;
OSStatus err;
Float32 vol;
switch (cmd) {
case AOCONTROL_GET_VOLUME:
control_vol = (ao_control_vol_t*)arg;
if (ao->b_digital) {
// Digital output has no volume adjust.
return CONTROL_FALSE;
}
err = AudioUnitGetParameter(ao->theOutputUnit, kHALOutputParam_Volume, kAudioUnitScope_Global, 0, &vol);
if(err==0) {
// printf("GET VOL=%f\n", vol);
control_vol->left=control_vol->right=vol*100.0/4.0;
return CONTROL_TRUE;
}
else {
ao_msg(MSGT_AO, MSGL_WARN, "could not get HAL output volume: [%4.4s]\n", (char *)&err);
return CONTROL_FALSE;
}
case AOCONTROL_SET_VOLUME:
control_vol = (ao_control_vol_t*)arg;
if (ao->b_digital) {
// Digital output can not set volume. Here we have to return true
// to make mixer forget it. Else mixer will add a soft filter,
// that's not we expected and the filter not support ac3 stream
// will cause mplayer die.
// Although not support set volume, but at least we support mute.
// MPlayer set mute by set volume to zero, we handle it.
if (control_vol->left == 0 && control_vol->right == 0)
ao->b_muted = 1;
else
ao->b_muted = 0;
return CONTROL_TRUE;
}
vol=(control_vol->left+control_vol->right)*4.0/200.0;
err = AudioUnitSetParameter(ao->theOutputUnit, kHALOutputParam_Volume, kAudioUnitScope_Global, 0, vol, 0);
if(err==0) {
// printf("SET VOL=%f\n", vol);
return CONTROL_TRUE;
}
else {
ao_msg(MSGT_AO, MSGL_WARN, "could not set HAL output volume: [%4.4s]\n", (char *)&err);
return CONTROL_FALSE;
}
/* Everything is currently unimplemented */
default:
return CONTROL_FALSE;
}
}
static void print_format(int lev, const char* str, const AudioStreamBasicDescription *f){
uint32_t flags=(uint32_t) f->mFormatFlags;
ao_msg(MSGT_AO,lev, "%s %7.1fHz %lubit [%c%c%c%c][%lu][%lu][%lu][%lu][%lu] %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,
f->mFormatFlags, f->mBytesPerPacket,
f->mFramesPerPacket, f->mBytesPerFrame,
f->mChannelsPerFrame,
(flags&kAudioFormatFlagIsFloat) ? "float" : "int",
(flags&kAudioFormatFlagIsBigEndian) ? "BE" : "LE",
(flags&kAudioFormatFlagIsSignedInteger) ? "S" : "U",
(flags&kAudioFormatFlagIsPacked) ? " packed" : "",
(flags&kAudioFormatFlagIsAlignedHigh) ? " aligned" : "",
(flags&kAudioFormatFlagIsNonInterleaved) ? " ni" : "" );
}
static int AudioDeviceSupportsDigital( AudioDeviceID i_dev_id );
static int AudioStreamSupportsDigital( AudioStreamID i_stream_id );
static int OpenSPDIF();
static int AudioStreamChangeFormat( AudioStreamID i_stream_id, AudioStreamBasicDescription change_format );
static OSStatus RenderCallbackSPDIF( AudioDeviceID inDevice,
const AudioTimeStamp * inNow,
const void * inInputData,
const AudioTimeStamp * inInputTime,
AudioBufferList * outOutputData,
const AudioTimeStamp * inOutputTime,
void * threadGlobals );
static OSStatus StreamListener( AudioStreamID inStream,
UInt32 inChannel,
AudioDevicePropertyID inPropertyID,
void * inClientData );
static OSStatus DeviceListener( AudioDeviceID inDevice,
UInt32 inChannel,
Boolean isInput,
AudioDevicePropertyID inPropertyID,
void* inClientData );
static int init(int rate,int channels,int format,int flags)
{
AudioStreamBasicDescription inDesc;
ComponentDescription desc;
Component comp;
AURenderCallbackStruct renderCallback;
OSStatus err;
UInt32 size, maxFrames, i_param_size;
char *psz_name;
AudioDeviceID devid_def = 0;
int b_alive;
ao_msg(MSGT_AO,MSGL_V, "init([%dHz][%dch][%s][%d])\n", rate, channels, af_fmt2str_short(format), flags);
ao = calloc(1, sizeof(ao_macosx_t));
ao->i_selected_dev = 0;
ao->b_supports_digital = 0;
ao->b_digital = 0;
ao->b_muted = 0;
ao->b_stream_format_changed = 0;
ao->i_hog_pid = -1;
ao->i_stream_id = 0;
ao->i_stream_index = -1;
ao->b_revert = 0;
ao->b_changed_mixing = 0;
/* Probe whether device support S/PDIF stream output if input is AC3 stream. */
if ((format & AF_FORMAT_SPECIAL_MASK) == AF_FORMAT_AC3)
{
/* Find the ID of the default Device. */
i_param_size = sizeof(AudioDeviceID);
err = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice,
&i_param_size, &devid_def);
if (err != noErr)
{
ao_msg(MSGT_AO, MSGL_WARN, "could not get default audio device: [%4.4s]\n", (char *)&err);
goto err_out;
}
/* Retrieve the length of the device name. */
i_param_size = 0;
err = AudioDeviceGetPropertyInfo(devid_def, 0, 0,
kAudioDevicePropertyDeviceName,
&i_param_size, NULL);
if (err != noErr)
{
ao_msg(MSGT_AO, MSGL_WARN, "could not get default audio device name length: [%4.4s]\n", (char *)&err);
goto err_out;
}
/* Retrieve the name of the device. */
psz_name = (char *)malloc(i_param_size);
err = AudioDeviceGetProperty(devid_def, 0, 0,
kAudioDevicePropertyDeviceName,
&i_param_size, psz_name);
if (err != noErr)
{
ao_msg(MSGT_AO, MSGL_WARN, "could not get default audio device name: [%4.4s]\n", (char *)&err);
free( psz_name);
goto err_out;
}
ao_msg(MSGT_AO,MSGL_V, "got default audio output device ID: %#lx Name: %s\n", devid_def, psz_name );
if (AudioDeviceSupportsDigital(devid_def))
{
ao->b_supports_digital = 1;
ao->i_selected_dev = devid_def;
}
ao_msg(MSGT_AO,MSGL_V, "probe default audio output device whether support digital s/pdif output:%d\n", ao->b_supports_digital );
free( psz_name);
}
// Build Description for the input format
inDesc.mSampleRate=rate;
inDesc.mFormatID=ao->b_supports_digital ? kAudioFormat60958AC3 : kAudioFormatLinearPCM;
inDesc.mChannelsPerFrame=channels;
switch(format&AF_FORMAT_BITS_MASK){
case AF_FORMAT_8BIT:
inDesc.mBitsPerChannel=8;
break;
case AF_FORMAT_16BIT:
inDesc.mBitsPerChannel=16;
break;
case AF_FORMAT_24BIT:
inDesc.mBitsPerChannel=24;
break;
case AF_FORMAT_32BIT:
inDesc.mBitsPerChannel=32;
break;
default:
ao_msg(MSGT_AO, MSGL_WARN, "Unsupported format (0x%08x)\n", format);
goto err_out;
}
if((format&AF_FORMAT_POINT_MASK)==AF_FORMAT_F) {
// float
inDesc.mFormatFlags = kAudioFormatFlagIsFloat|kAudioFormatFlagIsPacked;
}
else if((format&AF_FORMAT_SIGN_MASK)==AF_FORMAT_SI) {
// signed int
inDesc.mFormatFlags = kAudioFormatFlagIsSignedInteger|kAudioFormatFlagIsPacked;
}
else {
// unsigned int
inDesc.mFormatFlags = kAudioFormatFlagIsPacked;
}
if ((format & AF_FORMAT_SPECIAL_MASK) == AF_FORMAT_AC3) {
// Currently ac3 input (comes from hwac3) is always in native byte-order.
#ifdef WORDS_BIGENDIAN
inDesc.mFormatFlags |= kAudioFormatFlagIsBigEndian;
#endif
}
else if ((format & AF_FORMAT_END_MASK) == AF_FORMAT_BE)
inDesc.mFormatFlags |= kAudioFormatFlagIsBigEndian;
inDesc.mFramesPerPacket = 1;
ao->packetSize = inDesc.mBytesPerPacket = inDesc.mBytesPerFrame = inDesc.mFramesPerPacket*channels*(inDesc.mBitsPerChannel/8);
print_format(MSGL_V, "source:",&inDesc);
if (ao->b_supports_digital)
{
b_alive = 1;
i_param_size = sizeof(b_alive);
err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE,
kAudioDevicePropertyDeviceIsAlive,
&i_param_size, &b_alive);
if (err != noErr)
ao_msg(MSGT_AO, MSGL_WARN, "could not check whether device is alive: [%4.4s]\n", (char *)&err);
if (!b_alive)
ao_msg(MSGT_AO, MSGL_WARN, "device is not alive\n" );
/* S/PDIF output need device in HogMode. */
i_param_size = sizeof(ao->i_hog_pid);
err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE,
kAudioDevicePropertyHogMode,
&i_param_size, &ao->i_hog_pid);
if (err != noErr)
{
/* This is not a fatal error. Some drivers simply don't support this property. */
ao_msg(MSGT_AO, MSGL_WARN, "could not check whether device is hogged: [%4.4s]\n",
(char *)&err);
ao->i_hog_pid = -1;
}
if (ao->i_hog_pid != -1 && ao->i_hog_pid != getpid())
{
ao_msg(MSGT_AO, MSGL_WARN, "Selected audio device is exclusively in use by another program.\n" );
goto err_out;
}
ao->stream_format = inDesc;
return OpenSPDIF();
}
/* original analog output code */
desc.componentType = kAudioUnitType_Output;
desc.componentSubType = kAudioUnitSubType_DefaultOutput;
desc.componentManufacturer = kAudioUnitManufacturer_Apple;
desc.componentFlags = 0;
desc.componentFlagsMask = 0;
comp = FindNextComponent(NULL, &desc); //Finds an component that meets the desc spec's
if (comp == NULL) {
ao_msg(MSGT_AO, MSGL_WARN, "Unable to find Output Unit component\n");
goto err_out;
}
err = OpenAComponent(comp, &(ao->theOutputUnit)); //gains access to the services provided by the component
if (err) {
ao_msg(MSGT_AO, MSGL_WARN, "Unable to open Output Unit component: [%4.4s]\n", (char *)&err);
goto err_out;
}
// Initialize AudioUnit
err = AudioUnitInitialize(ao->theOutputUnit);
if (err) {
ao_msg(MSGT_AO, MSGL_WARN, "Unable to initialize Output Unit component: [%4.4s]\n", (char *)&err);
goto err_out1;
}
size = sizeof(AudioStreamBasicDescription);
err = AudioUnitSetProperty(ao->theOutputUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &inDesc, size);
if (err) {
ao_msg(MSGT_AO, MSGL_WARN, "Unable to set the input format: [%4.4s]\n", (char *)&err);
goto err_out2;
}
size = sizeof(UInt32);
err = AudioUnitGetProperty(ao->theOutputUnit, kAudioDevicePropertyBufferSize, kAudioUnitScope_Input, 0, &maxFrames, &size);
if (err)
{
ao_msg(MSGT_AO,MSGL_WARN, "AudioUnitGetProperty returned [%4.4s] when getting kAudioDevicePropertyBufferSize\n", (char *)&err);
goto err_out2;
}
ao->chunk_size = maxFrames;//*inDesc.mBytesPerFrame;
ao_data.samplerate = inDesc.mSampleRate;
ao_data.channels = inDesc.mChannelsPerFrame;
ao_data.bps = ao_data.samplerate * inDesc.mBytesPerFrame;
ao_data.outburst = ao->chunk_size;
ao_data.buffersize = ao_data.bps;
ao->num_chunks = (ao_data.bps+ao->chunk_size-1)/ao->chunk_size;
ao->buffer_len = (ao->num_chunks + 1) * ao->chunk_size;
ao->buffer = calloc(ao->num_chunks + 1, ao->chunk_size);
ao_msg(MSGT_AO,MSGL_V, "using %5d chunks of %d bytes (buffer len %d bytes)\n", (int)ao->num_chunks, (int)ao->chunk_size, (int)ao->buffer_len);
renderCallback.inputProc = theRenderProc;
renderCallback.inputProcRefCon = 0;
err = AudioUnitSetProperty(ao->theOutputUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, 0, &renderCallback, sizeof(AURenderCallbackStruct));
if (err) {
ao_msg(MSGT_AO, MSGL_WARN, "Unable to set the render callback: [%4.4s]\n", (char *)&err);
goto err_out2;
}
reset();
return CONTROL_OK;
err_out2:
AudioUnitUninitialize(ao->theOutputUnit);
err_out1:
CloseComponent(ao->theOutputUnit);
err_out:
free(ao->buffer);
free(ao);
ao = NULL;
return CONTROL_FALSE;
}
/*****************************************************************************
* Setup a encoded digital stream (SPDIF)
*****************************************************************************/
static int OpenSPDIF()
{
OSStatus err = noErr;
UInt32 i_param_size, b_mix = 0;
Boolean b_writeable = 0;
AudioStreamID *p_streams = NULL;
int i, i_streams = 0;
/* Start doing the SPDIF setup process. */
ao->b_digital = 1;
/* Hog the device. */
i_param_size = sizeof(ao->i_hog_pid);
ao->i_hog_pid = getpid() ;
err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE,
kAudioDevicePropertyHogMode, i_param_size, &ao->i_hog_pid);
if (err != noErr)
{
ao_msg(MSGT_AO, MSGL_WARN, "failed to set hogmode: [%4.4s]\n", (char *)&err);
ao->i_hog_pid = -1;
goto err_out;
}
/* Set mixable to false if we are allowed to. */
err = AudioDeviceGetPropertyInfo(ao->i_selected_dev, 0, FALSE,
kAudioDevicePropertySupportsMixing,
&i_param_size, &b_writeable);
err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE,
kAudioDevicePropertySupportsMixing,
&i_param_size, &b_mix);
if (err != noErr && b_writeable)
{
b_mix = 0;
err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE,
kAudioDevicePropertySupportsMixing,
i_param_size, &b_mix);
ao->b_changed_mixing = 1;
}
if (err != noErr)
{
ao_msg(MSGT_AO, MSGL_WARN, "failed to set mixmode: [%4.4s]\n", (char *)&err);
goto err_out;
}
/* Get a list of all the streams on this device. */
err = AudioDeviceGetPropertyInfo(ao->i_selected_dev, 0, FALSE,
kAudioDevicePropertyStreams,
&i_param_size, NULL);
if (err != noErr)
{
ao_msg(MSGT_AO, MSGL_WARN, "could not get number of streams: [%4.4s]\n", (char *)&err);
goto err_out;
}
i_streams = i_param_size / sizeof(AudioStreamID);
p_streams = (AudioStreamID *)malloc(i_param_size);
if (p_streams == NULL)
{
ao_msg(MSGT_AO, MSGL_WARN, "out of memory\n" );
goto err_out;
}
err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE,
kAudioDevicePropertyStreams,
&i_param_size, p_streams);
if (err != noErr)
{
ao_msg(MSGT_AO, MSGL_WARN, "could not get number of streams: [%4.4s]\n", (char *)&err);
if (p_streams) free(p_streams);
goto err_out;
}
ao_msg(MSGT_AO, MSGL_V, "current device stream number: %d\n", i_streams);
for (i = 0; i < i_streams && ao->i_stream_index < 0; ++i)
{
/* Find a stream with a cac3 stream. */
AudioStreamBasicDescription *p_format_list = NULL;
int i_formats = 0, j = 0, b_digital = 0;
/* Retrieve all the stream formats supported by each output stream. */
err = AudioStreamGetPropertyInfo(p_streams[i], 0,
kAudioStreamPropertyPhysicalFormats,
&i_param_size, NULL);
if (err != noErr)
{
ao_msg(MSGT_AO, MSGL_WARN, "could not get number of streamformats: [%4.4s]\n", (char *)&err);
continue;
}
i_formats = i_param_size / sizeof(AudioStreamBasicDescription);
p_format_list = (AudioStreamBasicDescription *)malloc(i_param_size);
if (p_format_list == NULL)
{
ao_msg(MSGT_AO, MSGL_WARN, "could not malloc the memory\n" );
continue;
}
err = AudioStreamGetProperty(p_streams[i], 0,
kAudioStreamPropertyPhysicalFormats,
&i_param_size, p_format_list);
if (err != noErr)
{
ao_msg(MSGT_AO, MSGL_WARN, "could not get the list of streamformats: [%4.4s]\n", (char *)&err);
if (p_format_list) free(p_format_list);
continue;
}
/* Check if one of the supported formats is a digital format. */
for (j = 0; j < i_formats; ++j)
{
if (p_format_list[j].mFormatID == 'IAC3' ||
p_format_list[j].mFormatID == kAudioFormat60958AC3)
{
b_digital = 1;
break;
}
}
if (b_digital)
{
/* If this stream supports a digital (cac3) format, then set it. */
int i_requested_rate_format = -1;
int i_current_rate_format = -1;
int i_backup_rate_format = -1;
ao->i_stream_id = p_streams[i];
ao->i_stream_index = i;
if (ao->b_revert == 0)
{
/* Retrieve the original format of this stream first if not done so already. */
i_param_size = sizeof(ao->sfmt_revert);
err = AudioStreamGetProperty(ao->i_stream_id, 0,
kAudioStreamPropertyPhysicalFormat,
&i_param_size,
&ao->sfmt_revert);
if (err != noErr)
{
ao_msg(MSGT_AO, MSGL_WARN, "could not retrieve the original streamformat: [%4.4s]\n", (char *)&err);
if (p_format_list) free(p_format_list);
continue;
}
ao->b_revert = 1;
}
for (j = 0; j < i_formats; ++j)
if (p_format_list[j].mFormatID == 'IAC3' ||
p_format_list[j].mFormatID == kAudioFormat60958AC3)
{
if (p_format_list[j].mSampleRate == ao->stream_format.mSampleRate)
{
i_requested_rate_format = j;
break;
}
if (p_format_list[j].mSampleRate == ao->sfmt_revert.mSampleRate)
i_current_rate_format = j;
else if (i_backup_rate_format < 0 || p_format_list[j].mSampleRate > p_format_list[i_backup_rate_format].mSampleRate)
i_backup_rate_format = j;
}
if (i_requested_rate_format >= 0) /* We prefer to output at the samplerate of the original audio. */
ao->stream_format = p_format_list[i_requested_rate_format];
else if (i_current_rate_format >= 0) /* If not possible, we will try to use the current samplerate of the device. */
ao->stream_format = p_format_list[i_current_rate_format];
else ao->stream_format = p_format_list[i_backup_rate_format]; /* And if we have to, any digital format will be just fine (highest rate possible). */
}
if (p_format_list) free(p_format_list);
}
if (p_streams) free(p_streams);
if (ao->i_stream_index < 0)
{
ao_msg(MSGT_AO, MSGL_WARN, "can not find any digital output stream format when OpenSPDIF().\n");
goto err_out;
}
print_format(MSGL_V, "original stream format:", &ao->sfmt_revert);
if (!AudioStreamChangeFormat(ao->i_stream_id, ao->stream_format))
goto err_out;
err = AudioDeviceAddPropertyListener(ao->i_selected_dev,
kAudioPropertyWildcardChannel,
0,
kAudioDevicePropertyDeviceHasChanged,
DeviceListener,
NULL);
if (err != noErr)
ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceAddPropertyListener for kAudioDevicePropertyDeviceHasChanged failed: [%4.4s]\n", (char *)&err);
/* FIXME: If output stream is not native byte-order, we need change endian somewhere. */
/* Although there's no such case reported. */
#ifdef WORDS_BIGENDIAN
if (!(ao->stream_format.mFormatFlags & kAudioFormatFlagIsBigEndian))
#else
if (ao->stream_format.mFormatFlags & kAudioFormatFlagIsBigEndian)
#endif
ao_msg(MSGT_AO, MSGL_WARN, "output stream has a no-native byte-order, digital output may failed.\n");
/* For ac3/dts, just use packet size 6144 bytes as chunk size. */
ao->chunk_size = ao->stream_format.mBytesPerPacket;
ao_data.samplerate = ao->stream_format.mSampleRate;
ao_data.channels = ao->stream_format.mChannelsPerFrame;
ao_data.bps = ao_data.samplerate * (ao->stream_format.mBytesPerPacket/ao->stream_format.mFramesPerPacket);
ao_data.outburst = ao->chunk_size;
ao_data.buffersize = ao_data.bps;
ao->num_chunks = (ao_data.bps+ao->chunk_size-1)/ao->chunk_size;
ao->buffer_len = (ao->num_chunks + 1) * ao->chunk_size;
ao->buffer = calloc(ao->num_chunks + 1, ao->chunk_size);
ao_msg(MSGT_AO,MSGL_V, "using %5d chunks of %d bytes (buffer len %d bytes)\n", (int)ao->num_chunks, (int)ao->chunk_size, (int)ao->buffer_len);
/* Add IOProc callback. */
err = AudioDeviceAddIOProc(ao->i_selected_dev,
(AudioDeviceIOProc)RenderCallbackSPDIF,
(void *)ao);
if (err != noErr)
{
ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceAddIOProc failed: [%4.4s]\n", (char *)&err);
goto err_out1;
}
reset();
return CONTROL_TRUE;
err_out1:
if (ao->b_revert)
AudioStreamChangeFormat(ao->i_stream_id, ao->sfmt_revert);
err_out:
if (ao->b_changed_mixing && ao->sfmt_revert.mFormatID != kAudioFormat60958AC3)
{
int b_mix = 1;
err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE,
kAudioDevicePropertySupportsMixing,
i_param_size, &b_mix);
if (err != noErr)
ao_msg(MSGT_AO, MSGL_WARN, "failed to set mixmode: [%4.4s]\n",
(char *)&err);
}
if (ao->i_hog_pid == getpid())
{
ao->i_hog_pid = -1;
i_param_size = sizeof(ao->i_hog_pid);
err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE,
kAudioDevicePropertyHogMode,
i_param_size, &ao->i_hog_pid);
if (err != noErr)
ao_msg(MSGT_AO, MSGL_WARN, "Could not release hogmode: [%4.4s]\n",
(char *)&err);
}
free(ao->buffer);
free(ao);
ao = NULL;
return CONTROL_FALSE;
}
/*****************************************************************************
* AudioDeviceSupportsDigital: Check i_dev_id for digital stream support.
*****************************************************************************/
static int AudioDeviceSupportsDigital( AudioDeviceID i_dev_id )
{
OSStatus err = noErr;
UInt32 i_param_size = 0;
AudioStreamID *p_streams = NULL;
int i = 0, i_streams = 0;
int b_return = CONTROL_FALSE;
/* Retrieve all the output streams. */
err = AudioDeviceGetPropertyInfo(i_dev_id, 0, FALSE,
kAudioDevicePropertyStreams,
&i_param_size, NULL);
if (err != noErr)
{
ao_msg(MSGT_AO,MSGL_V, "could not get number of streams: [%4.4s]\n", (char *)&err);
return CONTROL_FALSE;
}
i_streams = i_param_size / sizeof(AudioStreamID);
p_streams = (AudioStreamID *)malloc(i_param_size);
if (p_streams == NULL)
{
ao_msg(MSGT_AO,MSGL_V, "out of memory\n");
return CONTROL_FALSE;
}
err = AudioDeviceGetProperty(i_dev_id, 0, FALSE,
kAudioDevicePropertyStreams,
&i_param_size, p_streams);
if (err != noErr)
{
ao_msg(MSGT_AO,MSGL_V, "could not get number of streams: [%4.4s]\n", (char *)&err);
free(p_streams);
return CONTROL_FALSE;
}
for (i = 0; i < i_streams; ++i)
{
if (AudioStreamSupportsDigital(p_streams[i]))
b_return = CONTROL_OK;
}
free(p_streams);
return b_return;
}
/*****************************************************************************
* AudioStreamSupportsDigital: Check i_stream_id for digital stream support.
*****************************************************************************/
static int AudioStreamSupportsDigital( AudioStreamID i_stream_id )
{
OSStatus err = noErr;
UInt32 i_param_size;
AudioStreamBasicDescription *p_format_list = NULL;
int i, i_formats, b_return = CONTROL_FALSE;
/* Retrieve all the stream formats supported by each output stream. */
err = AudioStreamGetPropertyInfo(i_stream_id, 0,
kAudioStreamPropertyPhysicalFormats,
&i_param_size, NULL);
if (err != noErr)
{
ao_msg(MSGT_AO,MSGL_V, "could not get number of streamformats: [%4.4s]\n", (char *)&err);
return CONTROL_FALSE;
}
i_formats = i_param_size / sizeof(AudioStreamBasicDescription);
p_format_list = (AudioStreamBasicDescription *)malloc(i_param_size);
if (p_format_list == NULL)
{
ao_msg(MSGT_AO,MSGL_V, "could not malloc the memory\n" );
return CONTROL_FALSE;
}
err = AudioStreamGetProperty(i_stream_id, 0,
kAudioStreamPropertyPhysicalFormats,
&i_param_size, p_format_list);
if (err != noErr)
{
ao_msg(MSGT_AO,MSGL_V, "could not get the list of streamformats: [%4.4s]\n", (char *)&err);
free(p_format_list);
return CONTROL_FALSE;
}
for (i = 0; i < i_formats; ++i)
{
print_format(MSGL_V, "supported format:", &p_format_list[i]);
if (p_format_list[i].mFormatID == 'IAC3' ||
p_format_list[i].mFormatID == kAudioFormat60958AC3)
b_return = CONTROL_OK;
}
free(p_format_list);
return b_return;
}
/*****************************************************************************
* AudioStreamChangeFormat: Change i_stream_id to change_format
*****************************************************************************/
static int AudioStreamChangeFormat( AudioStreamID i_stream_id, AudioStreamBasicDescription change_format )
{
OSStatus err = noErr;
UInt32 i_param_size = 0;
int i;
static volatile int stream_format_changed;
stream_format_changed = 0;
print_format(MSGL_V, "setting stream format:", &change_format);
/* Install the callback. */
err = AudioStreamAddPropertyListener(i_stream_id, 0,
kAudioStreamPropertyPhysicalFormat,
StreamListener,
(void *)&stream_format_changed);
if (err != noErr)
{
ao_msg(MSGT_AO, MSGL_WARN, "AudioStreamAddPropertyListener failed: [%4.4s]\n", (char *)&err);
return CONTROL_FALSE;
}
/* Change the format. */
err = AudioStreamSetProperty(i_stream_id, 0, 0,
kAudioStreamPropertyPhysicalFormat,
sizeof(AudioStreamBasicDescription),
&change_format);
if (err != noErr)
{
ao_msg(MSGT_AO, MSGL_WARN, "could not set the stream format: [%4.4s]\n", (char *)&err);
return CONTROL_FALSE;
}
/* The AudioStreamSetProperty is not only asynchronious,
* it is also not Atomic, in its behaviour.
* Therefore we check 5 times before we really give up.
* FIXME: failing isn't actually implemented yet. */
for (i = 0; i < 5; ++i)
{
AudioStreamBasicDescription actual_format;
int j;
for (j = 0; !stream_format_changed && j < 50; ++j)
usec_sleep(10000);
if (stream_format_changed)
stream_format_changed = 0;
else
ao_msg(MSGT_AO, MSGL_V, "reached timeout\n" );
i_param_size = sizeof(AudioStreamBasicDescription);
err = AudioStreamGetProperty(i_stream_id, 0,
kAudioStreamPropertyPhysicalFormat,
&i_param_size,
&actual_format);
print_format(MSGL_V, "actual format in use:", &actual_format);
if (actual_format.mSampleRate == change_format.mSampleRate &&
actual_format.mFormatID == change_format.mFormatID &&
actual_format.mFramesPerPacket == change_format.mFramesPerPacket)
{
/* The right format is now active. */
break;
}
/* We need to check again. */
}
/* Removing the property listener. */
err = AudioStreamRemovePropertyListener(i_stream_id, 0,
kAudioStreamPropertyPhysicalFormat,
StreamListener);
if (err != noErr)
{
ao_msg(MSGT_AO, MSGL_WARN, "AudioStreamRemovePropertyListener failed: [%4.4s]\n", (char *)&err);
return CONTROL_FALSE;
}
return CONTROL_TRUE;
}
/*****************************************************************************
* RenderCallbackSPDIF: callback for SPDIF audio output
*****************************************************************************/
static OSStatus RenderCallbackSPDIF( AudioDeviceID inDevice,
const AudioTimeStamp * inNow,
const void * inInputData,
const AudioTimeStamp * inInputTime,
AudioBufferList * outOutputData,
const AudioTimeStamp * inOutputTime,
void * threadGlobals )
{
int amt = buf_used();
int req = outOutputData->mBuffers[ao->i_stream_index].mDataByteSize;
if (amt > req)
amt = req;
if (amt)
read_buffer(ao->b_muted ? NULL : (unsigned char *)outOutputData->mBuffers[ao->i_stream_index].mData, amt);
return noErr;
}
static int play(void* output_samples,int num_bytes,int flags)
{
int wrote, b_digital;
// Check whether we need to reset the digital output stream.
if (ao->b_digital && ao->b_stream_format_changed)
{
ao->b_stream_format_changed = 0;
b_digital = AudioStreamSupportsDigital(ao->i_stream_id);
if (b_digital)
{
/* Current stream support digital format output, let's set it. */
ao_msg(MSGT_AO, MSGL_V, "detected current stream support digital, try to restore digital output...\n");
if (!AudioStreamChangeFormat(ao->i_stream_id, ao->stream_format))
{
ao_msg(MSGT_AO, MSGL_WARN, "restore digital output failed.\n");
}
else
{
ao_msg(MSGT_AO, MSGL_WARN, "restore digital output succeed.\n");
reset();
}
}
else
ao_msg(MSGT_AO, MSGL_V, "detected current stream do not support digital.\n");
}
wrote=write_buffer(output_samples, num_bytes);
audio_resume();
return wrote;
}
/* set variables and buffer to initial state */
static void reset(void)
{
audio_pause();
/* reset ring-buffer state */
ao->buf_read_pos=0;
ao->buf_write_pos=0;
return;
}
/* return available space */
static int get_space(void)
{
return buf_free();
}
/* return delay until audio is played */
static float get_delay(void)
{
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)
{
OSStatus err = noErr;
UInt32 i_param_size = 0;
if (!immed) {
long long timeleft=(1000000LL*buf_used())/ao_data.bps;
ao_msg(MSGT_AO,MSGL_DBG2, "%d bytes left @%d bps (%d usec)\n", buf_used(), ao_data.bps, (int)timeleft);
usec_sleep((int)timeleft);
}
if (!ao->b_digital) {
AudioOutputUnitStop(ao->theOutputUnit);
AudioUnitUninitialize(ao->theOutputUnit);
CloseComponent(ao->theOutputUnit);
}
else {
/* Stop device. */
err = AudioDeviceStop(ao->i_selected_dev,
(AudioDeviceIOProc)RenderCallbackSPDIF);
if (err != noErr)
ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceStop failed: [%4.4s]\n", (char *)&err);
/* Remove IOProc callback. */
err = AudioDeviceRemoveIOProc(ao->i_selected_dev,
(AudioDeviceIOProc)RenderCallbackSPDIF);
if (err != noErr)
ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceRemoveIOProc failed: [%4.4s]\n", (char *)&err);
if (ao->b_revert)
AudioStreamChangeFormat(ao->i_stream_id, ao->sfmt_revert);
if (ao->b_changed_mixing && ao->sfmt_revert.mFormatID != kAudioFormat60958AC3)
{
int b_mix;
Boolean b_writeable;
/* Revert mixable to true if we are allowed to. */
err = AudioDeviceGetPropertyInfo(ao->i_selected_dev, 0, FALSE, kAudioDevicePropertySupportsMixing,
&i_param_size, &b_writeable);
err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE, kAudioDevicePropertySupportsMixing,
&i_param_size, &b_mix);
if (err != noErr && b_writeable)
{
b_mix = 1;
err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE,
kAudioDevicePropertySupportsMixing, i_param_size, &b_mix);
}
if (err != noErr)
ao_msg(MSGT_AO, MSGL_WARN, "failed to set mixmode: [%4.4s]\n", (char *)&err);
}
if (ao->i_hog_pid == getpid())
{
ao->i_hog_pid = -1;
i_param_size = sizeof(ao->i_hog_pid);
err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE,
kAudioDevicePropertyHogMode, i_param_size, &ao->i_hog_pid);
if (err != noErr) ao_msg(MSGT_AO, MSGL_WARN, "Could not release hogmode: [%4.4s]\n", (char *)&err);
}
}
free(ao->buffer);
free(ao);
ao = NULL;
}
/* stop playing, keep buffers (for pause) */
static void audio_pause(void)
{
OSErr err=noErr;
/* Stop callback. */
if (!ao->b_digital)
{
err=AudioOutputUnitStop(ao->theOutputUnit);
if (err != noErr)
ao_msg(MSGT_AO,MSGL_WARN, "AudioOutputUnitStop returned [%4.4s]\n", (char *)&err);
}
else
{
err = AudioDeviceStop(ao->i_selected_dev, (AudioDeviceIOProc)RenderCallbackSPDIF);
if (err != noErr)
ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceStop failed: [%4.4s]\n", (char *)&err);
}
ao->paused = 1;
}
/* resume playing, after audio_pause() */
static void audio_resume(void)
{
OSErr err=noErr;
if (!ao->paused)
return;
/* Start callback. */
if (!ao->b_digital)
{
err = AudioOutputUnitStart(ao->theOutputUnit);
if (err != noErr)
ao_msg(MSGT_AO,MSGL_WARN, "AudioOutputUnitStart returned [%4.4s]\n", (char *)&err);
}
else
{
err = AudioDeviceStart(ao->i_selected_dev, (AudioDeviceIOProc)RenderCallbackSPDIF);
if (err != noErr)
ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceStart failed: [%4.4s]\n", (char *)&err);
}
ao->paused = 0;
}
/*****************************************************************************
* StreamListener
*****************************************************************************/
static OSStatus StreamListener( AudioStreamID inStream,
UInt32 inChannel,
AudioDevicePropertyID inPropertyID,
void * inClientData )
{
switch (inPropertyID)
{
case kAudioStreamPropertyPhysicalFormat:
ao_msg(MSGT_AO, MSGL_V, "got notify kAudioStreamPropertyPhysicalFormat changed.\n");
if (inClientData)
*(volatile int *)inClientData = 1;
default:
break;
}
return noErr;
}
static OSStatus DeviceListener( AudioDeviceID inDevice,
UInt32 inChannel,
Boolean isInput,
AudioDevicePropertyID inPropertyID,
void* inClientData )
{
switch (inPropertyID)
{
case kAudioDevicePropertyDeviceHasChanged:
ao_msg(MSGT_AO, MSGL_WARN, "got notify kAudioDevicePropertyDeviceHasChanged.\n");
ao->b_stream_format_changed = 1;
default:
break;
}
return noErr;
}
Reference in New Issue View Git Blame Copy Permalink