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Commit c6d39ca3 authored by jehan's avatar jehan
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add audiot unit base iPhone sound integration

parent 6c8d5cea
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Showing with 678 additions and 2 deletions
configure.ac
View file @ c6d39ca3
......@@ -369,6 +369,22 @@ fi
AM_CONDITIONAL(BUILD_MACAQSND, test x$macaqsnd = xtrue)
AC_ARG_ENABLE(iounit,
[ --enable-iounit Turn on native apple iounit support (default=no)],
[case "${enableval}" in
yes) iounit=true ;;
no) iounit=false ;;
*) AC_MSG_ERROR(bad value ${enableval} for --enable-iounit) ;;
esac],[iounit=false])
if test "$iounit" = "true"; then
AC_DEFINE(__MACIOUNIT_ENABLED__,1,[defined if native apple iounit support is available])
iounit_enabled=true
found_sound=yes
fi
AM_CONDITIONAL(BUILD_MACIOUNIT, test x$iounit_enabled = xtrue)
AC_ARG_ENABLE(jack,
[ --disable-jack Disable jack support],
[case "${enableval}" in
......
include/mediastreamer2/allfilters.h
View file @ c6d39ca3
......@@ -97,7 +97,11 @@ typedef enum MSFilterId{
MS_ITC_SINK_ID,
MS_ITC_SOURCE_ID,
MS_EXT_DISPLAY_ID,
MS_H264_DEC_ID
MS_H264_DEC_ID,
MS_IOUNIT_READ_ID,
MS_IOUNIT_WRITE_ID,
MS_ANDROID_SOUND_READ_ID,
MS_ANDROID_SOUND_WRITE_ID
} MSFilterId;
......
src/Makefile.am
View file @ c6d39ca3
......@@ -88,6 +88,10 @@ if BUILD_MACSND
libmediastreamer_la_SOURCES+=macsnd.c
endif
if BUILD_MACIOUNIT
libmediastreamer_la_SOURCES+=msiounit.c
endif
if BUILD_MACAQSND
libmediastreamer_la_SOURCES+=aqsnd.c
endif
......
src/mscommon.c
View file @ c6d39ca3
......@@ -435,6 +435,10 @@ extern MSSndCardDesc aq_card_desc;
extern MSSndCardDesc pulse_card_desc;
#endif
#ifdef __MACIOUNIT_ENABLED__
extern MSSndCardDesc au_card_desc;
#endif
static MSSndCardDesc * ms_snd_card_descs[]={
#ifdef __ALSA_ENABLED__
&alsa_card_desc,
......@@ -463,7 +467,10 @@ static MSSndCardDesc * ms_snd_card_descs[]={
#ifdef __PULSEAUDIO_ENABLED__
&pulse_card_desc,
#endif
NULL
#ifdef __MACIOUNIT_ENABLED__
&au_card_desc,
#endif
NULL
};
#ifdef VIDEO_ENABLED
......
src/msiounit.c 0 → 100644
View file @ c6d39ca3
/*
* auiosnd.c -I/O unit Media plugin for Linphone-
*
*
* Copyright (C) 2009 Belledonne Comunications, Grenoble, France
*
* This program 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.
*
* This program 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 Library General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
*/
#include <AudioToolbox/AudioToolbox.h>
#include "mediastreamer2/mssndcard.h"
#include "mediastreamer2/msfilter.h"
#define PREFERRED_HW_SAMPLE_RATE 44100
#if 0
#undef ms_debug
#define ms_debug ms_message
#endif
static MSFilter *ms_au_read_new(MSSndCard *card);
static MSFilter *ms_au_write_new(MSSndCard *card);
typedef struct AUData_t{
AudioUnit io_unit;
unsigned int rate;
unsigned int bits;
unsigned int nchannels;
ms_mutex_t mutex;
queue_t rq;
MSBufferizer *bufferizer;
bool_t started;
bool_t read_started;
bool_t write_started;
AudioTimeStamp readTimeStamp;
unsigned int n_lost_frame;
bool_t io_unit_must_be_started;
} AUData;
/*
mediastreamer2 function
*/
static void au_set_level(MSSndCard *card, MSSndCardMixerElem e, int percent)
{
}
static int au_get_level(MSSndCard *card, MSSndCardMixerElem e)
{
return 0;
}
static void au_set_source(MSSndCard *card, MSSndCardCapture source)
{
}
static void au_init(MSSndCard *card){
ms_debug("au_init");
AUData *d=ms_new(AUData,1);
//OSStatus auresult;
d->bits=16;
d->rate=PREFERRED_HW_SAMPLE_RATE;
d->nchannels=1;
d->read_started=FALSE;
d->write_started=FALSE;
d->bufferizer=ms_bufferizer_new();
d->n_lost_frame=0;
d->started=FALSE;
d->io_unit_must_be_started=FALSE;
qinit(&d->rq);
d->readTimeStamp.mSampleTime=-1;
ms_mutex_init(&d->mutex,NULL);
card->data=d;
}
static void au_uninit(MSSndCard *card){
AUData *d=(AUData*)card->data;
ms_bufferizer_destroy(d->bufferizer);
ms_mutex_destroy(&d->mutex);
ms_free(d);
}
static void au_detect(MSSndCardManager *m);
static MSSndCard *au_duplicate(MSSndCard *obj);
MSSndCardDesc au_card_desc={
.driver_type="AU",
.detect=au_detect,
.init=au_init,
.set_level=au_set_level,
.get_level=au_get_level,
.set_capture=au_set_source,
.set_control=NULL,
.get_control=NULL,
.create_reader=ms_au_read_new,
.create_writer=ms_au_write_new,
.uninit=au_uninit,
.duplicate=au_duplicate
};
static MSSndCard *au_duplicate(MSSndCard *obj){
MSSndCard *card=ms_snd_card_new(&au_card_desc);
card->name=ms_strdup(obj->name);
return card;
}
#define check_auresult(au,method) \
if (au!=0) ms_error("AudioUnit error for %s: ret=%i",method,au)
static void au_interuption_listener(void* inClientData, UInt32 inInterruptionState) {
if (((MSSndCard*)inClientData)->data == NULL) return;
AUData *d=(AUData*)(((MSSndCard*)inClientData)->data);
if (d->started == FALSE) {
//nothing to do
return;
}
switch (inInterruptionState) {
case kAudioSessionBeginInterruption:
ms_message ("IO unit interruption begin");
AudioOutputUnitStop(d->io_unit);
break;
case kAudioSessionEndInterruption:
// make sure we are again the active session
ms_message ("IO unit interruption end");
OSStatus auresult = AudioSessionSetActive(true);
check_auresult(auresult,"AudioSessionSetActive");
d->io_unit_must_be_started=TRUE;
break;
default:
ms_warning ("unexpected interuption %i",inInterruptionState);
}
}
static MSSndCard *au_card_new(){
MSSndCard *card=ms_snd_card_new(&au_card_desc);
card->name=ms_strdup("Audio Unit");
OSStatus auresult = AudioSessionInitialize(NULL, NULL, au_interuption_listener, card);
check_auresult(auresult,"AudioSessionInitialize");
return card;
}
static void au_detect(MSSndCardManager *m){
ms_debug("au_detect");
MSSndCard *card=au_card_new();
ms_snd_card_manager_add_card(m,card);
}
static OSStatus au_read_cb (
void *inRefCon,
AudioUnitRenderActionFlags *ioActionFlags,
const AudioTimeStamp *inTimeStamp,
UInt32 inBusNumber,
UInt32 inNumberFrames,
AudioBufferList *ioData
)
{
AUData *d=(AUData*)inRefCon;
if (d->readTimeStamp.mSampleTime <0) {
d->readTimeStamp=*inTimeStamp;
}
OSStatus err=0;
mblk_t * rm=NULL;
if (d->read_started) {
rm=allocb(ioData->mBuffers[0].mDataByteSize,0);
ioData->mBuffers[0].mData=rm->b_wptr;
}
err = AudioUnitRender(d->io_unit, ioActionFlags, &d->readTimeStamp, inBusNumber,inNumberFrames, ioData);
if (d->read_started){
if (err == 0) {
rm->b_wptr += ioData->mBuffers[0].mDataByteSize;
ms_mutex_lock(&d->mutex);
putq(&d->rq,rm);
ms_mutex_unlock(&d->mutex);
d->readTimeStamp.mSampleTime+=ioData->mBuffers[0].mDataByteSize/(d->bits/2);
}else ms_warning("AudioUnitRender() failed: %i",err);
}
return err;
}
/********************read/write cb******************/
static OSStatus au_render_cb (
void *inRefCon,
AudioUnitRenderActionFlags *ioActionFlags,
const AudioTimeStamp *inTimeStamp,
UInt32 inBusNumber,
UInt32 inNumberFrames,
AudioBufferList *ioData
) {
ms_debug("render cb");
AUData *d=(AUData*)inRefCon;
if (d->write_started == TRUE) {
ms_mutex_lock(&d->mutex);
if(ms_bufferizer_get_avail(d->bufferizer) >= inNumberFrames*d->bits/8) {
ms_bufferizer_read(d->bufferizer, ioData->mBuffers[0].mData, inNumberFrames*d->bits/8);
if (ms_bufferizer_get_avail(d->bufferizer) >10*inNumberFrames*d->bits/8) {
ms_debug("we are late, bufferizer sise is %i bytes in framezize is %i bytes",ms_bufferizer_get_avail(d->bufferizer),inNumberFrames*d->bits/8);
ms_bufferizer_flush(d->bufferizer);
}
ioData->mBuffers[0].mDataByteSize=inNumberFrames*d->bits/8;
ms_mutex_unlock(&d->mutex);
} else {
ms_mutex_unlock(&d->mutex);
memset(ioData->mBuffers[0].mData, 0,ioData->mBuffers[0].mDataByteSize);
ms_debug("nothing to write, pushing silences, bufferizer sise is %i bytes in framezize is %i bytes mDataByteSize %i"
,ms_bufferizer_get_avail(d->bufferizer)
,inNumberFrames*d->bits/8
,ioData->mBuffers[0].mDataByteSize);
d->n_lost_frame+=inNumberFrames;
}
AudioBufferList readAudioBufferList;
readAudioBufferList.mBuffers[0].mDataByteSize=inNumberFrames*d->bits/8;
readAudioBufferList.mNumberBuffers=1;
readAudioBufferList.mBuffers[0].mData=NULL;
readAudioBufferList.mBuffers[0].mNumberChannels=d->nchannels;
AudioUnitElement inputBus = 1;
au_read_cb(d, ioActionFlags, inTimeStamp, inputBus, inNumberFrames, &readAudioBufferList);
}
return 0;
}
/****************config**************/
static void au_configure(AUData *d) {
AudioStreamBasicDescription audioFormat;
AudioComponentDescription au_description;
AudioComponent foundComponent;
OSStatus auresult;
if (d->started == TRUE) {
//nothing to do
return;
}
auresult = AudioSessionSetActive(true);
check_auresult(auresult,"AudioSessionSetActive");
UInt32 audioCategory = kAudioSessionCategory_PlayAndRecord;
auresult =AudioSessionSetProperty(kAudioSessionProperty_AudioCategory, sizeof(audioCategory), &audioCategory);
check_auresult(auresult,"Configuring audio session for play/record");
au_description.componentType = kAudioUnitType_Output;
au_description.componentSubType = kAudioUnitSubType_VoiceProcessingIO;
au_description.componentManufacturer = kAudioUnitManufacturer_Apple;
au_description.componentFlags = 0;
au_description.componentFlagsMask = 0;
foundComponent = AudioComponentFindNext (NULL,&au_description);
auresult=AudioComponentInstanceNew (foundComponent, &d->io_unit);
check_auresult(auresult,"AudioComponentInstanceNew");
audioFormat.mSampleRate = d->rate;
audioFormat.mFormatID = kAudioFormatLinearPCM;
audioFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked;
audioFormat.mFramesPerPacket = 1;
audioFormat.mChannelsPerFrame = d->nchannels;
audioFormat.mBitsPerChannel = d->bits;
audioFormat.mBytesPerPacket = d->bits / 8;
audioFormat.mBytesPerFrame = d->nchannels * d->bits / 8;
UInt32 doSetProperty = 1;
UInt32 doNotSetProperty = 0;
AudioUnitElement outputBus = 0;
AudioUnitElement inputBus = 1;
auresult=AudioUnitUninitialize (d->io_unit);
check_auresult(auresult,"AudioUnitUninitialize");
//read
auresult=AudioUnitSetProperty (
d->io_unit,
kAudioOutputUnitProperty_EnableIO,
kAudioUnitScope_Input ,
inputBus,
&doSetProperty,
sizeof (doSetProperty)
);
check_auresult(auresult,"kAudioOutputUnitProperty_EnableIO,kAudioUnitScope_Input");
auresult=AudioUnitSetProperty (
d->io_unit,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Output,
inputBus,
&audioFormat,
sizeof (audioFormat)
);
check_auresult(auresult,"kAudioUnitProperty_StreamFormat,kAudioUnitScope_Output");
//write
//enable output bus
auresult =AudioUnitSetProperty (
d->io_unit,
kAudioOutputUnitProperty_EnableIO,
kAudioUnitScope_Output ,
outputBus,
&doSetProperty,
sizeof (doSetProperty)
);
check_auresult(auresult,"kAudioOutputUnitProperty_EnableIO,kAudioUnitScope_Output");
//setup stream format
auresult=AudioUnitSetProperty (
d->io_unit,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Input,
outputBus,
&audioFormat,
sizeof (audioFormat)
);
check_auresult(auresult,"kAudioUnitProperty_StreamFormat,kAudioUnitScope_Input");
//disable unit buffer allocation
auresult=AudioUnitSetProperty (
d->io_unit,
kAudioUnitProperty_ShouldAllocateBuffer,
kAudioUnitScope_Output,
outputBus,
&doNotSetProperty,
sizeof (doNotSetProperty)
);
check_auresult(auresult,"kAudioUnitProperty_ShouldAllocateBuffer,kAudioUnitScope_Output");
AURenderCallbackStruct renderCallbackStruct;
renderCallbackStruct.inputProc = au_render_cb;
renderCallbackStruct.inputProcRefCon = d;
auresult=AudioUnitSetProperty (
d->io_unit,
kAudioUnitProperty_SetRenderCallback,
kAudioUnitScope_Input,
outputBus,
&renderCallbackStruct,
sizeof (renderCallbackStruct)
);
check_auresult(auresult,"kAudioUnitProperty_SetRenderCallback,kAudioUnitScope_Input");
const Float64 preferredSampleRate = d->rate;//PREFERRED_HW_SAMPLE_RATE; /*optimum to minimize delay, must put a software resampler to deal with 8khz*/
auresult=AudioSessionSetProperty(kAudioSessionProperty_PreferredHardwareSampleRate
,sizeof(preferredSampleRate)
, &preferredSampleRate);
check_auresult(auresult,"kAudioSessionProperty_PreferredHardwareSampleRate");
//start io unit
auresult=AudioUnitInitialize (d->io_unit);
Float32 preferredBufferSize;
switch (d->rate) {
case 11025:
case 22050:
preferredBufferSize= .020;
break;
default:
preferredBufferSize= .015;
}
auresult=AudioSessionSetProperty(kAudioSessionProperty_PreferredHardwareIOBufferDuration
,sizeof(preferredBufferSize)
, &preferredBufferSize);
if (auresult != 0) ms_message("kAudioSessionProperty_PreferredHardwareIOBufferDuration returns %i ",auresult);
Float64 delay;
UInt32 delaySize = sizeof(delay);
auresult=AudioUnitGetProperty(d->io_unit
,kAudioUnitProperty_Latency
, kAudioUnitScope_Global
, 0
, &delay
, &delaySize);
UInt32 quality;
UInt32 qualitySize = sizeof(quality);
auresult=AudioUnitGetProperty(d->io_unit
,kAudioUnitProperty_RenderQuality
, kAudioUnitScope_Global
, 0
, &quality
, &qualitySize);
ms_message("I/O unit latency [%f], quality [%i]",delay,quality);
Float32 hwoutputlatency;
UInt32 hwoutputlatencySize=sizeof(hwoutputlatency);
auresult=AudioSessionGetProperty(kAudioSessionProperty_CurrentHardwareOutputLatency
,&hwoutputlatencySize
, &hwoutputlatency);
Float32 hwinputlatency;
UInt32 hwinputlatencySize=sizeof(hwoutputlatency);
auresult=AudioSessionGetProperty(kAudioSessionProperty_CurrentHardwareInputLatency
,&hwinputlatencySize
, &hwinputlatency);
Float32 hwiobuf;
UInt32 hwiobufSize=sizeof(hwiobuf);
auresult=AudioSessionGetProperty(kAudioSessionProperty_CurrentHardwareIOBufferDuration
,&hwiobufSize
, &hwiobuf);
Float64 hwsamplerate;
UInt32 hwsamplerateSize=sizeof(hwsamplerate);
auresult=AudioSessionGetProperty(kAudioSessionProperty_CurrentHardwareSampleRate
,&hwsamplerateSize
,&hwsamplerate);
ms_message("current hw output latency [%f] input [%f] iobuf[%f] sample rate [%f]",hwoutputlatency,hwinputlatency,hwiobuf,hwsamplerate);
auresult=AudioOutputUnitStart(d->io_unit);
check_auresult(auresult,"AudioOutputUnitStart");
d->started=TRUE;
return;
}
static void au_configure_read(AUData *d) {
d->read_started=TRUE;
au_configure(d);
ms_mutex_lock(&d->mutex);
flushq(&d->rq,0);
ms_mutex_unlock(&d->mutex);
}
static void au_configure_write(AUData *d) {
d->write_started=TRUE;
au_configure(d);
ms_mutex_lock(&d->mutex);
ms_bufferizer_flush(d->bufferizer);
ms_mutex_unlock(&d->mutex);
}
static void au_unconfigure(AUData *d) {
if (d->write_started==FALSE && d->read_started==FALSE) {
AudioUnitUninitialize(d->io_unit);
AudioOutputUnitStop(d->io_unit);
AudioComponentInstanceDispose (d->io_unit);
d->started=FALSE;
}
}
static void au_unconfigure_read(AUData *d){
if(d->read_started == TRUE) {
d->read_started=FALSE;
AudioUnitElement inputBus = 1;
AudioUnitReset(d->io_unit,
kAudioUnitScope_Global,
inputBus);
}
au_unconfigure(d);
ms_mutex_lock(&d->mutex);
flushq(&d->rq,0);
ms_mutex_unlock(&d->mutex);
}
static void au_unconfigure_write(AUData *d){
if(d->write_started == TRUE) {
d->write_started=FALSE;
AudioUnitElement outputBus = 0;
AudioUnitReset(d->io_unit,
kAudioUnitScope_Global,
outputBus);
}
ms_message("[%i] frames of silence inserted for [%i] ms",d->n_lost_frame,(d->n_lost_frame*1000)/d->rate);
au_unconfigure(d);
ms_mutex_lock(&d->mutex);
ms_bufferizer_flush(d->bufferizer);
ms_mutex_unlock(&d->mutex);
}
/***********************************read function********************/
static void au_read_preprocess(MSFilter *f){
ms_debug("au_read_preprocess");
AUData *d=(AUData*)((MSSndCard*)f->data)->data;
au_configure_read(d);
}
static void au_read_postprocess(MSFilter *f){
AUData *d=(AUData*)((MSSndCard*)f->data)->data;
au_unconfigure_read(d);
}
static void au_read_process(MSFilter *f){
AUData *d=(AUData*)((MSSndCard*)f->data)->data;
mblk_t *m;
struct timeval tv;
if (d->io_unit_must_be_started) {
gettimeofday(&tv, 0);
if ((tv.tv_usec % 10) == 0) { /*more or less every 100ms*/
if (AudioOutputUnitStart(d->io_unit) == 0) {
d->io_unit_must_be_started=FALSE;
};
}
}
do {
ms_mutex_lock(&d->mutex);
m=getq(&d->rq);
ms_mutex_unlock(&d->mutex);
if (m != NULL) ms_queue_put(f->outputs[0],m);
}while(m!=NULL);
}
/***********************************write function********************/
static void au_write_preprocess(MSFilter *f){
ms_debug("au_write_preprocess");
AUData *d=(AUData*)((MSSndCard*)f->data)->data;
au_configure_write(d);
}
static void au_write_postprocess(MSFilter *f){
ms_debug("au_write_postprocess");
AUData *d=(AUData*)((MSSndCard*)f->data)->data;
au_unconfigure_write(d);
}
static void au_write_process(MSFilter *f){
ms_debug("au_write_process");
mblk_t *m;
AUData *d=(AUData*)((MSSndCard*)f->data)->data;
while((m=ms_queue_get(f->inputs[0]))!=NULL){
ms_mutex_lock(&d->mutex);
ms_bufferizer_put(d->bufferizer,m);
ms_mutex_unlock(&d->mutex);
}