Mercurial > mplayer.hg
changeset 29209:631aac106b7a
Rename macosx audio output driver to coreaudio.
| author | diego |
|---|---|
| date | Mon, 04 May 2009 14:53:47 +0000 |
| parents | 17eb5660b5e3 |
| children | 779281d8cd80 |
| files | AUTHORS DOCS/man/en/mplayer.1 DOCS/tech/MAINTAINERS Makefile configure libao2/ao_coreaudio.c libao2/ao_macosx.c libao2/audio_out.c |
| diffstat | 8 files changed, 1157 insertions(+), 1157 deletions(-) [+] |
line wrap: on
line diff
--- a/AUTHORS Mon May 04 14:49:42 2009 +0000 +++ b/AUTHORS Mon May 04 14:53:47 2009 +0000 @@ -168,7 +168,7 @@ Christiansen, Dan Villiom Podlaski (danchr) <danchr@daimi.au.dk> * timer-darwin.c - * ao_macosx + * CoreAudio audio output driver for Mac OS X * build system cleanups for Darwin Clagg, Jeff (snacky) <snacky@ikaruga.co.uk>
--- a/DOCS/man/en/mplayer.1 Mon May 04 14:49:42 2009 +0000 +++ b/DOCS/man/en/mplayer.1 Mon May 04 14:53:47 2009 +0000 @@ -2786,7 +2786,7 @@ audio output through NAS . .TP -.B macosx (Mac OS X only) +.B coreaudio (Mac OS X only) native Mac OS X audio output driver . .TP
--- a/DOCS/tech/MAINTAINERS Mon May 04 14:49:42 2009 +0000 +++ b/DOCS/tech/MAINTAINERS Mon May 04 14:53:47 2009 +0000 @@ -204,12 +204,12 @@ * ao_alsa5.c - None * ao_alsa.c - Clemens Ladisch * ao_arts.c - None + * ao_coreaudio.c - None * ao_dsound.c - None * ao_dxr2.c - None * ao_esd.c - None * ao_ivtv.c - Benjamin Zores * ao_jack.c - Reimar Döffinger - * ao_macosx.c - None * ao_mpegpes.c - None * ao_nas.c - Tobias Diedrich * ao_null.c - None
--- a/Makefile Mon May 04 14:49:42 2009 +0000 +++ b/Makefile Mon May 04 14:53:47 2009 +0000 @@ -531,7 +531,7 @@ SRCS_MPLAYER-$(ARTS) += libao2/ao_arts.c SRCS_MPLAYER-$(BL) += libvo/vo_bl.c SRCS_MPLAYER-$(CACA) += libvo/vo_caca.c -SRCS_MPLAYER-$(COREAUDIO) += libao2/ao_macosx.c +SRCS_MPLAYER-$(COREAUDIO) += libao2/ao_coreaudio.c SRCS_MPLAYER-$(COREVIDEO) += libvo/vo_macosx.m SRCS_MPLAYER-$(DART) += libao2/ao_dart.c SRCS_MPLAYER-$(DFBMGA) += libvo/vo_dfbmga.c
--- a/configure Mon May 04 14:49:42 2009 +0000 +++ b/configure Mon May 04 14:53:47 2009 +0000 @@ -3815,10 +3815,10 @@ if test "$_coreaudio" = yes ; then extra_ldflags="$extra_ldflags -framework CoreAudio -framework AudioUnit -framework AudioToolbox" def_coreaudio='#define CONFIG_COREAUDIO 1' - _aomodules="macosx $_aomodules" + _aomodules="coreaudio $_aomodules" else def_coreaudio='#undef CONFIG_COREAUDIO' - _noaomodules="macosx $_noaomodules" + _noaomodules="coreaudio $_noaomodules" fi echores $_coreaudio
--- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/libao2/ao_coreaudio.c Mon May 04 14:53:47 2009 +0000 @@ -0,0 +1,1149 @@ +/* + * CoreAudio audio output driver for Mac OS X + * + * 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" +#include "libavutil/fifo.h" + +static const ao_info_t info = + { + "Darwin/Mac OS X native audio output", + "coreaudio", + "Timothy J. Wood & Dan Christiansen & Chris Roccati", + "" + }; + +LIBAO_EXTERN(coreaudio) + +/* Prefix for all mp_msg() calls */ +#define ao_msg(a, b, c...) mp_msg(a, b, "AO: [coreaudio] " c) + +typedef struct ao_coreaudio_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 */ + AVFifoBuffer *buffer; + unsigned int buffer_len; ///< must always be num_chunks * chunk_size + unsigned int num_chunks; + unsigned int chunk_size; +} ao_coreaudio_t; + +static ao_coreaudio_t *ao = NULL; + +/** + * \brief add data to ringbuffer + */ +static int write_buffer(unsigned char* data, int len){ + int free = ao->buffer_len - av_fifo_size(ao->buffer); + if (len > free) len = free; + return av_fifo_generic_write(ao->buffer, data, len, NULL); +} + +/** + * \brief remove data from ringbuffer + */ +static int read_buffer(unsigned char* data,int len){ + int buffered = av_fifo_size(ao->buffer); + if (len > buffered) len = buffered; + return av_fifo_generic_read(ao->buffer, data, len, NULL); +} + +OSStatus theRenderProc(void *inRefCon, AudioUnitRenderActionFlags *inActionFlags, const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, UInt32 inNumFrames, AudioBufferList *ioData) +{ +int amt=av_fifo_size(ao->buffer); +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_coreaudio_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 * ao->chunk_size; + ao->buffer = av_fifo_alloc(ao->buffer_len); + + 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: + av_fifo_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 * ao->chunk_size; + ao->buffer = av_fifo_alloc(ao->buffer_len); + + 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); + } + av_fifo_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 = av_fifo_size(ao->buffer); + 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(); + av_fifo_reset(ao->buffer); +} + + +/* return available space */ +static int get_space(void) +{ + return ao->buffer_len - av_fifo_size(ao->buffer); +} + + +/* return delay until audio is played */ +static float get_delay(void) +{ + // inaccurate, should also contain the data buffered e.g. by the OS + return (float)av_fifo_size(ao->buffer)/(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*av_fifo_size(ao->buffer))/ao_data.bps; + ao_msg(MSGT_AO,MSGL_DBG2, "%d bytes left @%d bps (%d usec)\n", av_fifo_size(ao->buffer), 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); + } + } + + av_fifo_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; +}
--- a/libao2/ao_macosx.c Mon May 04 14:49:42 2009 +0000 +++ /dev/null Thu Jan 01 00:00:00 1970 +0000 @@ -1,1149 +0,0 @@ -/* - * 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" -#include "libavutil/fifo.h" - -static const 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 */ - AVFifoBuffer *buffer; - unsigned int buffer_len; ///< must always be num_chunks * chunk_size - unsigned int num_chunks; - unsigned int chunk_size; -} ao_macosx_t; - -static ao_macosx_t *ao = NULL; - -/** - * \brief add data to ringbuffer - */ -static int write_buffer(unsigned char* data, int len){ - int free = ao->buffer_len - av_fifo_size(ao->buffer); - if (len > free) len = free; - return av_fifo_generic_write(ao->buffer, data, len, NULL); -} - -/** - * \brief remove data from ringbuffer - */ -static int read_buffer(unsigned char* data,int len){ - int buffered = av_fifo_size(ao->buffer); - if (len > buffered) len = buffered; - return av_fifo_generic_read(ao->buffer, data, len, NULL); -} - -OSStatus theRenderProc(void *inRefCon, AudioUnitRenderActionFlags *inActionFlags, const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, UInt32 inNumFrames, AudioBufferList *ioData) -{ -int amt=av_fifo_size(ao->buffer); -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 * ao->chunk_size; - ao->buffer = av_fifo_alloc(ao->buffer_len); - - 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: - av_fifo_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 * ao->chunk_size; - ao->buffer = av_fifo_alloc(ao->buffer_len); - - 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); - } - av_fifo_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 = av_fifo_size(ao->buffer); - 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(); - av_fifo_reset(ao->buffer); -} - - -/* return available space */ -static int get_space(void) -{ - return ao->buffer_len - av_fifo_size(ao->buffer); -} - - -/* return delay until audio is played */ -static float get_delay(void) -{ - // inaccurate, should also contain the data buffered e.g. by the OS - return (float)av_fifo_size(ao->buffer)/(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*av_fifo_size(ao->buffer))/ao_data.bps; - ao_msg(MSGT_AO,MSGL_DBG2, "%d bytes left @%d bps (%d usec)\n", av_fifo_size(ao->buffer), 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); - } - } - - av_fifo_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; -}
--- a/libao2/audio_out.c Mon May 04 14:49:42 2009 +0000 +++ b/libao2/audio_out.c Mon May 04 14:53:47 2009 +0000 @@ -32,7 +32,7 @@ char *ao_subdevice = NULL; extern const ao_functions_t audio_out_oss; -extern const ao_functions_t audio_out_macosx; +extern const ao_functions_t audio_out_coreaudio; extern const ao_functions_t audio_out_arts; extern const ao_functions_t audio_out_esd; extern const ao_functions_t audio_out_pulse; @@ -68,7 +68,7 @@ &audio_out_dart, #endif #ifdef CONFIG_COREAUDIO - &audio_out_macosx, + &audio_out_coreaudio, #endif #ifdef CONFIG_OSS_AUDIO &audio_out_oss,