Mercurial > mplayer.hg
view libao2/ao_coreaudio.c @ 33928:27b1c4df84bd
Remove pointless pointer indirection for shader program strings.
author | reimar |
---|---|
date | Fri, 26 Aug 2011 19:35:46 +0000 |
parents | 5533ffce9c6b |
children | 8a80fb185741 |
line wrap: on
line source
/* * 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" #include "subopt-helper.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) #if MAC_OS_X_VERSION_MAX_ALLOWED <= 1040 /* AudioDeviceIOProcID does not exist in Mac OS X 10.4. We can emulate * this by using AudioDeviceAddIOProc() and AudioDeviceRemoveIOProc(). */ #define AudioDeviceIOProcID AudioDeviceIOProc #define AudioDeviceDestroyIOProcID AudioDeviceRemoveIOProc static OSStatus AudioDeviceCreateIOProcID(AudioDeviceID dev, AudioDeviceIOProc proc, void *data, AudioDeviceIOProcID *procid) { *procid = proc; return AudioDeviceAddIOProc(dev, proc, data); } #endif 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? */ AudioDeviceIOProcID renderCallback; /* Render callback used for SPDIF */ /* 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; av_fifo_generic_read(ao->buffer, data, len, NULL); return len; } static 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 %"PRIu32"bit [%c%c%c%c][%"PRIu32"][%"PRIu32"][%"PRIu32"][%"PRIu32"][%"PRIu32"] %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 OSStatus GetAudioProperty(AudioObjectID id, AudioObjectPropertySelector selector, UInt32 outSize, void *outData) { AudioObjectPropertyAddress property_address; property_address.mSelector = selector; property_address.mScope = kAudioObjectPropertyScopeGlobal; property_address.mElement = kAudioObjectPropertyElementMaster; return AudioObjectGetPropertyData(id, &property_address, 0, NULL, &outSize, outData); } static UInt32 GetAudioPropertyArray(AudioObjectID id, AudioObjectPropertySelector selector, AudioObjectPropertyScope scope, void **outData) { OSStatus err; AudioObjectPropertyAddress property_address; UInt32 i_param_size; property_address.mSelector = selector; property_address.mScope = scope; property_address.mElement = kAudioObjectPropertyElementMaster; err = AudioObjectGetPropertyDataSize(id, &property_address, 0, NULL, &i_param_size); if (err != noErr) return 0; *outData = malloc(i_param_size); err = AudioObjectGetPropertyData(id, &property_address, 0, NULL, &i_param_size, *outData); if (err != noErr) { free(*outData); return 0; } return i_param_size; } static UInt32 GetGlobalAudioPropertyArray(AudioObjectID id, AudioObjectPropertySelector selector, void **outData) { return GetAudioPropertyArray(id, selector, kAudioObjectPropertyScopeGlobal, outData); } static OSStatus GetAudioPropertyString(AudioObjectID id, AudioObjectPropertySelector selector, char **outData) { OSStatus err; AudioObjectPropertyAddress property_address; UInt32 i_param_size; CFStringRef string; CFIndex string_length; property_address.mSelector = selector; property_address.mScope = kAudioObjectPropertyScopeGlobal; property_address.mElement = kAudioObjectPropertyElementMaster; i_param_size = sizeof(CFStringRef); err = AudioObjectGetPropertyData(id, &property_address, 0, NULL, &i_param_size, &string); if (err != noErr) return err; string_length = CFStringGetMaximumSizeForEncoding(CFStringGetLength(string), kCFStringEncodingASCII); *outData = malloc(string_length + 1); CFStringGetCString(string, *outData, string_length + 1, kCFStringEncodingASCII); CFRelease(string); return err; } static OSStatus SetAudioProperty(AudioObjectID id, AudioObjectPropertySelector selector, UInt32 inDataSize, void *inData) { AudioObjectPropertyAddress property_address; property_address.mSelector = selector; property_address.mScope = kAudioObjectPropertyScopeGlobal; property_address.mElement = kAudioObjectPropertyElementMaster; return AudioObjectSetPropertyData(id, &property_address, 0, NULL, inDataSize, inData); } static Boolean IsAudioPropertySettable(AudioObjectID id, AudioObjectPropertySelector selector, Boolean *outData) { AudioObjectPropertyAddress property_address; property_address.mSelector = selector; property_address.mScope = kAudioObjectPropertyScopeGlobal; property_address.mElement = kAudioObjectPropertyElementMaster; return AudioObjectIsPropertySettable(id, &property_address, outData); } static int AudioDeviceSupportsDigital( AudioDeviceID i_dev_id ); static int AudioStreamSupportsDigital( AudioStreamID i_stream_id ); static int OpenSPDIF(void); 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( AudioObjectID inObjectID, UInt32 inNumberAddresses, const AudioObjectPropertyAddress inAddresses[], void *inClientData ); static OSStatus DeviceListener( AudioObjectID inObjectID, UInt32 inNumberAddresses, const AudioObjectPropertyAddress inAddresses[], void *inClientData ); static void print_help(void) { OSStatus err; UInt32 i_param_size; int num_devices; AudioDeviceID *devids; char *device_name; mp_msg(MSGT_AO, MSGL_FATAL, "\n-ao coreaudio commandline help:\n" "Example: mplayer -ao coreaudio:device_id=266\n" " open Core Audio with output device ID 266.\n" "\nOptions:\n" " device_id\n" " ID of output device to use (0 = default device)\n" " help\n" " This help including list of available devices.\n" "\n" "Available output devices:\n"); i_param_size = GetGlobalAudioPropertyArray(kAudioObjectSystemObject, kAudioHardwarePropertyDevices, (void **)&devids); if (!i_param_size) { mp_msg(MSGT_AO, MSGL_FATAL, "Failed to get list of output devices.\n"); return; } num_devices = i_param_size / sizeof(AudioDeviceID); for (int i = 0; i < num_devices; ++i) { err = GetAudioPropertyString(devids[i], kAudioObjectPropertyName, &device_name); if (err == noErr) { mp_msg(MSGT_AO, MSGL_FATAL, "%s (id: %"PRIu32")\n", device_name, devids[i]); free(device_name); } else mp_msg(MSGT_AO, MSGL_FATAL, "Unknown (id: %"PRIu32")\n", devids[i]); } mp_msg(MSGT_AO, MSGL_FATAL, "\n"); free(devids); } static int init(int rate,int channels,int format,int flags) { AudioStreamBasicDescription inDesc; ComponentDescription desc; Component comp; AURenderCallbackStruct renderCallback; OSStatus err; UInt32 size, maxFrames, b_alive; char *psz_name; AudioDeviceID devid_def = 0; int device_id, display_help = 0; const opt_t subopts[] = { {"device_id", OPT_ARG_INT, &device_id, NULL}, {"help", OPT_ARG_BOOL, &display_help, NULL}, {NULL} }; // set defaults device_id = 0; if (subopt_parse(ao_subdevice, subopts) != 0 || display_help) { print_help(); if (!display_help) return 0; } 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; if (device_id == 0) { /* Find the ID of the default Device. */ err = GetAudioProperty(kAudioObjectSystemObject, kAudioHardwarePropertyDefaultOutputDevice, sizeof(UInt32), &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; } } else { devid_def = device_id; } /* Retrieve the name of the device. */ err = GetAudioPropertyString(devid_def, kAudioObjectPropertyName, &psz_name); if (err != noErr) { ao_msg(MSGT_AO, MSGL_WARN, "could not get default audio device name: [%4.4s]\n", (char *)&err); goto err_out; } ao_msg(MSGT_AO,MSGL_V, "got audio output device ID: %"PRIu32" Name: %s\n", devid_def, psz_name ); /* Probe whether device support S/PDIF stream output if input is AC3 stream. */ if (AF_FORMAT_IS_AC3(format)) { if (AudioDeviceSupportsDigital(devid_def)) { ao->b_supports_digital = 1; } ao_msg(MSGT_AO, MSGL_V, "probe default audio output device about support for digital s/pdif output: %d\n", ao->b_supports_digital ); } free(psz_name); // Save selected device id ao->i_selected_dev = devid_def; // Build Description for the input format inDesc.mSampleRate=rate; inDesc.mFormatID=ao->b_supports_digital ? kAudioFormat60958AC3 : kAudioFormatLinearPCM; inDesc.mChannelsPerFrame=channels; inDesc.mBitsPerChannel=af_fmt2bits(format); 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_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; err = GetAudioProperty(ao->i_selected_dev, kAudioDevicePropertyDeviceIsAlive, sizeof(UInt32), &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. */ err = GetAudioProperty(ao->i_selected_dev, kAudioDevicePropertyHogMode, sizeof(pid_t), &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 = (device_id == 0) ? kAudioUnitSubType_DefaultOutput : kAudioUnitSubType_HALOutput; 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; } //Set the Current Device to the Default Output Unit. err = AudioUnitSetProperty(ao->theOutputUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0, &ao->i_selected_dev, sizeof(ao->i_selected_dev)); 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(void) { OSStatus err = noErr; UInt32 i_param_size, b_mix = 0; Boolean b_writeable = 0; AudioStreamID *p_streams = NULL; int i, i_streams = 0; AudioObjectPropertyAddress property_address; /* Start doing the SPDIF setup process. */ ao->b_digital = 1; /* Hog the device. */ ao->i_hog_pid = getpid() ; err = SetAudioProperty(ao->i_selected_dev, kAudioDevicePropertyHogMode, sizeof(ao->i_hog_pid), &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 = IsAudioPropertySettable(ao->i_selected_dev, kAudioDevicePropertySupportsMixing, &b_writeable); err = GetAudioProperty(ao->i_selected_dev, kAudioDevicePropertySupportsMixing, sizeof(UInt32), &b_mix); if (err != noErr && b_writeable) { b_mix = 0; err = SetAudioProperty(ao->i_selected_dev, kAudioDevicePropertySupportsMixing, sizeof(UInt32), &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. */ i_param_size = GetAudioPropertyArray(ao->i_selected_dev, kAudioDevicePropertyStreams, kAudioDevicePropertyScopeOutput, (void **)&p_streams); if (!i_param_size) { ao_msg(MSGT_AO, MSGL_WARN, "could not get number of streams.\n"); goto err_out; } i_streams = i_param_size / sizeof(AudioStreamID); 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; i_param_size = GetGlobalAudioPropertyArray(p_streams[i], kAudioStreamPropertyPhysicalFormats, (void **)&p_format_list); if (!i_param_size) { ao_msg(MSGT_AO, MSGL_WARN, "Could not get number of stream formats.\n"); continue; } i_formats = i_param_size / sizeof(AudioStreamBasicDescription); /* 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. */ err = GetAudioProperty(ao->i_stream_id, kAudioStreamPropertyPhysicalFormat, sizeof(ao->sfmt_revert), &ao->sfmt_revert); if (err != noErr) { ao_msg(MSGT_AO, MSGL_WARN, "Could not retrieve the original stream format: [%4.4s]\n", (char *)&err); 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). */ } free(p_format_list); } free(p_streams); if (ao->i_stream_index < 0) { ao_msg(MSGT_AO, MSGL_WARN, "Cannot 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; property_address.mSelector = kAudioDevicePropertyDeviceHasChanged; property_address.mScope = kAudioObjectPropertyScopeGlobal; property_address.mElement = kAudioObjectPropertyElementMaster; err = AudioObjectAddPropertyListener(ao->i_selected_dev, &property_address, 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. */ #if HAVE_BIGENDIAN if (!(ao->stream_format.mFormatFlags & kAudioFormatFlagIsBigEndian)) #else /* tell mplayer that we need a byteswap on AC3 streams, */ if (ao->stream_format.mFormatID & kAudioFormat60958AC3) ao_data.format = AF_FORMAT_AC3_LE; if (ao->stream_format.mFormatFlags & kAudioFormatFlagIsBigEndian) #endif ao_msg(MSGT_AO, MSGL_WARN, "Output stream has non-native byte order, digital output may fail.\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); /* Create IOProc callback. */ err = AudioDeviceCreateIOProcID(ao->i_selected_dev, (AudioDeviceIOProc)RenderCallbackSPDIF, (void *)ao, &ao->renderCallback); if (err != noErr || ao->renderCallback == NULL) { 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 = SetAudioProperty(ao->i_selected_dev, kAudioDevicePropertySupportsMixing, sizeof(int), &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; err = SetAudioProperty(ao->i_selected_dev, kAudioDevicePropertyHogMode, sizeof(ao->i_hog_pid), &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 ) { 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. */ i_param_size = GetAudioPropertyArray(i_dev_id, kAudioDevicePropertyStreams, kAudioDevicePropertyScopeOutput, (void **)&p_streams); if (!i_param_size) { ao_msg(MSGT_AO, MSGL_WARN, "could not get number of streams.\n"); return CONTROL_FALSE; } i_streams = i_param_size / sizeof(AudioStreamID); 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 ) { 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. */ i_param_size = GetGlobalAudioPropertyArray(i_stream_id, kAudioStreamPropertyPhysicalFormats, (void **)&p_format_list); if (!i_param_size) { ao_msg(MSGT_AO, MSGL_WARN, "Could not get number of stream formats.\n"); return CONTROL_FALSE; } i_formats = i_param_size / sizeof(AudioStreamBasicDescription); 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; int i; AudioObjectPropertyAddress property_address; static volatile int stream_format_changed; stream_format_changed = 0; print_format(MSGL_V, "setting stream format:", &change_format); /* Install the callback. */ property_address.mSelector = kAudioStreamPropertyPhysicalFormat; property_address.mScope = kAudioObjectPropertyScopeGlobal; property_address.mElement = kAudioObjectPropertyElementMaster; err = AudioObjectAddPropertyListener(i_stream_id, &property_address, 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 = SetAudioProperty(i_stream_id, 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" ); err = GetAudioProperty(i_stream_id, kAudioStreamPropertyPhysicalFormat, sizeof(AudioStreamBasicDescription), &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 = AudioObjectRemovePropertyListener(i_stream_id, &property_address, StreamListener, (void *)&stream_format_changed); 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; SInt32 exit_reason; // 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 supports digital format output, let's set it. */ ao_msg(MSGT_AO, MSGL_V, "Detected current stream supports digital, try to restore digital output...\n"); if (!AudioStreamChangeFormat(ao->i_stream_id, ao->stream_format)) { ao_msg(MSGT_AO, MSGL_WARN, "Restoring digital output failed.\n"); } else { ao_msg(MSGT_AO, MSGL_WARN, "Restoring digital output succeeded.\n"); reset(); } } else ao_msg(MSGT_AO, MSGL_V, "Detected current stream does not support digital.\n"); } wrote=write_buffer(output_samples, num_bytes); audio_resume(); do { exit_reason = CFRunLoopRunInMode(kCFRunLoopDefaultMode, 0.01, true); } while (exit_reason == kCFRunLoopRunHandledSource); 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; 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, ao->renderCallback); if (err != noErr) ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceStop failed: [%4.4s]\n", (char *)&err); /* Remove IOProc callback. */ err = AudioDeviceDestroyIOProcID(ao->i_selected_dev, ao->renderCallback); 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) { UInt32 b_mix; Boolean b_writeable; /* Revert mixable to true if we are allowed to. */ err = IsAudioPropertySettable(ao->i_selected_dev, kAudioDevicePropertySupportsMixing, &b_writeable); err = GetAudioProperty(ao->i_selected_dev, kAudioDevicePropertySupportsMixing, sizeof(UInt32), &b_mix); if (err != noErr && b_writeable) { b_mix = 1; err = SetAudioProperty(ao->i_selected_dev, kAudioDevicePropertySupportsMixing, sizeof(UInt32), &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; err = SetAudioProperty(ao->i_selected_dev, kAudioDevicePropertyHogMode, sizeof(ao->i_hog_pid), &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, ao->renderCallback); 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, ao->renderCallback); if (err != noErr) ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceStart failed: [%4.4s]\n", (char *)&err); } ao->paused = 0; } /***************************************************************************** * StreamListener *****************************************************************************/ static OSStatus StreamListener( AudioObjectID inObjectID, UInt32 inNumberAddresses, const AudioObjectPropertyAddress inAddresses[], void *inClientData ) { for (int i=0; i < inNumberAddresses; ++i) { if (inAddresses[i].mSelector == kAudioStreamPropertyPhysicalFormat) { ao_msg(MSGT_AO, MSGL_WARN, "got notify kAudioStreamPropertyPhysicalFormat changed.\n"); if (inClientData) *(volatile int *)inClientData = 1; break; } } return noErr; } static OSStatus DeviceListener( AudioObjectID inObjectID, UInt32 inNumberAddresses, const AudioObjectPropertyAddress inAddresses[], void *inClientData ) { for (int i=0; i < inNumberAddresses; ++i) { if (inAddresses[i].mSelector == kAudioDevicePropertyDeviceHasChanged) { ao_msg(MSGT_AO, MSGL_WARN, "got notify kAudioDevicePropertyDeviceHasChanged.\n"); ao->b_stream_format_changed = 1; break; } } return noErr; }