/* * * ao_macosx.c * * Original Copyright (C) Timothy J. Wood - Aug 2000 * * This file is part of libao, a cross-platform library. See * README for a history of this source code. * * libao is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2, or (at your option) * any later version. * * libao is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. * * You should have received a copy of the GNU General Public License along * with libao; 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). */ /* Change log: * * 14/5-2003: Ported to MPlayer libao2 by Dan Christiansen * * AC-3 and MPEG audio passthrough is possible, but I don't have * access to a sound card that supports it. */ #include <CoreServices/CoreServices.h> #include <AudioUnit/AudioUnit.h> #include <AudioToolbox/AudioToolbox.h> #include <stdio.h> #include <string.h> #include <stdlib.h> #include <inttypes.h> #include <pthread.h> #include <sys/types.h> #include <sys/time.h> #include <unistd.h> #include "config.h" #include "mp_msg.h" #include "audio_out.h" #include "audio_out_internal.h" #include "libaf/af_format.h" #include "osdep/timer.h" static ao_info_t info = { "Darwin/Mac OS X native audio output", "macosx", "Timothy J. Wood & Dan Christiansen & Chris Roccati", "" }; LIBAO_EXTERN(macosx) /* Prefix for all mp_msg() calls */ #define ao_msg(a, b, c...) mp_msg(a, b, "AO: [macosx] " c) /* This is large, but best (maybe it should be even larger). * CoreAudio supposedly has an internal latency in the order of 2ms */ #define NUM_BUFS 32 typedef struct ao_macosx_s { 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? */ /* AudioUnit */ AudioUnit theOutputUnit; /* CoreAudio SPDIF mode specific */ pid_t i_hog_pid; /* Keeps the pid of our hog status. */ AudioStreamID i_stream_id; /* The StreamID that has a cac3 streamformat */ int i_stream_index; /* The index of i_stream_id in an AudioBufferList */ AudioStreamBasicDescription stream_format;/* The format we changed the stream to */ AudioStreamBasicDescription sfmt_revert; /* The original format of the stream */ int b_revert; /* Whether we need to revert the stream format */ int b_changed_mixing; /* Whether we need to set the mixing mode back */ int b_stream_format_changed; /* Flag for main thread to reset stream's format to digital and reset buffer */ /* Original common part */ int packetSize; int paused; /* Ring-buffer */ /* does not need explicit synchronization, but needs to allocate * (num_chunks + 1) * chunk_size memory to store num_chunks * chunk_size * data */ unsigned char *buffer; unsigned int buffer_len; ///< must always be (num_chunks + 1) * chunk_size unsigned int num_chunks; unsigned int chunk_size; unsigned int buf_read_pos; unsigned int buf_write_pos; } ao_macosx_t; static ao_macosx_t *ao = NULL; /** * \brief return number of free bytes in the buffer * may only be called by mplayer's thread * \return minimum number of free bytes in buffer, value may change between * two immediately following calls, and the real number of free bytes * might actually be larger! */ static int buf_free(void) { int free = ao->buf_read_pos - ao->buf_write_pos - ao->chunk_size; if (free < 0) free += ao->buffer_len; return free; } /** * \brief return number of buffered bytes * may only be called by playback thread * \return minimum number of buffered bytes, value may change between * two immediately following calls, and the real number of buffered bytes * might actually be larger! */ static int buf_used(void) { int used = ao->buf_write_pos - ao->buf_read_pos; if (used < 0) used += ao->buffer_len; return used; } /** * \brief add data to ringbuffer */ static int write_buffer(unsigned char* data, int len){ int first_len = ao->buffer_len - ao->buf_write_pos; int free = buf_free(); if (len > free) len = free; if (first_len > len) first_len = len; // till end of buffer memcpy (&ao->buffer[ao->buf_write_pos], data, first_len); if (len > first_len) { // we have to wrap around // remaining part from beginning of buffer memcpy (ao->buffer, &data[first_len], len - first_len); } ao->buf_write_pos = (ao->buf_write_pos + len) % ao->buffer_len; return len; } /** * \brief remove data from ringbuffer */ static int read_buffer(unsigned char* data,int len){ int first_len = ao->buffer_len - ao->buf_read_pos; int buffered = buf_used(); if (len > buffered) len = buffered; if (first_len > len) first_len = len; // till end of buffer memcpy (data, &ao->buffer[ao->buf_read_pos], first_len); if (len > first_len) { // we have to wrap around // remaining part from beginning of buffer memcpy (&data[first_len], ao->buffer, len - first_len); } ao->buf_read_pos = (ao->buf_read_pos + len) % ao->buffer_len; return len; } OSStatus theRenderProc(void *inRefCon, AudioUnitRenderActionFlags *inActionFlags, const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, UInt32 inNumFrames, AudioBufferList *ioData) { int amt=buf_used(); int req=(inNumFrames)*ao->packetSize; if(amt>req) amt=req; if(amt) read_buffer((unsigned char *)ioData->mBuffers[0].mData, amt); else audio_pause(); ioData->mBuffers[0].mDataByteSize = amt; return noErr; } static int control(int cmd,void *arg){ ao_control_vol_t *control_vol; OSStatus err; Float32 vol; switch (cmd) { case AOCONTROL_GET_VOLUME: control_vol = (ao_control_vol_t*)arg; if (ao->b_digital) { // Digital output has no volume adjust. return CONTROL_FALSE; } err = AudioUnitGetParameter(ao->theOutputUnit, kHALOutputParam_Volume, kAudioUnitScope_Global, 0, &vol); if(err==0) { // printf("GET VOL=%f\n", vol); control_vol->left=control_vol->right=vol*100.0/4.0; return CONTROL_TRUE; } else { ao_msg(MSGT_AO, MSGL_WARN, "could not get HAL output volume: [%4.4s]\n", (char *)&err); return CONTROL_FALSE; } case AOCONTROL_SET_VOLUME: 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. return CONTROL_TRUE; control_vol = (ao_control_vol_t*)arg; 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; int aoIsCreated = ao != NULL; 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); if (!aoIsCreated) { ao = malloc(sizeof(ao_macosx_t)); ao->buffer = NULL;} ao->i_selected_dev = 0; ao->b_supports_digital = 0; ao->b_digital = 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); return CONTROL_FALSE; } /* 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); return CONTROL_FALSE; } /* 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); return CONTROL_FALSE; } 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); return CONTROL_FALSE; break; } 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" ); return CONTROL_FALSE; } ao->stream_format = inDesc; return OpenSPDIF(); } /* original analog output code */ if (!aoIsCreated) { 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"); return CONTROL_FALSE; } 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); return CONTROL_FALSE; } // 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); return CONTROL_FALSE; } } 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); return CONTROL_FALSE; } 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); return CONTROL_FALSE; } ao->chunk_size = maxFrames;//*inDesc.mBytesPerFrame; ao_data.samplerate = inDesc.mSampleRate; ao_data.channels = inDesc.mChannelsPerFrame; ao_data.bps = ao_data.samplerate * inDesc.mBytesPerFrame; ao_data.outburst = ao->chunk_size; ao_data.buffersize = ao_data.bps; ao->num_chunks = (ao_data.bps+ao->chunk_size-1)/ao->chunk_size; ao->buffer_len = (ao->num_chunks + 1) * ao->chunk_size; ao->buffer = aoIsCreated ? realloc(ao->buffer,(ao->num_chunks + 1)*ao->chunk_size) : calloc(ao->num_chunks + 1, ao->chunk_size); ao_msg(MSGT_AO,MSGL_V, "using %5d chunks of %d bytes (buffer len %d bytes)\n", (int)ao->num_chunks, (int)ao->chunk_size, (int)ao->buffer_len); renderCallback.inputProc = theRenderProc; renderCallback.inputProcRefCon = 0; err = AudioUnitSetProperty(ao->theOutputUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, 0, &renderCallback, sizeof(AURenderCallbackStruct)); if (err) { ao_msg(MSGT_AO, MSGL_WARN, "Unable to set the render callback: [%4.4s]\n", (char *)&err); return CONTROL_FALSE; } reset(); return CONTROL_OK; } /***************************************************************************** * 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); return CONTROL_FALSE; } /* 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); return CONTROL_FALSE; } /* 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); 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_WARN, "out of memory\n" ); return CONTROL_FALSE; } 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); return CONTROL_FALSE; } 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"); return CONTROL_FALSE; } print_format(MSGL_V, "original stream format:", &ao->sfmt_revert); if (!AudioStreamChangeFormat(ao->i_stream_id, ao->stream_format)) return CONTROL_FALSE; err = AudioDeviceAddPropertyListener(ao->i_selected_dev, kAudioPropertyWildcardChannel, 0, kAudioDevicePropertyDeviceHasChanged, DeviceListener, NULL); if (err != noErr) ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceAddPropertyListener for kAudioDevicePropertyDeviceHasChanged failed: [%4.4s]\n", (char *)&err); /* FIXME: If output stream is not native byte-order, we need change endian somewhere. */ /* Although there's no such case reported. */ #ifdef WORDS_BIGENDIAN if (!(ao->stream_format.mFormatFlags & kAudioFormatFlagIsBigEndian)) #else if (ao->stream_format.mFormatFlags & kAudioFormatFlagIsBigEndian) #endif ao_msg(MSGT_AO, MSGL_WARN, "output stream has a no-native byte-order, digital output may failed.\n"); /* For ac3/dts, just use packet size 6144 bytes as chunk size. */ ao->chunk_size = ao->stream_format.mBytesPerPacket; ao_data.samplerate = ao->stream_format.mSampleRate; ao_data.channels = ao->stream_format.mChannelsPerFrame; ao_data.bps = ao_data.samplerate * (ao->stream_format.mBytesPerPacket/ao->stream_format.mFramesPerPacket); ao_data.outburst = ao->chunk_size; ao_data.buffersize = ao_data.bps; ao->num_chunks = (ao_data.bps+ao->chunk_size-1)/ao->chunk_size; ao->buffer_len = (ao->num_chunks + 1) * ao->chunk_size; ao->buffer = NULL!=ao->buffer ? realloc(ao->buffer,(ao->num_chunks + 1)*ao->chunk_size) : calloc(ao->num_chunks + 1, ao->chunk_size); ao_msg(MSGT_AO,MSGL_V, "using %5d chunks of %d bytes (buffer len %d bytes)\n", (int)ao->num_chunks, (int)ao->chunk_size, (int)ao->buffer_len); /* Add IOProc callback. */ err = AudioDeviceAddIOProc(ao->i_selected_dev, (AudioDeviceIOProc)RenderCallbackSPDIF, (void *)ao); if (err != noErr) { ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceAddIOProc failed: [%4.4s]\n", (char *)&err); return CONTROL_FALSE; } reset(); return CONTROL_TRUE; } /***************************************************************************** * 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; struct timeval now; struct timespec timeout; struct { pthread_mutex_t lock; pthread_cond_t cond; } w; print_format(MSGL_V, "setting stream format:", &change_format); /* Condition because SetProperty is asynchronious. */ pthread_cond_init(&w.cond, NULL); pthread_mutex_init(&w.lock, NULL); pthread_mutex_lock(&w.lock); /* Install the callback. */ err = AudioStreamAddPropertyListener(i_stream_id, 0, kAudioStreamPropertyPhysicalFormat, StreamListener, (void *)&w); 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 (requiring the locks), * 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; gettimeofday(&now, NULL); timeout.tv_sec = now.tv_sec; timeout.tv_nsec = (now.tv_usec + 500000) * 1000; if (pthread_cond_timedwait(&w.cond, &w.lock, &timeout)) 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; } /* Destroy the lock and condition. */ pthread_mutex_unlock(&w.lock); pthread_mutex_destroy(&w.lock); pthread_cond_destroy(&w.cond); return CONTROL_TRUE; } /***************************************************************************** * RenderCallbackSPDIF: callback for SPDIF audio output *****************************************************************************/ static OSStatus RenderCallbackSPDIF( AudioDeviceID inDevice, const AudioTimeStamp * inNow, const void * inInputData, const AudioTimeStamp * inInputTime, AudioBufferList * outOutputData, const AudioTimeStamp * inOutputTime, void * threadGlobals ) { int amt = buf_used(); int req = outOutputData->mBuffers[ao->i_stream_index].mDataByteSize; if (amt > req) amt = req; if (amt) read_buffer((unsigned char *)outOutputData->mBuffers[ao->i_stream_index].mData, amt); return noErr; } static int play(void* output_samples,int num_bytes,int flags) { int wrote, b_digital; // Check whether we need to reset the digital output stream. if (ao->b_digital && ao->b_stream_format_changed) { ao->b_stream_format_changed = 0; b_digital = AudioStreamSupportsDigital(ao->i_stream_id); if (b_digital) { /* Current stream support digital format output, let's set it. */ ao_msg(MSGT_AO, MSGL_V, "detected current stream support digital, try to restore digital output...\n"); if (!AudioStreamChangeFormat(ao->i_stream_id, ao->stream_format)) { ao_msg(MSGT_AO, MSGL_WARN, "restore digital output failed.\n"); } else { ao_msg(MSGT_AO, MSGL_WARN, "restore digital output succeed.\n"); reset(); } } else ao_msg(MSGT_AO, MSGL_V, "detected current stream do not support digital.\n"); } wrote=write_buffer(output_samples, num_bytes); audio_resume(); return wrote; } /* set variables and buffer to initial state */ static void reset(void) { audio_pause(); /* reset ring-buffer state */ ao->buf_read_pos=0; ao->buf_write_pos=0; return; } /* return available space */ static int get_space(void) { return buf_free(); } /* return delay until audio is played */ static float get_delay(void) { int buffered = ao->buffer_len - ao->chunk_size - buf_free(); // could be less // inaccurate, should also contain the data buffered e.g. by the OS return (float)(buffered)/(float)ao_data.bps; } /* unload plugin and deregister from coreaudio */ static void uninit(int immed) { OSStatus err = noErr; UInt32 i_param_size = 0; if (!immed) { long long timeleft=(1000000LL*buf_used())/ao_data.bps; ao_msg(MSGT_AO,MSGL_DBG2, "%d bytes left @%d bps (%d usec)\n", buf_used(), ao_data.bps, (int)timeleft); usec_sleep((int)timeleft); } if (!ao->b_digital) { AudioOutputUnitStop(ao->theOutputUnit); AudioUnitUninitialize(ao->theOutputUnit); CloseComponent(ao->theOutputUnit); } else { /* Stop device. */ err = AudioDeviceStop(ao->i_selected_dev, (AudioDeviceIOProc)RenderCallbackSPDIF); if (err != noErr) ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceStop failed: [%4.4s]\n", (char *)&err); /* Remove IOProc callback. */ err = AudioDeviceRemoveIOProc(ao->i_selected_dev, (AudioDeviceIOProc)RenderCallbackSPDIF); if (err != noErr) ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceRemoveIOProc failed: [%4.4s]\n", (char *)&err); if (ao->b_revert) AudioStreamChangeFormat(ao->i_stream_id, ao->sfmt_revert); if (ao->b_changed_mixing && ao->sfmt_revert.mFormatID != kAudioFormat60958AC3) { int b_mix; Boolean b_writeable; /* Revert mixable to true if we are allowed to. */ err = AudioDeviceGetPropertyInfo(ao->i_selected_dev, 0, FALSE, kAudioDevicePropertySupportsMixing, &i_param_size, &b_writeable); err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE, kAudioDevicePropertySupportsMixing, &i_param_size, &b_mix); if (err != noErr && b_writeable) { b_mix = 1; err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE, kAudioDevicePropertySupportsMixing, i_param_size, &b_mix); } if (err != noErr) ao_msg(MSGT_AO, MSGL_WARN, "failed to set mixmode: [%4.4s]\n", (char *)&err); } if (ao->i_hog_pid == getpid()) { ao->i_hog_pid = -1; i_param_size = sizeof(ao->i_hog_pid); err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE, kAudioDevicePropertyHogMode, i_param_size, &ao->i_hog_pid); if (err != noErr) ao_msg(MSGT_AO, MSGL_WARN, "Could not release hogmode: [%4.4s]\n", (char *)&err); } } free(ao->buffer); free(ao); ao = NULL; } /* stop playing, keep buffers (for pause) */ static void audio_pause(void) { OSErr err=noErr; /* Stop callback. */ if (!ao->b_digital) { err=AudioOutputUnitStop(ao->theOutputUnit); if (err != noErr) ao_msg(MSGT_AO,MSGL_WARN, "AudioOutputUnitStop returned [%4.4s]\n", (char *)&err); } else { err = AudioDeviceStop(ao->i_selected_dev, (AudioDeviceIOProc)RenderCallbackSPDIF); if (err != noErr) ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceStop failed: [%4.4s]\n", (char *)&err); } ao->paused = 1; } /* resume playing, after audio_pause() */ static void audio_resume(void) { OSErr err=noErr; if (!ao->paused) return; /* Start callback. */ if (!ao->b_digital) { err = AudioOutputUnitStart(ao->theOutputUnit); if (err != noErr) ao_msg(MSGT_AO,MSGL_WARN, "AudioOutputUnitStart returned [%4.4s]\n", (char *)&err); } else { err = AudioDeviceStart(ao->i_selected_dev, (AudioDeviceIOProc)RenderCallbackSPDIF); if (err != noErr) ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceStart failed: [%4.4s]\n", (char *)&err); } ao->paused = 0; } /***************************************************************************** * StreamListener *****************************************************************************/ static OSStatus StreamListener( AudioStreamID inStream, UInt32 inChannel, AudioDevicePropertyID inPropertyID, void * inClientData ) { struct { pthread_mutex_t lock; pthread_cond_t cond; } * w = inClientData; switch (inPropertyID) { case kAudioStreamPropertyPhysicalFormat: if (NULL!=w) { pthread_mutex_lock(&w->lock); pthread_cond_signal(&w->cond); pthread_mutex_unlock(&w->lock); } 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; }