Index: libao2/ao_macosx.c =================================================================== --- libao2/ao_macosx.c (revision 24540) +++ libao2/ao_macosx.c (working copy) @@ -45,6 +45,8 @@ #include #include #include +#include +#include #include "config.h" #include "mp_msg.h" @@ -52,6 +54,7 @@ #include "audio_out.h" #include "audio_out_internal.h" #include "libaf/af_format.h" +#include "osdep/timer.h" static ao_info_t info = { @@ -72,8 +75,24 @@ 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; @@ -177,6 +196,12 @@ switch (cmd) { case AOCONTROL_GET_VOLUME: control_vol = (ao_control_vol_t*)arg; + if (ao->b_digital) { + // Digital output can not adjust output volume (always fixed 100%). + // Other ao modules such as ao_alsa also did this. + control_vol->left=control_vol->right = 100.0; + return CONTROL_TRUE; + } err = AudioUnitGetParameter(ao->theOutputUnit, kHALOutputParam_Volume, kAudioUnitScope_Global, 0, &vol); if(err==0) { @@ -185,10 +210,16 @@ 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. If return false, + // a soft volume filter will be inserted which will corrupt encoded audio data. + // So here we must return true. Other ao modules such as ao_alsa also did this. + return CONTROL_TRUE; control_vol = (ao_control_vol_t*)arg; vol=(control_vol->left+control_vol->right)*4.0/200.0; @@ -198,6 +229,7 @@ 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 */ @@ -208,61 +240,137 @@ } -static void print_format(const char* str,AudioStreamBasicDescription *f){ +static void print_format(int lev, const char* str, const AudioStreamBasicDescription *f){ uint32_t flags=(uint32_t) f->mFormatFlags; - ao_msg(MSGT_AO,MSGL_V, "%s %7.1fHz %dbit [%c%c%c%c] %s %s %s%s%s%s\n", + 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" : "" ); +} - ao_msg(MSGT_AO,MSGL_DBG2, "%5d mBytesPerPacket\n", - (int)f->mBytesPerPacket); - ao_msg(MSGT_AO,MSGL_DBG2, "%5d mFramesPerPacket\n", - (int)f->mFramesPerPacket); - ao_msg(MSGT_AO,MSGL_DBG2, "%5d mBytesPerFrame\n", - (int)f->mBytesPerFrame); - ao_msg(MSGT_AO,MSGL_DBG2, "%5d mChannelsPerFrame\n", - (int)f->mChannelsPerFrame); -} +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; +UInt32 size, maxFrames, i_param_size; +char *psz_name; int aoIsCreated = ao != NULL; +AudioDeviceID devid_def = 0; +int b_alive; - if (!aoIsCreated) ao = malloc(sizeof(ao_macosx_t)); + ao_msg(MSGT_AO,MSGL_V, "init([%dHz][%dch][%s][%d])\n", rate, channels, af_fmt2str_short(format), flags); - // Build Description for the input format - inDesc.mSampleRate=rate; - inDesc.mFormatID=kAudioFormatLinearPCM; - inDesc.mChannelsPerFrame=channels; + 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 + ao->stream_format.mSampleRate = rate; + ao->stream_format.mChannelsPerFrame = channels; + ao->stream_format.mFormatID = ao->b_supports_digital ? kAudioFormat60958AC3 : kAudioFormatLinearPCM; + switch(format&AF_FORMAT_BITS_MASK){ case AF_FORMAT_8BIT: - inDesc.mBitsPerChannel=8; + ao->stream_format.mBitsPerChannel=8; break; case AF_FORMAT_16BIT: - inDesc.mBitsPerChannel=16; + ao->stream_format.mBitsPerChannel=16; break; case AF_FORMAT_24BIT: - inDesc.mBitsPerChannel=24; + ao->stream_format.mBitsPerChannel=24; break; case AF_FORMAT_32BIT: - inDesc.mBitsPerChannel=32; + ao->stream_format.mBitsPerChannel=32; break; default: ao_msg(MSGT_AO, MSGL_WARN, "Unsupported format (0x%08x)\n", format); @@ -272,24 +380,64 @@ if((format&AF_FORMAT_POINT_MASK)==AF_FORMAT_F) { // float - inDesc.mFormatFlags = kAudioFormatFlagIsFloat|kAudioFormatFlagIsPacked; + ao->stream_format.mFormatFlags = kAudioFormatFlagIsFloat|kAudioFormatFlagIsPacked; } else if((format&AF_FORMAT_SIGN_MASK)==AF_FORMAT_SI) { // signed int - inDesc.mFormatFlags = kAudioFormatFlagIsSignedInteger|kAudioFormatFlagIsPacked; + ao->stream_format.mFormatFlags = kAudioFormatFlagIsSignedInteger|kAudioFormatFlagIsPacked; } else { // unsigned int - inDesc.mFormatFlags = kAudioFormatFlagIsPacked; + ao->stream_format.mFormatFlags = kAudioFormatFlagIsPacked; } - - if((format&AF_FORMAT_END_MASK)==AF_FORMAT_BE) - inDesc.mFormatFlags |= kAudioFormatFlagIsBigEndian; + if ((format & AF_FORMAT_SPECIAL_MASK) == AF_FORMAT_AC3) { + // Currently ac3 input (comes from hwac3) is always in native byte-order. +#ifdef WORDS_BIGENDIAN + ao->stream_format.mFormatFlags |= kAudioFormatFlagIsBigEndian; +#endif + } + else if ((format & AF_FORMAT_END_MASK) == AF_FORMAT_BE) + ao->stream_format.mFormatFlags |= kAudioFormatFlagIsBigEndian; - inDesc.mFramesPerPacket = 1; - ao->packetSize = inDesc.mBytesPerPacket = inDesc.mBytesPerFrame = inDesc.mFramesPerPacket*channels*(inDesc.mBitsPerChannel/8); - print_format("source: ",&inDesc); + ao->stream_format.mFramesPerPacket = 1; + ao->packetSize = ao->stream_format.mBytesPerPacket = ao->stream_format.mBytesPerFrame + = channels*(ao->stream_format.mBitsPerChannel/8); + print_format(MSGL_V, "source:",&ao->stream_format); + 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; + } + return OpenSPDIF(); + } + + /* original analog output code */ if (!aoIsCreated) { desc.componentType = kAudioUnitType_Output; desc.componentSubType = kAudioUnitSubType_DefaultOutput; @@ -305,23 +453,23 @@ 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 (err=%d)\n", 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 (err=%d)\n", 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); + err = AudioUnitSetProperty(ao->theOutputUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &ao->stream_format, size); if (err) { - ao_msg(MSGT_AO, MSGL_WARN, "Unable to set the input format (err=%d)\n", err); + ao_msg(MSGT_AO, MSGL_WARN, "Unable to set the input format: [%4.4s]\n", (char *)&err); return CONTROL_FALSE; } @@ -330,15 +478,15 @@ if (err) { - ao_msg(MSGT_AO,MSGL_WARN, "AudioUnitGetProperty returned %d when getting kAudioDevicePropertyBufferSize\n", (int)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.samplerate = ao->stream_format.mSampleRate; + ao_data.channels = ao->stream_format.mChannelsPerFrame; + ao_data.bps = ao_data.samplerate * ao->stream_format.mBytesPerFrame; ao_data.outburst = ao->chunk_size; ao_data.buffersize = ao_data.bps; @@ -353,7 +501,7 @@ 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 (err=%d)\n", err); + ao_msg(MSGT_AO, MSGL_WARN, "Unable to set the render callback: [%4.4s]\n", (char *)&err); return CONTROL_FALSE; } @@ -362,13 +510,489 @@ 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", (char *)&err); + + /* 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=write_buffer(output_samples, num_bytes); + int wrote, b_digital; - audio_resume(); - return wrote; + // 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 */ @@ -402,17 +1026,64 @@ /* 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 (%ld usec)\n", buf_used(), ao_data.bps, (int)timeleft); + 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); } - AudioOutputUnitStop(ao->theOutputUnit); - AudioUnitUninitialize(ao->theOutputUnit); - CloseComponent(ao->theOutputUnit); + 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; @@ -422,27 +1093,87 @@ /* stop playing, keep buffers (for pause) */ static void audio_pause(void) { - OSErr status=noErr; + OSErr err=noErr; - /* stop callback */ - status=AudioOutputUnitStop(ao->theOutputUnit); - if (status) - ao_msg(MSGT_AO,MSGL_WARN, "AudioOutputUnitStop returned %d\n", - (int)status); - ao->paused=1; + /* 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) { - if(ao->paused) { - OSErr status=noErr; - /* start callback */ - status=AudioOutputUnitStart(ao->theOutputUnit); - if (status) - ao_msg(MSGT_AO,MSGL_WARN, "AudioOutputUnitStart returned %d\n", - (int)status); - ao->paused=0; - } + 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; +}