diff options
author | wm4 <wm4@nowhere> | 2012-11-05 17:02:04 +0100 |
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committer | wm4 <wm4@nowhere> | 2012-11-12 20:06:14 +0100 |
commit | d4bdd0473d6f43132257c9fb3848d829755167a3 (patch) | |
tree | 8021c2f7da1841393c8c832105e20cd527826d6c /libao2/ao_coreaudio.c | |
parent | bd48deba77bd5582c5829d6fe73a7d2571088aba (diff) | |
download | mpv-d4bdd0473d6f43132257c9fb3848d829755167a3.tar.bz2 mpv-d4bdd0473d6f43132257c9fb3848d829755167a3.tar.xz |
Rename directories, move files (step 1 of 2) (does not compile)
Tis drops the silly lib prefixes, and attempts to organize the tree in
a more logical way. Make the top-level directory less cluttered as
well.
Renames the following directories:
libaf -> audio/filter
libao2 -> audio/out
libvo -> video/out
libmpdemux -> demux
Split libmpcodecs:
vf* -> video/filter
vd*, dec_video.* -> video/decode
mp_image*, img_format*, ... -> video/
ad*, dec_audio.* -> audio/decode
libaf/format.* is moved to audio/ - this is similar to how mp_image.*
is located in video/.
Move most top-level .c/.h files to core. (talloc.c/.h is left on top-
level, because it's external.) Park some of the more annoying files
in compat/. Some of these are relicts from the time mplayer used
ffmpeg internals.
sub/ is not split, because it's too much of a mess (subtitle code is
mixed with OSD display and rendering).
Maybe the organization of core is not ideal: it mixes playback core
(like mplayer.c) and utility helpers (like bstr.c/h). Should the need
arise, the playback core will be moved somewhere else, while core
contains all helper and common code.
Diffstat (limited to 'libao2/ao_coreaudio.c')
-rw-r--r-- | libao2/ao_coreaudio.c | 1283 |
1 files changed, 0 insertions, 1283 deletions
diff --git a/libao2/ao_coreaudio.c b/libao2/ao_coreaudio.c deleted file mode 100644 index 146cfd2a22..0000000000 --- a/libao2/ao_coreaudio.c +++ /dev/null @@ -1,1283 +0,0 @@ -/* - * 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/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; - if (data) - av_fifo_generic_read(ao->buffer, data, len, NULL); - else - av_fifo_drain(ao->buffer, len); - 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. - int vol = ao->b_muted ? 0 : 100; - *control_vol = (ao_control_vol_t) { - .left = vol, .right = vol, - }; - return CONTROL_TRUE; - } - 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: mpv -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; - - global_ao->per_application_mixer = true; - global_ao->no_persistent_volume = true; - - 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; - } - - property_address.mSelector = kAudioDevicePropertySupportsMixing; - property_address.mScope = kAudioObjectPropertyScopeGlobal; - property_address.mElement = kAudioObjectPropertyElementMaster; - - /* Set mixable to false if we are allowed to. */ - if (AudioObjectHasProperty(ao->i_selected_dev, &property_address)) { - /* 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. */ - AudioStreamRangedDescription *p_format_list = NULL; - int i_formats = 0, j = 0, b_digital = 0; - - i_param_size = GetGlobalAudioPropertyArray(p_streams[i], - kAudioStreamPropertyAvailablePhysicalFormats, - (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(AudioStreamRangedDescription); - - /* Check if one of the supported formats is a digital format. */ - for (j = 0; j < i_formats; ++j) - { - if (p_format_list[j].mFormat.mFormatID == 'IAC3' || - p_format_list[j].mFormat.mFormatID == 'iac3' || - p_format_list[j].mFormat.mFormatID == kAudioFormat60958AC3 || - p_format_list[j].mFormat.mFormatID == kAudioFormatAC3) - { - 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].mFormat.mFormatID == 'IAC3' || - p_format_list[j].mFormat.mFormatID == 'iac3' || - p_format_list[j].mFormat.mFormatID == kAudioFormat60958AC3 || - p_format_list[j].mFormat.mFormatID == kAudioFormatAC3) - { - if (p_format_list[j].mFormat.mSampleRate == ao->stream_format.mSampleRate) - { - i_requested_rate_format = j; - break; - } - if (p_format_list[j].mFormat.mSampleRate == ao->sfmt_revert.mSampleRate) - i_current_rate_format = j; - else if (i_backup_rate_format < 0 || p_format_list[j].mFormat.mSampleRate > p_format_list[i_backup_rate_format].mFormat.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].mFormat; - 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].mFormat; - else ao->stream_format = p_format_list[i_backup_rate_format].mFormat; /* 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 BYTE_ORDER == BIG_ENDIAN - 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; |