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authorwm4 <wm4@nowhere>2012-11-05 17:02:04 +0100
committerwm4 <wm4@nowhere>2012-11-12 20:06:14 +0100
commitd4bdd0473d6f43132257c9fb3848d829755167a3 (patch)
tree8021c2f7da1841393c8c832105e20cd527826d6c /libao2/ao_coreaudio.c
parentbd48deba77bd5582c5829d6fe73a7d2571088aba (diff)
downloadmpv-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.c1283
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;
- int b_return = CONTROL_FALSE;
-
- /* Retrieve all the output streams. */
- i_param_size = GetAudioPropertyArray(i_dev_id,
- kAudioDevicePropertyStreams,<