[MPlayer-cvslog] r24762 - trunk/libao2/ao_macosx.c
nplourde
subversion at mplayerhq.hu
Thu Oct 11 04:00:05 CEST 2007
Author: nplourde
Date: Thu Oct 11 04:00:05 2007
New Revision: 24762
Log:
ao_macosx: added support for ac3/dts passthrough
Modified:
trunk/libao2/ao_macosx.c
Modified: trunk/libao2/ao_macosx.c
==============================================================================
--- trunk/libao2/ao_macosx.c (original)
+++ trunk/libao2/ao_macosx.c Thu Oct 11 04:00:05 2007
@@ -45,6 +45,8 @@
#include <stdlib.h>
#include <inttypes.h>
#include <pthread.h>
+#include <sys/types.h>
+#include <unistd.h>
#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 @@ LIBAO_EXTERN(macosx)
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 +196,10 @@ Float32 vol;
switch (cmd) {
case AOCONTROL_GET_VOLUME:
control_vol = (ao_control_vol_t*)arg;
+ if (ao->b_digital) {
+ // Digital output has no volume adjust.
+ return CONTROL_FALSE;
+ }
err = AudioUnitGetParameter(ao->theOutputUnit, kHALOutputParam_Volume, kAudioUnitScope_Global, 0, &vol);
if(err==0) {
@@ -185,10 +208,17 @@ Float32 vol;
return CONTROL_TRUE;
}
else {
+ ao_msg(MSGT_AO, MSGL_WARN, "could not get HAL output volume: [%4.4s]\n", (char *)&err);
return CONTROL_FALSE;
}
case AOCONTROL_SET_VOLUME:
+ if (ao->b_digital)
+ // Digital output can not set volume. Here we have to return true
+ // to make mixer forget it. Else mixer will add a soft filter,
+ // that's not we expected and the filter not support ac3 stream
+ // will cause mplayer die.
+ return CONTROL_TRUE;
control_vol = (ao_control_vol_t*)arg;
vol=(control_vol->left+control_vol->right)*4.0/200.0;
@@ -198,6 +228,7 @@ Float32 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 */
@@ -208,33 +239,47 @@ Float32 vol;
}
-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;
@@ -242,14 +287,76 @@ 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);
+
+ if (!aoIsCreated) { ao = malloc(sizeof(ao_macosx_t)); ao->buffer = NULL;}
+
+ ao->i_selected_dev = 0;
+ ao->b_supports_digital = 0;
+ ao->b_digital = 0;
+ ao->b_stream_format_changed = 0;
+ ao->i_hog_pid = -1;
+ ao->i_stream_id = 0;
+ ao->i_stream_index = -1;
+ ao->b_revert = 0;
+ ao->b_changed_mixing = 0;
+
+ /* Probe whether device support S/PDIF stream output if input is AC3 stream. */
+ if ((format & AF_FORMAT_SPECIAL_MASK) == AF_FORMAT_AC3)
+ {
+ /* Find the ID of the default Device. */
+ i_param_size = sizeof(AudioDeviceID);
+ err = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice,
+ &i_param_size, &devid_def);
+ if (err != noErr)
+ {
+ ao_msg(MSGT_AO, MSGL_WARN, "could not get default audio device: [%4.4s]\n", (char *)&err);
+ return CONTROL_FALSE;
+ }
+
+ /* Retrieve the length of the device name. */
+ i_param_size = 0;
+ err = AudioDeviceGetPropertyInfo(devid_def, 0, 0,
+ kAudioDevicePropertyDeviceName,
+ &i_param_size, NULL);
+ if (err != noErr)
+ {
+ ao_msg(MSGT_AO, MSGL_WARN, "could not get default audio device name length: [%4.4s]\n", (char *)&err);
+ return CONTROL_FALSE;
+ }
+
+ /* Retrieve the name of the device. */
+ psz_name = (char *)malloc(i_param_size);
+ err = AudioDeviceGetProperty(devid_def, 0, 0,
+ kAudioDevicePropertyDeviceName,
+ &i_param_size, psz_name);
+ if (err != noErr)
+ {
+ ao_msg(MSGT_AO, MSGL_WARN, "could not get default audio device name: [%4.4s]\n", (char *)&err);
+ return CONTROL_FALSE;
+ }
+
+ ao_msg(MSGT_AO,MSGL_V, "got default audio output device ID: %#lx Name: %s\n", devid_def, psz_name );
+
+ if (AudioDeviceSupportsDigital(devid_def))
+ {
+ ao->b_supports_digital = 1;
+ ao->i_selected_dev = devid_def;
+ }
+ ao_msg(MSGT_AO,MSGL_V, "probe default audio output device whether support digital s/pdif output:%d\n", ao->b_supports_digital );
+
+ free( psz_name);
+ }
// Build Description for the input format
inDesc.mSampleRate=rate;
- inDesc.mFormatID=kAudioFormatLinearPCM;
+ inDesc.mFormatID=ao->b_supports_digital ? kAudioFormat60958AC3 : kAudioFormatLinearPCM;
inDesc.mChannelsPerFrame=channels;
switch(format&AF_FORMAT_BITS_MASK){
case AF_FORMAT_8BIT:
@@ -282,14 +389,54 @@ int aoIsCreated = ao != NULL;
// unsigned int
inDesc.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
+ inDesc.mFormatFlags |= kAudioFormatFlagIsBigEndian;
+#endif
+ }
+ else if ((format & AF_FORMAT_END_MASK) == AF_FORMAT_BE)
+ inDesc.mFormatFlags |= kAudioFormatFlagIsBigEndian;
inDesc.mFramesPerPacket = 1;
ao->packetSize = inDesc.mBytesPerPacket = inDesc.mBytesPerFrame = inDesc.mFramesPerPacket*channels*(inDesc.mBitsPerChannel/8);
- print_format("source: ",&inDesc);
+ print_format(MSGL_V, "source:",&inDesc);
+ if (ao->b_supports_digital)
+ {
+ b_alive = 1;
+ i_param_size = sizeof(b_alive);
+ err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE,
+ kAudioDevicePropertyDeviceIsAlive,
+ &i_param_size, &b_alive);
+ if (err != noErr)
+ ao_msg(MSGT_AO, MSGL_WARN, "could not check whether device is alive: [%4.4s]\n", (char *)&err);
+ if (!b_alive)
+ ao_msg(MSGT_AO, MSGL_WARN, "device is not alive\n" );
+ /* S/PDIF output need device in HogMode. */
+ i_param_size = sizeof(ao->i_hog_pid);
+ err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE,
+ kAudioDevicePropertyHogMode,
+ &i_param_size, &ao->i_hog_pid);
+
+ if (err != noErr)
+ {
+ /* This is not a fatal error. Some drivers simply don't support this property. */
+ ao_msg(MSGT_AO, MSGL_WARN, "could not check whether device is hogged: [%4.4s]\n",
+ (char *)&err);
+ ao->i_hog_pid = -1;
+ }
+
+ if (ao->i_hog_pid != -1 && ao->i_hog_pid != getpid())
+ {
+ ao_msg(MSGT_AO, MSGL_WARN, "Selected audio device is exclusively in use by another program.\n" );
+ return CONTROL_FALSE;
+ }
+ ao->stream_format = inDesc;
+ return OpenSPDIF();
+ }
+
+ /* original analog output code */
if (!aoIsCreated) {
desc.componentType = kAudioUnitType_Output;
desc.componentSubType = kAudioUnitSubType_DefaultOutput;
@@ -305,14 +452,14 @@ int aoIsCreated = ao != NULL;
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;
}
}
@@ -321,7 +468,7 @@ int aoIsCreated = ao != NULL;
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 (err=%d)\n", err);
+ ao_msg(MSGT_AO, MSGL_WARN, "Unable to set the input format: [%4.4s]\n", (char *)&err);
return CONTROL_FALSE;
}
@@ -330,7 +477,7 @@ int aoIsCreated = ao != NULL;
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;
}
@@ -353,7 +500,7 @@ int aoIsCreated = ao != NULL;
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 +509,489 @@ int aoIsCreated = ao != NULL;
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 */
@@ -402,16 +1025,63 @@ static float get_delay(void)
/* 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);
@@ -422,27 +1092,87 @@ static void uninit(int immed)
/* 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;
}
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