mirror of https://github.com/mpv-player/mpv
Add support for AC-3/DTS passthrough.
patch by Ulion, ulion2002 gmail com git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@24762 b3059339-0415-0410-9bf9-f77b7e298cf2
This commit is contained in:
nplourde
parent
2aa0a0cfc3
commit
3ec7b533fd
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@ -45,6 +45,8 @@
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#include <stdlib.h>
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#include <inttypes.h>
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#include <pthread.h>
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#include <sys/types.h>
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#include <unistd.h>
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#include "config.h"
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#include "mp_msg.h"
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@ -52,6 +54,7 @@
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#include "audio_out.h"
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#include "audio_out_internal.h"
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#include "libaf/af_format.h"
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#include "osdep/timer.h"
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static ao_info_t info =
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{
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@ -72,8 +75,24 @@ LIBAO_EXTERN(macosx)
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typedef struct ao_macosx_s
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{
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AudioDeviceID i_selected_dev; /* Keeps DeviceID of the selected device. */
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int b_supports_digital; /* Does the currently selected device support digital mode? */
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int b_digital; /* Are we running in digital mode? */
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/* AudioUnit */
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AudioUnit theOutputUnit;
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/* CoreAudio SPDIF mode specific */
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pid_t i_hog_pid; /* Keeps the pid of our hog status. */
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AudioStreamID i_stream_id; /* The StreamID that has a cac3 streamformat */
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int i_stream_index; /* The index of i_stream_id in an AudioBufferList */
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AudioStreamBasicDescription stream_format;/* The format we changed the stream to */
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AudioStreamBasicDescription sfmt_revert; /* The original format of the stream */
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int b_revert; /* Whether we need to revert the stream format */
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int b_changed_mixing; /* Whether we need to set the mixing mode back */
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int b_stream_format_changed; /* Flag for main thread to reset stream's format to digital and reset buffer */
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/* Original common part */
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int packetSize;
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int paused;
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@ -177,6 +196,10 @@ Float32 vol;
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switch (cmd) {
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case AOCONTROL_GET_VOLUME:
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control_vol = (ao_control_vol_t*)arg;
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if (ao->b_digital) {
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// Digital output has no volume adjust.
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return CONTROL_FALSE;
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}
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err = AudioUnitGetParameter(ao->theOutputUnit, kHALOutputParam_Volume, kAudioUnitScope_Global, 0, &vol);
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if(err==0) {
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@ -185,10 +208,17 @@ Float32 vol;
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return CONTROL_TRUE;
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}
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else {
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ao_msg(MSGT_AO, MSGL_WARN, "could not get HAL output volume: [%4.4s]\n", (char *)&err);
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return CONTROL_FALSE;
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}
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case AOCONTROL_SET_VOLUME:
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if (ao->b_digital)
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// Digital output can not set volume. Here we have to return true
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// to make mixer forget it. Else mixer will add a soft filter,
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// that's not we expected and the filter not support ac3 stream
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// will cause mplayer die.
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return CONTROL_TRUE;
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control_vol = (ao_control_vol_t*)arg;
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vol=(control_vol->left+control_vol->right)*4.0/200.0;
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@ -198,6 +228,7 @@ Float32 vol;
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return CONTROL_TRUE;
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}
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else {
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ao_msg(MSGT_AO, MSGL_WARN, "could not set HAL output volume: [%4.4s]\n", (char *)&err);
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return CONTROL_FALSE;
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}
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/* Everything is currently unimplemented */
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@ -208,33 +239,47 @@ Float32 vol;
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}
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static void print_format(const char* str,AudioStreamBasicDescription *f){
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static void print_format(int lev, const char* str, const AudioStreamBasicDescription *f){
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uint32_t flags=(uint32_t) f->mFormatFlags;
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ao_msg(MSGT_AO,MSGL_V, "%s %7.1fHz %dbit [%c%c%c%c] %s %s %s%s%s%s\n",
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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",
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str, f->mSampleRate, f->mBitsPerChannel,
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(int)(f->mFormatID & 0xff000000) >> 24,
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(int)(f->mFormatID & 0x00ff0000) >> 16,
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(int)(f->mFormatID & 0x0000ff00) >> 8,
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(int)(f->mFormatID & 0x000000ff) >> 0,
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f->mFormatFlags, f->mBytesPerPacket,
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f->mFramesPerPacket, f->mBytesPerFrame,
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f->mChannelsPerFrame,
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(flags&kAudioFormatFlagIsFloat) ? "float" : "int",
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(flags&kAudioFormatFlagIsBigEndian) ? "BE" : "LE",
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(flags&kAudioFormatFlagIsSignedInteger) ? "S" : "U",
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(flags&kAudioFormatFlagIsPacked) ? " packed" : "",
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(flags&kAudioFormatFlagIsAlignedHigh) ? " aligned" : "",
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(flags&kAudioFormatFlagIsNonInterleaved) ? " ni" : "" );
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ao_msg(MSGT_AO,MSGL_DBG2, "%5d mBytesPerPacket\n",
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(int)f->mBytesPerPacket);
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ao_msg(MSGT_AO,MSGL_DBG2, "%5d mFramesPerPacket\n",
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(int)f->mFramesPerPacket);
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ao_msg(MSGT_AO,MSGL_DBG2, "%5d mBytesPerFrame\n",
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(int)f->mBytesPerFrame);
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ao_msg(MSGT_AO,MSGL_DBG2, "%5d mChannelsPerFrame\n",
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(int)f->mChannelsPerFrame);
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}
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static int AudioDeviceSupportsDigital( AudioDeviceID i_dev_id );
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static int AudioStreamSupportsDigital( AudioStreamID i_stream_id );
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static int OpenSPDIF();
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static int AudioStreamChangeFormat( AudioStreamID i_stream_id, AudioStreamBasicDescription change_format );
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static OSStatus RenderCallbackSPDIF( AudioDeviceID inDevice,
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const AudioTimeStamp * inNow,
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const void * inInputData,
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const AudioTimeStamp * inInputTime,
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AudioBufferList * outOutputData,
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const AudioTimeStamp * inOutputTime,
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void * threadGlobals );
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static OSStatus StreamListener( AudioStreamID inStream,
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UInt32 inChannel,
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AudioDevicePropertyID inPropertyID,
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void * inClientData );
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static OSStatus DeviceListener( AudioDeviceID inDevice,
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UInt32 inChannel,
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Boolean isInput,
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AudioDevicePropertyID inPropertyID,
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void* inClientData );
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static int init(int rate,int channels,int format,int flags)
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{
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AudioStreamBasicDescription inDesc;
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@ -242,14 +287,76 @@ ComponentDescription desc;
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Component comp;
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AURenderCallbackStruct renderCallback;
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OSStatus err;
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UInt32 size, maxFrames;
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UInt32 size, maxFrames, i_param_size;
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char *psz_name;
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int aoIsCreated = ao != NULL;
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AudioDeviceID devid_def = 0;
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int b_alive;
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if (!aoIsCreated) ao = malloc(sizeof(ao_macosx_t));
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ao_msg(MSGT_AO,MSGL_V, "init([%dHz][%dch][%s][%d])\n", rate, channels, af_fmt2str_short(format), flags);
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if (!aoIsCreated) { ao = malloc(sizeof(ao_macosx_t)); ao->buffer = NULL;}
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ao->i_selected_dev = 0;
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ao->b_supports_digital = 0;
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ao->b_digital = 0;
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ao->b_stream_format_changed = 0;
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ao->i_hog_pid = -1;
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ao->i_stream_id = 0;
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ao->i_stream_index = -1;
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ao->b_revert = 0;
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ao->b_changed_mixing = 0;
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/* Probe whether device support S/PDIF stream output if input is AC3 stream. */
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if ((format & AF_FORMAT_SPECIAL_MASK) == AF_FORMAT_AC3)
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{
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/* Find the ID of the default Device. */
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i_param_size = sizeof(AudioDeviceID);
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err = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice,
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&i_param_size, &devid_def);
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if (err != noErr)
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{
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ao_msg(MSGT_AO, MSGL_WARN, "could not get default audio device: [%4.4s]\n", (char *)&err);
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return CONTROL_FALSE;
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}
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/* Retrieve the length of the device name. */
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i_param_size = 0;
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err = AudioDeviceGetPropertyInfo(devid_def, 0, 0,
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kAudioDevicePropertyDeviceName,
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&i_param_size, NULL);
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if (err != noErr)
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{
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ao_msg(MSGT_AO, MSGL_WARN, "could not get default audio device name length: [%4.4s]\n", (char *)&err);
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return CONTROL_FALSE;
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}
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/* Retrieve the name of the device. */
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psz_name = (char *)malloc(i_param_size);
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err = AudioDeviceGetProperty(devid_def, 0, 0,
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kAudioDevicePropertyDeviceName,
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&i_param_size, psz_name);
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if (err != noErr)
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{
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ao_msg(MSGT_AO, MSGL_WARN, "could not get default audio device name: [%4.4s]\n", (char *)&err);
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return CONTROL_FALSE;
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}
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ao_msg(MSGT_AO,MSGL_V, "got default audio output device ID: %#lx Name: %s\n", devid_def, psz_name );
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if (AudioDeviceSupportsDigital(devid_def))
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{
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ao->b_supports_digital = 1;
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ao->i_selected_dev = devid_def;
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}
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ao_msg(MSGT_AO,MSGL_V, "probe default audio output device whether support digital s/pdif output:%d\n", ao->b_supports_digital );
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free( psz_name);
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}
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// Build Description for the input format
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inDesc.mSampleRate=rate;
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inDesc.mFormatID=kAudioFormatLinearPCM;
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inDesc.mFormatID=ao->b_supports_digital ? kAudioFormat60958AC3 : kAudioFormatLinearPCM;
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inDesc.mChannelsPerFrame=channels;
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switch(format&AF_FORMAT_BITS_MASK){
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case AF_FORMAT_8BIT:
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@ -282,14 +389,54 @@ int aoIsCreated = ao != NULL;
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// unsigned int
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inDesc.mFormatFlags = kAudioFormatFlagIsPacked;
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}
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if((format&AF_FORMAT_END_MASK)==AF_FORMAT_BE)
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inDesc.mFormatFlags |= kAudioFormatFlagIsBigEndian;
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if ((format & AF_FORMAT_SPECIAL_MASK) == AF_FORMAT_AC3) {
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// Currently ac3 input (comes from hwac3) is always in native byte-order.
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#ifdef WORDS_BIGENDIAN
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inDesc.mFormatFlags |= kAudioFormatFlagIsBigEndian;
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#endif
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}
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else if ((format & AF_FORMAT_END_MASK) == AF_FORMAT_BE)
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inDesc.mFormatFlags |= kAudioFormatFlagIsBigEndian;
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inDesc.mFramesPerPacket = 1;
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ao->packetSize = inDesc.mBytesPerPacket = inDesc.mBytesPerFrame = inDesc.mFramesPerPacket*channels*(inDesc.mBitsPerChannel/8);
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print_format("source: ",&inDesc);
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print_format(MSGL_V, "source:",&inDesc);
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if (ao->b_supports_digital)
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{
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b_alive = 1;
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i_param_size = sizeof(b_alive);
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err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE,
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kAudioDevicePropertyDeviceIsAlive,
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&i_param_size, &b_alive);
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if (err != noErr)
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ao_msg(MSGT_AO, MSGL_WARN, "could not check whether device is alive: [%4.4s]\n", (char *)&err);
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if (!b_alive)
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ao_msg(MSGT_AO, MSGL_WARN, "device is not alive\n" );
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/* S/PDIF output need device in HogMode. */
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i_param_size = sizeof(ao->i_hog_pid);
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err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE,
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kAudioDevicePropertyHogMode,
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&i_param_size, &ao->i_hog_pid);
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if (err != noErr)
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{
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/* This is not a fatal error. Some drivers simply don't support this property. */
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ao_msg(MSGT_AO, MSGL_WARN, "could not check whether device is hogged: [%4.4s]\n",
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(char *)&err);
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ao->i_hog_pid = -1;
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}
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if (ao->i_hog_pid != -1 && ao->i_hog_pid != getpid())
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{
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ao_msg(MSGT_AO, MSGL_WARN, "Selected audio device is exclusively in use by another program.\n" );
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return CONTROL_FALSE;
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}
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ao->stream_format = inDesc;
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return OpenSPDIF();
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}
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/* original analog output code */
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if (!aoIsCreated) {
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desc.componentType = kAudioUnitType_Output;
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desc.componentSubType = kAudioUnitSubType_DefaultOutput;
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@ -305,14 +452,14 @@ int aoIsCreated = ao != NULL;
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err = OpenAComponent(comp, &(ao->theOutputUnit)); //gains access to the services provided by the component
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if (err) {
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ao_msg(MSGT_AO, MSGL_WARN, "Unable to open Output Unit component (err=%d)\n", err);
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ao_msg(MSGT_AO, MSGL_WARN, "Unable to open Output Unit component: [%4.4s]\n", (char *)&err);
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return CONTROL_FALSE;
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}
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// Initialize AudioUnit
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err = AudioUnitInitialize(ao->theOutputUnit);
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if (err) {
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ao_msg(MSGT_AO, MSGL_WARN, "Unable to initialize Output Unit component (err=%d)\n", err);
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ao_msg(MSGT_AO, MSGL_WARN, "Unable to initialize Output Unit component: [%4.4s]\n", (char *)&err);
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return CONTROL_FALSE;
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}
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}
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@ -321,7 +468,7 @@ int aoIsCreated = ao != NULL;
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err = AudioUnitSetProperty(ao->theOutputUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &inDesc, size);
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if (err) {
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ao_msg(MSGT_AO, MSGL_WARN, "Unable to set the input format (err=%d)\n", err);
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ao_msg(MSGT_AO, MSGL_WARN, "Unable to set the input format: [%4.4s]\n", (char *)&err);
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return CONTROL_FALSE;
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}
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@ -330,7 +477,7 @@ int aoIsCreated = ao != NULL;
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if (err)
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{
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ao_msg(MSGT_AO,MSGL_WARN, "AudioUnitGetProperty returned %d when getting kAudioDevicePropertyBufferSize\n", (int)err);
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ao_msg(MSGT_AO,MSGL_WARN, "AudioUnitGetProperty returned [%4.4s] when getting kAudioDevicePropertyBufferSize\n", (char *)&err);
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return CONTROL_FALSE;
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}
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@ -353,7 +500,7 @@ int aoIsCreated = ao != NULL;
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renderCallback.inputProcRefCon = 0;
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err = AudioUnitSetProperty(ao->theOutputUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, 0, &renderCallback, sizeof(AURenderCallbackStruct));
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if (err) {
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ao_msg(MSGT_AO, MSGL_WARN, "Unable to set the render callback (err=%d)\n", err);
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ao_msg(MSGT_AO, MSGL_WARN, "Unable to set the render callback: [%4.4s]\n", (char *)&err);
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return CONTROL_FALSE;
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}
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@ -362,13 +509,489 @@ int aoIsCreated = ao != NULL;
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return CONTROL_OK;
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}
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/*****************************************************************************
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* Setup a encoded digital stream (SPDIF)
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*****************************************************************************/
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static int OpenSPDIF()
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{
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OSStatus err = noErr;
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UInt32 i_param_size, b_mix = 0;
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Boolean b_writeable = 0;
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AudioStreamID *p_streams = NULL;
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int i, i_streams = 0;
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/* Start doing the SPDIF setup process. */
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ao->b_digital = 1;
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/* Hog the device. */
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i_param_size = sizeof(ao->i_hog_pid);
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ao->i_hog_pid = getpid() ;
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err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE,
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kAudioDevicePropertyHogMode, i_param_size, &ao->i_hog_pid);
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if (err != noErr)
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{
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ao_msg(MSGT_AO, MSGL_WARN, "failed to set hogmode: [%4.4s]\n", (char *)&err);
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return CONTROL_FALSE;
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}
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/* Set mixable to false if we are allowed to. */
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err = AudioDeviceGetPropertyInfo(ao->i_selected_dev, 0, FALSE,
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kAudioDevicePropertySupportsMixing,
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&i_param_size, &b_writeable);
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err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE,
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kAudioDevicePropertySupportsMixing,
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&i_param_size, &b_mix);
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if (err != noErr && b_writeable)
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{
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b_mix = 0;
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err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE,
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kAudioDevicePropertySupportsMixing,
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i_param_size, &b_mix);
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ao->b_changed_mixing = 1;
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}
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if (err != noErr)
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{
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ao_msg(MSGT_AO, MSGL_WARN, "failed to set mixmode: [%4.4s]\n", (char *)&err);
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return CONTROL_FALSE;
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}
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/* Get a list of all the streams on this device. */
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err = AudioDeviceGetPropertyInfo(ao->i_selected_dev, 0, FALSE,
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kAudioDevicePropertyStreams,
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&i_param_size, NULL);
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if (err != noErr)
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{
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ao_msg(MSGT_AO, MSGL_WARN, "could not get number of streams: [%4.4s]\n", (char *)&err);
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return CONTROL_FALSE;
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}
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i_streams = i_param_size / sizeof(AudioStreamID);
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p_streams = (AudioStreamID *)malloc(i_param_size);
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if (p_streams == NULL)
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{
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ao_msg(MSGT_AO, MSGL_WARN, "out of memory\n" );
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return CONTROL_FALSE;
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}
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err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE,
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kAudioDevicePropertyStreams,
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&i_param_size, p_streams);
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if (err != noErr)
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{
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ao_msg(MSGT_AO, MSGL_WARN, "could not get number of streams: [%4.4s]\n", (char *)&err);
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if (p_streams) free(p_streams);
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return CONTROL_FALSE;
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}
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|
||||
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;
|
||||
}
|
||||
|
|
|
|||
Loading…
Reference in New Issue