mirror of https://github.com/mpv-player/mpv
503 lines
19 KiB
C
503 lines
19 KiB
C
/*
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*
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* ao_macosx.c
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*
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* Original Copyright (C) Timothy J. Wood - Aug 2000
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*
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* This file is part of libao, a cross-platform library. See
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* README for a history of this source code.
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*
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* libao is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2, or (at your option)
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* any later version.
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*
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* libao is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with GNU Make; see the file COPYING. If not, write to
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* the Free Software Foundation, 675 Mass Ave, Cambridge, MA 02139, USA.
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*/
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/*
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* The MacOS X CoreAudio framework doesn't mesh as simply as some
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* simpler frameworks do. This is due to the fact that CoreAudio pulls
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* audio samples rather than having them pushed at it (which is nice
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* when you are wanting to do good buffering of audio).
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*/
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/* Change log:
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*
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* 14/5-2003: Ported to MPlayer libao2 by Dan Christiansen
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*
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* AC-3 and MPEG audio passthrough is possible, but I don't have
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* access to a sound card that supports it.
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*/
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#include <CoreAudio/AudioHardware.h>
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#include <stdio.h>
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#include <string.h>
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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 "config.h"
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#include "mp_msg.h"
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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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static ao_info_t info =
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{
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"Darwin/Mac OS X native audio output",
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"macosx",
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"Timothy J. Wood & Dan Christiansen",
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""
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};
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LIBAO_EXTERN(macosx)
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/* Prefix for all mp_msg() calls */
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#define ao_msg(a, b, c...) mp_msg(a, b, "AO: [macosx] " c)
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/* This is large, but best (maybe it should be even larger).
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* CoreAudio supposedly has an internal latency in the order of 2ms */
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#define NUM_BUFS 32
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typedef struct ao_macosx_s
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{
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/* CoreAudio */
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AudioDeviceID outputDeviceID;
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AudioStreamBasicDescription outputStreamBasicDescription;
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/* Ring-buffer */
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/* does not need explicit synchronization, but needs to allocate
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* (num_chunks + 1) * chunk_size memory to store num_chunks * chunk_size
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* data */
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unsigned char *buffer;
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unsigned int buffer_len; ///< must always be (num_chunks + 1) * chunk_size
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unsigned int num_chunks;
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unsigned int chunk_size;
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unsigned int buf_read_pos;
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unsigned int buf_write_pos;
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} ao_macosx_t;
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static ao_macosx_t *ao;
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/**
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* \brief return number of free bytes in the buffer
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* may only be called by mplayer's thread
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* \return minimum number of free bytes in buffer, value may change between
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* two immediately following calls, and the real number of free bytes
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* might actually be larger!
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*/
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static int buf_free() {
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int free = ao->buf_read_pos - ao->buf_write_pos - ao->chunk_size;
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if (free < 0) free += ao->buffer_len;
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return free;
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}
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/**
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* \brief return number of buffered bytes
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* may only be called by playback thread
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* \return minimum number of buffered bytes, value may change between
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* two immediately following calls, and the real number of buffered bytes
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* might actually be larger!
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*/
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static int buf_used() {
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int used = ao->buf_write_pos - ao->buf_read_pos;
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if (used < 0) used += ao->buffer_len;
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return used;
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}
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/**
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* \brief add data to ringbuffer
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*/
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static int write_buffer(unsigned char* data, int len){
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int first_len = ao->buffer_len - ao->buf_write_pos;
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int free = buf_free();
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if (len > free) len = free;
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if (first_len > len) first_len = len;
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// till end of buffer
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memcpy (&ao->buffer[ao->buf_write_pos], data, first_len);
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if (len > first_len) { // we have to wrap around
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// remaining part from beginning of buffer
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memcpy (ao->buffer, &data[first_len], len - first_len);
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}
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ao->buf_write_pos = (ao->buf_write_pos + len) % ao->buffer_len;
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return len;
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}
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/**
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* \brief remove data from ringbuffer
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*/
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static int read_buffer(unsigned char* data,int len){
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int first_len = ao->buffer_len - ao->buf_read_pos;
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int buffered = buf_used();
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if (len > buffered) len = buffered;
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if (first_len > len) first_len = len;
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// till end of buffer
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memcpy (data, &ao->buffer[ao->buf_read_pos], first_len);
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if (len > first_len) { // we have to wrap around
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// remaining part from beginning of buffer
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memcpy (&data[first_len], ao->buffer, len - first_len);
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}
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ao->buf_read_pos = (ao->buf_read_pos + len) % ao->buffer_len;
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return len;
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}
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/* end ring buffer stuff */
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/* The function that the CoreAudio thread calls when it wants more data */
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static OSStatus audioDeviceIOProc(AudioDeviceID inDevice, const AudioTimeStamp *inNow, const AudioBufferList *inInputData, const AudioTimeStamp *inInputTime, AudioBufferList *outOutputData, const AudioTimeStamp *inOutputTime, void *inClientData)
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{
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outOutputData->mBuffers[0].mDataByteSize =
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read_buffer((char *)outOutputData->mBuffers[0].mData, ao->chunk_size);
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return 0;
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}
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static int control(int cmd,void *arg){
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OSStatus status;
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UInt32 propertySize;
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ao_control_vol_t* vol = (ao_control_vol_t*)arg;
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UInt32 stereoChannels[2];
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static float volume=0.5;
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switch (cmd) {
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case AOCONTROL_SET_DEVICE:
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case AOCONTROL_GET_DEVICE:
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/* unimplemented/meaningless */
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return CONTROL_FALSE;
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case AOCONTROL_GET_VOLUME:
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propertySize=sizeof(stereoChannels);
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status = AudioDeviceGetProperty(ao->outputDeviceID, NULL, 0,
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kAudioDevicePropertyPreferredChannelsForStereo, &propertySize,
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&stereoChannels);
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// printf("OSX: stereochannels %d ; %d \n",stereoChannels[0],stereoChannels[1]);
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propertySize=sizeof(volume);
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status = AudioDeviceGetProperty(ao->outputDeviceID, stereoChannels[0], false, kAudioDevicePropertyVolumeScalar, &propertySize, &volume);
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// printf("OSX: get volume=%5.3f status=%d \n",volume,status);
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vol->left=(int)(volume*100.0);
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status = AudioDeviceGetProperty(ao->outputDeviceID, stereoChannels[1], false, kAudioDevicePropertyVolumeScalar, &propertySize, &volume);
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vol->right=(int)(volume*100.0);
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return CONTROL_TRUE;
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case AOCONTROL_SET_VOLUME:
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propertySize=sizeof(stereoChannels);
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status = AudioDeviceGetProperty(ao->outputDeviceID, NULL, 0,
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kAudioDevicePropertyPreferredChannelsForStereo, &propertySize,
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&stereoChannels);
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// printf("OSX: stereochannels %d ; %d \n",stereoChannels[0],stereoChannels[1]);
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propertySize=sizeof(volume);
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volume=vol->left/100.0;
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status = AudioDeviceSetProperty(ao->outputDeviceID, 0, stereoChannels[0], 0, kAudioDevicePropertyVolumeScalar, propertySize, &volume);
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// printf("OSX: set volume=%5.3f status=%d\n",volume,status);
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volume=vol->right/100.0;
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status = AudioDeviceSetProperty(ao->outputDeviceID, 0, stereoChannels[1], 0, kAudioDevicePropertyVolumeScalar, propertySize, &volume);
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return CONTROL_TRUE;
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case AOCONTROL_QUERY_FORMAT:
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/* stick with what CoreAudio requests */
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return CONTROL_FALSE;
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default:
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return CONTROL_FALSE;
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}
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}
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static void print_format(const char* str,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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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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(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 init(int rate,int channels,int format,int flags)
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{
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OSStatus status;
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UInt32 propertySize;
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int rc;
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int i;
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ao = (ao_macosx_t *)malloc(sizeof(ao_macosx_t));
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/* get default output device */
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propertySize = sizeof(ao->outputDeviceID);
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status = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice, &propertySize, &(ao->outputDeviceID));
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if (status) {
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ao_msg(MSGT_AO,MSGL_WARN,
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"AudioHardwareGetProperty returned %d\n",
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(int)status);
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return CONTROL_FALSE;
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}
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if (ao->outputDeviceID == kAudioDeviceUnknown) {
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ao_msg(MSGT_AO,MSGL_WARN, "AudioHardwareGetProperty: ao->outputDeviceID is kAudioDeviceUnknown\n");
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return CONTROL_FALSE;
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}
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propertySize = sizeof(ao->outputStreamBasicDescription);
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status = AudioDeviceGetProperty(ao->outputDeviceID, 0, false, kAudioDevicePropertyStreamFormat, &propertySize, &ao->outputStreamBasicDescription);
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if(!status) print_format("default:",&ao->outputStreamBasicDescription);
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#if 1
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// dump supported format list:
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{ AudioStreamBasicDescription* p;
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Boolean ow;
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int i;
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propertySize=0; //sizeof(p);
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// status = AudioDeviceGetPropertyInfo(ao->outputDeviceID, 0, false, kAudioStreamPropertyPhysicalFormats, &propertySize, &ow);
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status = AudioDeviceGetPropertyInfo(ao->outputDeviceID, 0, false, kAudioDevicePropertyStreamFormats, &propertySize, &ow);
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if (status) {
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ao_msg(MSGT_AO,MSGL_WARN, "AudioDeviceGetPropertyInfo returned 0x%X when getting kAudioDevicePropertyStreamFormats\n", (int)status);
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}
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p=malloc(propertySize);
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// status = AudioDeviceGetProperty(ao->outputDeviceID, 0, false, kAudioStreamPropertyPhysicalFormats, &propertySize, p);
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status = AudioDeviceGetProperty(ao->outputDeviceID, 0, false, kAudioDevicePropertyStreamFormats, &propertySize, p);
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if (status) {
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ao_msg(MSGT_AO,MSGL_WARN, "AudioDeviceGetProperty returned 0x%X when getting kAudioDevicePropertyStreamFormats\n", (int)status);
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// return CONTROL_FALSE;
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}
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for(i=0;i<propertySize/sizeof(AudioStreamBasicDescription);i++)
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print_format("support:",&p[i]);
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// printf("FORMATS: (%d) %p %p %p %p\n",propertySize,p[0],p[1],p[2],p[3]);
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free(p);
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}
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#endif
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// fill in our wanted format, and let's see if the driver accepts it or
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// offers some similar alternative:
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propertySize = sizeof(ao->outputStreamBasicDescription);
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memset(&ao->outputStreamBasicDescription,0,propertySize);
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ao->outputStreamBasicDescription.mSampleRate=rate;
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ao->outputStreamBasicDescription.mFormatID=kAudioFormatLinearPCM;
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ao->outputStreamBasicDescription.mChannelsPerFrame=channels;
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switch(format&AF_FORMAT_BITS_MASK){
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case AF_FORMAT_8BIT: ao->outputStreamBasicDescription.mBitsPerChannel=8; break;
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case AF_FORMAT_16BIT: ao->outputStreamBasicDescription.mBitsPerChannel=16; break;
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case AF_FORMAT_24BIT: ao->outputStreamBasicDescription.mBitsPerChannel=24; break;
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case AF_FORMAT_32BIT: ao->outputStreamBasicDescription.mBitsPerChannel=32; break;
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}
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if((format&AF_FORMAT_POINT_MASK)==AF_FORMAT_F){
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// float
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ao->outputStreamBasicDescription.mFormatFlags=kAudioFormatFlagIsFloat|kAudioFormatFlagIsPacked;
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} else if((format&AF_FORMAT_SIGN_MASK)==AF_FORMAT_SI){
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// signed int
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ao->outputStreamBasicDescription.mFormatFlags=kAudioFormatFlagIsSignedInteger|kAudioFormatFlagIsPacked;
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} else {
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// unsigned int
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ao->outputStreamBasicDescription.mFormatFlags=kAudioFormatFlagIsPacked;
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}
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if((format&AF_FORMAT_END_MASK)==AF_FORMAT_BE)
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ao->outputStreamBasicDescription.mFormatFlags|=kAudioFormatFlagIsBigEndian;
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ao->outputStreamBasicDescription.mBytesPerPacket=
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ao->outputStreamBasicDescription.mBytesPerFrame=channels*(ao->outputStreamBasicDescription.mBitsPerChannel/8);
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ao->outputStreamBasicDescription.mFramesPerPacket=1;
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print_format("wanted: ",&ao->outputStreamBasicDescription);
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// try 1: ask if it accepts our specific requirements?
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propertySize = sizeof(ao->outputStreamBasicDescription);
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// status = AudioDeviceGetProperty(ao->outputDeviceID, 0, false, kAudioStreamPropertyPhysicalFormatMatch, &propertySize, &ao->outputStreamBasicDescription);
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status = AudioDeviceGetProperty(ao->outputDeviceID, 0, false, kAudioDevicePropertyStreamFormatMatch, &propertySize, &ao->outputStreamBasicDescription);
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if (status || ao->outputStreamBasicDescription.mSampleRate!=rate
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|| ao->outputStreamBasicDescription.mFormatID!=kAudioFormatLinearPCM) {
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ao_msg(MSGT_AO,MSGL_V, "AudioDeviceGetProperty returned 0x%X when getting kAudioDevicePropertyStreamFormatMatch\n", (int)status);
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// failed (error, bad rate or bad type)
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// try 2: set only rate & type, no format details (bits, channels etc)
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propertySize = sizeof(ao->outputStreamBasicDescription);
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memset(&ao->outputStreamBasicDescription,0,propertySize);
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ao->outputStreamBasicDescription.mSampleRate=rate;
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ao->outputStreamBasicDescription.mFormatID=kAudioFormatLinearPCM;
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// status = AudioDeviceGetProperty(ao->outputDeviceID, 0, false, kAudioStreamPropertyPhysicalFormatMatch, &propertySize, &ao->outputStreamBasicDescription);
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status = AudioDeviceGetProperty(ao->outputDeviceID, 0, false, kAudioDevicePropertyStreamFormatMatch, &propertySize, &ao->outputStreamBasicDescription);
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if (status || ao->outputStreamBasicDescription.mFormatID!=kAudioFormatLinearPCM) {
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ao_msg(MSGT_AO,MSGL_V, "AudioDeviceGetProperty returned 0x%X when getting kAudioDevicePropertyStreamFormatMatch\n", (int)status);
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// failed again. (error or bad type)
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// giving up... just read the default.
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propertySize = sizeof(ao->outputStreamBasicDescription);
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// status = AudioDeviceGetProperty(ao->outputDeviceID, 0, false, kAudioStreamPropertyPhysicalFormat, &propertySize, &ao->outputStreamBasicDescription);
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status = AudioDeviceGetProperty(ao->outputDeviceID, 0, false, kAudioDevicePropertyStreamFormat, &propertySize, &ao->outputStreamBasicDescription);
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if (status) {
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// failed to read the default format - WTF?
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ao_msg(MSGT_AO,MSGL_WARN, "AudioDeviceGetProperty returned 0x%X when getting kAudioDevicePropertyStreamFormat\n", (int)status);
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return CONTROL_FALSE;
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}
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}
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}
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// propertySize = sizeof(ao->outputStreamBasicDescription);
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// status = AudioDeviceGetProperty(ao->outputDeviceID, 0, false, kAudioDevicePropertyStreamFormatSupported, &propertySize, &ao->outputStreamBasicDescription);
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// if (status) {
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// ao_msg(MSGT_AO,MSGL_V, "AudioDeviceGetProperty returned 0x%X when getting kAudioDevicePropertyStreamFormatSupported\n", (int)status);
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// }
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// ok, now try to set the new (default or matched) audio format:
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print_format("best: ",&ao->outputStreamBasicDescription);
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propertySize = sizeof(ao->outputStreamBasicDescription);
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status = AudioDeviceSetProperty(ao->outputDeviceID, 0, 0, false, kAudioDevicePropertyStreamFormat, propertySize, &ao->outputStreamBasicDescription);
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if(status)
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ao_msg(MSGT_AO,MSGL_WARN, "AudioDeviceSetProperty returned 0x%X when getting kAudioDevicePropertyStreamFormat\n", (int)status);
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// see what did we get finally... we'll be forced to use this anyway :(
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propertySize = sizeof(ao->outputStreamBasicDescription);
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status = AudioDeviceGetProperty(ao->outputDeviceID, 0, false, kAudioDevicePropertyStreamFormat, &propertySize, &ao->outputStreamBasicDescription);
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print_format("final: ",&ao->outputStreamBasicDescription);
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/* get requested buffer length */
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// TODO: set NUM_BUFS dinamically, based on buffer size!
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propertySize = sizeof(ao->chunk_size);
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status = AudioDeviceGetProperty(ao->outputDeviceID, 0, false, kAudioDevicePropertyBufferSize, &propertySize, &ao->chunk_size);
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if (status) {
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ao_msg(MSGT_AO,MSGL_WARN, "AudioDeviceGetProperty returned %d when getting kAudioDevicePropertyBufferSize\n", (int)status);
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return CONTROL_FALSE;
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}
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ao_msg(MSGT_AO,MSGL_V, "%5d chunk size\n", (int)ao->chunk_size);
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ao_data.samplerate = ao->outputStreamBasicDescription.mSampleRate;
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ao_data.channels = channels;
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ao_data.outburst = ao_data.buffersize = ao->chunk_size;
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ao_data.bps =
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ao_data.samplerate * ao->outputStreamBasicDescription.mBytesPerFrame;
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if (ao->outputStreamBasicDescription.mFormatID == kAudioFormatLinearPCM) {
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uint32_t flags = ao->outputStreamBasicDescription.mFormatFlags;
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if (flags & kAudioFormatFlagIsFloat) {
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ao_data.format = (flags&kAudioFormatFlagIsBigEndian) ? AF_FORMAT_FLOAT_BE : AF_FORMAT_FLOAT_LE;
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} else {
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ao_msg(MSGT_AO,MSGL_WARN, "Unsupported audio output "
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"format 0x%X. Please report this to the developer\n", format);
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return CONTROL_FALSE;
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}
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} else {
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/* TODO: handle AFMT_AC3, AFMT_MPEG & friends */
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ao_msg(MSGT_AO,MSGL_WARN, "Default Audio Device doesn't "
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"support Linear PCM!\n");
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return CONTROL_FALSE;
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}
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/* Allocate ring-buffer memory */
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ao->num_chunks = NUM_BUFS;
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ao->buffer_len = (ao->num_chunks + 1) * ao->chunk_size;
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ao->buffer = (unsigned char *)malloc(ao->buffer_len);
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/* Prepare for playback */
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/* Set the IO proc that CoreAudio will call when it needs data */
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status = AudioDeviceAddIOProc(ao->outputDeviceID, audioDeviceIOProc, NULL);
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if (status) {
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ao_msg(MSGT_AO,MSGL_WARN, "AudioDeviceAddIOProc returned %d\n", (int)status);
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return CONTROL_FALSE;
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}
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/* Start callback */
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reset();
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|
return CONTROL_OK;
|
|
}
|
|
|
|
|
|
static int play(void* output_samples,int num_bytes,int flags)
|
|
{
|
|
return write_buffer(output_samples, num_bytes);
|
|
}
|
|
|
|
/* set variables and buffer to initial state */
|
|
static void reset()
|
|
{
|
|
audio_pause();
|
|
/* reset ring-buffer state */
|
|
ao->buf_read_pos=0;
|
|
ao->buf_write_pos=0;
|
|
audio_resume();
|
|
|
|
return;
|
|
}
|
|
|
|
|
|
/* return available space */
|
|
static int get_space()
|
|
{
|
|
return buf_free();
|
|
}
|
|
|
|
|
|
/* return delay until audio is played */
|
|
static float get_delay()
|
|
{
|
|
int buffered = ao->buffer_len - ao->chunk_size - buf_free(); // could be less
|
|
// inaccurate, should also contain the data buffered e.g. by the OS
|
|
return (float)(buffered)/(float)ao_data.bps;
|
|
}
|
|
|
|
|
|
/* unload plugin and deregister from coreaudio */
|
|
static void uninit(int immed)
|
|
{
|
|
int i;
|
|
OSErr status;
|
|
|
|
reset();
|
|
|
|
status = AudioDeviceRemoveIOProc(ao->outputDeviceID, audioDeviceIOProc);
|
|
if (status)
|
|
ao_msg(MSGT_AO,MSGL_WARN, "AudioDeviceRemoveIOProc "
|
|
"returned %d\n", (int)status);
|
|
|
|
free(ao->buffer);
|
|
free(ao);
|
|
}
|
|
|
|
|
|
/* stop playing, keep buffers (for pause) */
|
|
static void audio_pause()
|
|
{
|
|
OSErr status;
|
|
|
|
/* stop callback */
|
|
status = AudioDeviceStop(ao->outputDeviceID, audioDeviceIOProc);
|
|
if (status)
|
|
ao_msg(MSGT_AO,MSGL_WARN, "AudioDeviceStop returned %d\n",
|
|
(int)status);
|
|
}
|
|
|
|
|
|
/* resume playing, after audio_pause() */
|
|
static void audio_resume()
|
|
{
|
|
OSErr status = AudioDeviceStart(ao->outputDeviceID, audioDeviceIOProc);
|
|
if (status)
|
|
ao_msg(MSGT_AO,MSGL_WARN, "AudioDeviceStart returned %d\n",
|
|
(int)status);
|
|
}
|