Mercurial > mplayer.hg
view libao2/ao_macosx.c @ 15069:3f5daa60e049
support for negative strides (fixes -vf spp,flip crash)
author | henry |
---|---|
date | Fri, 08 Apr 2005 10:31:18 +0000 |
parents | 21f44596f356 |
children | 324046793f7c |
line wrap: on
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/* * * ao_macosx.c * * Original Copyright (C) Timothy J. Wood - Aug 2000 * * This file is part of libao, a cross-platform library. See * README for a history of this source code. * * libao is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2, or (at your option) * any later version. * * libao is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. * * You should have received a copy of the GNU General Public License * along with GNU Make; see the file COPYING. If not, write to * the Free Software Foundation, 675 Mass Ave, Cambridge, MA 02139, USA. */ /* * The MacOS X CoreAudio framework doesn't mesh as simply as some * simpler frameworks do. This is due to the fact that CoreAudio pulls * audio samples rather than having them pushed at it (which is nice * when you are wanting to do good buffering of audio). */ /* Change log: * * 14/5-2003: Ported to MPlayer libao2 by Dan Christiansen * * AC-3 and MPEG audio passthrough is possible, but I don't have * access to a sound card that supports it. */ #include <CoreAudio/AudioHardware.h> #include <stdio.h> #include <string.h> #include <stdlib.h> #include <inttypes.h> #include <pthread.h> #include "config.h" #include "mp_msg.h" #include "audio_out.h" #include "audio_out_internal.h" #include "libaf/af_format.h" static ao_info_t info = { "Darwin/Mac OS X native audio output", "macosx", "Timothy J. Wood & Dan Christiansen", "" }; LIBAO_EXTERN(macosx) /* Prefix for all mp_msg() calls */ #define ao_msg(a, b, c...) mp_msg(a, b, "AO: [macosx] " c) /* This is large, but best (maybe it should be even larger). * CoreAudio supposedly has an internal latency in the order of 2ms */ #define NUM_BUFS 32 typedef struct ao_macosx_s { /* CoreAudio */ AudioDeviceID outputDeviceID; AudioStreamBasicDescription outputStreamBasicDescription; /* Ring-buffer */ /* does not need explicit synchronization, but needs to allocate * (num_chunks + 1) * chunk_size memory to store num_chunks * chunk_size * data */ unsigned char *buffer; unsigned int buffer_len; ///< must always be (num_chunks + 1) * chunk_size unsigned int num_chunks; unsigned int chunk_size; unsigned int buf_read_pos; unsigned int buf_write_pos; } ao_macosx_t; static ao_macosx_t *ao; /** * \brief return number of free bytes in the buffer * may only be called by mplayer's thread * \return minimum number of free bytes in buffer, value may change between * two immediately following calls, and the real number of free bytes * might actually be larger! */ static int buf_free() { int free = ao->buf_read_pos - ao->buf_write_pos - ao->chunk_size; if (free < 0) free += ao->buffer_len; return free; } /** * \brief return number of buffered bytes * may only be called by playback thread * \return minimum number of buffered bytes, value may change between * two immediately following calls, and the real number of buffered bytes * might actually be larger! */ static int buf_used() { int used = ao->buf_write_pos - ao->buf_read_pos; if (used < 0) used += ao->buffer_len; return used; } /** * \brief add data to ringbuffer */ static int write_buffer(unsigned char* data, int len){ int first_len = ao->buffer_len - ao->buf_write_pos; int free = buf_free(); if (len > free) len = free; if (first_len > len) first_len = len; // till end of buffer memcpy (&ao->buffer[ao->buf_write_pos], data, first_len); if (len > first_len) { // we have to wrap around // remaining part from beginning of buffer memcpy (ao->buffer, &data[first_len], len - first_len); } ao->buf_write_pos = (ao->buf_write_pos + len) % ao->buffer_len; return len; } /** * \brief remove data from ringbuffer */ static int read_buffer(unsigned char* data,int len){ int first_len = ao->buffer_len - ao->buf_read_pos; int buffered = buf_used(); if (len > buffered) len = buffered; if (first_len > len) first_len = len; // till end of buffer memcpy (data, &ao->buffer[ao->buf_read_pos], first_len); if (len > first_len) { // we have to wrap around // remaining part from beginning of buffer memcpy (&data[first_len], ao->buffer, len - first_len); } ao->buf_read_pos = (ao->buf_read_pos + len) % ao->buffer_len; return len; } /* end ring buffer stuff */ /* The function that the CoreAudio thread calls when it wants more data */ static OSStatus audioDeviceIOProc(AudioDeviceID inDevice, const AudioTimeStamp *inNow, const AudioBufferList *inInputData, const AudioTimeStamp *inInputTime, AudioBufferList *outOutputData, const AudioTimeStamp *inOutputTime, void *inClientData) { outOutputData->mBuffers[0].mDataByteSize = read_buffer((char *)outOutputData->mBuffers[0].mData, ao->chunk_size); return 0; } static int control(int cmd,void *arg){ OSStatus status; UInt32 propertySize; ao_control_vol_t* vol = (ao_control_vol_t*)arg; UInt32 stereoChannels[2]; static float volume=0.5; switch (cmd) { case AOCONTROL_SET_DEVICE: case AOCONTROL_GET_DEVICE: /* unimplemented/meaningless */ return CONTROL_FALSE; case AOCONTROL_GET_VOLUME: propertySize=sizeof(stereoChannels); status = AudioDeviceGetProperty(ao->outputDeviceID, NULL, 0, kAudioDevicePropertyPreferredChannelsForStereo, &propertySize, &stereoChannels); // printf("OSX: stereochannels %d ; %d \n",stereoChannels[0],stereoChannels[1]); propertySize=sizeof(volume); status = AudioDeviceGetProperty(ao->outputDeviceID, stereoChannels[0], false, kAudioDevicePropertyVolumeScalar, &propertySize, &volume); // printf("OSX: get volume=%5.3f status=%d \n",volume,status); vol->left=(int)(volume*100.0); status = AudioDeviceGetProperty(ao->outputDeviceID, stereoChannels[1], false, kAudioDevicePropertyVolumeScalar, &propertySize, &volume); vol->right=(int)(volume*100.0); return CONTROL_TRUE; case AOCONTROL_SET_VOLUME: propertySize=sizeof(stereoChannels); status = AudioDeviceGetProperty(ao->outputDeviceID, NULL, 0, kAudioDevicePropertyPreferredChannelsForStereo, &propertySize, &stereoChannels); // printf("OSX: stereochannels %d ; %d \n",stereoChannels[0],stereoChannels[1]); propertySize=sizeof(volume); volume=vol->left/100.0; status = AudioDeviceSetProperty(ao->outputDeviceID, 0, stereoChannels[0], 0, kAudioDevicePropertyVolumeScalar, propertySize, &volume); // printf("OSX: set volume=%5.3f status=%d\n",volume,status); volume=vol->right/100.0; status = AudioDeviceSetProperty(ao->outputDeviceID, 0, stereoChannels[1], 0, kAudioDevicePropertyVolumeScalar, propertySize, &volume); return CONTROL_TRUE; case AOCONTROL_QUERY_FORMAT: /* stick with what CoreAudio requests */ return CONTROL_FALSE; default: return CONTROL_FALSE; } } static void print_format(const char* str,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", 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, (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 init(int rate,int channels,int format,int flags) { OSStatus status; UInt32 propertySize; int rc; int i; ao = (ao_macosx_t *)malloc(sizeof(ao_macosx_t)); /* get default output device */ propertySize = sizeof(ao->outputDeviceID); status = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice, &propertySize, &(ao->outputDeviceID)); if (status) { ao_msg(MSGT_AO,MSGL_WARN, "AudioHardwareGetProperty returned %d\n", (int)status); return CONTROL_FALSE; } if (ao->outputDeviceID == kAudioDeviceUnknown) { ao_msg(MSGT_AO,MSGL_WARN, "AudioHardwareGetProperty: ao->outputDeviceID is kAudioDeviceUnknown\n"); return CONTROL_FALSE; } propertySize = sizeof(ao->outputStreamBasicDescription); status = AudioDeviceGetProperty(ao->outputDeviceID, 0, false, kAudioDevicePropertyStreamFormat, &propertySize, &ao->outputStreamBasicDescription); if(!status) print_format("default:",&ao->outputStreamBasicDescription); #if 1 // dump supported format list: { AudioStreamBasicDescription* p; Boolean ow; int i; propertySize=0; //sizeof(p); // status = AudioDeviceGetPropertyInfo(ao->outputDeviceID, 0, false, kAudioStreamPropertyPhysicalFormats, &propertySize, &ow); status = AudioDeviceGetPropertyInfo(ao->outputDeviceID, 0, false, kAudioDevicePropertyStreamFormats, &propertySize, &ow); if (status) { ao_msg(MSGT_AO,MSGL_WARN, "AudioDeviceGetPropertyInfo returned 0x%X when getting kAudioDevicePropertyStreamFormats\n", (int)status); } p=malloc(propertySize); // status = AudioDeviceGetProperty(ao->outputDeviceID, 0, false, kAudioStreamPropertyPhysicalFormats, &propertySize, p); status = AudioDeviceGetProperty(ao->outputDeviceID, 0, false, kAudioDevicePropertyStreamFormats, &propertySize, p); if (status) { ao_msg(MSGT_AO,MSGL_WARN, "AudioDeviceGetProperty returned 0x%X when getting kAudioDevicePropertyStreamFormats\n", (int)status); // return CONTROL_FALSE; } for(i=0;i<propertySize/sizeof(AudioStreamBasicDescription);i++) print_format("support:",&p[i]); // printf("FORMATS: (%d) %p %p %p %p\n",propertySize,p[0],p[1],p[2],p[3]); free(p); } #endif // fill in our wanted format, and let's see if the driver accepts it or // offers some similar alternative: propertySize = sizeof(ao->outputStreamBasicDescription); memset(&ao->outputStreamBasicDescription,0,propertySize); ao->outputStreamBasicDescription.mSampleRate=rate; ao->outputStreamBasicDescription.mFormatID=kAudioFormatLinearPCM; ao->outputStreamBasicDescription.mChannelsPerFrame=channels; switch(format&AF_FORMAT_BITS_MASK){ case AF_FORMAT_8BIT: ao->outputStreamBasicDescription.mBitsPerChannel=8; break; case AF_FORMAT_16BIT: ao->outputStreamBasicDescription.mBitsPerChannel=16; break; case AF_FORMAT_24BIT: ao->outputStreamBasicDescription.mBitsPerChannel=24; break; case AF_FORMAT_32BIT: ao->outputStreamBasicDescription.mBitsPerChannel=32; break; } if((format&AF_FORMAT_POINT_MASK)==AF_FORMAT_F){ // float ao->outputStreamBasicDescription.mFormatFlags=kAudioFormatFlagIsFloat|kAudioFormatFlagIsPacked; } else if((format&AF_FORMAT_SIGN_MASK)==AF_FORMAT_SI){ // signed int ao->outputStreamBasicDescription.mFormatFlags=kAudioFormatFlagIsSignedInteger|kAudioFormatFlagIsPacked; } else { // unsigned int ao->outputStreamBasicDescription.mFormatFlags=kAudioFormatFlagIsPacked; } if((format&AF_FORMAT_END_MASK)==AF_FORMAT_BE) ao->outputStreamBasicDescription.mFormatFlags|=kAudioFormatFlagIsBigEndian; ao->outputStreamBasicDescription.mBytesPerPacket= ao->outputStreamBasicDescription.mBytesPerFrame=channels*(ao->outputStreamBasicDescription.mBitsPerChannel/8); ao->outputStreamBasicDescription.mFramesPerPacket=1; print_format("wanted: ",&ao->outputStreamBasicDescription); // try 1: ask if it accepts our specific requirements? propertySize = sizeof(ao->outputStreamBasicDescription); // status = AudioDeviceGetProperty(ao->outputDeviceID, 0, false, kAudioStreamPropertyPhysicalFormatMatch, &propertySize, &ao->outputStreamBasicDescription); status = AudioDeviceGetProperty(ao->outputDeviceID, 0, false, kAudioDevicePropertyStreamFormatMatch, &propertySize, &ao->outputStreamBasicDescription); if (status || ao->outputStreamBasicDescription.mSampleRate!=rate || ao->outputStreamBasicDescription.mFormatID!=kAudioFormatLinearPCM) { ao_msg(MSGT_AO,MSGL_V, "AudioDeviceGetProperty returned 0x%X when getting kAudioDevicePropertyStreamFormatMatch\n", (int)status); // failed (error, bad rate or bad type) // try 2: set only rate & type, no format details (bits, channels etc) propertySize = sizeof(ao->outputStreamBasicDescription); memset(&ao->outputStreamBasicDescription,0,propertySize); ao->outputStreamBasicDescription.mSampleRate=rate; ao->outputStreamBasicDescription.mFormatID=kAudioFormatLinearPCM; // status = AudioDeviceGetProperty(ao->outputDeviceID, 0, false, kAudioStreamPropertyPhysicalFormatMatch, &propertySize, &ao->outputStreamBasicDescription); status = AudioDeviceGetProperty(ao->outputDeviceID, 0, false, kAudioDevicePropertyStreamFormatMatch, &propertySize, &ao->outputStreamBasicDescription); if (status || ao->outputStreamBasicDescription.mFormatID!=kAudioFormatLinearPCM) { ao_msg(MSGT_AO,MSGL_V, "AudioDeviceGetProperty returned 0x%X when getting kAudioDevicePropertyStreamFormatMatch\n", (int)status); // failed again. (error or bad type) // giving up... just read the default. propertySize = sizeof(ao->outputStreamBasicDescription); // status = AudioDeviceGetProperty(ao->outputDeviceID, 0, false, kAudioStreamPropertyPhysicalFormat, &propertySize, &ao->outputStreamBasicDescription); status = AudioDeviceGetProperty(ao->outputDeviceID, 0, false, kAudioDevicePropertyStreamFormat, &propertySize, &ao->outputStreamBasicDescription); if (status) { // failed to read the default format - WTF? ao_msg(MSGT_AO,MSGL_WARN, "AudioDeviceGetProperty returned 0x%X when getting kAudioDevicePropertyStreamFormat\n", (int)status); return CONTROL_FALSE; } } } // propertySize = sizeof(ao->outputStreamBasicDescription); // status = AudioDeviceGetProperty(ao->outputDeviceID, 0, false, kAudioDevicePropertyStreamFormatSupported, &propertySize, &ao->outputStreamBasicDescription); // if (status) { // ao_msg(MSGT_AO,MSGL_V, "AudioDeviceGetProperty returned 0x%X when getting kAudioDevicePropertyStreamFormatSupported\n", (int)status); // } // ok, now try to set the new (default or matched) audio format: print_format("best: ",&ao->outputStreamBasicDescription); propertySize = sizeof(ao->outputStreamBasicDescription); status = AudioDeviceSetProperty(ao->outputDeviceID, 0, 0, false, kAudioDevicePropertyStreamFormat, propertySize, &ao->outputStreamBasicDescription); if(status) ao_msg(MSGT_AO,MSGL_WARN, "AudioDeviceSetProperty returned 0x%X when getting kAudioDevicePropertyStreamFormat\n", (int)status); // see what did we get finally... we'll be forced to use this anyway :( propertySize = sizeof(ao->outputStreamBasicDescription); status = AudioDeviceGetProperty(ao->outputDeviceID, 0, false, kAudioDevicePropertyStreamFormat, &propertySize, &ao->outputStreamBasicDescription); print_format("final: ",&ao->outputStreamBasicDescription); /* get requested buffer length */ // TODO: set NUM_BUFS dinamically, based on buffer size! propertySize = sizeof(ao->chunk_size); status = AudioDeviceGetProperty(ao->outputDeviceID, 0, false, kAudioDevicePropertyBufferSize, &propertySize, &ao->chunk_size); if (status) { ao_msg(MSGT_AO,MSGL_WARN, "AudioDeviceGetProperty returned %d when getting kAudioDevicePropertyBufferSize\n", (int)status); return CONTROL_FALSE; } ao_msg(MSGT_AO,MSGL_V, "%5d chunk size\n", (int)ao->chunk_size); ao_data.samplerate = ao->outputStreamBasicDescription.mSampleRate; ao_data.channels = channels; ao_data.outburst = ao_data.buffersize = ao->chunk_size; ao_data.bps = ao_data.samplerate * ao->outputStreamBasicDescription.mBytesPerFrame; if (ao->outputStreamBasicDescription.mFormatID == kAudioFormatLinearPCM) { uint32_t flags = ao->outputStreamBasicDescription.mFormatFlags; if (flags & kAudioFormatFlagIsFloat) { ao_data.format = (flags&kAudioFormatFlagIsBigEndian) ? AF_FORMAT_FLOAT_BE : AF_FORMAT_FLOAT_LE; } else { ao_msg(MSGT_AO,MSGL_WARN, "Unsupported audio output " "format 0x%X. Please report this to the developer\n", format); return CONTROL_FALSE; } } else { /* TODO: handle AFMT_AC3, AFMT_MPEG & friends */ ao_msg(MSGT_AO,MSGL_WARN, "Default Audio Device doesn't " "support Linear PCM!\n"); return CONTROL_FALSE; } /* Allocate ring-buffer memory */ ao->num_chunks = NUM_BUFS; ao->buffer_len = (ao->num_chunks + 1) * ao->chunk_size; ao->buffer = (unsigned char *)malloc(ao->buffer_len); /* Prepare for playback */ /* Set the IO proc that CoreAudio will call when it needs data */ status = AudioDeviceAddIOProc(ao->outputDeviceID, audioDeviceIOProc, NULL); if (status) { ao_msg(MSGT_AO,MSGL_WARN, "AudioDeviceAddIOProc returned %d\n", (int)status); return CONTROL_FALSE; } /* Start callback */ reset(); 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); }