/*
- * Copyright (C) 2006 Andre Noll <maan@systemlinux.org>
+ * Copyright (C) 2006-2010 Andre Noll <maan@systemlinux.org>
*
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program 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 this program; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111, USA.
+ * Licensed under the GPL v2. For licencing details see COPYING.
*/
/** \file osx_write.c paraslash's output plugin for MacOs */
* <skortze@sourceforge.net>
*/
-#include <CoreAudio/CoreAudio.h>
+#include <regex.h>
+#include <sys/types.h>
+#include <dirent.h>
+#include <stdbool.h>
+
#include "para.h"
#include "fd.h"
#include "string.h"
#include "list.h"
#include "sched.h"
+#include "ggo.h"
+#include "buffer_tree.h"
#include "write.h"
+#include "write_common.h"
#include "osx_write.cmdline.h"
#include "error.h"
-
-#include <CoreAudio/CoreAudio.h>
+#include <CoreServices/CoreServices.h>
#include <AudioUnit/AudioUnit.h>
-#include <AudioToolbox/DefaultAudioOutput.h>
+#include <AudioToolbox/AudioToolbox.h>
+
+/** describes one input buffer for the osx writer */
struct osx_buffer {
+ /** pointer to the beginning of the buffer */
float *buffer;
+ /** the size of this buffer */
long size;
- float *ptr; /* Where in the buffer are we? */
+ /** current position in the buffer */
+ float *ptr;
+ /** number of floats not yet consuned */
long remaining;
+ /** pointer to the next audio buffer */
struct osx_buffer *next;
};
-typedef struct osx_buffer osx_buffer;
+/** data specific to the osx writer */
struct private_osx_write_data {
- long size;
- short *ptr;
- AudioUnit output;
+ /** the main control structure for audio data manipulation */
+ AudioUnit audio_unit;
+ /** non-zero if playback has started */
char play;
- osx_buffer *from; /* Current buffers */
- osx_buffer *to;
- unsigned samplerate;
+ /** callback reads audio data from this buffer */
+ struct osx_buffer *from;
+ /** the post_select writes audio data here */
+ struct osx_buffer *to;
+ /** sample rate of the current audio stream */
+ unsigned sample_rate;
+ /** Sample format of the current audio stream */
+ unsigned sample_format;
+ /** number of channels of the current audio stream */
unsigned channels;
};
-
-/*
- * Tried with 3 buffers, but then any little window move is sufficient to
- * stop the sound (OK, on a G3 400 with a Public Beta. Perhaps now we can
- * go down to 2 buffers). With 16 buffers we have 1.5 seconds music
- * buffered (or, if you're pessimistic, 1.5 seconds latency). Note 0
- * buffers don't work much further than the Bus error.
- */
-#define NUMBER_BUFFERS 2
-
static void destroy_buffers(struct private_osx_write_data *powd)
{
- osx_buffer *ptr;
- osx_buffer *ptr2;
+ struct osx_buffer *ptr;
+ struct osx_buffer *ptr2;
ptr = powd->to->next;
powd->to->next = NULL;
while (ptr) {
}
}
-static void init_buffers(struct private_osx_write_data *powd)
+static void init_buffers(struct writer_node *wn)
{
+ struct private_osx_write_data *powd = wn->private_data;
+ struct osx_write_args_info *conf = wn->conf;
+ struct osx_buffer **ptrptr;
int i;
- osx_buffer ** ptrptr;
ptrptr = &powd->to;
- for (i = 0; i < NUMBER_BUFFERS; i++) {
- *ptrptr = malloc(sizeof(osx_buffer));
+ for (i = 0; i < conf->numbuffers_arg; i++) {
+ *ptrptr = para_malloc(sizeof(struct osx_buffer));
(*ptrptr)->size = 0;
(*ptrptr)->remaining = 0;
(*ptrptr)->buffer = NULL;
*ptrptr = powd->from = powd->to;
}
-static void fill_buffer(osx_buffer *b, short *source, long size)
+static void fill_buffer(struct private_osx_write_data *powd, char *data, long bytes)
{
+ struct osx_buffer *b = powd->to;
float *dest;
+ long samples;
+ enum sample_format sf = powd->sample_format;
- if (b->remaining) /* Non empty buffer, must still be playing */
- return;
- PARA_INFO_LOG("%ld\n", size);
- if (b->size != size) {
- /*
- * Hey! What's that? Coudn't this buffer size be fixed
- * once (well, perhaps we just didn't allocate it yet)
- */
- free(b->buffer);
- b->buffer = malloc(size * sizeof(float));
- b->size = size;
+ samples = (sf == SF_S8 || sf == SF_U8)? bytes : bytes / 2;
+ assert(b->remaining == 0 || samples > 0);
+ if (b->size != samples) {
+ b->buffer = para_realloc(b->buffer, samples * sizeof(float));
+ b->size = samples;
}
dest = b->buffer;
- while (size--) {
- char *tmp = (char *)source;
- char c = *tmp;
- *tmp = *(tmp + 1);
- *(tmp + 1) = c;
- /* *dest++ = ((*source++) + 32768) / 65536.0; */
- *dest++ = (*source++) / 32768.0;
+ switch (powd->sample_format) {
+ case SF_U8: {
+ uint8_t *src = (uint8_t *)data;
+ while (samples--) {
+ *dest++ = (*src++) / 256.0;
+ }
+ break;
+ }
+ case SF_S8: {
+ int8_t *src = (int8_t *)data;
+ while (samples--) {
+ *dest++ = ((*src++) + 128) / 256.0;
+ }
+ break;
+ }
+ default: {
+ short *src = (short *)data;
+ while (samples--)
+ *dest++ = (*src++) / 32768.0;
+ }
}
b->ptr = b->buffer;
b->remaining = b->size;
m = outOutputData->mBuffers[i].mDataByteSize / sizeof(float);
dest = outOutputData->mBuffers[i].mData;
while (m > 0) {
- if ((n = powd->from->remaining) <= 0) {
- PARA_INFO_LOG("%s", "buffer underrun\n");
- /* no more bytes in the current read buffer! */
- while ((n = powd->from->remaining) <= 0)
- /* wait for the results */
- usleep(2000);
+ n = powd->from->remaining;
+ if (n <= 0) {
+ n = powd->from->next->remaining;
+ if (n <= 0) {
+ PARA_INFO_LOG("buffer underrun\n");
+ return 0;
+ }
+ powd->from = powd->from->next;
}
- PARA_INFO_LOG("buf %p: n = %ld, m= %ld\n", powd->from->buffer, n, m);
+// PARA_INFO_LOG("buf %p: n = %ld, m= %ld\n", powd->from->buffer, n, m);
/*
* we dump what we can. In fact, just the necessary
* should be sufficient
return 0;
}
+#ifdef WORDS_BIGENDIAN /* ppc */
+#define ENDIAN_FLAGS kLinearPCMFormatFlagIsBigEndian
+#else
+#define ENDIAN_FLAGS 0
+#endif
+
static int osx_write_open(struct writer_node *wn)
{
- struct private_osx_write_data *powd = para_calloc(
- sizeof(struct private_osx_write_data));
+ struct private_osx_write_data *powd = para_calloc(sizeof(*powd));
+
+ wn->private_data = powd;
+ init_buffers(wn);
+ return 0;
+}
+
+static int core_audio_init(struct writer_node *wn)
+{
+ struct private_osx_write_data *powd = wn->private_data;
ComponentDescription desc;
Component comp;
AURenderCallbackStruct inputCallback = {osx_callback, powd};
AudioStreamBasicDescription format;
int ret;
- struct writer_node_group *wng = wn->wng;
- struct osx_write_args_info *conf = wn->conf;
+ struct btr_node *btrn = wn->btrn;
+ int32_t val;
- wn->private_data = powd;
/* where did that default audio output go? */
desc.componentType = kAudioUnitType_Output;
desc.componentSubType = kAudioUnitSubType_DefaultOutput;
if (!comp)
goto e0;
ret = -E_OPEN_COMP;
- if (OpenAComponent(comp, &powd->output))
+ if (OpenAComponent(comp, &powd->audio_unit))
goto e0;
ret = -E_UNIT_INIT;
- if (AudioUnitInitialize(powd->output))
+ if (AudioUnitInitialize(powd->audio_unit))
goto e1;
- powd->size = 0;
- powd->ptr = NULL;
powd->play = 0;
- /* Hmmm, let's choose PCM format */
- /* We tell the Output Unit what format we're going to supply data to it.
- * This is necessary if you're providing data through an input callback
- * AND you want the DefaultOutputUnit to do any format conversions
- * necessary from your format to the device's format.
+ get_btr_sample_rate(btrn, &val);
+ powd->sample_rate = val;
+ get_btr_channels(btrn, &val);
+ powd->channels = val;
+ get_btr_sample_format(btrn, &val);
+ powd->sample_format = val;
+ /*
+ * Choose PCM format. We tell the Output Unit what format we're going
+ * to supply data to it. This is necessary if you're providing data
+ * through an input callback AND you want the DefaultOutputUnit to do
+ * any format conversions necessary from your format to the device's
+ * format.
*/
- if (!conf->samplerate_given && wng->samplerate)
- powd->samplerate = *wng->samplerate;
- else
- powd->samplerate = conf->samplerate_arg;
- format.mSampleRate = powd->samplerate;
- /* The specific encoding type of audio stream*/
format.mFormatID = kAudioFormatLinearPCM;
+ format.mFramesPerPacket = 1;
+ format.mSampleRate = powd->sample_rate;
/* flags specific to each format */
format.mFormatFlags = kLinearPCMFormatFlagIsFloat
| kLinearPCMFormatFlagIsPacked
- | kLinearPCMFormatFlagIsBigEndian;
- if (!conf->channels_given && wng->channels)
- powd->channels = *wng->channels;
- else
- powd->channels = conf->channels_arg;
+ | ENDIAN_FLAGS;
+ switch (powd->sample_format) {
+ case SF_S8:
+ case SF_U8:
+ wn->min_iqs = powd->channels;
+ break;
+ default:
+ wn->min_iqs = powd->channels * 2;
+ }
+ format.mBitsPerChannel = 8 * sizeof(float);
+ format.mBytesPerPacket = powd->channels * sizeof(float);
+ format.mBytesPerFrame = format.mBytesPerPacket;
format.mChannelsPerFrame = powd->channels;
- format.mFramesPerPacket = 1;
- format.mBytesPerPacket = format.mChannelsPerFrame * sizeof(float);
- format.mBytesPerFrame = format.mFramesPerPacket * format.mBytesPerPacket;
- /* one of the most constant constants of the whole computer history */
- format.mBitsPerChannel = sizeof(float) * 8;
+
ret = -E_STREAM_FORMAT;
- if (AudioUnitSetProperty(powd->output, kAudioUnitProperty_StreamFormat,
+ if (AudioUnitSetProperty(powd->audio_unit, kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Input, 0, &format,
sizeof(AudioStreamBasicDescription)))
goto e2;
- init_buffers(powd);
ret = -E_ADD_CALLBACK;
- if (AudioUnitSetProperty(powd->output, kAudioUnitProperty_SetRenderCallback,
+ if (AudioUnitSetProperty(powd->audio_unit, kAudioUnitProperty_SetRenderCallback,
kAudioUnitScope_Input, 0, &inputCallback,
sizeof(inputCallback)) < 0)
goto e3;
e3:
destroy_buffers(powd);
e2:
- AudioUnitUninitialize(powd->output);
+ AudioUnitUninitialize(powd->audio_unit);
e1:
- CloseComponent(powd->output);
+ CloseComponent(powd->audio_unit);
e0:
return ret;
}
-__malloc void *osx_write_parse_config(char *options)
+__malloc static void *osx_write_parse_config(const char *options)
{
struct osx_write_args_info *conf
= para_calloc(sizeof(struct osx_write_args_info));
}
+static void osx_free_config(void *conf)
+{
+ osx_cmdline_parser_free(conf);
+}
+
static void osx_write_close(struct writer_node *wn)
{
struct private_osx_write_data *powd = wn->private_data;
PARA_INFO_LOG("closing writer node %p\n", wn);
- AudioOutputUnitStop(powd->output);
- AudioUnitUninitialize(powd->output);
- CloseComponent(powd->output);
+ AudioOutputUnitStop(powd->audio_unit);
+ AudioUnitUninitialize(powd->audio_unit);
+ CloseComponent(powd->audio_unit);
destroy_buffers(powd);
free(powd);
}
-static int need_new_buffer(struct writer_node *wn)
+static void osx_write_post_select(__a_unused struct sched *s, struct task *t)
{
- struct writer_node_group *wng = wn->wng;
+ struct writer_node *wn = container_of(t, struct writer_node, task);
struct private_osx_write_data *powd = wn->private_data;
+ struct btr_node *btrn = wn->btrn;
+ char *data;
+ size_t bytes;
+ int ret = 0;
- if (*wng->loaded < sizeof(short))
- return 0;
- if (powd->to->remaining) /* Non empty buffer, must still be playing */
- return 0;
- return 1;
+ while (powd->to->remaining <= 0) {
+ ret = btr_node_status(wn->btrn, wn->min_iqs, BTR_NT_LEAF);
+ if (ret <= 0)
+ break;
+ if (powd->sample_rate == 0) {
+ ret = core_audio_init(wn);
+ if (ret < 0)
+ break;
+ }
+ btr_merge(btrn, 8192);
+ bytes = btr_next_buffer(btrn, &data);
+ //PARA_CRIT_LOG("have: %zu\n", bytes);
+ fill_buffer(powd, data, bytes);
+ btr_consume(btrn, bytes);
+ if (!powd->play) {
+ ret = -E_UNIT_START;
+ if (AudioOutputUnitStart(powd->audio_unit))
+ break;
+ powd->play = 1;
+ }
+ powd->to = powd->to->next;
+ }
+ if (ret < 0 && powd->from->remaining <= 0) {
+ btr_remove_node(btrn);
+ t->error = ret;
+ }
}
-static int osx_write_post_select(__a_unused struct sched *s,
- struct writer_node *wn)
+static void osx_write_pre_select(struct sched *s, struct task *t)
{
+ struct writer_node *wn = container_of(t, struct writer_node, task);
struct private_osx_write_data *powd = wn->private_data;
- struct writer_node_group *wng = wn->wng;
- short *data = (short*)wng->buf;
-
- if (!need_new_buffer(wn))
- return 1;
- fill_buffer(powd->to, data, *wng->loaded / sizeof(short));
- powd->to = powd->to->next;
- wn->written = *wng->loaded;
- if (!powd->play) {
- if (AudioOutputUnitStart(powd->output))
- return -E_UNIT_START;
- powd->play = 1;
- }
- return 1;
-}
+ struct timeval tmp = {.tv_sec = 1, .tv_usec = 0}, delay = tmp;
+ unsigned long divisor;
+ size_t numbytes = powd->to->remaining * sizeof(short);
+ int ret = btr_node_status(wn->btrn, wn->min_iqs, BTR_NT_LEAF);
-static int osx_write_pre_select(struct sched *s, __a_unused struct writer_node *wn)
-{
- s->timeout.tv_sec = 0;
- s->timeout.tv_usec = 20;
- return 1;
+ if (ret < 0)
+ sched_min_delay(s);
+ if (ret <= 0 || numbytes < wn->min_iqs)
+ return;
+ divisor = powd->sample_rate * wn->min_iqs / numbytes;
+ if (divisor)
+ tv_divide(divisor, &tmp, &delay);
+ sched_request_timeout(&delay, s);
}
+/** the init function of the osx writer */
void osx_write_init(struct writer *w)
{
+ struct osx_write_args_info dummy;
+
+ osx_cmdline_parser_init(&dummy);
w->open = osx_write_open;
w->close = osx_write_close;
w->pre_select = osx_write_pre_select;
w->post_select = osx_write_post_select;
w->parse_config = osx_write_parse_config;
+ w->free_config = osx_free_config;
w->shutdown = NULL; /* nothing to do */
+ w->help = (struct ggo_help) {
+ .short_help = osx_write_args_info_help,
+ .detailed_help = osx_write_args_info_detailed_help
+ };
+ osx_cmdline_parser_free(&dummy);
}
projects / paraslash.git / blobdiff