Commit c1b85795 authored by Diego Biurrun's avatar Diego Biurrun

Remove broken BeOS audio interface.

Originally committed as revision 23568 to svn://svn.ffmpeg.org/ffmpeg/trunk
parent 108caaaa
...@@ -1387,10 +1387,6 @@ alsa_indev_deps="alsa_asoundlib_h snd_pcm_htimestamp" ...@@ -1387,10 +1387,6 @@ alsa_indev_deps="alsa_asoundlib_h snd_pcm_htimestamp"
alsa_indev_extralibs="-lasound" alsa_indev_extralibs="-lasound"
alsa_outdev_deps="alsa_asoundlib_h" alsa_outdev_deps="alsa_asoundlib_h"
alsa_outdev_extralibs="-lasound" alsa_outdev_extralibs="-lasound"
audio_beos_indev_deps="audio_beos"
audio_beos_indev_extralibs="-lmedia -lbe"
audio_beos_outdev_deps="audio_beos"
audio_beos_outdev_extralibs="-lmedia -lbe"
bktr_indev_deps_any="dev_bktr_ioctl_bt848_h machine_ioctl_bt848_h dev_video_bktr_ioctl_bt848_h dev_ic_bt8xx_h" bktr_indev_deps_any="dev_bktr_ioctl_bt848_h machine_ioctl_bt848_h dev_video_bktr_ioctl_bt848_h dev_ic_bt8xx_h"
dv1394_indev_deps="dv1394 dv_demuxer" dv1394_indev_deps="dv1394 dv_demuxer"
jack_indev_deps="jack_jack_h" jack_indev_deps="jack_jack_h"
...@@ -2127,7 +2123,6 @@ enabled spic && enable pic ...@@ -2127,7 +2123,6 @@ enabled spic && enable pic
case $target_os in case $target_os in
haiku) haiku)
prefix_default="/boot/common" prefix_default="/boot/common"
disable audio_beos
network_extralibs="-lnetwork" network_extralibs="-lnetwork"
;; ;;
sunos) sunos)
......
...@@ -708,7 +708,6 @@ performance on systems without hardware floating point support). ...@@ -708,7 +708,6 @@ performance on systems without hardware floating point support).
@multitable @columnfractions .4 .1 .1 @multitable @columnfractions .4 .1 .1
@item Name @tab Input @tab Output @item Name @tab Input @tab Output
@item ALSA @tab X @tab X @item ALSA @tab X @tab X
@item BEOS audio @tab X @tab X
@item BKTR @tab X @tab @item BKTR @tab X @tab
@item DV1394 @tab X @tab @item DV1394 @tab X @tab
@item JACK @tab X @tab @item JACK @tab X @tab
......
...@@ -25,9 +25,6 @@ OBJS-$(CONFIG_X11_GRAB_DEVICE_INDEV) += x11grab.o ...@@ -25,9 +25,6 @@ OBJS-$(CONFIG_X11_GRAB_DEVICE_INDEV) += x11grab.o
# external libraries # external libraries
OBJS-$(CONFIG_LIBDC1394_INDEV) += libdc1394.o OBJS-$(CONFIG_LIBDC1394_INDEV) += libdc1394.o
OBJS-$(CONFIG_AUDIO_BEOS_INDEV) += beosaudio.o
OBJS-$(CONFIG_AUDIO_BEOS_OUTDEV) += beosaudio.o
SKIPHEADERS-$(HAVE_ALSA_ASOUNDLIB_H) += alsa-audio.h SKIPHEADERS-$(HAVE_ALSA_ASOUNDLIB_H) += alsa-audio.h
include $(SUBDIR)../subdir.mak include $(SUBDIR)../subdir.mak
...@@ -40,7 +40,6 @@ void avdevice_register_all(void) ...@@ -40,7 +40,6 @@ void avdevice_register_all(void)
/* devices */ /* devices */
REGISTER_INOUTDEV (ALSA, alsa); REGISTER_INOUTDEV (ALSA, alsa);
REGISTER_INOUTDEV (AUDIO_BEOS, audio_beos);
REGISTER_INDEV (BKTR, bktr); REGISTER_INDEV (BKTR, bktr);
REGISTER_INDEV (DV1394, dv1394); REGISTER_INDEV (DV1394, dv1394);
REGISTER_INDEV (JACK, jack); REGISTER_INDEV (JACK, jack);
......
/*
* BeOS audio play interface
* Copyright (c) 2000, 2001 Fabrice Bellard
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <signal.h>
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <unistd.h>
#include <sys/time.h>
#include <Application.h>
#include <SoundPlayer.h>
extern "C" {
#include "libavformat/avformat.h"
}
#if HAVE_BSOUNDRECORDER
#include <SoundRecorder.h>
using namespace BPrivate::Media::Experimental;
#endif
/* enable performance checks */
//#define PERF_CHECK
/* enable Media Kit latency checks */
//#define LATENCY_CHECK
#define AUDIO_BLOCK_SIZE 4096
#define AUDIO_BLOCK_COUNT 8
#define AUDIO_BUFFER_SIZE (AUDIO_BLOCK_SIZE*AUDIO_BLOCK_COUNT)
typedef struct {
int fd; // UNUSED
int sample_rate;
int channels;
int frame_size; /* in bytes ! */
CodecID codec_id;
uint8_t buffer[AUDIO_BUFFER_SIZE];
int buffer_ptr;
/* ring buffer */
sem_id input_sem;
int input_index;
sem_id output_sem;
int output_index;
BSoundPlayer *player;
#if HAVE_BSOUNDRECORDER
BSoundRecorder *recorder;
#endif
int has_quit; /* signal callbacks not to wait */
volatile bigtime_t starve_time;
} AudioData;
static thread_id main_thid;
static thread_id bapp_thid;
static int own_BApp_created = 0;
static int refcount = 0;
/* create the BApplication and Run() it */
static int32 bapp_thread(void *arg)
{
new BApplication("application/x-vnd.ffmpeg");
own_BApp_created = 1;
be_app->Run();
/* kill the process group */
// kill(0, SIGINT);
// kill(main_thid, SIGHUP);
return B_OK;
}
/* create the BApplication only if needed */
static void create_bapp_if_needed(void)
{
if (refcount++ == 0) {
/* needed by libmedia */
if (be_app == NULL) {
bapp_thid = spawn_thread(bapp_thread, "ffmpeg BApplication", B_NORMAL_PRIORITY, NULL);
resume_thread(bapp_thid);
while (!own_BApp_created)
snooze(50000);
}
}
}
static void destroy_bapp_if_needed(void)
{
if (--refcount == 0 && own_BApp_created) {
be_app->Lock();
be_app->Quit();
be_app = NULL;
}
}
/* called back by BSoundPlayer */
static void audioplay_callback(void *cookie, void *buffer, size_t bufferSize, const media_raw_audio_format &format)
{
AudioData *s;
size_t len, amount;
unsigned char *buf = (unsigned char *)buffer;
s = (AudioData *)cookie;
if (s->has_quit)
return;
while (bufferSize > 0) {
#ifdef PERF_CHECK
bigtime_t t;
t = system_time();
#endif
len = MIN(AUDIO_BLOCK_SIZE, bufferSize);
if (acquire_sem_etc(s->output_sem, len, B_CAN_INTERRUPT, 0LL) < B_OK) {
s->has_quit = 1;
s->player->SetHasData(false);
return;
}
amount = MIN(len, (AUDIO_BUFFER_SIZE - s->output_index));
memcpy(buf, &s->buffer[s->output_index], amount);
s->output_index += amount;
if (s->output_index >= AUDIO_BUFFER_SIZE) {
s->output_index %= AUDIO_BUFFER_SIZE;
memcpy(buf + amount, &s->buffer[s->output_index], len - amount);
s->output_index += len-amount;
s->output_index %= AUDIO_BUFFER_SIZE;
}
release_sem_etc(s->input_sem, len, 0);
#ifdef PERF_CHECK
t = system_time() - t;
s->starve_time = MAX(s->starve_time, t);
#endif
buf += len;
bufferSize -= len;
}
}
#if HAVE_BSOUNDRECORDER
/* called back by BSoundRecorder */
static void audiorecord_callback(void *cookie, bigtime_t timestamp, void *buffer, size_t bufferSize, const media_multi_audio_format &format)
{
AudioData *s;
size_t len, amount;
unsigned char *buf = (unsigned char *)buffer;
s = (AudioData *)cookie;
if (s->has_quit)
return;
while (bufferSize > 0) {
len = MIN(bufferSize, AUDIO_BLOCK_SIZE);
//printf("acquire_sem(input, %d)\n", len);
if (acquire_sem_etc(s->input_sem, len, B_CAN_INTERRUPT, 0LL) < B_OK) {
s->has_quit = 1;
return;
}
amount = MIN(len, (AUDIO_BUFFER_SIZE - s->input_index));
memcpy(&s->buffer[s->input_index], buf, amount);
s->input_index += amount;
if (s->input_index >= AUDIO_BUFFER_SIZE) {
s->input_index %= AUDIO_BUFFER_SIZE;
memcpy(&s->buffer[s->input_index], buf + amount, len - amount);
s->input_index += len - amount;
}
release_sem_etc(s->output_sem, len, 0);
//printf("release_sem(output, %d)\n", len);
buf += len;
bufferSize -= len;
}
}
#endif
static int audio_open(AudioData *s, int is_output, const char *audio_device)
{
int p[2];
int ret;
media_raw_audio_format format;
media_multi_audio_format iformat;
#if !HAVE_BSOUNDRECORDER
if (!is_output)
return AVERROR(EIO); /* not for now */
#endif
s->input_sem = create_sem(AUDIO_BUFFER_SIZE, "ffmpeg_ringbuffer_input");
if (s->input_sem < B_OK)
return AVERROR(EIO);
s->output_sem = create_sem(0, "ffmpeg_ringbuffer_output");
if (s->output_sem < B_OK) {
delete_sem(s->input_sem);
return AVERROR(EIO);
}
s->input_index = 0;
s->output_index = 0;
create_bapp_if_needed();
s->frame_size = AUDIO_BLOCK_SIZE;
/* bump up the priority (avoid realtime though) */
set_thread_priority(find_thread(NULL), B_DISPLAY_PRIORITY+1);
#if HAVE_BSOUNDRECORDER
if (!is_output) {
bool wait_for_input = false;
if (audio_device && !strcmp(audio_device, "wait:"))
wait_for_input = true;
s->recorder = new BSoundRecorder(&iformat, wait_for_input, "ffmpeg input", audiorecord_callback);
if (wait_for_input && (s->recorder->InitCheck() == B_OK)) {
s->recorder->WaitForIncomingConnection(&iformat);
}
if (s->recorder->InitCheck() != B_OK || iformat.format != media_raw_audio_format::B_AUDIO_SHORT) {
delete s->recorder;
s->recorder = NULL;
if (s->input_sem)
delete_sem(s->input_sem);
if (s->output_sem)
delete_sem(s->output_sem);
return AVERROR(EIO);
}
s->codec_id = (iformat.byte_order == B_MEDIA_LITTLE_ENDIAN)?CODEC_ID_PCM_S16LE:CODEC_ID_PCM_S16BE;
s->channels = iformat.channel_count;
s->sample_rate = (int)iformat.frame_rate;
s->frame_size = iformat.buffer_size;
s->recorder->SetCookie(s);
s->recorder->SetVolume(1.0);
s->recorder->Start();
return 0;
}
#endif
format = media_raw_audio_format::wildcard;
format.format = media_raw_audio_format::B_AUDIO_SHORT;
format.byte_order = B_HOST_IS_LENDIAN ? B_MEDIA_LITTLE_ENDIAN : B_MEDIA_BIG_ENDIAN;
format.channel_count = s->channels;
format.buffer_size = s->frame_size;
format.frame_rate = s->sample_rate;
s->player = new BSoundPlayer(&format, "ffmpeg output", audioplay_callback);
if (s->player->InitCheck() != B_OK) {
delete s->player;
s->player = NULL;
if (s->input_sem)
delete_sem(s->input_sem);
if (s->output_sem)
delete_sem(s->output_sem);
return AVERROR(EIO);
}
s->player->SetCookie(s);
s->player->SetVolume(1.0);
s->player->Start();
s->player->SetHasData(true);
return 0;
}
static int audio_close(AudioData *s)
{
if (s->input_sem)
delete_sem(s->input_sem);
if (s->output_sem)
delete_sem(s->output_sem);
s->has_quit = 1;
if (s->player) {
s->player->Stop();
}
if (s->player)
delete s->player;
#if HAVE_BSOUNDRECORDER
if (s->recorder)
delete s->recorder;
#endif
destroy_bapp_if_needed();
return 0;
}
/* sound output support */
static int audio_write_header(AVFormatContext *s1)
{
AudioData *s = (AudioData *)s1->priv_data;
AVStream *st;
int ret;
st = s1->streams[0];
s->sample_rate = st->codec->sample_rate;
s->channels = st->codec->channels;
ret = audio_open(s, 1, NULL);
if (ret < 0)
return AVERROR(EIO);
return 0;
}
static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
{
AudioData *s = (AudioData *)s1->priv_data;
int len, ret;
const uint8_t *buf = pkt->data;
int size = pkt->size;
#ifdef LATENCY_CHECK
bigtime_t lat1, lat2;
lat1 = s->player->Latency();
#endif
#ifdef PERF_CHECK
bigtime_t t = s->starve_time;
s->starve_time = 0;
printf("starve_time: %lld \n", t);
#endif
while (size > 0) {
int amount;
len = MIN(size, AUDIO_BLOCK_SIZE);
if (acquire_sem_etc(s->input_sem, len, B_CAN_INTERRUPT, 0LL) < B_OK)
return AVERROR(EIO);
amount = MIN(len, (AUDIO_BUFFER_SIZE - s->input_index));
memcpy(&s->buffer[s->input_index], buf, amount);
s->input_index += amount;
if (s->input_index >= AUDIO_BUFFER_SIZE) {
s->input_index %= AUDIO_BUFFER_SIZE;
memcpy(&s->buffer[s->input_index], buf + amount, len - amount);
s->input_index += len - amount;
}
release_sem_etc(s->output_sem, len, 0);
buf += len;
size -= len;
}
#ifdef LATENCY_CHECK
lat2 = s->player->Latency();
printf("#### BSoundPlayer::Latency(): before= %lld, after= %lld\n", lat1, lat2);
#endif
return 0;
}
static int audio_write_trailer(AVFormatContext *s1)
{
AudioData *s = (AudioData *)s1->priv_data;
audio_close(s);
return 0;
}
/* grab support */
static int audio_read_header(AVFormatContext *s1, AVFormatParameters *ap)
{
AudioData *s = (AudioData *)s1->priv_data;
AVStream *st;
int ret;
if (!ap || ap->sample_rate <= 0 || ap->channels <= 0)
return -1;
st = av_new_stream(s1, 0);
if (!st) {
return AVERROR(ENOMEM);
}
s->sample_rate = ap->sample_rate;
s->channels = ap->channels;
ret = audio_open(s, 0, s1->filename);
if (ret < 0) {
av_free(st);
return AVERROR(EIO);
}
/* take real parameters */
st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
st->codec->codec_id = s->codec_id;
st->codec->sample_rate = s->sample_rate;
st->codec->channels = s->channels;
return 0;
av_set_pts_info(st, 48, 1, 1000000); /* 48 bits pts in us */
}
static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
{
AudioData *s = (AudioData *)s1->priv_data;
int size;
size_t len, amount;
unsigned char *buf;
status_t err;
if (av_new_packet(pkt, s->frame_size) < 0)
return AVERROR(EIO);
buf = (unsigned char *)pkt->data;
size = pkt->size;
while (size > 0) {
len = MIN(AUDIO_BLOCK_SIZE, size);
//printf("acquire_sem(output, %d)\n", len);
while ((err=acquire_sem_etc(s->output_sem, len, B_CAN_INTERRUPT, 0LL)) == B_INTERRUPTED);
if (err < B_OK) {
av_free_packet(pkt);
return AVERROR(EIO);
}
amount = MIN(len, (AUDIO_BUFFER_SIZE - s->output_index));
memcpy(buf, &s->buffer[s->output_index], amount);
s->output_index += amount;
if (s->output_index >= AUDIO_BUFFER_SIZE) {
s->output_index %= AUDIO_BUFFER_SIZE;
memcpy(buf + amount, &s->buffer[s->output_index], len - amount);
s->output_index += len-amount;
s->output_index %= AUDIO_BUFFER_SIZE;
}
release_sem_etc(s->input_sem, len, 0);
//printf("release_sem(input, %d)\n", len);
buf += len;
size -= len;
}
//XXX: add pts info
return 0;
}
static int audio_read_close(AVFormatContext *s1)
{
AudioData *s = (AudioData *)s1->priv_data;
audio_close(s);
return 0;
}
static AVInputFormat audio_beos_demuxer = {
"audio_beos",
NULL_IF_CONFIG_SMALL("audio grab and output"),
sizeof(AudioData),
NULL,
audio_read_header,
audio_read_packet,
audio_read_close,
NULL,
NULL,
AVFMT_NOFILE,
};
AVOutputFormat audio_beos_muxer = {
"audio_beos",
NULL_IF_CONFIG_SMALL("audio grab and output"),
"",
"",
sizeof(AudioData),
#if HAVE_BIGENDIAN
CODEC_ID_PCM_S16BE,
#else
CODEC_ID_PCM_S16LE,
#endif
CODEC_ID_NONE,
audio_write_header,
audio_write_packet,
audio_write_trailer,
AVFMT_NOFILE,
};
extern "C" {
int audio_init(void)
{
main_thid = find_thread(NULL);
av_register_input_format(&audio_beos_demuxer);
av_register_output_format(&audio_beos_muxer);
return 0;
}
} // "C"
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