* BeOS audio play interface
* Copyright (c) 2000, 2001 Fabrice Bellard.
*
- * This library is free software; you can redistribute it and/or
+ * 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 of the License, or (at your option) any later version.
+ * version 2.1 of the License, or (at your option) any later version.
*
- * This library is distributed in the hope that it will be useful,
+ * 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 this library; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ * 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 "avformat.h"
}
+#ifdef HAVE_BSOUNDRECORDER
+#include <SoundRecorder.h>
+using namespace BPrivate::Media::Experimental;
+#endif
+
/* enable performance checks */
//#define PERF_CHECK
-//const char *audio_device = "/dev/dsp";
-const char *audio_device = "beosaudio:";
+/* enable Media Kit latency checks */
+//#define LATENCY_CHECK
-/* Pipes are 4k in BeOS IIRC */
#define AUDIO_BLOCK_SIZE 4096
-//#define AUDIO_BLOCK_SIZE 2048
#define AUDIO_BLOCK_COUNT 8
#define AUDIO_BUFFER_SIZE (AUDIO_BLOCK_SIZE*AUDIO_BLOCK_COUNT)
-/* pipes suck for realtime */
-#define USE_RING_BUFFER 1
-
typedef struct {
- int fd;
+ int fd; // UNUSED
int sample_rate;
int channels;
int frame_size; /* in bytes ! */
CodecID codec_id;
- int flip_left : 1;
- UINT8 buffer[AUDIO_BUFFER_SIZE];
+ uint8_t buffer[AUDIO_BUFFER_SIZE];
int buffer_ptr;
- int pipefd; /* the other end of the pipe */
/* ring buffer */
sem_id input_sem;
int input_index;
sem_id output_sem;
int output_index;
- int queued;
BSoundPlayer *player;
+#ifdef HAVE_BSOUNDRECORDER
+ BSoundRecorder *recorder;
+#endif
int has_quit; /* signal callbacks not to wait */
volatile bigtime_t starve_time;
} AudioData;
bigtime_t t;
t = system_time();
#endif
-#ifdef USE_RING_BUFFER
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->output_index %= AUDIO_BUFFER_SIZE;
}
release_sem_etc(s->input_sem, len, 0);
-#else
- len = read(s->pipefd, buf, bufferSize);
-#endif
#ifdef PERF_CHECK
t = system_time() - t;
s->starve_time = MAX(s->starve_time, t);
#endif
-#ifndef USE_RING_BUFFER
- if (len < B_OK) {
- puts("EPIPE");
- s->player->SetHasData(false);
- snooze(100000);
+ buf += len;
+ bufferSize -= len;
+ }
+}
+
+#ifdef 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;
}
- if (len == 0) {
- s->player->SetHasData(false);
- snooze(100000);
- 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;
}
-#endif
+ 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)
+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;
+#ifndef HAVE_BSOUNDRECORDER
if (!is_output)
- return -EIO; /* not for now */
-#ifdef USE_RING_BUFFER
+ return AVERROR(EIO); /* not for now */
+#endif
s->input_sem = create_sem(AUDIO_BUFFER_SIZE, "ffmpeg_ringbuffer_input");
-// s->input_sem = create_sem(AUDIO_BLOCK_SIZE, "ffmpeg_ringbuffer_input");
if (s->input_sem < B_OK)
- return -EIO;
+ return AVERROR(EIO);
s->output_sem = create_sem(0, "ffmpeg_ringbuffer_output");
if (s->output_sem < B_OK) {
delete_sem(s->input_sem);
- return -EIO;
+ return AVERROR(EIO);
}
s->input_index = 0;
s->output_index = 0;
- s->queued = 0;
-#else
- ret = pipe(p);
- if (ret < 0)
- return -EIO;
- s->fd = p[is_output?1:0];
- s->pipefd = p[is_output?0:1];
- if (s->fd < 0) {
- perror(is_output?"audio out":"audio in");
- return -EIO;
- }
-#endif
create_bapp_if_needed();
- /* non blocking mode */
-// fcntl(s->fd, F_SETFL, O_NONBLOCK);
-// fcntl(s->pipefd, F_SETFL, O_NONBLOCK);
s->frame_size = AUDIO_BLOCK_SIZE;
+ /* bump up the priority (avoid realtime though) */
+ set_thread_priority(find_thread(NULL), B_DISPLAY_PRIORITY+1);
+#ifdef 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;
if (s->player->InitCheck() != B_OK) {
delete s->player;
s->player = NULL;
-#ifdef USE_RING_BUFFER
- if (s->input_sem)
- delete_sem(s->input_sem);
- if (s->output_sem)
- delete_sem(s->output_sem);
-#else
- close(s->fd);
- close(s->pipefd);
-#endif
- return -EIO;
+ 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);
- /* bump up the priority (avoid realtime though) */
- set_thread_priority(find_thread(NULL), B_DISPLAY_PRIORITY+1);
return 0;
}
static int audio_close(AudioData *s)
{
-#ifdef USE_RING_BUFFER
if (s->input_sem)
delete_sem(s->input_sem);
if (s->output_sem)
delete_sem(s->output_sem);
-#endif
s->has_quit = 1;
if (s->player) {
s->player->Stop();
}
if (s->player)
delete s->player;
-#ifndef USE_RING_BUFFER
- close(s->pipefd);
- close(s->fd);
+#ifdef HAVE_BSOUNDRECORDER
+ if (s->recorder)
+ delete s->recorder;
#endif
destroy_bapp_if_needed();
return 0;
int ret;
st = s1->streams[0];
- s->sample_rate = st->codec.sample_rate;
- s->channels = st->codec.channels;
- ret = audio_open(s, 1);
+ s->sample_rate = st->codec->sample_rate;
+ s->channels = st->codec->channels;
+ ret = audio_open(s, 1, NULL);
if (ret < 0)
- return -EIO;
+ return AVERROR(EIO);
return 0;
}
static int audio_write_packet(AVFormatContext *s1, int stream_index,
- UINT8 *buf, int size, int force_pts)
+ const uint8_t *buf, int size, int64_t force_pts)
{
AudioData *s = (AudioData *)s1->priv_data;
int len, ret;
+#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
-#ifdef USE_RING_BUFFER
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 -EIO;
+ return AVERROR(EIO);
amount = MIN(len, (AUDIO_BUFFER_SIZE - s->input_index));
memcpy(&s->buffer[s->input_index], buf, amount);
s->input_index += amount;
buf += len;
size -= len;
}
-#else
- while (size > 0) {
- len = AUDIO_BLOCK_SIZE - s->buffer_ptr;
- if (len > size)
- len = size;
- memcpy(s->buffer + s->buffer_ptr, buf, len);
- s->buffer_ptr += len;
- if (s->buffer_ptr >= AUDIO_BLOCK_SIZE) {
- for(;;) {
-//snooze(1000);
- ret = write(s->fd, s->buffer, AUDIO_BLOCK_SIZE);
- if (ret != 0)
- break;
- if (ret < 0 && (errno != EAGAIN && errno != EINTR))
- return -EIO;
- }
- s->buffer_ptr = 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;
}
st = av_new_stream(s1, 0);
if (!st) {
- return -ENOMEM;
+ return AVERROR(ENOMEM);
}
s->sample_rate = ap->sample_rate;
s->channels = ap->channels;
- ret = audio_open(s, 0);
+ ret = audio_open(s, 0, s1->filename);
if (ret < 0) {
av_free(st);
- return -EIO;
- } else {
- /* take real parameters */
- st->codec.codec_type = CODEC_TYPE_AUDIO;
- st->codec.codec_id = s->codec_id;
- st->codec.sample_rate = s->sample_rate;
- st->codec.channels = s->channels;
- return 0;
+ return AVERROR(EIO);
}
+ /* take real parameters */
+ st->codec->codec_type = CODEC_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(s1, 48, 1, 1000000); /* 48 bits pts in us */
}
static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
{
AudioData *s = (AudioData *)s1->priv_data;
- int ret;
+ int size;
+ size_t len, amount;
+ unsigned char *buf;
+ status_t err;
if (av_new_packet(pkt, s->frame_size) < 0)
- return -EIO;
- for(;;) {
- ret = read(s->fd, pkt->data, pkt->size);
- if (ret > 0)
- break;
- if (ret == -1 && (errno == EAGAIN || errno == EINTR)) {
- av_free_packet(pkt);
- pkt->size = 0;
- return 0;
- }
- if (!(ret == 0 || (ret == -1 && (errno == EAGAIN || errno == EINTR)))) {
+ 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 -EIO;
+ return AVERROR(EIO);
}
- }
- pkt->size = ret;
- if (s->flip_left && s->channels == 2) {
- int i;
- short *p = (short *) pkt->data;
-
- for (i = 0; i < ret; i += 4) {
- *p = ~*p;
- p += 2;
+ 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;
}
return 0;
}
-AVInputFormat audio_in_format = {
- "audio_device",
+static AVInputFormat audio_beos_demuxer = {
+ "audio_beos",
"audio grab and output",
sizeof(AudioData),
NULL,
AVFMT_NOFILE,
};
-AVOutputFormat audio_out_format = {
- "audio_device",
+AVOutputFormat audio_beos_muxer = {
+ "audio_beos",
"audio grab and output",
"",
"",
int audio_init(void)
{
main_thid = find_thread(NULL);
- av_register_input_format(&audio_in_format);
- av_register_output_format(&audio_out_format);
+ av_register_input_format(&audio_beos_demuxer);
+ av_register_output_format(&audio_beos_muxer);
return 0;
}
projects / ffmpeg / blobdiff