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experimental BeOS audio input support. (needs unreleased library)

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Originally committed as revision 1712 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
François Revol
2003-03-27 14:44:30 +00:00
parent 50515cddf3
commit 788a2d3d8b
1 changed files with 110 additions and 26 deletions
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136
libavformat/beosaudio.cpp
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@@ -31,6 +31,10 @@ extern "C" {
#include "avformat.h"
}
#ifdef HAVE_BSOUNDRECORDER
#include <SoundRecorder.h>
#endif
/* enable performance checks */
//#define PERF_CHECK
@@ -55,6 +59,9 @@ typedef struct {
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;
@@ -139,14 +146,52 @@ static void audioplay_callback(void *cookie, void *buffer, size_t bufferSize, co
}
}
#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;
}
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)
{
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 */
#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)
@@ -161,6 +206,30 @@ static int audio_open(AudioData *s, int is_output)
s->queued = 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);
#ifdef HAVE_BSOUNDRECORDER
if (!is_output) {
s->recorder = new BSoundRecorder(&iformat, false, "ffmpeg input", audiorecord_callback);
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 -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;
@@ -171,18 +240,16 @@ static int audio_open(AudioData *s, int is_output)
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);
if (s->input_sem)
delete_sem(s->input_sem);
if (s->output_sem)
delete_sem(s->output_sem);
return -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;
}
@@ -198,6 +265,10 @@ static int audio_close(AudioData *s)
}
if (s->player)
delete s->player;
#ifdef HAVE_BSOUNDRECORDER
if (s->recorder)
delete s->recorder;
#endif
destroy_bapp_if_needed();
return 0;
}
@@ -278,38 +349,51 @@ static int audio_read_header(AVFormatContext *s1, AVFormatParameters *ap)
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;
}
/* 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)))) {
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;
}
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;
}
pkt->size = ret;
//XXX: add pts info
return 0;
}
@@ -321,7 +405,7 @@ static int audio_read_close(AVFormatContext *s1)
return 0;
}
AVInputFormat audio_in_format = {
static AVInputFormat audio_in_format = {
"audio_device",
"audio grab and output",
sizeof(AudioData),