kolibrios-gitea/contrib/media/fplay/audio.c
Sergey Semyonov (Serge) 2b4519e34d Fplay: fixed broken rewind in software decoder mode
git-svn-id: svn://kolibrios.org@6301 a494cfbc-eb01-0410-851d-a64ba20cac60
2016-03-03 02:14:18 +00:00

405 lines
11 KiB
C

#include <stdint.h>
#include <stdio.h>
#include <string.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libswresample/swresample.h>
#include <kos32sys.h>
#include "winlib/winlib.h"
#include "sound.h"
#include "fplay.h"
astream_t astream;
extern uint8_t *decoder_buffer;
int resampler_size;
volatile int sound_level_0;
volatile int sound_level_1;
volatile enum player_state player_state;
volatile enum player_state decoder_state;
volatile enum player_state sound_state;
static SNDBUF hBuff;
int sample_rate;
static uint32_t samples_written = 0;
int init_audio(vst_t* vst)
{
int err;
int version =-1;
char *errstr;
if((err = InitSound(&version)) !=0 )
{
errstr = "Sound service not installed\n\r";
goto exit_whith_error;
};
if( (SOUND_VERSION>(version&0xFFFF)) ||
(SOUND_VERSION<(version >> 16)))
{
errstr = "Sound service version mismatch\n\r";
goto exit_whith_error;
}
create_thread(audio_thread, vst, 32768);
return 1;
exit_whith_error:
printf(errstr);
return 0;
};
void set_audio_volume(int left, int right)
{
SetVolume(hBuff, left, right);
};
static uint64_t samples_lost;
static double audio_delta;
static double last_time_stamp;
double get_master_clock(void)
{
double tstamp;
GetTimeStamp(hBuff, &tstamp);
return tstamp - audio_delta;
};
int decode_audio(AVCodecContext *ctx, queue_t *qa)
{
static struct SwrContext *swr_ctx;
static int64_t src_layout;
static int src_freq;
static int src_channels;
static enum AVSampleFormat src_fmt = -1;
static AVFrame *aFrame;
AVPacket pkt;
AVPacket pkt_tmp;
int64_t dec_channel_layout;
int len, len2;
int got_frame;
int data_size;
if( astream.count > 192000*2)
return -1;
if( get_packet(qa, &pkt) == 0 )
return 0;
if (!aFrame)
{
if (!(aFrame = av_frame_alloc()))
return -1;
} else
avcodec_get_frame_defaults(aFrame);
pkt_tmp = pkt;
while(pkt_tmp.size > 0)
{
data_size = 192000;
got_frame = 0;
len = avcodec_decode_audio4(ctx, aFrame, &got_frame, &pkt_tmp);
if(len >= 0 && got_frame)
{
char *samples;
int ch, plane_size;
int planar = av_sample_fmt_is_planar(ctx->sample_fmt);
int data_size = av_samples_get_buffer_size(&plane_size, ctx->channels,
aFrame->nb_samples,
ctx->sample_fmt, 1);
pkt_tmp.data += len;
pkt_tmp.size -= len;
dec_channel_layout =
(aFrame->channel_layout && aFrame->channels == av_get_channel_layout_nb_channels(aFrame->channel_layout)) ?
aFrame->channel_layout : av_get_default_channel_layout(aFrame->channels);
if (aFrame->format != src_fmt ||
dec_channel_layout != src_layout ||
aFrame->sample_rate != src_freq ||
!swr_ctx)
{
swr_free(&swr_ctx);
swr_ctx = swr_alloc_set_opts(NULL, AV_CH_LAYOUT_STEREO, AV_SAMPLE_FMT_S16,
aFrame->sample_rate, dec_channel_layout,aFrame->format,
aFrame->sample_rate, 0, NULL);
if (!swr_ctx || swr_init(swr_ctx) < 0)
{
printf("Cannot create sample rate converter for conversion of %d Hz %s %d channels to %d Hz %s %d channels!\n",
aFrame->sample_rate, av_get_sample_fmt_name(aFrame->format), (int)aFrame->channels,
aFrame->sample_rate, av_get_sample_fmt_name(AV_SAMPLE_FMT_S16), 2);
break;
}
src_layout = dec_channel_layout;
src_channels = aFrame->channels;
src_freq = aFrame->sample_rate;
src_fmt = aFrame->format;
};
if (swr_ctx)
{
const uint8_t **in = (const uint8_t **)aFrame->extended_data;
uint8_t *out[] = {decoder_buffer};
int out_count = 192000 * 3 / 2 / av_get_bytes_per_sample(AV_SAMPLE_FMT_S16);
len2 = swr_convert(swr_ctx, out, out_count, in, aFrame->nb_samples);
if (len2 < 0) {
printf("swr_convert() failed\n");
break;
}
if (len2 == out_count) {
printf("warning: audio buffer is probably too small\n");
swr_init(swr_ctx);
}
data_size = len2 * 2 * av_get_bytes_per_sample(AV_SAMPLE_FMT_S16);
mutex_lock(&astream.lock);
samples = astream.buffer+astream.count;
memcpy(samples, decoder_buffer, data_size);
astream.count += data_size;
mutex_unlock(&astream.lock);
};
}
else pkt_tmp.size = 0;
}
av_free_packet(&pkt);
return 1;
};
static void sync_audio(SNDBUF hbuff, int buffsize)
{
SND_EVENT evnt;
uint32_t offset;
double time_stamp;
#ifdef BLACK_MAGIC_SOUND
while( player_state != CLOSED)
{
GetNotify(&evnt);
if(evnt.code != 0xFF000001)
{
printf("invalid event code %d\n\r", evnt.code);
continue;
}
if(evnt.stream != hbuff)
{
printf("invalid stream %x hBuff= %x\n\r",
evnt.stream, hbuff);
continue;
}
GetTimeStamp(hbuff, &time_stamp);
audio_delta = time_stamp - last_time_stamp;
offset = evnt.offset;
mutex_lock(&astream.lock);
{
if(astream.count < buffsize)
{
memset(astream.buffer+astream.count,
0, buffsize-astream.count);
astream.count = buffsize;
};
SetBuffer(hbuff, astream.buffer, offset, buffsize);
samples_written+= buffsize/4;
astream.count -= buffsize;
if(astream.count)
memcpy(astream.buffer, astream.buffer+buffsize, astream.count);
mutex_unlock(&astream.lock);
};
break;
};
#endif
};
int audio_thread(void *param)
{
vst_t *vst = param;
SND_EVENT evnt;
int buffsize;
int samples;
int err;
char *errstr;
int active;
if((err = CreateBuffer(vst->snd_format|PCM_RING,0, &hBuff)) != 0)
{
errstr = "Cannot create sound buffer\n\r";
goto exit_whith_error;
};
SetVolume(hBuff,-900,-900);
if((err = GetBufferSize(hBuff, &buffsize)) != 0)
{
errstr = "Cannot get buffer size\n\r";
goto exit_whith_error;
};
__sync_or_and_fetch(&threads_running,AUDIO_THREAD);
resampler_size = buffsize = buffsize/2;
samples = buffsize/4;
while( player_state != CLOSED)
{
uint32_t offset;
double event_stamp, wait_stamp;
int too_late = 0;
switch(sound_state)
{
case PREPARE:
mutex_lock(&astream.lock);
if(astream.count < buffsize*2)
{
memset(astream.buffer+astream.count,
0, buffsize*2-astream.count);
astream.count = buffsize*2;
};
SetBuffer(hBuff, astream.buffer, 0, buffsize*2);
astream.count -= buffsize*2;
if(astream.count)
memcpy(astream.buffer, astream.buffer+buffsize*2, astream.count);
mutex_unlock(&astream.lock);
SetTimeBase(hBuff, vst->audio_timer_base);
case PAUSE_2_PLAY:
GetTimeStamp(hBuff, &last_time_stamp);
if((err = PlayBuffer(hBuff, 0)) !=0 )
{
errstr = "Cannot play buffer\n\r";
goto exit_whith_error;
};
active = 1;
sync_audio(hBuff, buffsize);
sound_state = PLAY;
/* breaktrough */
case PLAY:
GetNotify(&evnt);
if(evnt.code != 0xFF000001)
{
printf("invalid event code %d\n\r", evnt.code);
continue;
}
if(evnt.stream != hBuff)
{
printf("invalid stream %x hBuff= %x\n\r",
evnt.stream, hBuff);
continue;
};
offset = evnt.offset;
mutex_lock(&astream.lock);
if(astream.count < buffsize)
{
memset(astream.buffer+astream.count,
0, buffsize-astream.count);
astream.count = buffsize;
};
SetBuffer(hBuff, astream.buffer, offset, buffsize);
{
double val = 0;
int16_t *src = (int16_t*)astream.buffer;
int samples = buffsize/2;
int i;
for(i = 0, val = 0; i < samples/2; i++, src++)
if(val < abs(*src))
val= abs(*src); // * *src;
sound_level_0 = val; //sqrt(val / (samples/2));
for(i = 0, val = 0; i < samples/2; i++, src++)
if(val < abs(*src))
val= abs(*src); // * *src;
sound_level_1 = val; //sqrt(val / (samples/2));
// printf("%d\n", sound_level);
};
samples_written+= buffsize/4;
astream.count -= buffsize;
if(astream.count)
memcpy(astream.buffer, astream.buffer+buffsize, astream.count);
mutex_unlock(&astream.lock);
break;
case PLAY_2_STOP:
if( active )
{
ResetBuffer(hBuff, SND_RESET_ALL);
vst->audio_timer_valid = 0;
active = 0;
}
sound_state = STOP;
break;
case PLAY_2_PAUSE:
if( active )
{
StopBuffer(hBuff);
};
sound_state = PAUSE;
case PAUSE:
case STOP:
delay(1);
};
}
__sync_and_and_fetch(&threads_running,~AUDIO_THREAD);
StopBuffer(hBuff);
DestroyBuffer(hBuff);
return 0;
exit_whith_error:
printf(errstr);
return -1;
};