forked from KolibriOS/kolibrios
421 lines
12 KiB
C
421 lines
12 KiB
C
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#include <stdint.h>
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#include <stdio.h>
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#include <string.h>
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#include <libavcodec/avcodec.h>
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#include <libavformat/avformat.h>
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#include <libswresample/swresample.h>
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#include <kos32sys.h>
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#include "winlib/winlib.h"
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#include "sound.h"
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#include "fplay.h"
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astream_t astream;
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extern uint8_t *decoder_buffer;
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int resampler_size;
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volatile int sound_level_0;
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volatile int sound_level_1;
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volatile enum player_state player_state;
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volatile enum player_state decoder_state;
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volatile enum player_state sound_state;
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static SNDBUF hBuff;
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static int snd_format;
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int sample_rate;
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static uint32_t samples_written = 0;
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double audio_base = -1.0;
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double get_audio_base();
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int init_audio(int format)
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{
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int err;
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int version =-1;
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char *errstr;
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if((err = InitSound(&version)) !=0 )
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{
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errstr = "Sound service not installed\n\r";
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goto exit_whith_error;
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};
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if( (SOUND_VERSION>(version&0xFFFF)) ||
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(SOUND_VERSION<(version >> 16)))
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{
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errstr = "Sound service version mismatch\n\r";
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goto exit_whith_error;
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}
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snd_format = format;
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create_thread(audio_thread, 0, 163840);
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return 1;
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exit_whith_error:
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printf(errstr);
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return 0;
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};
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void set_audio_volume(int left, int right)
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{
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SetVolume(hBuff, left, right);
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};
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static uint64_t samples_lost;
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static double audio_delta;
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static double last_time_stamp;
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double get_master_clock(void)
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{
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double tstamp;
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GetTimeStamp(hBuff, &tstamp);
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return tstamp - audio_delta;
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};
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int decode_audio(AVCodecContext *ctx, queue_t *qa)
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{
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static struct SwrContext *swr_ctx;
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static int64_t src_layout;
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static int src_freq;
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static int src_channels;
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static enum AVSampleFormat src_fmt = -1;
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static AVFrame *aFrame;
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AVPacket pkt;
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AVPacket pkt_tmp;
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int64_t dec_channel_layout;
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int len, len2;
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int got_frame;
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int data_size;
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if( astream.count > 192000*2)
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return -1;
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if( get_packet(qa, &pkt) == 0 )
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return 0;
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if (!aFrame)
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{
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if (!(aFrame = av_frame_alloc()))
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return -1;
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} else
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avcodec_get_frame_defaults(aFrame);
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pkt_tmp = pkt;
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while(pkt_tmp.size > 0)
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{
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data_size = 192000;
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got_frame = 0;
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len = avcodec_decode_audio4(ctx, aFrame, &got_frame, &pkt_tmp);
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if(len >= 0 && got_frame)
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{
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char *samples;
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int ch, plane_size;
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int planar = av_sample_fmt_is_planar(ctx->sample_fmt);
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int data_size = av_samples_get_buffer_size(&plane_size, ctx->channels,
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aFrame->nb_samples,
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ctx->sample_fmt, 1);
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// if(audio_base == -1.0)
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// {
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// if (pkt.pts != AV_NOPTS_VALUE)
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// audio_base = get_audio_base() * pkt.pts;
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// printf("audio base %f\n", audio_base);
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// };
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pkt_tmp.data += len;
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pkt_tmp.size -= len;
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dec_channel_layout =
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(aFrame->channel_layout && aFrame->channels == av_get_channel_layout_nb_channels(aFrame->channel_layout)) ?
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aFrame->channel_layout : av_get_default_channel_layout(aFrame->channels);
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if (aFrame->format != src_fmt ||
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dec_channel_layout != src_layout ||
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aFrame->sample_rate != src_freq ||
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!swr_ctx)
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{
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swr_free(&swr_ctx);
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swr_ctx = swr_alloc_set_opts(NULL, AV_CH_LAYOUT_STEREO, AV_SAMPLE_FMT_S16,
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aFrame->sample_rate, dec_channel_layout,aFrame->format,
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aFrame->sample_rate, 0, NULL);
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if (!swr_ctx || swr_init(swr_ctx) < 0)
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{
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printf("Cannot create sample rate converter for conversion of %d Hz %s %d channels to %d Hz %s %d channels!\n",
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aFrame->sample_rate, av_get_sample_fmt_name(aFrame->format), (int)aFrame->channels,
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aFrame->sample_rate, av_get_sample_fmt_name(AV_SAMPLE_FMT_S16), 2);
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break;
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}
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src_layout = dec_channel_layout;
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src_channels = aFrame->channels;
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src_freq = aFrame->sample_rate;
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src_fmt = aFrame->format;
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};
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if (swr_ctx)
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{
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const uint8_t **in = (const uint8_t **)aFrame->extended_data;
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uint8_t *out[] = {decoder_buffer};
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int out_count = 192000 * 3 / 2 / av_get_bytes_per_sample(AV_SAMPLE_FMT_S16);
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len2 = swr_convert(swr_ctx, out, out_count, in, aFrame->nb_samples);
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if (len2 < 0) {
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printf("swr_convert() failed\n");
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break;
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}
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if (len2 == out_count) {
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printf("warning: audio buffer is probably too small\n");
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swr_init(swr_ctx);
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}
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data_size = len2 * 2 * av_get_bytes_per_sample(AV_SAMPLE_FMT_S16);
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mutex_lock(&astream.lock);
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samples = astream.buffer+astream.count;
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memcpy(samples, decoder_buffer, data_size);
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astream.count += data_size;
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mutex_unlock(&astream.lock);
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};
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}
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else pkt_tmp.size = 0;
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}
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av_free_packet(&pkt);
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return 1;
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};
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static void sync_audio(SNDBUF hbuff, int buffsize)
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{
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SND_EVENT evnt;
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uint32_t offset;
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double time_stamp;
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#ifdef BLACK_MAGIC_SOUND
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while( player_state != CLOSED)
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{
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GetNotify(&evnt);
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if(evnt.code != 0xFF000001)
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{
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printf("invalid event code %d\n\r", evnt.code);
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continue;
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}
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if(evnt.stream != hbuff)
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{
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printf("invalid stream %x hBuff= %x\n\r",
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evnt.stream, hbuff);
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continue;
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}
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GetTimeStamp(hbuff, &time_stamp);
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audio_delta = time_stamp - last_time_stamp;
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offset = evnt.offset;
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mutex_lock(&astream.lock);
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{
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if(astream.count < buffsize)
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{
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memset(astream.buffer+astream.count,
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0, buffsize-astream.count);
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astream.count = buffsize;
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};
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SetBuffer(hbuff, astream.buffer, offset, buffsize);
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samples_written+= buffsize/4;
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astream.count -= buffsize;
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if(astream.count)
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memcpy(astream.buffer, astream.buffer+buffsize, astream.count);
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mutex_unlock(&astream.lock);
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};
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break;
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};
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#endif
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};
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int audio_thread(void *param)
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{
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SND_EVENT evnt;
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int buffsize;
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int samples;
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int err;
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char *errstr;
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int active;
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if((err = CreateBuffer(snd_format|PCM_RING,0, &hBuff)) != 0)
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{
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errstr = "Cannot create sound buffer\n\r";
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goto exit_whith_error;
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};
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SetVolume(hBuff,-900,-900);
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if((err = GetBufferSize(hBuff, &buffsize)) != 0)
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{
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errstr = "Cannot get buffer size\n\r";
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goto exit_whith_error;
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};
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__sync_or_and_fetch(&threads_running,AUDIO_THREAD);
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resampler_size = buffsize = buffsize/2;
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samples = buffsize/4;
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while( player_state != CLOSED)
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{
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uint32_t offset;
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double event_stamp, wait_stamp;
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int too_late = 0;
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switch(sound_state)
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{
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case PREPARE:
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mutex_lock(&astream.lock);
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if(astream.count < buffsize*2)
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{
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memset(astream.buffer+astream.count,
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0, buffsize*2-astream.count);
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astream.count = buffsize*2;
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};
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SetBuffer(hBuff, astream.buffer, 0, buffsize*2);
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astream.count -= buffsize*2;
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if(astream.count)
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memcpy(astream.buffer, astream.buffer+buffsize*2, astream.count);
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mutex_unlock(&astream.lock);
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SetTimeBase(hBuff, audio_base);
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case PAUSE_2_PLAY:
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GetTimeStamp(hBuff, &last_time_stamp);
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// printf("last audio time stamp %f\n", last_time_stamp);
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if((err = PlayBuffer(hBuff, 0)) !=0 )
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{
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errstr = "Cannot play buffer\n\r";
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goto exit_whith_error;
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};
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active = 1;
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sync_audio(hBuff, buffsize);
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sound_state = PLAY;
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// printf("render: set audio latency to %f\n", audio_delta);
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/* breaktrough */
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case PLAY:
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GetNotify(&evnt);
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if(evnt.code != 0xFF000001)
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{
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printf("invalid event code %d\n\r", evnt.code);
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continue;
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}
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if(evnt.stream != hBuff)
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{
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printf("invalid stream %x hBuff= %x\n\r",
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evnt.stream, hBuff);
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continue;
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};
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offset = evnt.offset;
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mutex_lock(&astream.lock);
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if(astream.count < buffsize)
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{
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memset(astream.buffer+astream.count,
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0, buffsize-astream.count);
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astream.count = buffsize;
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};
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SetBuffer(hBuff, astream.buffer, offset, buffsize);
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{
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double val = 0;
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int16_t *src = (int16_t*)astream.buffer;
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int samples = buffsize/2;
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int i;
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for(i = 0, val = 0; i < samples/2; i++, src++)
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if(val < abs(*src))
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val= abs(*src); // * *src;
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sound_level_0 = val; //sqrt(val / (samples/2));
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for(i = 0, val = 0; i < samples/2; i++, src++)
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if(val < abs(*src))
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val= abs(*src); // * *src;
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sound_level_1 = val; //sqrt(val / (samples/2));
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// printf("%d\n", sound_level);
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};
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samples_written+= buffsize/4;
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astream.count -= buffsize;
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if(astream.count)
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memcpy(astream.buffer, astream.buffer+buffsize, astream.count);
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mutex_unlock(&astream.lock);
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break;
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case PLAY_2_STOP:
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if( active )
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{
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ResetBuffer(hBuff, SND_RESET_ALL);
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audio_base = -1.0;
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active = 0;
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}
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sound_state = STOP;
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break;
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case PLAY_2_PAUSE:
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if( active )
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{
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StopBuffer(hBuff);
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};
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sound_state = PAUSE;
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case PAUSE:
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case STOP:
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delay(1);
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};
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}
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__sync_and_and_fetch(&threads_running,~AUDIO_THREAD);
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StopBuffer(hBuff);
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DestroyBuffer(hBuff);
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return 0;
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exit_whith_error:
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printf(errstr);
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return -1;
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};
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