diff --git a/codec.c b/codec.c
index 41b0774..ce1e500 100644
--- a/codec.c
+++ b/codec.c
@@ -30,11 +30,6 @@
 ///		many bugs and incompatiblity in it.  Don't use this shit.
 ///
 
-    /**
-    **	use av_parser to support insane dvb audio streams.
-    */
-#define noUSE_AVPARSER
-
     /// compile with passthrough support (experimental)
 #define USE_PASSTHROUGH
 
@@ -603,10 +598,6 @@ struct _audio_decoder_
     AVCodec *AudioCodec;		///< audio codec
     AVCodecContext *AudioCtx;		///< audio codec context
 
-#ifdef USE_AVPARSER
-    /// audio parser to support insane dvb streaks
-    AVCodecParserContext *AudioParser;
-#endif
     int PassthroughAC3;			///< current ac-3 pass-through
     int SampleRate;			///< current stream sample rate
     int Channels;			///< current stream channels
@@ -615,7 +606,6 @@ struct _audio_decoder_
     int HwChannels;			///< hw channels
 
     ReSampleContext *ReSample;		///< audio resampling context
-
 };
 
 #ifdef USE_PASSTHROUGH
@@ -667,7 +657,6 @@ void CodecAudioOpen(AudioDecoder * audio_decoder, const char *name,
     int codec_id)
 {
     AVCodec *audio_codec;
-    //AVDictionary *av_dict;
 
     if (name && (audio_codec = avcodec_find_decoder_by_name(name))) {
 	Debug(3, "codec: audio decoder '%s' found\n", name);
@@ -694,15 +683,20 @@ void CodecAudioOpen(AudioDecoder * audio_decoder, const char *name,
 	Fatal(_("codec: can't open audio codec\n"));
     }
 #else
-    //av_dict = NULL;
-    //av_dict_set(&av_dict, "dmix_mode", "0", 0);
-    //av_dict_set(&av_dict, "ltrt_cmixlev", "1.414", 0);
-    //av_dict_set(&av_dict, "loro_cmixlev", "1.414", 0);
-    if (avcodec_open2(audio_decoder->AudioCtx, audio_codec, NULL) < 0) {
-	pthread_mutex_unlock(&CodecLockMutex);
-	Fatal(_("codec: can't open audio codec\n"));
+    if (1) {
+	AVDictionary *av_dict;
+
+	av_dict = NULL;
+	// FIXME: import settings
+	//av_dict_set(&av_dict, "dmix_mode", "0", 0);
+	//av_dict_set(&av_dict, "ltrt_cmixlev", "1.414", 0);
+	//av_dict_set(&av_dict, "loro_cmixlev", "1.414", 0);
+	if (avcodec_open2(audio_decoder->AudioCtx, audio_codec, &av_dict) < 0) {
+	    pthread_mutex_unlock(&CodecLockMutex);
+	    Fatal(_("codec: can't open audio codec\n"));
+	}
+	av_dict_free(&av_dict);
     }
-    //av_dict_free(&av_dict);
 #endif
     pthread_mutex_unlock(&CodecLockMutex);
     Debug(3, "codec: audio '%s'\n", audio_decoder->AudioCtx->codec_name);
@@ -712,12 +706,6 @@ void CodecAudioOpen(AudioDecoder * audio_decoder, const char *name,
 	// we do not send complete frames
 	audio_decoder->AudioCtx->flags |= CODEC_FLAG_TRUNCATED;
     }
-#ifdef USE_AVPARSER
-    if (!(audio_decoder->AudioParser =
-	    av_parser_init(audio_decoder->AudioCtx->codec_id))) {
-	Fatal(_("codec: can't init audio parser\n"));
-    }
-#endif
     audio_decoder->SampleRate = 0;
     audio_decoder->Channels = 0;
     audio_decoder->HwSampleRate = 0;
@@ -736,12 +724,6 @@ void CodecAudioClose(AudioDecoder * audio_decoder)
 	audio_resample_close(audio_decoder->ReSample);
 	audio_decoder->ReSample = NULL;
     }
-#ifdef USE_AVPARSER
-    if (audio_decoder->AudioParser) {
-	av_parser_close(audio_decoder->AudioParser);
-	audio_decoder->AudioParser = NULL;
-    }
-#endif
     if (audio_decoder->AudioCtx) {
 	pthread_mutex_lock(&CodecLockMutex);
 	avcodec_close(audio_decoder->AudioCtx);
@@ -827,268 +809,6 @@ static void CodecReorderAudioFrame(int16_t * buf, int size, int channels)
     }
 }
 
-#ifdef USE_AVPARSER
-
-/**
-**	Decode an audio packet.
-**
-**	PTS must be handled self.
-**
-**	@param audio_decoder	audio decoder data
-**	@param avpkt		audio packet
-*/
-void CodecAudioDecodeOld(AudioDecoder * audio_decoder, const AVPacket * avpkt)
-{
-    int16_t buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 4 +
-	FF_INPUT_BUFFER_PADDING_SIZE] __attribute__ ((aligned(16)));
-    AVCodecContext *audio_ctx;
-    int index;
-
-//#define spkt avpkt
-#if 1					// didn't fix crash in av_parser_parse2
-    AVPacket spkt[1];
-
-    // av_new_packet reserves FF_INPUT_BUFFER_PADDING_SIZE and clears it
-    if (av_new_packet(spkt, avpkt->size)) {
-	Error(_("codec: out of memory\n"));
-	return;
-    }
-    memcpy(spkt->data, avpkt->data, avpkt->size);
-    spkt->pts = avpkt->pts;
-    spkt->dts = avpkt->dts;
-#endif
-#ifdef DEBUG
-    if (!audio_decoder->AudioParser) {
-	Fatal(_("codec: internal error parser freeded while running\n"));
-    }
-#endif
-
-    audio_ctx = audio_decoder->AudioCtx;
-    index = 0;
-    while (spkt->size > index) {
-	int n;
-	int l;
-	AVPacket dpkt[1];
-
-	av_init_packet(dpkt);
-	n = av_parser_parse2(audio_decoder->AudioParser, audio_ctx,
-	    &dpkt->data, &dpkt->size, spkt->data + index, spkt->size - index,
-	    !index ? spkt->pts : (int64_t) AV_NOPTS_VALUE,
-	    !index ? spkt->dts : (int64_t) AV_NOPTS_VALUE, -1);
-
-	// FIXME: make this a function for both #ifdef cases
-	if (dpkt->size) {
-	    int buf_sz;
-
-	    dpkt->pts = audio_decoder->AudioParser->pts;
-	    dpkt->dts = audio_decoder->AudioParser->dts;
-	    buf_sz = sizeof(buf);
-	    l = avcodec_decode_audio3(audio_ctx, buf, &buf_sz, dpkt);
-	    if (l == AVERROR(EAGAIN)) {
-		index += n;		// this is needed for aac latm
-		continue;
-	    }
-	    if (l < 0) {		// no audio frame could be decompressed
-		Error(_("codec: error audio data at %d\n"), index);
-		break;
-	    }
-#ifdef notyetFF_API_OLD_DECODE_AUDIO
-	    // FIXME: ffmpeg git comeing
-	    int got_frame;
-
-	    avcodec_decode_audio4(audio_ctx, frame, &got_frame, dpkt);
-#else
-#endif
-	    // Update audio clock
-	    if (dpkt->pts != (int64_t) AV_NOPTS_VALUE) {
-		AudioSetClock(dpkt->pts);
-	    }
-	    // FIXME: must first play remainings bytes, than change and play new.
-	    if (audio_decoder->PassthroughAC3 != CodecPassthroughAC3
-		|| audio_decoder->SampleRate != audio_ctx->sample_rate
-		|| audio_decoder->Channels != audio_ctx->channels) {
-		int err;
-		int isAC3;
-
-		audio_decoder->PassthroughAC3 = CodecPassthroughAC3;
-		// FIXME: use swr_convert from swresample (only in ffmpeg!)
-		if (audio_decoder->ReSample) {
-		    audio_resample_close(audio_decoder->ReSample);
-		    audio_decoder->ReSample = NULL;
-		}
-
-		audio_decoder->SampleRate = audio_ctx->sample_rate;
-		audio_decoder->HwSampleRate = audio_ctx->sample_rate;
-		audio_decoder->Channels = audio_ctx->channels;
-		// SPDIF/HDMI passthrough
-		if (CodecPassthroughAC3 && audio_ctx->codec_id == CODEC_ID_AC3) {
-		    audio_decoder->HwChannels = 2;
-		    isAC3 = 1;
-		} else {
-		    audio_decoder->HwChannels = audio_ctx->channels;
-		    isAC3 = 0;
-		}
-
-		// channels not support?
-		if ((err =
-			AudioSetup(&audio_decoder->HwSampleRate,
-			    &audio_decoder->HwChannels, isAC3))) {
-		    Debug(3, "codec/audio: resample %dHz *%d -> %dHz *%d\n",
-			audio_ctx->sample_rate, audio_ctx->channels,
-			audio_decoder->HwSampleRate,
-			audio_decoder->HwChannels);
-
-		    if (err == 1) {
-			audio_decoder->ReSample =
-			    av_audio_resample_init(audio_decoder->HwChannels,
-			    audio_ctx->channels, audio_decoder->HwSampleRate,
-			    audio_ctx->sample_rate, audio_ctx->sample_fmt,
-			    audio_ctx->sample_fmt, 16, 10, 0, 0.8);
-			// libav-0.8_pre didn't support 6 -> 2 channels
-			if (!audio_decoder->ReSample) {
-			    Error(_("codec/audio: resample setup error\n"));
-			    audio_decoder->HwChannels = 0;
-			    audio_decoder->HwSampleRate = 0;
-			}
-		    } else {
-			Debug(3, "codec/audio: audio setup error\n");
-			// FIXME: handle errors
-			audio_decoder->HwChannels = 0;
-			audio_decoder->HwSampleRate = 0;
-			break;
-		    }
-		}
-	    }
-
-	    if (audio_decoder->HwSampleRate && audio_decoder->HwChannels) {
-		// need to resample audio
-		if (audio_decoder->ReSample) {
-		    int16_t outbuf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 4 +
-			FF_INPUT_BUFFER_PADDING_SIZE]
-			__attribute__ ((aligned(16)));
-		    int outlen;
-
-		    // FIXME: libav-0.7.2 crash here
-		    outlen =
-			audio_resample(audio_decoder->ReSample, outbuf, buf,
-			buf_sz);
-#ifdef DEBUG
-		    if (outlen != buf_sz) {
-			Debug(3, "codec/audio: possible fixed ffmpeg\n");
-		    }
-#endif
-		    if (outlen) {
-			// outlen seems to be wrong in ffmpeg-0.9
-			outlen /= audio_decoder->Channels *
-			    av_get_bytes_per_sample(audio_ctx->sample_fmt);
-			outlen *=
-			    audio_decoder->HwChannels *
-			    av_get_bytes_per_sample(audio_ctx->sample_fmt);
-			Debug(4, "codec/audio: %d -> %d\n", buf_sz, outlen);
-			CodecReorderAudioFrame(outbuf, outlen,
-			    audio_decoder->HwChannels);
-			AudioEnqueue(outbuf, outlen);
-		    }
-		} else {
-#ifdef USE_PASSTHROUGH
-		    // SPDIF/HDMI passthrough
-		    if (CodecPassthroughAC3
-			&& audio_ctx->codec_id == CODEC_ID_AC3) {
-			// build SPDIF header and append A52 audio to it
-			// dpkt is the original data
-			buf_sz = 6144;
-			if (buf_sz < dpkt->size + 8) {
-			    Error(_
-				("codec/audio: decoded data smaller than encoded\n"));
-			    break;
-			}
-			// copy original data for output
-			// FIXME: not 100% sure, if endian is correct
-			buf[0] = htole16(0xF872);	// iec 61937 sync word
-			buf[1] = htole16(0x4E1F);
-			buf[2] = htole16(0x01 | (dpkt->data[5] & 0x07) << 8);
-			buf[3] = htole16(dpkt->size * 8);
-			swab(dpkt->data, buf + 4, dpkt->size);
-			memset(buf + 4 + dpkt->size / 2, 0,
-			    buf_sz - 8 - dpkt->size);
-		    }
-#if 0
-		    //
-		    //	old experimental code
-		    //
-		    if (1) {
-			// FIXME: need to detect dts
-			// copy original data for output
-			// FIXME: buf is sint
-			buf[0] = 0x72;
-			buf[1] = 0xF8;
-			buf[2] = 0x1F;
-			buf[3] = 0x4E;
-			buf[4] = 0x00;
-			switch (dpkt->size) {
-			    case 512:
-				buf[5] = 0x0B;
-				break;
-			    case 1024:
-				buf[5] = 0x0C;
-				break;
-			    case 2048:
-				buf[5] = 0x0D;
-				break;
-			    default:
-				Debug(3,
-				    "codec/audio: dts sample burst not supported\n");
-				buf[5] = 0x00;
-				break;
-			}
-			buf[6] = (dpkt->size * 8);
-			buf[7] = (dpkt->size * 8) >> 8;
-			//buf[8] = 0x0B;
-			//buf[9] = 0x77;
-			//printf("%x %x\n", dpkt->data[0],dpkt->data[1]);
-			// swab?
-			memcpy(buf + 8, dpkt->data, dpkt->size);
-			memset(buf + 8 + dpkt->size, 0,
-			    buf_sz - 8 - dpkt->size);
-		    } else if (1) {
-			// FIXME: need to detect mp2
-			// FIXME: mp2 passthrough
-			// see softhddev.c version/layer
-			// 0x04 mpeg1 layer1
-			// 0x05 mpeg1 layer23
-			// 0x06 mpeg2 ext
-			// 0x07 mpeg2.5 layer 1
-			// 0x08 mpeg2.5 layer 2
-			// 0x09 mpeg2.5 layer 3
-		    }
-		    // DTS HD?
-		    // True HD?
-#endif
-#endif
-		    CodecReorderAudioFrame(buf, buf_sz,
-			audio_decoder->HwChannels);
-		    AudioEnqueue(buf, buf_sz);
-		}
-	    }
-
-	    if (dpkt->size > l) {
-		Error(_("codec: error more than one frame data\n"));
-	    }
-	}
-
-	index += n;
-    }
-
-#if 1
-    // or av_free_packet, make no difference here
-    av_destruct_packet(spkt);
-#endif
-}
-
-#else
-
-#endif
-
 /**
 **	Decode an audio packet.
 **
@@ -1301,17 +1021,6 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
 */
 void CodecAudioFlushBuffers(AudioDecoder * decoder)
 {
-#ifdef USE_AVPARSER
-    // FIXME: reset audio parser
-    if (decoder->AudioParser) {
-	av_parser_close(decoder->AudioParser);
-	decoder->AudioParser = NULL;
-	if (!(decoder->AudioParser =
-		av_parser_init(decoder->AudioCtx->codec_id))) {
-	    Fatal(_("codec: can't init audio parser\n"));
-	}
-    }
-#endif
     avcodec_flush_buffers(decoder->AudioCtx);
 }