added parameter total_time to decoder_initialized()

Similar to the previous patch: pass total_time instead of manipulating
dc->totalTime directly.

src/decoder_api.c

@@ -25,7 +25,8 @@
 #include "gcc.h"
 
 void decoder_initialized(mpd_unused struct decoder * decoder,
-			 const AudioFormat * audio_format)
+			 const AudioFormat * audio_format,
+			 float total_time)
 {
 	assert(dc.state == DECODE_STATE_START);
 
@@ -35,6 +36,8 @@ void decoder_initialized(mpd_unused struct decoder * decoder,
 				     &(ob.audioFormat));
 	}
 
+	dc.totalTime = total_time;
+
 	dc.state = DECODE_STATE_DECODE;
 	notify_signal(&pc.notify);
 }

src/decoder_api.h

@@ -41,7 +41,8 @@ struct decoder;
  * that it has read the song's meta data.
  */
 void decoder_initialized(struct decoder * decoder,
-			 const AudioFormat * audio_format);
+			 const AudioFormat * audio_format,
+			 float total_time);
 
 /**
  * This function is called by the decoder plugin when it has

src/inputPlugins/_flac_common.c

@@ -165,7 +165,7 @@ void flac_metadata_common_cb(const FLAC__StreamMetadata * block,
 		data->audio_format.bits = (mpd_sint8)si->bits_per_sample;
 		data->audio_format.sampleRate = si->sample_rate;
 		data->audio_format.channels = (mpd_sint8)si->channels;
-		dc.totalTime = ((float)si->total_samples) / (si->sample_rate);
+		data->total_time = ((float)si->total_samples) / (si->sample_rate);
 		break;
 	case FLAC__METADATA_TYPE_VORBIS_COMMENT:
 		flacParseReplayGain(block, data);

src/inputPlugins/_flac_common.h

@@ -144,6 +144,7 @@ typedef struct {
 	float time;
 	unsigned int bitRate;
 	AudioFormat audio_format;
+	float total_time;
 	FLAC__uint64 position;
 	struct decoder *decoder;
 	InputStream *inStream;

src/inputPlugins/aac_plugin.c

@@ -338,8 +338,6 @@ static int aac_decode(struct decoder * mpd_decoder, char *path)
 
 	audio_format.bits = 16;
 
-	dc.totalTime = totalTime;
-
 	file_time = 0.0;
 
 	advanceAacBuffer(&b, bread);
@@ -372,7 +370,8 @@ static int aac_decode(struct decoder * mpd_decoder, char *path)
 		if (dc.state != DECODE_STATE_DECODE) {
 			audio_format.channels = frameInfo.channels;
 			audio_format.sampleRate = sampleRate;
-			decoder_initialized(mpd_decoder, &audio_format);
+			decoder_initialized(mpd_decoder, &audio_format,
+					    totalTime);
 		}
 
 		advanceAacBuffer(&b, frameInfo.bytesconsumed);

src/inputPlugins/audiofile_plugin.c

@@ -46,6 +46,7 @@ static int audiofile_decode(struct decoder * decoder, char *path)
 	AFfilehandle af_fp;
 	int bits;
 	AudioFormat audio_format;
+	float total_time;
 	mpd_uint16 bitRate;
 	struct stat st;
 
@@ -71,10 +72,9 @@ static int audiofile_decode(struct decoder * decoder, char *path)
 
 	frame_count = afGetFrameCount(af_fp, AF_DEFAULT_TRACK);
 
-	dc.totalTime =
+	total_time = ((float)frame_count / (float)audio_format.sampleRate);
-	    ((float)frame_count / (float)audio_format.sampleRate);
 
-	bitRate = (mpd_uint16)(st.st_size * 8.0 / dc.totalTime / 1000.0 + 0.5);
+	bitRate = (mpd_uint16)(st.st_size * 8.0 / total_time / 1000.0 + 0.5);
 
 	if (audio_format.bits != 8 && audio_format.bits != 16) {
 		ERROR("Only 8 and 16-bit files are supported. %s is %i-bit\n",
@@ -85,7 +85,7 @@ static int audiofile_decode(struct decoder * decoder, char *path)
 
 	fs = (int)afGetVirtualFrameSize(af_fp, AF_DEFAULT_TRACK, 1);
 
-	decoder_initialized(decoder, &audio_format);
+	decoder_initialized(decoder, &audio_format, total_time);
 
 	{
 		int ret, eof = 0, current = 0;

src/inputPlugins/flac_plugin.c

@@ -415,7 +415,7 @@ static int flac_decode_internal(struct decoder * decoder,
 		}
 	}
 
-	decoder_initialized(decoder, &data.audio_format);
+	decoder_initialized(decoder, &data.audio_format, data.total_time);
 
 	while (1) {
 		if (!flac_process_single(flacDec))

src/inputPlugins/mod_plugin.c

@@ -176,7 +176,6 @@ static int mod_decode(struct decoder * decoder, char *path)
 		return -1;
 	}
 
-	dc.totalTime = 0;
 	audio_format.bits = 16;
 	audio_format.sampleRate = 44100;
 	audio_format.channels = 2;
@@ -185,7 +184,7 @@ static int mod_decode(struct decoder * decoder, char *path)
 	    1.0 / ((audio_format.bits * audio_format.channels / 8.0) *
 		   (float)audio_format.sampleRate);
 
-	decoder_initialized(decoder, &audio_format);
+	decoder_initialized(decoder, &audio_format, 0);
 
 	while (1) {
 		if (dc.command == DECODE_COMMAND_SEEK) {

src/inputPlugins/mp3_plugin.c

@@ -1035,8 +1035,6 @@ static int mp3_decode(struct decoder * decoder, InputStream * inStream)
 
 	initAudioFormatFromMp3DecodeData(&data, &audio_format);
 
-	dc.totalTime = data.totalTime;
-
 	if (inStream->metaTitle) {
 		if (tag)
 			freeMpdTag(tag);
@@ -1062,7 +1060,7 @@ static int mp3_decode(struct decoder * decoder, InputStream * inStream)
 		freeMpdTag(tag);
 	}
 
-	decoder_initialized(decoder, &audio_format);
+	decoder_initialized(decoder, &audio_format, data.totalTime);
 
 	while (mp3Read(&data, decoder, &replayGainInfo) != DECODE_BREAK) ;
 	/* send last little bit if not DECODE_COMMAND_STOP */

src/inputPlugins/mp4_plugin.c

@@ -83,7 +83,7 @@ static int mp4_decode(struct decoder * mpd_decoder, InputStream * inStream)
 	mp4ff_t *mp4fh;
 	mp4ff_callback_t *mp4cb;
 	int32_t track;
-	float file_time;
+	float file_time, total_time;
 	int32_t scale;
 	faacDecHandle decoder;
 	faacDecFrameInfo frameInfo;
@@ -170,7 +170,7 @@ static int mp4_decode(struct decoder * mpd_decoder, InputStream * inStream)
 		free(mp4cb);
 		return -1;
 	}
-	dc.totalTime = ((float)file_time) / scale;
+	total_time = ((float)file_time) / scale;
 
 	numSamples = mp4ff_num_samples(mp4fh, track);
 
@@ -248,7 +248,8 @@ static int mp4_decode(struct decoder * mpd_decoder, InputStream * inStream)
 #endif
 			audio_format.sampleRate = scale;
 			audio_format.channels = frameInfo.channels;
-			decoder_initialized(mpd_decoder, &audio_format);
+			decoder_initialized(mpd_decoder, &audio_format,
+					    total_time);
 		}
 
 		if (channels * (unsigned long)(dur + offset) > frameInfo.samples) {

src/inputPlugins/mpc_plugin.c

@@ -160,8 +160,6 @@ static int mpc_decode(struct decoder * mpd_decoder, InputStream * inStream)
 		return 0;
 	}
 
-	dc.totalTime = mpc_streaminfo_get_length(&info);
-
 	audio_format.bits = 16;
 	audio_format.channels = info.channels;
 	audio_format.sampleRate = info.sample_freq;
@@ -172,7 +170,8 @@ static int mpc_decode(struct decoder * mpd_decoder, InputStream * inStream)
 	replayGainInfo->trackGain = info.gain_title * 0.01;
 	replayGainInfo->trackPeak = info.peak_title / 32767.0;
 
-	decoder_initialized(mpd_decoder, &audio_format);
+	decoder_initialized(mpd_decoder, &audio_format,
+			    mpc_streaminfo_get_length(&info));
 
 	while (!eof) {
 		if (dc.command == DECODE_COMMAND_SEEK) {

src/inputPlugins/oggflac_plugin.c

@@ -345,7 +345,7 @@ static int oggflac_decode(struct decoder * mpd_decoder, InputStream * inStream)
 		goto fail;
 	}
 
-	decoder_initialized(mpd_decoder, &data.audio_format);
+	decoder_initialized(mpd_decoder, &data.audio_format, data.total_time);
 
 	while (1) {
 		OggFLAC__seekable_stream_decoder_process_single(decoder);

src/inputPlugins/oggvorbis_plugin.c

@@ -262,9 +262,6 @@ static int oggvorbis_decode(struct decoder * decoder, InputStream * inStream)
 		}
 		return 0;
 	}
-	dc.totalTime = ov_time_total(&vf, -1);
-	if (dc.totalTime < 0)
-		dc.totalTime = 0;
 	audio_format.bits = 16;
 
 	while (1) {
@@ -285,7 +282,11 @@ static int oggvorbis_decode(struct decoder * decoder, InputStream * inStream)
 			audio_format.channels = vi->channels;
 			audio_format.sampleRate = vi->rate;
 			if (dc.state == DECODE_STATE_START) {
-				decoder_initialized(decoder, &audio_format);
+				float total_time = ov_time_total(&vf, -1);
+				if (total_time < 0)
+					total_time = 0;
+				decoder_initialized(decoder, &audio_format,
+						    total_time);
 			}
 			comments = ov_comment(&vf, -1)->user_comments;
 			putOggCommentsIntoOutputBuffer(inStream->metaName,

src/inputPlugins/wavpack_plugin.c

@@ -164,10 +164,10 @@ static void wavpack_decode(struct decoder * decoder,
 
 	samplesreq = sizeof(chunk) / (4 * audio_format.channels);
 
-	dc.totalTime = (float)allsamples / audio_format.sampleRate;
 	dc.seekable = canseek;
 
-	decoder_initialized(decoder, &audio_format);
+	decoder_initialized(decoder, &audio_format,
+			    (float)allsamples / audio_format.sampleRate);
 
 	position = 0;