1 files changed, 66 insertions, 31 deletions

diff --git a/src/decoder/audiofile_plugin.c b/src/decoder/audiofile_plugin.c
index f66d90dc1..18cfdda5d 100644
--- a/src/decoder/audiofile_plugin.c
+++ b/src/decoder/audiofile_plugin.c
@@ -17,7 +17,9 @@
  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
-#include "../decoder_api.h"
+#include "config.h"
+#include "decoder_api.h"
+#include "audio_check.h"
 
 #include <audiofile.h>
 #include <af_vfs.h>
@@ -45,10 +47,20 @@ static int audiofile_get_duration(const char *file)
 }
 
 static ssize_t
-audiofile_file_read(AFvirtualfile *vfile, void *data, size_t nbytes)
+audiofile_file_read(AFvirtualfile *vfile, void *data, size_t length)
 {
 	struct input_stream *is = (struct input_stream *) vfile->closure;
-	return input_stream_read(is, data, nbytes);
+	GError *error = NULL;
+	size_t nbytes;
+
+	nbytes = input_stream_read(is, data, length, &error);
+	if (nbytes == 0 && error != NULL) {
+		g_warning("%s", error->message);
+		g_error_free(error);
+		return -1;
+	}
+
+	return nbytes;
 }
 
 static long
@@ -78,7 +90,7 @@ audiofile_file_seek(AFvirtualfile *vfile, long offset, int is_relative)
 {
 	struct input_stream *is = (struct input_stream *) vfile->closure;
 	int whence = (is_relative ? SEEK_CUR : SEEK_SET);
-	if (input_stream_seek(is, offset, whence)) {
+	if (input_stream_seek(is, offset, whence, NULL)) {
 		return is->offset;
 	} else {
 		return -1;
@@ -99,17 +111,56 @@ setup_virtual_fops(struct input_stream *stream)
 	return vf;
 }
 
+static enum sample_format
+audiofile_bits_to_sample_format(int bits)
+{
+	switch (bits) {
+	case 8:
+		return SAMPLE_FORMAT_S8;
+
+	case 16:
+		return SAMPLE_FORMAT_S16;
+
+	case 24:
+		return SAMPLE_FORMAT_S24_P32;
+
+	case 32:
+		return SAMPLE_FORMAT_S32;
+	}
+
+	return SAMPLE_FORMAT_UNDEFINED;
+}
+
+static enum sample_format
+audiofile_setup_sample_format(AFfilehandle af_fp)
+{
+	int fs, bits;
+
+	afGetSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits);
+	if (!audio_valid_sample_format(audiofile_bits_to_sample_format(bits))) {
+		g_debug("input file has %d bit samples, converting to 16",
+			bits);
+		bits = 16;
+	}
+
+	afSetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK,
+	                         AF_SAMPFMT_TWOSCOMP, bits);
+	afGetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits);
+
+	return audiofile_bits_to_sample_format(bits);
+}
+
 static void
 audiofile_stream_decode(struct decoder *decoder, struct input_stream *is)
 {
+	GError *error = NULL;
 	AFvirtualfile *vf;
 	int fs, frame_count;
 	AFfilehandle af_fp;
-	int bits;
 	struct audio_format audio_format;
 	float total_time;
 	uint16_t bit_rate;
-	int ret, current = 0;
+	int ret;
 	char chunk[CHUNK_SIZE];
 	enum decoder_command cmd;
 
@@ -126,26 +177,13 @@ audiofile_stream_decode(struct decoder *decoder, struct input_stream *is)
 		return;
 	}
 
-	afGetSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits);
-	if (!audio_valid_sample_format(bits)) {
-		g_debug("input file has %d bit samples, converting to 16",
-			bits);
-		bits = 16;
-	}
-
-	afSetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK,
-	                         AF_SAMPFMT_TWOSCOMP, bits);
-	afGetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits);
-	audio_format.bits = (uint8_t)bits;
-	audio_format.sample_rate =
-	                      (unsigned int)afGetRate(af_fp, AF_DEFAULT_TRACK);
-	audio_format.channels =
-	              (uint8_t)afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK);
-
-	if (!audio_format_valid(&audio_format)) {
-		g_warning("Invalid audio format: %u:%u:%u\n",
-			  audio_format.sample_rate, audio_format.bits,
-			  audio_format.channels);
+	if (!audio_format_init_checked(&audio_format,
+				       afGetRate(af_fp, AF_DEFAULT_TRACK),
+				       audiofile_setup_sample_format(af_fp),
+				       afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK),
+				       &error)) {
+		g_warning("%s", error->message);
+		g_error_free(error);
 		afCloseFile(af_fp);
 		return;
 	}
@@ -166,17 +204,14 @@ audiofile_stream_decode(struct decoder *decoder, struct input_stream *is)
 		if (ret <= 0)
 			break;
 
-		current += ret;
 		cmd = decoder_data(decoder, NULL,
 				   chunk, ret * fs,
-				   (float)current /
-				   (float)audio_format.sample_rate,
 				   bit_rate, NULL);
 
 		if (cmd == DECODE_COMMAND_SEEK) {
-			current = decoder_seek_where(decoder) *
+			AFframecount frame = decoder_seek_where(decoder) *
 				audio_format.sample_rate;
-			afSeekFrame(af_fp, AF_DEFAULT_TRACK, current);
+			afSeekFrame(af_fp, AF_DEFAULT_TRACK, frame);
 
 			decoder_command_finished(decoder);
 			cmd = DECODE_COMMAND_NONE;