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authorAvuton Olrich <avuton@gmail.com>2006-07-26 03:05:50 +0000
committerAvuton Olrich <avuton@gmail.com>2006-07-26 03:05:50 +0000
commitbf3848ef94f3c4e09c150b51dab07e14c4435f05 (patch)
treeea8ac1030902973ab58eb769b2af4fa6469a3f6f /src
parent1523e8a84290690e67474534a99a250be47d751b (diff)
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Put mp4ff back into the tree
git-svn-id: https://svn.musicpd.org/mpd/trunk@4461 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Diffstat (limited to '')
-rw-r--r--src/Makefile.am5
-rw-r--r--src/inputPlugins/mp4_plugin.c55
2 files changed, 29 insertions, 31 deletions
diff --git a/src/Makefile.am b/src/Makefile.am
index c4d4a6fd3..805403165 100644
--- a/src/Makefile.am
+++ b/src/Makefile.am
@@ -1,4 +1,5 @@
bin_PROGRAMS = mpd
+SUBDIRS = $(MP4FF_SUBDIR)
mpd_audioOutputs = \
audioOutputs/audioOutput_alsa.c \
@@ -115,7 +116,9 @@ mpd_SOURCES = \
mpd_CFLAGS = $(MPD_CFLAGS)
-mpd_LDADD = $(MPD_LIBS)
+mpd_LDADD = $(MPD_LIBS) $(MP4FF_LIB)
+
+DIST_SUBDIRS = mp4ff
# sparse is a semantic parser
# URL: git://www.kernel.org/pub/scm/devel/sparse/sparse.git
diff --git a/src/inputPlugins/mp4_plugin.c b/src/inputPlugins/mp4_plugin.c
index 0a68dcc98..c13d38b43 100644
--- a/src/inputPlugins/mp4_plugin.c
+++ b/src/inputPlugins/mp4_plugin.c
@@ -28,12 +28,13 @@
#include "../outputBuffer.h"
#include "../decode.h"
+#include "../mp4ff/mp4ff.h"
+
#include <stdio.h>
#include <unistd.h>
#include <stdlib.h>
#include <string.h>
#include <faad.h>
-#include <mp4ff.h>
/* all code here is either based on or copied from FAAD2's frontend code */
@@ -45,7 +46,7 @@ static int mp4_getAACTrack(mp4ff_t * infile)
for (i = 0; i < numTracks; i++) {
unsigned char *buff = NULL;
- unsigned int *buff_size = 0;
+ int buff_size = 0;
#ifdef HAVE_MP4AUDIOSPECIFICCONFIG
mp4AudioSpecificConfig mp4ASC;
#else
@@ -54,11 +55,11 @@ static int mp4_getAACTrack(mp4ff_t * infile)
dummy7_8, dummy8_8;
#endif
- mp4ff_get_decoder_config(infile, i, &buff, buff_size);
+ mp4ff_get_decoder_config(infile, i, &buff, &buff_size);
if (buff) {
#ifdef HAVE_MP4AUDIOSPECIFICCONFIG
- rc = AudioSpecificConfig(buff, *buff_size, &mp4ASC);
+ rc = AudioSpecificConfig(buff, buff_size, &mp4ASC);
#else
rc = AudioSpecificConfig(buff, &dummy1_32, &dummy2_8,
&dummy3_8, &dummy4_8,
@@ -87,24 +88,6 @@ static uint32_t mp4_inputStreamSeekCallback(void *inStream, uint64_t position)
return seekInputStream((InputStream *) inStream, position, SEEK_SET);
}
-static faacDecHandle *openConfigureFaad()
-{
- faacDecConfigurationPtr config;
- faacDecHandle decoder = faacDecOpen();
-
- config = faacDecGetCurrentConfiguration(decoder);
- config->outputFormat = FAAD_FMT_16BIT;
-#ifdef HAVE_FAACDECCONFIGURATION_DOWNMATRIX
- config->downMatrix = 1;
-#endif
-#ifdef HAVE_FAACDECCONFIGURATION_DONTUPSAMPLEIMPLICITSBR
- config->dontUpSampleImplicitSBR = 0;
-#endif
- faacDecSetConfiguration(decoder, config);
-
- return decoder;
-}
-
static int mp4_decode(OutputBuffer * cb, DecoderControl * dc, char *path)
{
mp4ff_t *mp4fh;
@@ -112,11 +95,12 @@ static int mp4_decode(OutputBuffer * cb, DecoderControl * dc, char *path)
int32_t track;
float time;
int32_t scale;
- faacDecHandle *decoder;
+ faacDecHandle decoder;
faacDecFrameInfo frameInfo;
+ faacDecConfigurationPtr config;
unsigned char *mp4Buffer;
- unsigned int mp4BufferSize;
- uint32_t sampleRate;
+ int mp4BufferSize;
+ unsigned long sampleRate;
unsigned char channels;
long sampleId;
long numSamples;
@@ -161,7 +145,17 @@ static int mp4_decode(OutputBuffer * cb, DecoderControl * dc, char *path)
return -1;
}
- decoder = openConfigureFaad();
+ decoder = faacDecOpen();
+
+ config = faacDecGetCurrentConfiguration(decoder);
+ config->outputFormat = FAAD_FMT_16BIT;
+#ifdef HAVE_FAACDECCONFIGURATION_DOWNMATRIX
+ config->downMatrix = 1;
+#endif
+#ifdef HAVE_FAACDECCONFIGURATION_DONTUPSAMPLEIMPLICITSBR
+ config->dontUpSampleImplicitSBR = 0;
+#endif
+ faacDecSetConfiguration(decoder, config);
dc->audioFormat.bits = 16;
@@ -170,8 +164,7 @@ static int mp4_decode(OutputBuffer * cb, DecoderControl * dc, char *path)
mp4ff_get_decoder_config(mp4fh, track, &mp4Buffer, &mp4BufferSize);
if (faacDecInit2
- (decoder, mp4Buffer, mp4BufferSize, &sampleRate, &channels)
- < 0) {
+ (decoder, mp4Buffer, mp4BufferSize, &sampleRate, &channels) < 0) {
ERROR("Error not a AAC stream.\n");
faacDecClose(decoder);
mp4ff_close(mp4fh);
@@ -321,9 +314,11 @@ static int mp4_decode(OutputBuffer * cb, DecoderControl * dc, char *path)
}
flushOutputBuffer(cb);
- if (dc->stop)
+ if (dc->stop) {
+ dc->state = DECODE_STATE_STOP;
dc->stop = 0;
- dc->state = DECODE_STATE_STOP;
+ } else
+ dc->state = DECODE_STATE_STOP;
return 0;
}