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authorDiego Pettenò <flameeyes@gmail.com>2008-10-02 16:03:00 +0000
committerBenoit Fouet <benoit.fouet@free.fr>2008-10-02 16:03:00 +0000
commitfb65d2ca84d79fb1c5a5708555c23e1d289b5c92 (patch)
tree933cef560872486d2c23628aef5bfa37d0eaf315
parent529dae12f786ed8840a8ccec75d66c7d27cdf9d9 (diff)
downloadffmpeg-fb65d2ca84d79fb1c5a5708555c23e1d289b5c92.tar.gz
Use enum typers instead of int.
Patch by Diego 'Flameeyes' Pettenò: flameeyes gmail Originally committed as revision 15517 to svn://svn.ffmpeg.org/ffmpeg/trunk
-rw-r--r--libavdevice/audio.c2
-rw-r--r--libavformat/aiff.c4
-rw-r--r--libavformat/asf.c3
-rw-r--r--libavformat/au.c3
-rw-r--r--libavformat/mov.c4
-rw-r--r--libavformat/mpeg.c4
-rw-r--r--libavformat/mpegts.c3
-rw-r--r--libavformat/rtsp.c3
-rw-r--r--libavformat/segafilm.c8
-rw-r--r--libavformat/sol.c5
-rw-r--r--libavformat/westwood.c2
11 files changed, 24 insertions, 17 deletions
diff --git a/libavdevice/audio.c b/libavdevice/audio.c
index 1cb3d69409..8520a5f3ea 100644
--- a/libavdevice/audio.c
+++ b/libavdevice/audio.c
@@ -46,7 +46,7 @@ typedef struct {
int sample_rate;
int channels;
int frame_size; /* in bytes ! */
- int codec_id;
+ enum CodecID codec_id;
unsigned int flip_left : 1;
uint8_t buffer[AUDIO_BLOCK_SIZE];
int buffer_ptr;
diff --git a/libavformat/aiff.c b/libavformat/aiff.c
index 911814bf92..3a4bf9f9dd 100644
--- a/libavformat/aiff.c
+++ b/libavformat/aiff.c
@@ -46,7 +46,7 @@ static const AVCodecTag codec_aiff_tags[] = {
#define AIFF 0
#define AIFF_C_VERSION1 0xA2805140
-static int aiff_codec_get_id(int bps)
+static enum CodecID aiff_codec_get_id(int bps)
{
if (bps <= 8)
return CODEC_ID_PCM_S8;
@@ -58,7 +58,7 @@ static int aiff_codec_get_id(int bps)
return CODEC_ID_PCM_S32BE;
/* bigger than 32 isn't allowed */
- return 0;
+ return CODEC_ID_NONE;
}
/* returns the size of the found tag */
diff --git a/libavformat/asf.c b/libavformat/asf.c
index b837e76891..e8b9144609 100644
--- a/libavformat/asf.c
+++ b/libavformat/asf.c
@@ -197,7 +197,8 @@ static int asf_read_header(AVFormatContext *s, AVFormatParameters *ap)
asf->hdr.max_bitrate = get_le32(pb);
asf->packet_size = asf->hdr.max_pktsize;
} else if (!memcmp(&g, &stream_header, sizeof(GUID))) {
- int type, type_specific_size, sizeX;
+ enum CodecType type;
+ int type_specific_size, sizeX;
uint64_t total_size;
unsigned int tag1;
int64_t pos1, pos2, start_time;
diff --git a/libavformat/au.c b/libavformat/au.c
index 3dfd31e761..ffb26ed54e 100644
--- a/libavformat/au.c
+++ b/libavformat/au.c
@@ -122,7 +122,8 @@ static int au_read_header(AVFormatContext *s,
int size;
unsigned int tag;
ByteIOContext *pb = s->pb;
- unsigned int id, codec, channels, rate;
+ unsigned int id, channels, rate;
+ enum CodecID codec;
AVStream *st;
/* check ".snd" header */
diff --git a/libavformat/mov.c b/libavformat/mov.c
index 79db9fc52c..27ddb1c718 100644
--- a/libavformat/mov.c
+++ b/libavformat/mov.c
@@ -676,7 +676,7 @@ static int mov_read_stco(MOVContext *c, ByteIOContext *pb, MOV_atom_t atom)
* Compute codec id for 'lpcm' tag.
* See CoreAudioTypes and AudioStreamBasicDescription at Apple.
*/
-static int mov_get_lpcm_codec_id(int bps, int flags)
+static enum CodecID mov_get_lpcm_codec_id(int bps, int flags)
{
if (flags & 1) { // floating point
if (flags & 2) { // big endian
@@ -704,7 +704,7 @@ static int mov_get_lpcm_codec_id(int bps, int flags)
else if (bps == 32) return CODEC_ID_PCM_S32LE;
}
}
- return 0;
+ return CODEC_ID_NONE;
}
static int mov_read_stsd(MOVContext *c, ByteIOContext *pb, MOV_atom_t atom)
diff --git a/libavformat/mpeg.c b/libavformat/mpeg.c
index 289be8caf6..ec89d5a9a2 100644
--- a/libavformat/mpeg.c
+++ b/libavformat/mpeg.c
@@ -409,7 +409,9 @@ static int mpegps_read_packet(AVFormatContext *s,
{
MpegDemuxContext *m = s->priv_data;
AVStream *st;
- int len, startcode, i, type, codec_id = 0, es_type;
+ int len, startcode, i, es_type;
+ enum CodecID codec_id = CODEC_ID_NONE;
+ enum CodecType type;
int64_t pts, dts, dummy_pos; //dummy_pos is needed for the index building to work
redo:
diff --git a/libavformat/mpegts.c b/libavformat/mpegts.c
index 80b604c5ae..9f41d396a9 100644
--- a/libavformat/mpegts.c
+++ b/libavformat/mpegts.c
@@ -934,7 +934,8 @@ static void mpegts_push_data(MpegTSFilter *filter,
static AVStream* new_pes_av_stream(PESContext *pes, uint32_t code)
{
AVStream *st;
- int codec_type, codec_id;
+ enum CodecID codec_id;
+ enum CodecType codec_type;
switch(pes->stream_type){
case STREAM_TYPE_AUDIO_MPEG1:
diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c
index b1fafedf01..413aadde95 100644
--- a/libavformat/rtsp.c
+++ b/libavformat/rtsp.c
@@ -389,7 +389,8 @@ static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
RTSPState *rt = s->priv_data;
char buf1[64], st_type[64];
const char *p;
- int codec_type, payload_type, i;
+ enum CodecType codec_type;
+ int payload_type, i;
AVStream *st;
RTSPStream *rtsp_st;
struct in_addr sdp_ip;
diff --git a/libavformat/segafilm.c b/libavformat/segafilm.c
index 4bc46c2773..1ea296db95 100644
--- a/libavformat/segafilm.c
+++ b/libavformat/segafilm.c
@@ -46,12 +46,12 @@ typedef struct FilmDemuxContext {
int video_stream_index;
int audio_stream_index;
- unsigned int audio_type;
+ enum CodecID audio_type;
unsigned int audio_samplerate;
unsigned int audio_bits;
unsigned int audio_channels;
- unsigned int video_type;
+ enum CodecID video_type;
unsigned int sample_count;
film_sample_t *sample_table;
unsigned int current_sample;
@@ -115,7 +115,7 @@ static int film_read_header(AVFormatContext *s,
else if (film->audio_bits == 16)
film->audio_type = CODEC_ID_PCM_S16BE;
else
- film->audio_type = 0;
+ film->audio_type = CODEC_ID_NONE;
}
if (AV_RB32(&scratch[0]) != FDSC_TAG)
@@ -124,7 +124,7 @@ static int film_read_header(AVFormatContext *s,
if (AV_RB32(&scratch[8]) == CVID_TAG) {
film->video_type = CODEC_ID_CINEPAK;
} else
- film->video_type = 0;
+ film->video_type = CODEC_ID_NONE;
/* initialize the decoder streams */
if (film->video_type) {
diff --git a/libavformat/sol.c b/libavformat/sol.c
index 775b4e0adf..8b680be742 100644
--- a/libavformat/sol.c
+++ b/libavformat/sol.c
@@ -47,7 +47,7 @@ static int sol_probe(AVProbeData *p)
#define SOL_16BIT 4
#define SOL_STEREO 16
-static int sol_codec_id(int magic, int type)
+static enum CodecID sol_codec_id(int magic, int type)
{
if (magic == 0x0B8D)
{
@@ -88,7 +88,8 @@ static int sol_read_header(AVFormatContext *s,
int size;
unsigned int magic,tag;
ByteIOContext *pb = s->pb;
- unsigned int id, codec, channels, rate, type;
+ unsigned int id, channels, rate, type;
+ enum CodecID codec;
AVStream *st;
/* check ".snd" header */
diff --git a/libavformat/westwood.c b/libavformat/westwood.c
index 600863e837..d88199d690 100644
--- a/libavformat/westwood.c
+++ b/libavformat/westwood.c
@@ -66,7 +66,7 @@ typedef struct WsAudDemuxContext {
int audio_samplerate;
int audio_channels;
int audio_bits;
- int audio_type;
+ enum CodecID audio_type;
int audio_stream_index;
int64_t audio_frame_counter;
} WsAudDemuxContext;