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authorMike Melanson <mike@multimedia.cx>2004-06-19 03:59:34 +0000
committerMike Melanson <mike@multimedia.cx>2004-06-19 03:59:34 +0000
commit0bd586c50dc27ec38ca94e58f50030544d729463 (patch)
treedaa391f5efede8e58b88431b9b453dcbd1e6c2e0 /libavformat/segafilm.c
parentcbf5374fc0f733cefe304fd4d11c7b0fa21fba61 (diff)
sweeping change from -EIO -> AVERROR_IO
Originally committed as revision 3239 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavformat/segafilm.c')
-rw-r--r--libavformat/segafilm.c18
1 files changed, 9 insertions, 9 deletions
diff --git a/libavformat/segafilm.c b/libavformat/segafilm.c
index 7ddedf930e..9e94de103c 100644
--- a/libavformat/segafilm.c
+++ b/libavformat/segafilm.c
@@ -92,7 +92,7 @@ static int film_read_header(AVFormatContext *s,
/* load the main FILM header */
if (get_buffer(pb, scratch, 16) != 16)
- return -EIO;
+ return AVERROR_IO;
data_offset = BE_32(&scratch[4]);
film->version = BE_32(&scratch[8]);
@@ -100,7 +100,7 @@ static int film_read_header(AVFormatContext *s,
if (film->version == 0) {
/* special case for Lemmings .film files; 20-byte header */
if (get_buffer(pb, scratch, 20) != 20)
- return -EIO;
+ return AVERROR_IO;
/* make some assumptions about the audio parameters */
film->audio_type = CODEC_ID_PCM_S8;
film->audio_samplerate = 22050;
@@ -109,7 +109,7 @@ static int film_read_header(AVFormatContext *s,
} else {
/* normal Saturn .cpk files; 32-byte header */
if (get_buffer(pb, scratch, 32) != 32)
- return -EIO;
+ return AVERROR_IO;
film->audio_samplerate = BE_16(&scratch[24]);;
film->audio_channels = scratch[21];
film->audio_bits = scratch[22];
@@ -166,7 +166,7 @@ static int film_read_header(AVFormatContext *s,
/* load the sample table */
if (get_buffer(pb, scratch, 16) != 16)
- return -EIO;
+ return AVERROR_IO;
if (BE_32(&scratch[0]) != STAB_TAG)
return AVERROR_INVALIDDATA;
film->base_clock = BE_32(&scratch[8]);
@@ -181,7 +181,7 @@ static int film_read_header(AVFormatContext *s,
/* load the next sample record and transfer it to an internal struct */
if (get_buffer(pb, scratch, 16) != 16) {
av_free(film->sample_table);
- return -EIO;
+ return AVERROR_IO;
}
film->sample_table[i].sample_offset =
data_offset + BE_32(&scratch[0]);
@@ -217,7 +217,7 @@ static int film_read_packet(AVFormatContext *s,
int left, right;
if (film->current_sample >= film->sample_count)
- return -EIO;
+ return AVERROR_IO;
sample = &film->sample_table[film->current_sample];
@@ -235,7 +235,7 @@ static int film_read_packet(AVFormatContext *s,
ret += get_buffer(pb, pkt->data + 10,
sample->sample_size - 10 - film->cvid_extra_bytes);
if (ret != sample->sample_size - film->cvid_extra_bytes)
- ret = -EIO;
+ ret = AVERROR_IO;
} else if ((sample->stream == film->audio_stream_index) &&
(film->audio_channels == 2)) {
/* stereo PCM needs to be interleaved */
@@ -252,7 +252,7 @@ static int film_read_packet(AVFormatContext *s,
ret = get_buffer(pb, film->stereo_buffer, sample->sample_size);
if (ret != sample->sample_size)
- ret = -EIO;
+ ret = AVERROR_IO;
left = 0;
right = sample->sample_size / 2;
@@ -272,7 +272,7 @@ static int film_read_packet(AVFormatContext *s,
return AVERROR_NOMEM;
ret = get_buffer(pb, pkt->data, sample->sample_size);
if (ret != sample->sample_size)
- ret = -EIO;
+ ret = AVERROR_IO;
}
pkt->stream_index = sample->stream;