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authorPeter Ross <pross@xvid.org>2008-08-18 12:39:57 +0000
committerPeter Ross <pross@xvid.org>2008-08-18 12:39:57 +0000
commitfc7ed9a6f60860fb19c54ae2482a0588ca8054f6 (patch)
tree11d938515f7c670107ea3d051d7b44e32b0880dc /libavformat/electronicarts.c
parent97ceb75218742b410ddb6f74d955a80a96162117 (diff)
downloadffmpeg-fc7ed9a6f60860fb19c54ae2482a0588ca8054f6.tar.gz
Support Electronic Arts files containing MP3 audio.
Originally committed as revision 14824 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavformat/electronicarts.c')
-rw-r--r--libavformat/electronicarts.c12
1 files changed, 10 insertions, 2 deletions
diff --git a/libavformat/electronicarts.c b/libavformat/electronicarts.c
index 7bc42805db..2e627792c3 100644
--- a/libavformat/electronicarts.c
+++ b/libavformat/electronicarts.c
@@ -185,6 +185,7 @@ static int process_audio_header_elements(AVFormatContext *s)
switch (revision2) {
case 8: ea->audio_codec = CODEC_ID_PCM_S16LE_PLANAR; break;
case 10: ea->audio_codec = CODEC_ID_ADPCM_EA_R2; break;
+ case 16: ea->audio_codec = CODEC_ID_MP3; break;
case -1: break;
default:
av_log(s, AV_LOG_ERROR, "unsupported stream type; revision2=%i\n", revision2);
@@ -430,6 +431,7 @@ static int ea_read_packet(AVFormatContext *s,
int packet_read = 0;
unsigned int chunk_type, chunk_size;
int key = 0;
+ int num_samples;
while (!packet_read) {
chunk_type = get_le32(pb);
@@ -448,8 +450,10 @@ static int ea_read_packet(AVFormatContext *s,
if (!ea->audio_codec) {
url_fskip(pb, chunk_size);
break;
- } else if (ea->audio_codec == CODEC_ID_PCM_S16LE_PLANAR) {
- url_fskip(pb, 12); /* planar header */
+ } else if (ea->audio_codec == CODEC_ID_PCM_S16LE_PLANAR ||
+ ea->audio_codec == CODEC_ID_MP3) {
+ num_samples = get_le32(pb);
+ url_fskip(pb, 8);
chunk_size -= 12;
}
ret = av_get_packet(pb, pkt, chunk_size);
@@ -468,6 +472,10 @@ static int ea_read_packet(AVFormatContext *s,
ea->audio_frame_counter += ((chunk_size - 12) * 2) /
ea->num_channels;
break;
+ case CODEC_ID_PCM_S16LE_PLANAR:
+ case CODEC_ID_MP3:
+ ea->audio_frame_counter += num_samples;
+ break;
default:
ea->audio_frame_counter += chunk_size /
(ea->bytes * ea->num_channels);