The time base is 1 / sample_rate, not 90000.
Several more codecs also encode the sample count in the first 4 bytes of the
chunk.
---
libavformat/electronicarts.c | 25 +++++++++++++++++--------
1 files changed, 17 insertions(+), 8 deletions(-)
diff --git a/libavformat/electronicarts.c b/libavformat/electronicarts.c
index a10e645..d772f95 100644
--- a/libavformat/electronicarts.c
+++ b/libavformat/electronicarts.c
@@ -493,22 +493,31 @@ static int ea_read_packet(AVFormatContext *s,
num_samples = avio_rl32(pb);
avio_skip(pb, 8);
chunk_size -= 12;
+ } else if (ea->audio_codec == CODEC_ID_ADPCM_EA ||
+ ea->audio_codec == CODEC_ID_ADPCM_EA_R1 ||
+ ea->audio_codec == CODEC_ID_ADPCM_EA_R2 ||
+ ea->audio_codec == CODEC_ID_ADPCM_IMA_EA_EACS) {
+ num_samples = avio_rl32(pb);
+ avio_seek(pb, -4, SEEK_CUR);
+ } else if (ea->audio_codec == CODEC_ID_ADPCM_EA_R3) {
+ num_samples = avio_rb32(pb);
+ avio_seek(pb, -4, SEEK_CUR);
+ } else if (ea->audio_codec == CODEC_ID_ADPCM_IMA_EA_SEAD) {
+ num_samples = chunk_size * 2 / ea->num_channels;
}
ret = av_get_packet(pb, pkt, chunk_size);
if (ret < 0)
return ret;
pkt->stream_index = ea->audio_stream_index;
- pkt->pts = 90000;
- pkt->pts *= ea->audio_frame_counter;
- pkt->pts /= ea->sample_rate;
+ pkt->pts = ea->audio_frame_counter;
switch (ea->audio_codec) {
case CODEC_ID_ADPCM_EA:
- /* 2 samples/byte, 1 or 2 samples per frame depending
- * on stereo; chunk also has 12-byte header */
- ea->audio_frame_counter += ((chunk_size - 12) * 2) /
- ea->num_channels;
- break;
+ case CODEC_ID_ADPCM_EA_R1:
+ case CODEC_ID_ADPCM_EA_R2:
+ case CODEC_ID_ADPCM_EA_R3:
+ case CODEC_ID_ADPCM_IMA_EA_EACS:
+ case CODEC_ID_ADPCM_IMA_EA_SEAD:
case CODEC_ID_PCM_S16LE_PLANAR:
case CODEC_ID_MP3:
ea->audio_frame_counter += num_samples;
--
1.7.1
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