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authorChia-chi Yeh <chiachi@android.com>2011-09-06 14:18:37 -0700
committerChia-chi Yeh <chiachi@android.com>2011-09-06 14:34:44 -0700
commit35d05dcba1e829782813b6ec21afceb5cffc22e6 (patch)
tree4f5287ebc5edea42a12ff4f772c5574f57cf2542 /voip/jni/rtp/AudioCodec.h
parent1ed7a407fafe50b1eb0878f560bb0618706e4e82 (diff)
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RTP: support payloads with larger packetization interval.
RFC 3551 section 4.2 said that a receiver should accept packets representing between 0 and 200ms of audio data. Now we add the ability to decode multiple frames in a payload as long as the jitter buffer is not full. This change covers G711, GSM, and GSM-EFR. AMR will be added later. Bug: 3029736 Change-Id: Ifd194596766d14f02177925c58432cd620e44dd7
Diffstat (limited to 'voip/jni/rtp/AudioCodec.h')
-rw-r--r--voip/jni/rtp/AudioCodec.h2
1 files changed, 1 insertions, 1 deletions
diff --git a/voip/jni/rtp/AudioCodec.h b/voip/jni/rtp/AudioCodec.h
index e389255..741730b 100644
--- a/voip/jni/rtp/AudioCodec.h
+++ b/voip/jni/rtp/AudioCodec.h
@@ -30,7 +30,7 @@ public:
// Returns the length of payload in bytes.
virtual int encode(void *payload, int16_t *samples) = 0;
// Returns the number of decoded samples.
- virtual int decode(int16_t *samples, void *payload, int length) = 0;
+ virtual int decode(int16_t *samples, int count, void *payload, int length) = 0;
};
AudioCodec *newAudioCodec(const char *codecName);