From 35d05dcba1e829782813b6ec21afceb5cffc22e6 Mon Sep 17 00:00:00 2001 From: Chia-chi Yeh Date: Tue, 6 Sep 2011 14:18:37 -0700 Subject: 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 --- voip/jni/rtp/AudioCodec.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'voip/jni/rtp/AudioCodec.h') 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); -- cgit v1.1