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author | Chia-chi Yeh <chiachi@android.com> | 2011-09-06 14:18:37 -0700 |
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committer | Chia-chi Yeh <chiachi@android.com> | 2011-09-06 14:34:44 -0700 |
commit | 35d05dcba1e829782813b6ec21afceb5cffc22e6 (patch) | |
tree | 4f5287ebc5edea42a12ff4f772c5574f57cf2542 /voip/jni/rtp/AudioCodec.h | |
parent | 1ed7a407fafe50b1eb0878f560bb0618706e4e82 (diff) | |
download | frameworks_base-35d05dcba1e829782813b6ec21afceb5cffc22e6.zip frameworks_base-35d05dcba1e829782813b6ec21afceb5cffc22e6.tar.gz frameworks_base-35d05dcba1e829782813b6ec21afceb5cffc22e6.tar.bz2 |
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.h | 2 |
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); |