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Cleaned up the data path for payload data, made callbacks to rtp_receiver nonoptional.

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The audio receiver is now completely independent of rtp_receiver: video will hopefully be too in the next patch.

BUG=
TEST=vie & voe_auto_test full runs

Review URL: https://webrtc-codereview.appspot.com/1014006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3372 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
phoglund@webrtc.org
2013-01-14 10:01:55 +00:00
parent 49273ffa79
commit a22a9bd9ca
12 changed files with 184 additions and 152 deletions
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17
webrtc/modules/rtp_rtcp/source/rtp_receiver.h
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@ -32,11 +32,15 @@ class RTPReceiverStrategy;
class RTPReceiver : public Bitrate {
public:
// Callbacks passed in here may not be NULL (use Null object callbacks if you
// want callbacks to do nothing).
RTPReceiver(const WebRtc_Word32 id,
const bool audio,
RtpRtcpClock* clock,
ModuleRtpRtcpImpl* owner,
RtpAudioFeedback* incoming_messages_callback);
RtpAudioFeedback* incoming_audio_messages_callback,
RtpData* incoming_payload_callback,
RtpFeedback* incoming_messages_callback);
virtual ~RTPReceiver();
@ -50,10 +54,6 @@ class RTPReceiver : public Bitrate {
void ProcessBitrate();
WebRtc_Word32 RegisterIncomingDataCallback(RtpData* incoming_data_callback);
WebRtc_Word32 RegisterIncomingRTPCallback(
RtpFeedback* incoming_messages_callback);
WebRtc_Word32 RegisterReceivePayload(
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
const WebRtc_Word8 payload_type,
@ -161,11 +161,6 @@ class RTPReceiver : public Bitrate {
return rtp_receiver_audio_;
}
virtual WebRtc_Word32 CallbackOfReceivedPayloadData(
const WebRtc_UWord8* payload_data,
const WebRtc_UWord16 payload_size,
const WebRtcRTPHeader* rtp_header);
virtual WebRtc_Word8 REDPayloadType() const;
bool HaveNotReceivedPackets() const;
@ -202,9 +197,7 @@ class RTPReceiver : public Bitrate {
WebRtc_Word32 id_;
ModuleRtpRtcpImpl& rtp_rtcp_;
CriticalSectionWrapper* critical_section_cbs_;
RtpFeedback* cb_rtp_feedback_;
RtpData* cb_rtp_data_;
CriticalSectionWrapper* critical_section_rtp_receiver_;
mutable WebRtc_Word64 last_receive_time_;