zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Replaced the _audio parameter with a strategy.

Browse Source 
The purpose is to make _rtpReceiver mostly agnostic to if it processes audio or video, and make its delegates responsible for that. This patch makes the actual interfaces and interactions between the classes a lot clearer which will probably help straighten out the rather convoluted business logic in here. There are a number of rough edges I hope to address in coming patches.

In particular, I think there are a lot of audio-specific hacks, especially when it comes to telephone event handling. I think we will see a lot of benefit once that stuff moves out of rtp_receiver altogether. The new strategy I introduced doesn't quite pull its own weight yet, but I think I will be able to remove a lot of that interface later once the responsibilities of the classes becomes move cohesive (e.g. that audio specific stuff actually lives in the audio class, and so on). Also I think it should be possible to extract payload type management to a helper class later on.

BUG=
TEST=vie/voe_auto_test, trybots

Review URL: https://webrtc-codereview.appspot.com/1001006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3306 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
phoglund@webrtc.org
2012-12-18 15:40:53 +00:00
parent 59ad541e57
commit 07bf43c673
13 changed files with 581 additions and 242 deletions
Download Patch File Download Diff File Expand all files Collapse all files

64
webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h
View File

@ -11,21 +11,20 @@
#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_VIDEO_H_
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_VIDEO_H_
#include "Bitrate.h"
#include "rtp_receiver.h"
#include "rtp_receiver_strategy.h"
#include "rtp_rtcp_defines.h"
#include "rtp_utility.h"
#include "typedefs.h"
#include "Bitrate.h"
#include "scoped_ptr.h"
#include "typedefs.h"
namespace webrtc {
class CriticalSectionWrapper;
class ModuleRtpRtcpImpl;
class ReceiverFEC;
class RTPReceiver;
class RTPReceiverVideo {
class RTPReceiverVideo : public RTPReceiverStrategy {
public:
RTPReceiverVideo(const WebRtc_Word32 id,
RTPReceiver* parent,
@ -33,20 +32,41 @@ class RTPReceiverVideo {
virtual ~RTPReceiverVideo();
ModuleRTPUtility::Payload* RegisterReceiveVideoPayload(
WebRtc_Word32 ParseRtpPacket(
WebRtcRTPHeader* rtp_header,
const ModuleRTPUtility::PayloadUnion& specificPayload,
const bool is_red,
const WebRtc_UWord8* packet,
const WebRtc_UWord16 packet_length,
const WebRtc_Word64 timestamp);
WebRtc_Word32 GetFrequencyHz() const;
RTPAliveType ProcessDeadOrAlive(WebRtc_UWord16 lastPayloadLength) const;
bool PayloadIsCompatible(
const ModuleRTPUtility::Payload& payload,
const WebRtc_UWord32 frequency,
const WebRtc_UWord8 channels,
const WebRtc_UWord32 rate) const;
void UpdatePayloadRate(
ModuleRTPUtility::Payload* payload,
const WebRtc_UWord32 rate) const;
ModuleRTPUtility::Payload* CreatePayloadType(
const char payloadName[RTP_PAYLOAD_NAME_SIZE],
const WebRtc_Word8 payloadType,
const WebRtc_UWord32 maxRate);
const WebRtc_UWord32 frequency,
const WebRtc_UWord8 channels,
const WebRtc_UWord32 rate);
WebRtc_Word32 ParseVideoCodecSpecific(
WebRtcRTPHeader* rtpHeader,
const WebRtc_UWord8* payloadData,
const WebRtc_UWord16 payloadDataLength,
const RtpVideoCodecTypes videoType,
const bool isRED,
const WebRtc_UWord8* incomingRtpPacket,
const WebRtc_UWord16 incomingRtpPacketSize,
const WebRtc_Word64 nowMS);
WebRtc_Word32 InvokeOnInitializeDecoder(
RtpFeedback* callback,
const WebRtc_Word32 id,
const WebRtc_Word8 payloadType,
const char payloadName[RTP_PAYLOAD_NAME_SIZE],
const ModuleRTPUtility::PayloadUnion& specificPayload) const;
virtual WebRtc_Word32 ReceiveRecoveredPacketCallback(
WebRtcRTPHeader* rtpHeader,
@ -77,6 +97,16 @@ class RTPReceiverVideo {
WebRtc_UWord8* dataBuffer) const;
private:
WebRtc_Word32 ParseVideoCodecSpecific(
WebRtcRTPHeader* rtpHeader,
const WebRtc_UWord8* payloadData,
const WebRtc_UWord16 payloadDataLength,
const RtpVideoCodecTypes videoType,
const bool isRED,
const WebRtc_UWord8* incomingRtpPacket,
const WebRtc_UWord16 incomingRtpPacketSize,
const WebRtc_Word64 nowMS);
WebRtc_Word32 _id;
RTPReceiver* _parent;