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

Update talk to 50918584.

Browse Source 
Together with Stefan's http://review.webrtc.org/1960004/.

R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2048004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4556 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
wu@webrtc.org
2013-08-15 23:38:54 +00:00
parent dde7d4c6ed
commit 822fbd8b68
108 changed files with 2926 additions and 4301 deletions
Download Patch File Download Diff File Expand all files Collapse all files

85
webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h
View File

@ -14,23 +14,24 @@
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class TelephoneEventHandler;
// This strategy deals with media-specific RTP packet processing.
// This class is not thread-safe and must be protected by its caller.
class RTPReceiverStrategy {
public:
// The data callback is where we should send received payload data.
// See ParseRtpPacket. This class does not claim ownership of the callback.
// Implementations must NOT hold any critical sections while calling the
// callback.
//
// Note: Implementations may call the callback for other reasons than calls
// to ParseRtpPacket, for instance if the implementation somehow recovers a
// packet.
RTPReceiverStrategy(RtpData* data_callback);
static RTPReceiverStrategy* CreateVideoStrategy(int32_t id,
RtpData* data_callback);
static RTPReceiverStrategy* CreateAudioStrategy(
int32_t id, RtpData* data_callback,
RtpAudioFeedback* incoming_messages_callback);
virtual ~RTPReceiverStrategy() {}
// Parses the RTP packet and calls the data callback with the payload data.
@ -39,21 +40,22 @@ class RTPReceiverStrategy {
// make changes in the data as necessary. The specific_payload argument
// provides audio or video-specific data. The is_first_packet argument is true
// if this packet is either the first packet ever or the first in its frame.
virtual int32_t ParseRtpPacket(
WebRtcRTPHeader* rtp_header,
const ModuleRTPUtility::PayloadUnion& specific_payload,
const bool is_red,
const uint8_t* packet,
const uint16_t packet_length,
const int64_t timestamp_ms,
const bool is_first_packet) = 0;
virtual int32_t ParseRtpPacket(WebRtcRTPHeader* rtp_header,
const PayloadUnion& specific_payload,
bool is_red,
const uint8_t* packet,
uint16_t packet_length,
int64_t timestamp_ms,
bool is_first_packet) = 0;
virtual TelephoneEventHandler* GetTelephoneEventHandler() = 0;
// Retrieves the last known applicable frequency.
virtual int32_t GetFrequencyHz() const = 0;
virtual int GetPayloadTypeFrequency() const = 0;
// Computes the current dead-or-alive state.
virtual RTPAliveType ProcessDeadOrAlive(
uint16_t last_payload_length) const = 0;
uint16_t last_payload_length) const = 0;
// Returns true if we should report CSRC changes for this payload type.
// TODO(phoglund): should move out of here along with other payload stuff.
@ -63,36 +65,45 @@ class RTPReceiverStrategy {
// the payload registry.
virtual int32_t OnNewPayloadTypeCreated(
const char payloadName[RTP_PAYLOAD_NAME_SIZE],
const int8_t payloadType,
const uint32_t frequency) = 0;
int8_t payloadType,
uint32_t frequency) = 0;
// Invokes the OnInitializeDecoder callback in a media-specific way.
virtual int32_t InvokeOnInitializeDecoder(
RtpFeedback* callback,
const int32_t id,
const int8_t payload_type,
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
const ModuleRTPUtility::PayloadUnion& specific_payload) const = 0;
RtpFeedback* callback,
int32_t id,
int8_t payload_type,
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
const PayloadUnion& specific_payload) const = 0;
// Checks if the payload type has changed, and returns whether we should
// reset statistics and/or discard this packet.
virtual void CheckPayloadChanged(
const int8_t payload_type,
ModuleRTPUtility::PayloadUnion* specific_payload,
bool* should_reset_statistics,
bool* should_discard_changes);
virtual void CheckPayloadChanged(int8_t payload_type,
PayloadUnion* specific_payload,
bool* should_reset_statistics,
bool* should_discard_changes);
virtual int Energy(uint8_t array_of_energy[kRtpCsrcSize]) const;
// Stores / retrieves the last media specific payload for later reference.
void GetLastMediaSpecificPayload(
ModuleRTPUtility::PayloadUnion* payload) const;
void SetLastMediaSpecificPayload(
const ModuleRTPUtility::PayloadUnion& payload);
void GetLastMediaSpecificPayload(PayloadUnion* payload) const;
void SetLastMediaSpecificPayload(const PayloadUnion& payload);
protected:
ModuleRTPUtility::PayloadUnion last_payload_;
// The data callback is where we should send received payload data.
// See ParseRtpPacket. This class does not claim ownership of the callback.
// Implementations must NOT hold any critical sections while calling the
// callback.
//
// Note: Implementations may call the callback for other reasons than calls
// to ParseRtpPacket, for instance if the implementation somehow recovers a
// packet.
RTPReceiverStrategy(RtpData* data_callback);
scoped_ptr<CriticalSectionWrapper> crit_sect_;
PayloadUnion last_payload_;
RtpData* data_callback_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_