blob: f909b85aecd0078cb6cdbb7ade5fa40f90038612 [file] [log] [blame]
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/rtc_base/criticalsection.h"
#include "webrtc/typedefs.h"
namespace webrtc {
struct CodecInst;
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 {
static RTPReceiverStrategy* CreateVideoStrategy(RtpData* data_callback);
static RTPReceiverStrategy* CreateAudioStrategy(RtpData* data_callback);
virtual ~RTPReceiverStrategy() {}
// Parses the RTP packet and calls the data callback with the payload data.
// Implementations are encouraged to use the provided packet buffer and RTP
// header as arguments to the callback; implementations are also allowed to
// 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 PayloadUnion& specific_payload,
bool is_red,
const uint8_t* payload,
size_t payload_length,
int64_t timestamp_ms,
bool is_first_packet) = 0;
virtual TelephoneEventHandler* GetTelephoneEventHandler() = 0;
// Computes the current dead-or-alive state.
virtual RTPAliveType ProcessDeadOrAlive(
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.
virtual bool ShouldReportCsrcChanges(uint8_t payload_type) const = 0;
// Notifies the strategy that we have created a new non-RED audio payload type
// in the payload registry.
virtual int32_t OnNewPayloadTypeCreated(const CodecInst& audio_codec) = 0;
// Invokes the OnInitializeDecoder callback in a media-specific way.
virtual int32_t InvokeOnInitializeDecoder(
RtpFeedback* callback,
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(int8_t payload_type,
PayloadUnion* specific_payload,
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(PayloadUnion* payload) const;
void SetLastMediaSpecificPayload(const PayloadUnion& 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.
explicit RTPReceiverStrategy(RtpData* data_callback);
rtc::CriticalSection crit_sect_;
PayloadUnion last_payload_;
RtpData* data_callback_;
} // namespace webrtc