425a6ccac3
CheckPayloadChanged. Removed last_received_frequency_, cng_payload_type_, g722_payload_type_ and last_received_g722_ from RTPReceiverAudio and cleaned up most of the related, now dead code. Since g722_payload_type_ was never set, neither was last_received_g722_, which means the frequency change in CNGPayloadType was never done. Setting the frequency to the standard values also proved unnecessary, since they were already set before the call. Even if frequency would have been changed by RTPReceiverAudio, I was not able to find a place where that would actually have mattered. The ACM and NetEq, for example, which eventually gets these packages, don't care about that value. Also, GetPayloadTypeFrequency was never called, so keeping track of last_received_frequency_ proved unnecessary. cng_payload_type_ was stored to be able to check in CNGPayloadType if cng_payload_type_has_changed. This flag was also never read, so these all disappear. The main reason for starting this change was to root out any G722 specific code we have sprinkled around the code base (specifically dealing with the fact that for G722 clock rate != sample rate). In this case, once I started pulling at one end of the string, the whole thing came unraveled. BUG=webrtc:5805 Review-Url: https://codereview.webrtc.org/2383103002 Cr-Commit-Position: refs/heads/master@{#14530}
101 lines
4.2 KiB
C++
101 lines
4.2 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_
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#include "webrtc/base/criticalsection.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class TelephoneEventHandler;
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// This strategy deals with media-specific RTP packet processing.
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// This class is not thread-safe and must be protected by its caller.
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class RTPReceiverStrategy {
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public:
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static RTPReceiverStrategy* CreateVideoStrategy(RtpData* data_callback);
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static RTPReceiverStrategy* CreateAudioStrategy(RtpData* data_callback);
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virtual ~RTPReceiverStrategy() {}
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// Parses the RTP packet and calls the data callback with the payload data.
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// Implementations are encouraged to use the provided packet buffer and RTP
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// header as arguments to the callback; implementations are also allowed to
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// make changes in the data as necessary. The specific_payload argument
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// provides audio or video-specific data. The is_first_packet argument is true
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// if this packet is either the first packet ever or the first in its frame.
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virtual int32_t ParseRtpPacket(WebRtcRTPHeader* rtp_header,
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const PayloadUnion& specific_payload,
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bool is_red,
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const uint8_t* payload,
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size_t payload_length,
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int64_t timestamp_ms,
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bool is_first_packet) = 0;
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virtual TelephoneEventHandler* GetTelephoneEventHandler() = 0;
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// Computes the current dead-or-alive state.
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virtual RTPAliveType ProcessDeadOrAlive(
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uint16_t last_payload_length) const = 0;
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// Returns true if we should report CSRC changes for this payload type.
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// TODO(phoglund): should move out of here along with other payload stuff.
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virtual bool ShouldReportCsrcChanges(uint8_t payload_type) const = 0;
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// Notifies the strategy that we have created a new non-RED payload type in
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// the payload registry.
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virtual int32_t OnNewPayloadTypeCreated(
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const char payloadName[RTP_PAYLOAD_NAME_SIZE],
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int8_t payloadType,
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uint32_t frequency) = 0;
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// Invokes the OnInitializeDecoder callback in a media-specific way.
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virtual int32_t InvokeOnInitializeDecoder(
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RtpFeedback* callback,
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int8_t payload_type,
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const char payload_name[RTP_PAYLOAD_NAME_SIZE],
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const PayloadUnion& specific_payload) const = 0;
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// Checks if the payload type has changed, and returns whether we should
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// reset statistics and/or discard this packet.
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virtual void CheckPayloadChanged(int8_t payload_type,
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PayloadUnion* specific_payload,
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bool* should_discard_changes);
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virtual int Energy(uint8_t array_of_energy[kRtpCsrcSize]) const;
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// Stores / retrieves the last media specific payload for later reference.
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void GetLastMediaSpecificPayload(PayloadUnion* payload) const;
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void SetLastMediaSpecificPayload(const PayloadUnion& payload);
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protected:
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// The data callback is where we should send received payload data.
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// See ParseRtpPacket. This class does not claim ownership of the callback.
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// Implementations must NOT hold any critical sections while calling the
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// callback.
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//
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// Note: Implementations may call the callback for other reasons than calls
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// to ParseRtpPacket, for instance if the implementation somehow recovers a
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// packet.
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explicit RTPReceiverStrategy(RtpData* data_callback);
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rtc::CriticalSection crit_sect_;
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PayloadUnion last_payload_;
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RtpData* data_callback_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_
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