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}
61 lines
2.0 KiB
C++
61 lines
2.0 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_VIDEO_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_VIDEO_H_
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#include "webrtc/base/onetimeevent.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_receiver_strategy.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 RTPReceiverVideo : public RTPReceiverStrategy {
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public:
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explicit RTPReceiverVideo(RtpData* data_callback);
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virtual ~RTPReceiverVideo();
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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* packet,
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size_t packet_length,
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int64_t timestamp,
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bool is_first_packet) override;
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TelephoneEventHandler* GetTelephoneEventHandler() override { return NULL; }
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RTPAliveType ProcessDeadOrAlive(uint16_t last_payload_length) const override;
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bool ShouldReportCsrcChanges(uint8_t payload_type) const override;
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int32_t OnNewPayloadTypeCreated(
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const char payload_name[RTP_PAYLOAD_NAME_SIZE],
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int8_t payload_type,
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uint32_t frequency) override;
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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 override;
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void SetPacketOverHead(uint16_t packet_over_head);
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private:
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OneTimeEvent first_packet_received_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_VIDEO_H_
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