webrtc_m130/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h
magjed f3feeffe03 Reland of move RTPPayloadStrategy and simplify RTPPayloadRegistry (patchset #1 id:1 of https://codereview.webrtc.org/2528993002/ )
Reason for revert:
Downstream code has been updated.

Original issue's description:
> Revert of Remove RTPPayloadStrategy and simplify RTPPayloadRegistry (patchset #7 id:240001 of https://codereview.webrtc.org/2524923002/ )
>
> Reason for revert:
> Breaks downstream projects.
>
> Original issue's description:
> > Remove RTPPayloadStrategy and simplify RTPPayloadRegistry
> >
> > This CL removes RTPPayloadStrategy that is currently used to handle
> > audio/video specific aspects of payload handling. Instead, the audio and
> > video specific aspects will now have different functions, with linear
> > code flow.
> >
> > This CL does not contain any functional changes, and is just a
> > preparation for future CL:s.
> >
> > The main purpose with this CL is to add this function:
> > bool PayloadIsCompatible(const RtpUtility::Payload& payload,
> >                          const webrtc::VideoCodec& video_codec);
> > that can easily be extended in a future CL to look at video codec
> > specific information.
> >
> > BUG=webrtc:6743
> >
> > Committed: https://crrev.com/b881254dc86d2cc80a52e08155433458be002166
> > Cr-Commit-Position: refs/heads/master@{#15232}
>
> TBR=danilchap@webrtc.org,solenberg@webrtc.org,mflodman@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6743
>
> Committed: https://crrev.com/33c81d05613f45f65ee17224ed381c6cdd1c6c6f
> Cr-Commit-Position: refs/heads/master@{#15234}

TBR=danilchap@webrtc.org,solenberg@webrtc.org,mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6743

Review-Url: https://codereview.webrtc.org/2531043002
Cr-Commit-Position: refs/heads/master@{#15245}
2016-11-25 14:40:30 +00:00

99 lines
3.7 KiB
C++

/*
* 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.
*/
#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_
#include <set>
#include "webrtc/base/onetimeevent.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/typedefs.h"
namespace webrtc {
// Handles audio RTP packets. This class is thread-safe.
class RTPReceiverAudio : public RTPReceiverStrategy,
public TelephoneEventHandler {
public:
explicit RTPReceiverAudio(RtpData* data_callback);
virtual ~RTPReceiverAudio() {}
// The following three methods implement the TelephoneEventHandler interface.
// Forward DTMFs to decoder for playout.
void SetTelephoneEventForwardToDecoder(bool forward_to_decoder) override;
// Is forwarding of outband telephone events turned on/off?
bool TelephoneEventForwardToDecoder() const override;
// Is TelephoneEvent configured with |payload_type|.
bool TelephoneEventPayloadType(const int8_t payload_type) const override;
TelephoneEventHandler* GetTelephoneEventHandler() override { return this; }
// Returns true if CNG is configured with |payload_type|.
bool CNGPayloadType(const int8_t payload_type);
int32_t ParseRtpPacket(WebRtcRTPHeader* rtp_header,
const PayloadUnion& specific_payload,
bool is_red,
const uint8_t* packet,
size_t payload_length,
int64_t timestamp_ms,
bool is_first_packet) override;
RTPAliveType ProcessDeadOrAlive(uint16_t last_payload_length) const override;
bool ShouldReportCsrcChanges(uint8_t payload_type) const override;
int32_t OnNewPayloadTypeCreated(const CodecInst& audio_codec) override;
int32_t InvokeOnInitializeDecoder(
RtpFeedback* callback,
int8_t payload_type,
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
const PayloadUnion& specific_payload) const override;
// We need to look out for special payload types here and sometimes reset
// statistics. In addition we sometimes need to tweak the frequency.
void CheckPayloadChanged(int8_t payload_type,
PayloadUnion* specific_payload,
bool* should_discard_changes) override;
int Energy(uint8_t array_of_energy[kRtpCsrcSize]) const override;
private:
int32_t ParseAudioCodecSpecific(WebRtcRTPHeader* rtp_header,
const uint8_t* payload_data,
size_t payload_length,
const AudioPayload& audio_specific,
bool is_red);
bool telephone_event_forward_to_decoder_;
int8_t telephone_event_payload_type_;
std::set<uint8_t> telephone_event_reported_;
int8_t cng_nb_payload_type_;
int8_t cng_wb_payload_type_;
int8_t cng_swb_payload_type_;
int8_t cng_fb_payload_type_;
uint8_t num_energy_;
uint8_t current_remote_energy_[kRtpCsrcSize];
ThreadUnsafeOneTimeEvent first_packet_received_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_