822fbd8b68
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
138 lines
4.7 KiB
C++
138 lines
4.7 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_AUDIO_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_
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#include <set>
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#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
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#include "webrtc/modules/rtp_rtcp/interface/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/system_wrappers/interface/scoped_ptr.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class CriticalSectionWrapper;
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// Handles audio RTP packets. This class is thread-safe.
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class RTPReceiverAudio : public RTPReceiverStrategy,
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public TelephoneEventHandler {
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public:
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RTPReceiverAudio(const int32_t id,
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RtpData* data_callback,
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RtpAudioFeedback* incoming_messages_callback);
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virtual ~RTPReceiverAudio() {}
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// The following three methods implement the TelephoneEventHandler interface.
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// Forward DTMFs to decoder for playout.
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void SetTelephoneEventForwardToDecoder(bool forward_to_decoder);
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// Is forwarding of outband telephone events turned on/off?
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bool TelephoneEventForwardToDecoder() const;
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// Is TelephoneEvent configured with payload type payload_type
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bool TelephoneEventPayloadType(const int8_t payload_type) const;
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TelephoneEventHandler* GetTelephoneEventHandler() {
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return this;
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}
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// Returns true if CNG is configured with payload type payload_type. If so,
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// the frequency and cng_payload_type_has_changed are filled in.
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bool CNGPayloadType(const int8_t payload_type,
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uint32_t* frequency,
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bool* cng_payload_type_has_changed);
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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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uint16_t packet_length,
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int64_t timestamp_ms,
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bool is_first_packet);
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int GetPayloadTypeFrequency() const OVERRIDE;
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virtual RTPAliveType ProcessDeadOrAlive(uint16_t last_payload_length) const
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OVERRIDE;
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virtual bool ShouldReportCsrcChanges(uint8_t payload_type) const OVERRIDE;
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virtual 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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virtual int32_t InvokeOnInitializeDecoder(
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RtpFeedback* callback,
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int32_t id,
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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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// We do not allow codecs to have multiple payload types for audio, so we
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// need to override the default behavior (which is to do nothing).
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void PossiblyRemoveExistingPayloadType(
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ModuleRTPUtility::PayloadTypeMap* payload_type_map,
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const char payload_name[RTP_PAYLOAD_NAME_SIZE],
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size_t payload_name_length,
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uint32_t frequency,
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uint8_t channels,
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uint32_t rate) const;
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// We need to look out for special payload types here and sometimes reset
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// statistics. In addition we sometimes need to tweak the frequency.
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void CheckPayloadChanged(int8_t payload_type,
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PayloadUnion* specific_payload,
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bool* should_reset_statistics,
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bool* should_discard_changes) OVERRIDE;
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int Energy(uint8_t array_of_energy[kRtpCsrcSize]) const OVERRIDE;
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private:
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int32_t ParseAudioCodecSpecific(
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WebRtcRTPHeader* rtp_header,
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const uint8_t* payload_data,
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uint16_t payload_length,
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const AudioPayload& audio_specific,
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bool is_red);
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int32_t id_;
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uint32_t last_received_frequency_;
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bool telephone_event_forward_to_decoder_;
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int8_t telephone_event_payload_type_;
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std::set<uint8_t> telephone_event_reported_;
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int8_t cng_nb_payload_type_;
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int8_t cng_wb_payload_type_;
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int8_t cng_swb_payload_type_;
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int8_t cng_fb_payload_type_;
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int8_t cng_payload_type_;
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// G722 is special since it use the wrong number of RTP samples in timestamp
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// VS. number of samples in the frame
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int8_t g722_payload_type_;
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bool last_received_g722_;
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uint8_t num_energy_;
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uint8_t current_remote_energy_[kRtpCsrcSize];
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RtpAudioFeedback* cb_audio_feedback_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_
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