1bd66642c3
All users call SetTelephoneEventForwardToDecoder(true). Setting the flag to true on construction, enables deletion of those calls, followed by deletion of the flag itself. The unused getter method TelephoneEventForwardToDecoder() is deleted right away. Bug: webrtc:7135 Change-Id: I8c52c957b3f074be7ffc425b3588402d1e42b844 Reviewed-on: https://webrtc-review.googlesource.com/90402 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24141}
80 lines
3.0 KiB
C++
80 lines
3.0 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 MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_
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#define MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_
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#include <set>
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#include "modules/rtp_rtcp/include/rtp_receiver.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "modules/rtp_rtcp/source/rtp_receiver_strategy.h"
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#include "modules/rtp_rtcp/source/rtp_utility.h"
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#include "rtc_base/onetimeevent.h"
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namespace webrtc {
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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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explicit RTPReceiverAudio(RtpData* data_callback);
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~RTPReceiverAudio() override;
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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) override;
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// Is TelephoneEvent configured with |payload_type|.
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bool TelephoneEventPayloadType(const int8_t payload_type) const override;
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TelephoneEventHandler* GetTelephoneEventHandler() override;
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// Returns true if CNG is configured with |payload_type|.
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bool CNGPayloadType(const int8_t payload_type);
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int32_t ParseRtpPacket(WebRtcRTPHeader* rtp_header,
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const PayloadUnion& specific_payload,
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const uint8_t* packet,
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size_t payload_length,
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int64_t timestamp_ms) override;
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RTPAliveType ProcessDeadOrAlive(uint16_t last_payload_length) const override;
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int32_t OnNewPayloadTypeCreated(int payload_type,
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const SdpAudioFormat& audio_format) override;
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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_discard_changes) override;
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private:
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int32_t ParseAudioCodecSpecific(WebRtcRTPHeader* rtp_header,
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const uint8_t* payload_data,
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size_t payload_length,
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const AudioPayload& audio_specific);
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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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ThreadUnsafeOneTimeEvent first_packet_received_;
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};
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_
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