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3ed46bd83b
Bug: webrtc:7135 Change-Id: Ic20d98cbfb8154f5abbc2501cbcccb950148e732 Reviewed-on: https://webrtc-review.googlesource.com/92396 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24219}
46 lines
1.6 KiB
C++
46 lines
1.6 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:
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explicit RTPReceiverAudio(RtpData* data_callback);
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~RTPReceiverAudio() override;
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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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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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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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