525df3ffd1
OnEncodedImage() is going to replace Encoded(), which is deprecated now. The new OnEncodedImage() returns Result struct that contains frame_id, which tells the encoder RTP timestamp for the frame. BUG=chromium:621691 R=niklas.enbom@webrtc.org, sprang@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/2089773002 . Committed: https://crrev.com/4c7f4cd2ef76821edca6d773d733a924b0bedd25 Committed: https://crrev.com/ad34dbe934d47f88011045671b4aea00dbd5a795 Cr-Original-Original-Commit-Position: refs/heads/master@{#13613} Cr-Original-Commit-Position: refs/heads/master@{#13615} Cr-Commit-Position: refs/heads/master@{#13617}
112 lines
3.8 KiB
C++
112 lines
3.8 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_SENDER_AUDIO_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
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#include "webrtc/common_types.h"
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#include "webrtc/base/criticalsection.h"
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#include "webrtc/base/onetimeevent.h"
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#include "webrtc/modules/rtp_rtcp/source/dtmf_queue.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_sender.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 RTPSenderAudio : public DTMFqueue {
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public:
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RTPSenderAudio(Clock* clock, RTPSender* rtp_sender);
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virtual ~RTPSenderAudio();
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int32_t RegisterAudioPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE],
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int8_t payload_type,
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uint32_t frequency,
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size_t channels,
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uint32_t rate,
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RtpUtility::Payload** payload);
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bool SendAudio(FrameType frame_type,
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int8_t payload_type,
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uint32_t capture_timestamp,
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const uint8_t* payload_data,
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size_t payload_size,
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const RTPFragmentationHeader* fragmentation);
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// set audio packet size, used to determine when it's time to send a DTMF
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// packet in silence (CNG)
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int32_t SetAudioPacketSize(uint16_t packet_size_samples);
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// Store the audio level in dBov for
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// header-extension-for-audio-level-indication.
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// Valid range is [0,100]. Actual value is negative.
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int32_t SetAudioLevel(uint8_t level_dbov);
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// Send a DTMF tone using RFC 2833 (4733)
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int32_t SendTelephoneEvent(uint8_t key, uint16_t time_ms, uint8_t level);
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int AudioFrequency() const;
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// Set payload type for Redundant Audio Data RFC 2198
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int32_t SetRED(int8_t payload_type);
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// Get payload type for Redundant Audio Data RFC 2198
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int32_t RED(int8_t* payload_type) const;
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protected:
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bool SendTelephoneEventPacket(
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bool ended,
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int8_t dtmf_payload_type,
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uint32_t dtmf_timestamp,
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uint16_t duration,
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bool marker_bit); // set on first packet in talk burst
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bool MarkerBit(FrameType frame_type, int8_t payload_type);
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private:
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Clock* const clock_;
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RTPSender* const rtp_sender_;
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rtc::CriticalSection send_audio_critsect_;
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uint16_t packet_size_samples_ GUARDED_BY(send_audio_critsect_);
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// DTMF.
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bool dtmf_event_is_on_;
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bool dtmf_event_first_packet_sent_;
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int8_t dtmf_payload_type_ GUARDED_BY(send_audio_critsect_);
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uint32_t dtmf_timestamp_;
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uint8_t dtmf_key_;
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uint32_t dtmf_length_samples_;
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uint8_t dtmf_level_;
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int64_t dtmf_time_last_sent_;
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uint32_t dtmf_timestamp_last_sent_;
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int8_t red_payload_type_ GUARDED_BY(send_audio_critsect_);
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// VAD detection, used for marker bit.
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bool inband_vad_active_ GUARDED_BY(send_audio_critsect_);
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int8_t cngnb_payload_type_ GUARDED_BY(send_audio_critsect_);
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int8_t cngwb_payload_type_ GUARDED_BY(send_audio_critsect_);
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int8_t cngswb_payload_type_ GUARDED_BY(send_audio_critsect_);
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int8_t cngfb_payload_type_ GUARDED_BY(send_audio_critsect_);
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int8_t last_payload_type_ GUARDED_BY(send_audio_critsect_);
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// Audio level indication.
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// (https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/)
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uint8_t audio_level_dbov_ GUARDED_BY(send_audio_critsect_);
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OneTimeEvent first_packet_sent_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
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