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}
127 lines
4.5 KiB
C++
127 lines
4.5 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_SENDER_VIDEO_H_
|
|
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_
|
|
|
|
#include <list>
|
|
|
|
#include "webrtc/base/criticalsection.h"
|
|
#include "webrtc/base/onetimeevent.h"
|
|
#include "webrtc/base/rate_statistics.h"
|
|
#include "webrtc/base/thread_annotations.h"
|
|
#include "webrtc/common_types.h"
|
|
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
|
|
#include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
|
|
#include "webrtc/modules/rtp_rtcp/source/producer_fec.h"
|
|
#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
|
|
#include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
|
|
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
|
|
#include "webrtc/modules/rtp_rtcp/source/video_codec_information.h"
|
|
#include "webrtc/typedefs.h"
|
|
|
|
namespace webrtc {
|
|
|
|
class RTPSenderVideo {
|
|
public:
|
|
RTPSenderVideo(Clock* clock, RTPSenderInterface* rtp_sender);
|
|
virtual ~RTPSenderVideo();
|
|
|
|
virtual RtpVideoCodecTypes VideoCodecType() const;
|
|
|
|
size_t FECPacketOverhead() const;
|
|
|
|
static RtpUtility::Payload* CreateVideoPayload(
|
|
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
|
|
int8_t payload_type);
|
|
|
|
bool SendVideo(RtpVideoCodecTypes video_type,
|
|
FrameType frame_type,
|
|
int8_t payload_type,
|
|
uint32_t capture_timestamp,
|
|
int64_t capture_time_ms,
|
|
const uint8_t* payload_data,
|
|
size_t payload_size,
|
|
const RTPFragmentationHeader* fragmentation,
|
|
const RTPVideoHeader* video_header);
|
|
|
|
int32_t SendRTPIntraRequest();
|
|
|
|
void SetVideoCodecType(RtpVideoCodecTypes type);
|
|
|
|
// FEC
|
|
void SetGenericFECStatus(bool enable,
|
|
uint8_t payload_type_red,
|
|
uint8_t payload_type_fec);
|
|
|
|
void GenericFECStatus(bool* enable,
|
|
uint8_t* payload_type_red,
|
|
uint8_t* payload_type_fec) const;
|
|
|
|
void SetFecParameters(const FecProtectionParams* delta_params,
|
|
const FecProtectionParams* key_params);
|
|
|
|
uint32_t VideoBitrateSent() const;
|
|
uint32_t FecOverheadRate() const;
|
|
|
|
int SelectiveRetransmissions() const;
|
|
void SetSelectiveRetransmissions(uint8_t settings);
|
|
|
|
private:
|
|
void SendVideoPacket(uint8_t* data_buffer,
|
|
size_t payload_length,
|
|
size_t rtp_header_length,
|
|
uint16_t seq_num,
|
|
uint32_t capture_timestamp,
|
|
int64_t capture_time_ms,
|
|
StorageType storage);
|
|
|
|
void SendVideoPacketAsRed(uint8_t* data_buffer,
|
|
size_t payload_length,
|
|
size_t rtp_header_length,
|
|
uint16_t video_seq_num,
|
|
uint32_t capture_timestamp,
|
|
int64_t capture_time_ms,
|
|
StorageType media_packet_storage,
|
|
bool protect);
|
|
|
|
RTPSenderInterface* const rtp_sender_;
|
|
Clock* const clock_;
|
|
|
|
// Should never be held when calling out of this class.
|
|
rtc::CriticalSection crit_;
|
|
|
|
RtpVideoCodecTypes video_type_ = kRtpVideoGeneric;
|
|
int32_t retransmission_settings_ GUARDED_BY(crit_) = kRetransmitBaseLayer;
|
|
|
|
// FEC
|
|
ForwardErrorCorrection fec_;
|
|
bool fec_enabled_ GUARDED_BY(crit_) = false;
|
|
int8_t red_payload_type_ GUARDED_BY(crit_) = 0;
|
|
int8_t fec_payload_type_ GUARDED_BY(crit_) = 0;
|
|
FecProtectionParams delta_fec_params_ GUARDED_BY(crit_) = FecProtectionParams{
|
|
0, 1, kFecMaskRandom};
|
|
FecProtectionParams key_fec_params_ GUARDED_BY(crit_) = FecProtectionParams{
|
|
0, 1, kFecMaskRandom};
|
|
ProducerFec producer_fec_ GUARDED_BY(crit_);
|
|
|
|
rtc::CriticalSection stats_crit_;
|
|
// Bitrate used for FEC payload, RED headers, RTP headers for FEC packets
|
|
// and any padding overhead.
|
|
RateStatistics fec_bitrate_ GUARDED_BY(stats_crit_);
|
|
// Bitrate used for video payload and RTP headers.
|
|
RateStatistics video_bitrate_ GUARDED_BY(stats_crit_);
|
|
OneTimeEvent first_frame_sent_;
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_
|