| Index: webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.h
|
| diff --git a/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.h b/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.h
|
| new file mode 100644
|
| index 0000000000000000000000000000000000000000..ade88f413b0392582069e6f4ad363b92f5efc799
|
| --- /dev/null
|
| +++ b/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.h
|
| @@ -0,0 +1,68 @@
|
| +/*
|
| + * Copyright (c) 2015 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_LOGGING_RTC_EVENT_LOG_ENCODER_RTC_EVENT_LOG_ENCODER_LEGACY_H_
|
| +#define WEBRTC_LOGGING_RTC_EVENT_LOG_ENCODER_RTC_EVENT_LOG_ENCODER_LEGACY_H_
|
| +
|
| +#include <memory>
|
| +#include <string>
|
| +#include <vector>
|
| +
|
| +#include "webrtc/api/array_view.h"
|
| +#include "webrtc/api/rtpparameters.h"
|
| +#include "webrtc/common_types.h"
|
| +#include "webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder.h"
|
| +#include "webrtc/rtc_base/platform_file.h"
|
| +
|
| +namespace webrtc {
|
| +
|
| +class RtcEventLogEncoderLegacy final : public RtcEventLogEncoder {
|
| + public:
|
| + ~RtcEventLogEncoderLegacy() override = default;
|
| +
|
| + void LoggingStart() override;
|
| + void LoggingStopped() override;
|
| + void VideoReceiveStreamConfig(const rtclog::StreamConfig& config) override;
|
| + void VideoSendStreamConfig(const rtclog::StreamConfig& config) override;
|
| + void AudioReceiveStreamConfig(const rtclog::StreamConfig& config) override;
|
| + void AudioSendStreamConfig(const rtclog::StreamConfig& config) override;
|
| + void RtpHeader(PacketDirection direction,
|
| + const uint8_t* header,
|
| + size_t packet_length) override;
|
| + void RtpHeader(PacketDirection direction,
|
| + const uint8_t* header,
|
| + size_t packet_length,
|
| + int probe_cluster_id) override;
|
| + void IncomingRtpHeader(const RtpPacketReceived& packet) override;
|
| + void OutgoingRtpHeader(const RtpPacketToSend& packet,
|
| + int probe_cluster_id) override;
|
| + void RtcpPacket(PacketDirection direction,
|
| + const uint8_t* packet,
|
| + size_t length) override;
|
| + void IncomingRtcpPacket(rtc::ArrayView<const uint8_t> packet) override;
|
| + void OutgoingRtcpPacket(rtc::ArrayView<const uint8_t> packet) override;
|
| + void AudioPlayout(uint32_t ssrc) override;
|
| + void LossBasedBweUpdate(int32_t bitrate_bps,
|
| + uint8_t fraction_loss,
|
| + int32_t total_packets) override;
|
| + void DelayBasedBweUpdate(int32_t bitrate_bps,
|
| + BandwidthUsage detector_state) override;
|
| + void AudioNetworkAdaptation(const AudioEncoderRuntimeConfig& config) override;
|
| + void ProbeClusterCreated(int id,
|
| + int bitrate_bps,
|
| + int min_probes,
|
| + int min_bytes) override;
|
| + void ProbeResultSuccess(int id, int bitrate_bps) override;
|
| + void ProbeResultFailure(int id, ProbeFailureReason failure_reason) override;
|
| +};
|
| +
|
| +} // namespace webrtc
|
| +
|
| +#endif // WEBRTC_LOGGING_RTC_EVENT_LOG_ENCODER_RTC_EVENT_LOG_ENCODER_LEGACY_H_
|
|
|