mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-16 15:20:42 +01:00

In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
84 lines
2.5 KiB
C++
84 lines
2.5 KiB
C++
/*
|
|
* Copyright (c) 2016 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_AUDIO_CODING_NETEQ_TOOLS_NETEQ_PACKET_SOURCE_INPUT_H_
|
|
#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_PACKET_SOURCE_INPUT_H_
|
|
|
|
#include <map>
|
|
#include <string>
|
|
|
|
#include "webrtc/modules/audio_coding/neteq/tools/neteq_input.h"
|
|
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
|
|
|
|
namespace webrtc {
|
|
namespace test {
|
|
|
|
class RtpFileSource;
|
|
class RtcEventLogSource;
|
|
|
|
// An adapter class to dress up a PacketSource object as a NetEqInput.
|
|
class NetEqPacketSourceInput : public NetEqInput {
|
|
public:
|
|
using RtpHeaderExtensionMap = std::map<int, webrtc::RTPExtensionType>;
|
|
|
|
NetEqPacketSourceInput();
|
|
rtc::Optional<int64_t> NextPacketTime() const override;
|
|
std::unique_ptr<PacketData> PopPacket() override;
|
|
rtc::Optional<RTPHeader> NextHeader() const override;
|
|
bool ended() const override { return !next_output_event_ms_; }
|
|
|
|
protected:
|
|
virtual PacketSource* source() = 0;
|
|
void LoadNextPacket();
|
|
|
|
rtc::Optional<int64_t> next_output_event_ms_;
|
|
|
|
private:
|
|
std::unique_ptr<Packet> packet_;
|
|
};
|
|
|
|
// Implementation of NetEqPacketSourceInput to be used with an RtpFileSource.
|
|
class NetEqRtpDumpInput final : public NetEqPacketSourceInput {
|
|
public:
|
|
NetEqRtpDumpInput(const std::string& file_name,
|
|
const RtpHeaderExtensionMap& hdr_ext_map);
|
|
|
|
rtc::Optional<int64_t> NextOutputEventTime() const override;
|
|
void AdvanceOutputEvent() override;
|
|
|
|
protected:
|
|
PacketSource* source() override;
|
|
|
|
private:
|
|
static constexpr int64_t kOutputPeriodMs = 10;
|
|
|
|
std::unique_ptr<RtpFileSource> source_;
|
|
};
|
|
|
|
// Implementation of NetEqPacketSourceInput to be used with an
|
|
// RtcEventLogSource.
|
|
class NetEqEventLogInput final : public NetEqPacketSourceInput {
|
|
public:
|
|
NetEqEventLogInput(const std::string& file_name,
|
|
const RtpHeaderExtensionMap& hdr_ext_map);
|
|
|
|
rtc::Optional<int64_t> NextOutputEventTime() const override;
|
|
void AdvanceOutputEvent() override;
|
|
|
|
protected:
|
|
PacketSource* source() override;
|
|
|
|
private:
|
|
std::unique_ptr<RtcEventLogSource> source_;
|
|
};
|
|
|
|
} // namespace test
|
|
} // namespace webrtc
|
|
#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_PACKET_SOURCE_INPUT_H_
|