Index: webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h |
diff --git a/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h b/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h |
similarity index 50% |
copy from webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h |
copy to webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h |
index d711685950e8b84094ba948dd81e854604848cad..03828429300fb432b8af0340da5bdfc5e1b35624 100644 |
--- a/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h |
+++ b/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h |
@@ -1,5 +1,5 @@ |
/* |
- * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. |
+ * 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 |
@@ -7,16 +7,13 @@ |
* 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_RTC_EVENT_LOG_SOURCE_H_ |
minyue-webrtc
2015/08/28 14:50:21
a line break before
ivoc
2015/09/01 10:03:50
Done.
|
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTC_EVENT_LOG_SOURCE_H_ |
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_ |
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_ |
- |
-#include <stdio.h> |
#include <string> |
#include "webrtc/base/constructormagic.h" |
#include "webrtc/base/scoped_ptr.h" |
-#include "webrtc/common_types.h" |
#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h" |
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" |
@@ -24,40 +21,48 @@ namespace webrtc { |
class RtpHeaderParser; |
+namespace rtclog { |
+class EventStream; |
+} // namespace rtclog |
+ |
namespace test { |
-class RtpFileReader; |
+class Packet; |
-class RtpFileSource : public PacketSource { |
+class RtcEventLogSource : public PacketSource { |
public: |
- // Creates an RtpFileSource reading from |file_name|. If the file cannot be |
- // opened, or has the wrong format, NULL will be returned. |
- static RtpFileSource* Create(const std::string& file_name); |
+ // Creates an RtcEventLogSource reading from |file_name|. If the file cannot |
+ // be opened, or has the wrong format, NULL will be returned. |
+ static RtcEventLogSource* Create(const std::string& file_name); |
- virtual ~RtpFileSource(); |
+ virtual ~RtcEventLogSource(); |
// Registers an RTP header extension and binds it to |id|. |
virtual bool RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id); |
// Returns a pointer to the next packet. Returns NULL if end of file was |
- // reached, or if a the data was corrupt. |
+ // reached. |
Packet* NextPacket() override; |
- private: |
- static const int kFirstLineLength = 40; |
- static const int kRtpFileHeaderSize = 4 + 4 + 4 + 2 + 2; |
- static const size_t kPacketHeaderSize = 8; |
+ // Returns the timestamp of the next audio output event, in milliseconds. A |
+ // value of -1 is returned if the end of the file is reached. |
hlundin-webrtc
2015/08/28 12:06:25
Is it end of file that is reached, or are there si
ivoc
2015/09/01 10:03:50
Updated to say that there are no more audio output
|
+ int NextAudioOutputEventMs(); |
- RtpFileSource(); |
+ private: |
+ RtcEventLogSource(); |
bool OpenFile(const std::string& file_name); |
- rtc::scoped_ptr<RtpFileReader> rtp_reader_; |
+ int rtp_packet_index_ = 0; |
+ int audio_output_index_ = 0; |
+ |
+ rtc::scoped_ptr<rtclog::EventStream> event_log_; |
rtc::scoped_ptr<RtpHeaderParser> parser_; |
- DISALLOW_COPY_AND_ASSIGN(RtpFileSource); |
+ DISALLOW_COPY_AND_ASSIGN(RtcEventLogSource); |
}; |
} // namespace test |
} // namespace webrtc |
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_ |
+ |
+#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTC_EVENT_LOG_SOURCE_H_ |