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Unified Diff: webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h

Issue 1316903002: Update to the neteq_rtpplay utility to support RtcEventLog input files. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Initial version Created 5 years, 4 months ago
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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_

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