Index: webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h |
diff --git a/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h b/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h |
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index 0000000000000000000000000000000000000000..d144b516a28cf9e8030cb1a0e9e29882e359a981 |
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+++ b/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h |
@@ -0,0 +1,70 @@ |
+/* |
+ * 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_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTC_EVENT_LOG_SOURCE_H_ |
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTC_EVENT_LOG_SOURCE_H_ |
+ |
+#include <string> |
+ |
+#include "webrtc/base/constructormagic.h" |
+#include "webrtc/base/scoped_ptr.h" |
+#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h" |
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" |
+ |
+namespace webrtc { |
+ |
+class RtpHeaderParser; |
+ |
+namespace rtclog { |
+class EventStream; |
+} // namespace rtclog |
+ |
+namespace test { |
+ |
+class Packet; |
+ |
+class RtcEventLogSource : public PacketSource { |
+ public: |
+ // 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 ~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. |
+ Packet* NextPacket() override; |
+ |
+ // Returns the timestamp of the next audio output event, in milliseconds. The |
+ // maximum value of int64_t is returned if there are no more audio output |
+ // events available. |
+ int64_t NextAudioOutputEventMs(); |
+ |
+ private: |
+ RtcEventLogSource(); |
+ |
+ bool OpenFile(const std::string& file_name); |
+ |
+ 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(RtcEventLogSource); |
+}; |
+ |
+} // namespace test |
+} // namespace webrtc |
+ |
+#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTC_EVENT_LOG_SOURCE_H_ |