webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h - Issue 2005873002: Let PacketSource::NextPacket() return an std::unique_ptr

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

Issue 2005873002: Let PacketSource::NextPacket() return an std::unique_ptr (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: rebase Created 4 years, 7 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/rtc_event_log_source.h b/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h

index ad7add154c5a1d3371f78ad85e0f4b89933223c1..71bf841bde8871947ba866f91a85815a55786a5a 100644

--- a/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h

+++ b/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h

@@ -38,9 +38,7 @@ class RtcEventLogSource : public PacketSource {

// 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;

+ std::unique_ptr<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