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Use size_t more consistently for packet/payload lengths.

Browse Source 
See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information.

This CL was reviewed and approved in pieces in the following CLs:
https://webrtc-codereview.appspot.com/24209004/
https://webrtc-codereview.appspot.com/24229004/
https://webrtc-codereview.appspot.com/24259004/
https://webrtc-codereview.appspot.com/25109004/
https://webrtc-codereview.appspot.com/26099004/
https://webrtc-codereview.appspot.com/27069004/
https://webrtc-codereview.appspot.com/27969004/
https://webrtc-codereview.appspot.com/27989004/
https://webrtc-codereview.appspot.com/29009004/
https://webrtc-codereview.appspot.com/30929004/
https://webrtc-codereview.appspot.com/30939004/
https://webrtc-codereview.appspot.com/31999004/
Committing as TBR to the original reviewers.

BUG=chromium:81439
TEST=none
TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom

Review URL: https://webrtc-codereview.appspot.com/23129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pkasting@chromium.org
2014-11-20 22:28:14 +00:00
parent edc6e57a92
commit 4591fbd09f
341 changed files with 2610 additions and 2613 deletions
Download Patch File Download Diff File Expand all files Collapse all files

8
webrtc/modules/audio_coding/neteq/mock/mock_payload_splitter.h
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@ -28,11 +28,11 @@ class MockPayloadSplitter : public PayloadSplitter {
MOCK_METHOD2(SplitAudio,
int(PacketList* packet_list, const DecoderDatabase& decoder_database));
MOCK_METHOD4(SplitBySamples,
void(const Packet* packet, int bytes_per_ms, int timestamps_per_ms,
PacketList* new_packets));
void(const Packet* packet, size_t bytes_per_ms,
uint32_t timestamps_per_ms, PacketList* new_packets));
MOCK_METHOD4(SplitByFrames,
int(const Packet* packet, int bytes_per_frame, int timestamps_per_frame,
PacketList* new_packets));
int(const Packet* packet, size_t bytes_per_frame,
uint32_t timestamps_per_frame, PacketList* new_packets));
};
} // namespace webrtc