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Use size_t more consistently for packet/payload lengths.

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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
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27
webrtc/modules/audio_coding/main/interface/audio_coding_module.h
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@ -36,13 +36,12 @@ class AudioPacketizationCallback {
public:
virtual ~AudioPacketizationCallback() {}
virtual int32_t SendData(
FrameType frame_type,
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
uint16_t payload_len_bytes,
const RTPFragmentationHeader* fragmentation) = 0;
virtual int32_t SendData(FrameType frame_type,
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
size_t payload_len_bytes,
const RTPFragmentationHeader* fragmentation) = 0;
};
// Callback class used for inband Dtmf detection
@ -668,8 +667,8 @@ class AudioCodingModule: public Module {
// 0 if payload is successfully pushed in.
//
virtual int32_t IncomingPacket(const uint8_t* incoming_payload,
const int32_t payload_len_bytes,
const WebRtcRTPHeader& rtp_info) = 0;
const size_t payload_len_bytes,
const WebRtcRTPHeader& rtp_info) = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t IncomingPayload()
@ -696,9 +695,9 @@ class AudioCodingModule: public Module {
// 0 if payload is successfully pushed in.
//
virtual int32_t IncomingPayload(const uint8_t* incoming_payload,
const int32_t payload_len_byte,
const uint8_t payload_type,
const uint32_t timestamp = 0) = 0;
const size_t payload_len_byte,
const uint8_t payload_type,
const uint32_t timestamp = 0) = 0;
///////////////////////////////////////////////////////////////////////////
// int SetMinimumPlayoutDelay()
@ -1090,12 +1089,12 @@ class AudioCoding {
// |incoming_payload| contains the RTP payload after the RTP header. Return
// true if successful, false if not.
virtual bool InsertPacket(const uint8_t* incoming_payload,
int32_t payload_len_bytes,
size_t payload_len_bytes,
const WebRtcRTPHeader& rtp_info) = 0;
// TODO(henrik.lundin): Remove this method?
virtual bool InsertPayload(const uint8_t* incoming_payload,
int32_t payload_len_byte,
size_t payload_len_byte,
uint8_t payload_type,
uint32_t timestamp) = 0;