Use size_t more consistently for packet/payload lengths.
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
diff --git a/webrtc/modules/audio_coding/main/test/TestAllCodecs.h b/webrtc/modules/audio_coding/main/test/TestAllCodecs.h
index 2fbf9ef..42d65a1 100644
--- a/webrtc/modules/audio_coding/main/test/TestAllCodecs.h
+++ b/webrtc/modules/audio_coding/main/test/TestAllCodecs.h
@@ -29,12 +29,14 @@
void RegisterReceiverACM(AudioCodingModule* acm);
virtual int32_t SendData(
- FrameType frame_type, uint8_t payload_type,
- uint32_t timestamp, const uint8_t* payload_data,
- uint16_t payload_size,
+ FrameType frame_type,
+ uint8_t payload_type,
+ uint32_t timestamp,
+ const uint8_t* payload_data,
+ size_t payload_size,
const RTPFragmentationHeader* fragmentation) OVERRIDE;
- uint16_t payload_size();
+ size_t payload_size();
uint32_t timestamp_diff();
void reset_payload_size();
@@ -45,7 +47,7 @@
uint32_t timestamp_diff_;
uint32_t last_in_timestamp_;
uint64_t total_bytes_;
- uint16_t payload_size_;
+ size_t payload_size_;
};
class TestAllCodecs : public ACMTest {
@@ -61,7 +63,7 @@
// This is useful for codecs which support several sampling frequency.
// Note! Only mono mode is tested in this test.
void RegisterSendCodec(char side, char* codec_name, int32_t sampling_freq_hz,
- int rate, int packet_size, int extra_byte);
+ int rate, int packet_size, size_t extra_byte);
void Run(TestPack* channel);
void OpenOutFile(int test_number);
@@ -75,7 +77,7 @@
PCMFile outfile_b_;
int test_count_;
int packet_size_samples_;
- int packet_size_bytes_;
+ size_t packet_size_bytes_;
};
} // namespace webrtc