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