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
This commit is contained in:
@ -18,6 +18,8 @@
|
||||
#include "webrtc/modules/audio_coding/main/test/Channel.h"
|
||||
#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
|
||||
|
||||
#define PCMA_AND_PCMU
|
||||
|
||||
namespace webrtc {
|
||||
|
||||
enum StereoMonoMode {
|
||||
@ -38,7 +40,7 @@ class TestPackStereo : public AudioPacketizationCallback {
|
||||
const uint8_t payload_type,
|
||||
const uint32_t timestamp,
|
||||
const uint8_t* payload_data,
|
||||
const uint16_t payload_size,
|
||||
const size_t payload_size,
|
||||
const RTPFragmentationHeader* fragmentation) OVERRIDE;
|
||||
|
||||
uint16_t payload_size();
|
||||
@ -78,11 +80,6 @@ class TestStereo : public ACMTest {
|
||||
void OpenOutFile(int16_t test_number);
|
||||
void DisplaySendReceiveCodec();
|
||||
|
||||
int32_t SendData(const FrameType frame_type, const uint8_t payload_type,
|
||||
const uint32_t timestamp, const uint8_t* payload_data,
|
||||
const uint16_t payload_size,
|
||||
const RTPFragmentationHeader* fragmentation);
|
||||
|
||||
int test_mode_;
|
||||
|
||||
scoped_ptr<AudioCodingModule> acm_a_;
|
||||
@ -100,17 +97,24 @@ class TestStereo : public ACMTest {
|
||||
char* send_codec_name_;
|
||||
|
||||
// Payload types for stereo codecs and CNG
|
||||
#ifdef WEBRTC_CODEC_G722
|
||||
int g722_pltype_;
|
||||
#endif
|
||||
#ifdef WEBRTC_CODEC_PCM16
|
||||
int l16_8khz_pltype_;
|
||||
int l16_16khz_pltype_;
|
||||
int l16_32khz_pltype_;
|
||||
#endif
|
||||
#ifdef PCMA_AND_PCMU
|
||||
int pcma_pltype_;
|
||||
int pcmu_pltype_;
|
||||
#endif
|
||||
#ifdef WEBRTC_CODEC_CELT
|
||||
int celt_pltype_;
|
||||
#endif
|
||||
#ifdef WEBRTC_CODEC_OPUS
|
||||
int opus_pltype_;
|
||||
int cn_8khz_pltype_;
|
||||
int cn_16khz_pltype_;
|
||||
int cn_32khz_pltype_;
|
||||
#endif
|
||||
};
|
||||
|
||||
} // namespace webrtc
|
||||
|
||||
Reference in New Issue
Block a user