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/acm2/acm_receive_test.cc b/webrtc/modules/audio_coding/main/acm2/acm_receive_test.cc
index 7e41328..08ece69 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receive_test.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receive_test.cc
@@ -113,10 +113,9 @@
header.header = packet->header();
header.frameType = kAudioFrameSpeech;
memset(&header.type.Audio, 0, sizeof(RTPAudioHeader));
- EXPECT_TRUE(
- acm_->InsertPacket(packet->payload(),
- static_cast<int32_t>(packet->payload_length_bytes()),
- header))
+ EXPECT_TRUE(acm_->InsertPacket(packet->payload(),
+ packet->payload_length_bytes(),
+ header))
<< "Failure when inserting packet:" << std::endl
<< " PT = " << static_cast<int>(header.header.payloadType) << std::endl
<< " TS = " << header.header.timestamp << std::endl
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc b/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
index 0744754..bbe5a16 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
@@ -261,7 +261,7 @@
int AcmReceiver::InsertPacket(const WebRtcRTPHeader& rtp_header,
const uint8_t* incoming_payload,
- int length_payload) {
+ size_t length_payload) {
uint32_t receive_timestamp = 0;
InitialDelayManager::PacketType packet_type =
InitialDelayManager::kUndefinedPacket;
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver.h b/webrtc/modules/audio_coding/main/acm2/acm_receiver.h
index 6d31b9a..057cb5a 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver.h
@@ -67,7 +67,7 @@
//
int InsertPacket(const WebRtcRTPHeader& rtp_header,
const uint8_t* incoming_payload,
- int length_payload);
+ size_t length_payload);
//
// Asks NetEq for 10 milliseconds of decoded audio.
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc b/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc
index 9cfef3a..ff43899 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc
@@ -115,12 +115,12 @@
}
}
- virtual int SendData(
+ virtual int32_t SendData(
FrameType frame_type,
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
- uint16_t payload_len_bytes,
+ size_t payload_len_bytes,
const RTPFragmentationHeader* fragmentation) OVERRIDE {
if (frame_type == kFrameEmpty)
return 0;
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc b/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc
index ef890ec..8c37c96 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc
@@ -124,7 +124,7 @@
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
- uint16_t payload_len_bytes,
+ size_t payload_len_bytes,
const RTPFragmentationHeader* fragmentation) OVERRIDE {
if (frame_type == kFrameEmpty)
return 0;
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_send_test.cc b/webrtc/modules/audio_coding/main/acm2/acm_send_test.cc
index ec3c254..d2ecb16 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_send_test.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_send_test.cc
@@ -94,7 +94,7 @@
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
- uint16_t payload_len_bytes,
+ size_t payload_len_bytes,
const RTPFragmentationHeader* fragmentation) {
// Store the packet locally.
frame_type_ = frame_type;
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_send_test.h b/webrtc/modules/audio_coding/main/acm2/acm_send_test.h
index 8bc0cde..ac20cc7 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_send_test.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_send_test.h
@@ -49,7 +49,7 @@
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
- uint16_t payload_len_bytes,
+ size_t payload_len_bytes,
const RTPFragmentationHeader* fragmentation) OVERRIDE;
private:
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc b/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc
index 2f5178e..42dc628 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc
@@ -98,7 +98,7 @@
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
- uint16_t payload_len_bytes,
+ size_t payload_len_bytes,
const RTPFragmentationHeader* fragmentation) {
// Store the packet locally.
frame_type_ = frame_type;
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h b/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h
index ff229a0..e990284 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h
@@ -51,7 +51,7 @@
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
- uint16_t payload_len_bytes,
+ size_t payload_len_bytes,
const RTPFragmentationHeader* fragmentation) OVERRIDE;
private:
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
index bee1f66..458e5c8 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
@@ -314,7 +314,7 @@
int AudioCodingModuleImpl::ProcessDualStream() {
uint8_t stream[kMaxNumFragmentationVectors * MAX_PAYLOAD_SIZE_BYTE];
uint32_t current_timestamp;
- int16_t length_bytes = 0;
+ size_t length_bytes = 0;
RTPFragmentationHeader my_fragmentation;
uint8_t my_red_payload_type;
@@ -336,8 +336,7 @@
// Nothing to send.
return 0;
}
- int len_bytes_previous_secondary = static_cast<int>(
- fragmentation_.fragmentationLength[2]);
+ size_t len_bytes_previous_secondary = fragmentation_.fragmentationLength[2];
assert(len_bytes_previous_secondary <= MAX_PAYLOAD_SIZE_BYTE);
bool has_previous_payload = len_bytes_previous_secondary > 0;
@@ -1689,13 +1688,8 @@
// Incoming packet from network parsed and ready for decode.
int AudioCodingModuleImpl::IncomingPacket(const uint8_t* incoming_payload,
- const int payload_length,
+ const size_t payload_length,
const WebRtcRTPHeader& rtp_header) {
- if (payload_length < 0) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
- "IncomingPacket() Error, payload-length cannot be negative");
- return -1;
- }
int last_audio_pltype = receiver_.last_audio_payload_type();
if (receiver_.InsertPacket(rtp_header, incoming_payload, payload_length) <
0) {
@@ -1797,16 +1791,9 @@
// TODO(tlegrand): Modify this function to work for stereo, and add tests.
int AudioCodingModuleImpl::IncomingPayload(const uint8_t* incoming_payload,
- int payload_length,
+ size_t payload_length,
uint8_t payload_type,
uint32_t timestamp) {
- if (payload_length < 0) {
- // Log error in trace file.
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
- "IncomingPacket() Error, payload-length cannot be negative");
- return -1;
- }
-
// We are not acquiring any lock when interacting with |aux_rtp_header_| no
// other method uses this member variable.
if (aux_rtp_header_ == NULL) {
@@ -1960,7 +1947,7 @@
}
void AudioCodingModuleImpl::ResetFragmentation(int vector_size) {
- for (int n = 0; n < kMaxNumFragmentationVectors; n++) {
+ for (size_t n = 0; n < kMaxNumFragmentationVectors; n++) {
fragmentation_.fragmentationOffset[n] = n * MAX_PAYLOAD_SIZE_BYTE;
}
memset(fragmentation_.fragmentationLength, 0, kMaxNumFragmentationVectors *
@@ -2116,14 +2103,14 @@
}
bool AudioCodingImpl::InsertPacket(const uint8_t* incoming_payload,
- int32_t payload_len_bytes,
+ size_t payload_len_bytes,
const WebRtcRTPHeader& rtp_info) {
return acm_old_->IncomingPacket(
incoming_payload, payload_len_bytes, rtp_info) == 0;
}
bool AudioCodingImpl::InsertPayload(const uint8_t* incoming_payload,
- int32_t payload_len_byte,
+ size_t payload_len_byte,
uint8_t payload_type,
uint32_t timestamp) {
FATAL() << "Not implemented yet.";
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
index b8d128f..949ce33 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
@@ -156,13 +156,13 @@
// Incoming packet from network parsed and ready for decode.
virtual int IncomingPacket(const uint8_t* incoming_payload,
- int payload_length,
+ const size_t payload_length,
const WebRtcRTPHeader& rtp_info) OVERRIDE;
// Incoming payloads, without rtp-info, the rtp-info will be created in ACM.
// One usage for this API is when pre-encoded files are pushed in ACM.
virtual int IncomingPayload(const uint8_t* incoming_payload,
- int payload_length,
+ const size_t payload_length,
uint8_t payload_type,
uint32_t timestamp) OVERRIDE;
@@ -423,11 +423,11 @@
uint8_t payload_type) OVERRIDE;
virtual bool InsertPacket(const uint8_t* incoming_payload,
- int32_t payload_len_bytes,
+ size_t payload_len_bytes,
const WebRtcRTPHeader& rtp_info) OVERRIDE;
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) OVERRIDE;
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc
index 828b772..b64c74d 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc
@@ -42,7 +42,7 @@
const int kNumSamples10ms = kSampleRateHz / 100;
const int kFrameSizeMs = 10; // Multiple of 10.
const int kFrameSizeSamples = kFrameSizeMs / 10 * kNumSamples10ms;
-const int kPayloadSizeBytes = kFrameSizeSamples * sizeof(int16_t);
+const size_t kPayloadSizeBytes = kFrameSizeSamples * sizeof(int16_t);
const uint8_t kPayloadType = 111;
class RtpUtility {
@@ -87,7 +87,7 @@
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
- uint16_t payload_len_bytes,
+ size_t payload_len_bytes,
const RTPFragmentationHeader* fragmentation) OVERRIDE {
CriticalSectionScoped lock(crit_sect_.get());
++num_calls_;
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
index d9ed32c..e887317 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
@@ -87,7 +87,7 @@
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
- uint16_t payload_len_bytes,
+ size_t payload_len_bytes,
const RTPFragmentationHeader* fragmentation) OVERRIDE {
CriticalSectionScoped lock(crit_sect_.get());
++num_calls_;
diff --git a/webrtc/modules/audio_coding/main/interface/audio_coding_module.h b/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
index 8d73285..8dd5cdc 100644
--- a/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
+++ b/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
@@ -36,13 +36,12 @@
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 @@
// 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 @@
// 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 @@
// |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;
diff --git a/webrtc/modules/audio_coding/main/test/Channel.cc b/webrtc/modules/audio_coding/main/test/Channel.cc
index 20ecf3a..aa9e6cd 100644
--- a/webrtc/modules/audio_coding/main/test/Channel.cc
+++ b/webrtc/modules/audio_coding/main/test/Channel.cc
@@ -13,18 +13,21 @@
#include <assert.h>
#include <iostream>
+#include "webrtc/base/format_macros.h"
#include "webrtc/system_wrappers/interface/tick_util.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
namespace webrtc {
-int32_t Channel::SendData(const FrameType frameType, const uint8_t payloadType,
- const uint32_t timeStamp, const uint8_t* payloadData,
- const uint16_t payloadSize,
+int32_t Channel::SendData(FrameType frameType,
+ uint8_t payloadType,
+ uint32_t timeStamp,
+ const uint8_t* payloadData,
+ size_t payloadSize,
const RTPFragmentationHeader* fragmentation) {
WebRtcRTPHeader rtpInfo;
int32_t status;
- uint16_t payloadDataSize = payloadSize;
+ size_t payloadDataSize = payloadSize;
rtpInfo.header.markerBit = false;
rtpInfo.header.ssrc = 0;
@@ -52,8 +55,8 @@
(fragmentation->fragmentationVectorSize == 2)) {
// only 0x80 if we have multiple blocks
_payloadData[0] = 0x80 + fragmentation->fragmentationPlType[1];
- uint32_t REDheader = (((uint32_t) fragmentation->fragmentationTimeDiff[1])
- << 10) + fragmentation->fragmentationLength[1];
+ size_t REDheader = (fragmentation->fragmentationTimeDiff[1] << 10) +
+ fragmentation->fragmentationLength[1];
_payloadData[1] = uint8_t((REDheader >> 16) & 0x000000FF);
_payloadData[2] = uint8_t((REDheader >> 8) & 0x000000FF);
_payloadData[3] = uint8_t(REDheader & 0x000000FF);
@@ -72,7 +75,7 @@
// single block (newest one)
memcpy(_payloadData, payloadData + fragmentation->fragmentationOffset[0],
fragmentation->fragmentationLength[0]);
- payloadDataSize = uint16_t(fragmentation->fragmentationLength[0]);
+ payloadDataSize = fragmentation->fragmentationLength[0];
rtpInfo.header.payloadType = fragmentation->fragmentationPlType[0];
}
} else {
@@ -121,7 +124,7 @@
}
// TODO(turajs): rewite this method.
-void Channel::CalcStatistics(WebRtcRTPHeader& rtpInfo, uint16_t payloadSize) {
+void Channel::CalcStatistics(WebRtcRTPHeader& rtpInfo, size_t payloadSize) {
int n;
if ((rtpInfo.header.payloadType != _lastPayloadType)
&& (_lastPayloadType != -1)) {
@@ -371,7 +374,7 @@
payloadStats.frameSizeStats[k].frameSizeSample);
printf("Average Rate.................. %.0f bits/sec\n",
payloadStats.frameSizeStats[k].rateBitPerSec);
- printf("Maximum Payload-Size.......... %d Bytes\n",
+ printf("Maximum Payload-Size.......... %" PRIuS " Bytes\n",
payloadStats.frameSizeStats[k].maxPayloadLen);
printf(
"Maximum Instantaneous Rate.... %.0f bits/sec\n",
diff --git a/webrtc/modules/audio_coding/main/test/Channel.h b/webrtc/modules/audio_coding/main/test/Channel.h
index cdb99c0..4ab32b9 100644
--- a/webrtc/modules/audio_coding/main/test/Channel.h
+++ b/webrtc/modules/audio_coding/main/test/Channel.h
@@ -27,7 +27,7 @@
// TODO(turajs): Write constructor for this structure.
struct ACMTestFrameSizeStats {
uint16_t frameSizeSample;
- int16_t maxPayloadLen;
+ size_t maxPayloadLen;
uint32_t numPackets;
uint64_t totalPayloadLenByte;
uint64_t totalEncodedSamples;
@@ -39,7 +39,7 @@
struct ACMTestPayloadStats {
bool newPacket;
int16_t payloadType;
- int16_t lastPayloadLenByte;
+ size_t lastPayloadLenByte;
uint32_t lastTimestamp;
ACMTestFrameSizeStats frameSizeStats[MAX_NUM_FRAMESIZES];
};
@@ -51,9 +51,11 @@
~Channel();
virtual int32_t SendData(
- const FrameType frameType, const uint8_t payloadType,
- const uint32_t timeStamp, const uint8_t* payloadData,
- const uint16_t payloadSize,
+ FrameType frameType,
+ uint8_t payloadType,
+ uint32_t timeStamp,
+ const uint8_t* payloadData,
+ size_t payloadSize,
const RTPFragmentationHeader* fragmentation) OVERRIDE;
void RegisterReceiverACM(AudioCodingModule *acm);
@@ -93,7 +95,7 @@
}
private:
- void CalcStatistics(WebRtcRTPHeader& rtpInfo, uint16_t payloadSize);
+ void CalcStatistics(WebRtcRTPHeader& rtpInfo, size_t payloadSize);
AudioCodingModule* _receiverACM;
uint16_t _seqNo;
diff --git a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc b/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
index 66fd220..27f5500 100644
--- a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
+++ b/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
@@ -37,7 +37,7 @@
int32_t TestPacketization::SendData(
const FrameType /* frameType */, const uint8_t payloadType,
const uint32_t timeStamp, const uint8_t* payloadData,
- const uint16_t payloadSize,
+ const size_t payloadSize,
const RTPFragmentationHeader* /* fragmentation */) {
_rtpStream->Write(payloadType, timeStamp, _seqNo++, payloadData, payloadSize,
_frequency);
diff --git a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h b/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h
index f6b5553..4ee4fa2 100644
--- a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h
+++ b/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h
@@ -30,9 +30,11 @@
TestPacketization(RTPStream *rtpStream, uint16_t frequency);
~TestPacketization();
virtual int32_t SendData(
- const FrameType frameType, const uint8_t payloadType,
- const uint32_t timeStamp, const uint8_t* payloadData,
- const uint16_t payloadSize,
+ const FrameType frameType,
+ const uint8_t payloadType,
+ const uint32_t timeStamp,
+ const uint8_t* payloadData,
+ const size_t payloadSize,
const RTPFragmentationHeader* fragmentation) OVERRIDE;
private:
@@ -92,8 +94,8 @@
uint8_t _incomingPayload[MAX_INCOMING_PAYLOAD];
RTPStream* _rtpStream;
WebRtcRTPHeader _rtpInfo;
- uint16_t _realPayloadSizeBytes;
- uint16_t _payloadSizeBytes;
+ size_t _realPayloadSizeBytes;
+ size_t _payloadSizeBytes;
uint32_t _nextTime;
};
diff --git a/webrtc/modules/audio_coding/main/test/RTPFile.cc b/webrtc/modules/audio_coding/main/test/RTPFile.cc
index b7f587b..6f0c74e 100644
--- a/webrtc/modules/audio_coding/main/test/RTPFile.cc
+++ b/webrtc/modules/audio_coding/main/test/RTPFile.cc
@@ -11,6 +11,7 @@
#include "RTPFile.h"
#include <stdlib.h>
+#include <limits>
#ifdef WIN32
# include <Winsock2.h>
@@ -60,7 +61,7 @@
}
RTPPacket::RTPPacket(uint8_t payloadType, uint32_t timeStamp, int16_t seqNo,
- const uint8_t* payloadData, uint16_t payloadSize,
+ const uint8_t* payloadData, size_t payloadSize,
uint32_t frequency)
: payloadType(payloadType),
timeStamp(timeStamp),
@@ -87,7 +88,7 @@
void RTPBuffer::Write(const uint8_t payloadType, const uint32_t timeStamp,
const int16_t seqNo, const uint8_t* payloadData,
- const uint16_t payloadSize, uint32_t frequency) {
+ const size_t payloadSize, uint32_t frequency) {
RTPPacket *packet = new RTPPacket(payloadType, timeStamp, seqNo, payloadData,
payloadSize, frequency);
_queueRWLock->AcquireLockExclusive();
@@ -95,8 +96,8 @@
_queueRWLock->ReleaseLockExclusive();
}
-uint16_t RTPBuffer::Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
- uint16_t payloadSize, uint32_t* offset) {
+size_t RTPBuffer::Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
+ size_t payloadSize, uint32_t* offset) {
_queueRWLock->AcquireLockShared();
RTPPacket *packet = _rtpQueue.front();
_rtpQueue.pop();
@@ -143,21 +144,11 @@
fprintf(_rtpFile, "#!RTPencode%s\n", "1.0");
uint32_t dummy_variable = 0;
// should be converted to network endian format, but does not matter when 0
- if (fwrite(&dummy_variable, 4, 1, _rtpFile) != 1) {
- return;
- }
- if (fwrite(&dummy_variable, 4, 1, _rtpFile) != 1) {
- return;
- }
- if (fwrite(&dummy_variable, 4, 1, _rtpFile) != 1) {
- return;
- }
- if (fwrite(&dummy_variable, 2, 1, _rtpFile) != 1) {
- return;
- }
- if (fwrite(&dummy_variable, 2, 1, _rtpFile) != 1) {
- return;
- }
+ EXPECT_EQ(1u, fwrite(&dummy_variable, 4, 1, _rtpFile));
+ EXPECT_EQ(1u, fwrite(&dummy_variable, 4, 1, _rtpFile));
+ EXPECT_EQ(1u, fwrite(&dummy_variable, 4, 1, _rtpFile));
+ EXPECT_EQ(1u, fwrite(&dummy_variable, 2, 1, _rtpFile));
+ EXPECT_EQ(1u, fwrite(&dummy_variable, 2, 1, _rtpFile));
fflush(_rtpFile);
}
@@ -180,35 +171,26 @@
void RTPFile::Write(const uint8_t payloadType, const uint32_t timeStamp,
const int16_t seqNo, const uint8_t* payloadData,
- const uint16_t payloadSize, uint32_t frequency) {
+ const size_t payloadSize, uint32_t frequency) {
/* write RTP packet to file */
uint8_t rtpHeader[12];
MakeRTPheader(rtpHeader, payloadType, seqNo, timeStamp, 0);
- uint16_t lengthBytes = htons(12 + payloadSize + 8);
- uint16_t plen = htons(12 + payloadSize);
+ ASSERT_LE(12 + payloadSize + 8, std::numeric_limits<u_short>::max());
+ uint16_t lengthBytes = htons(static_cast<u_short>(12 + payloadSize + 8));
+ uint16_t plen = htons(static_cast<u_short>(12 + payloadSize));
uint32_t offsetMs;
offsetMs = (timeStamp / (frequency / 1000));
offsetMs = htonl(offsetMs);
- if (fwrite(&lengthBytes, 2, 1, _rtpFile) != 1) {
- return;
- }
- if (fwrite(&plen, 2, 1, _rtpFile) != 1) {
- return;
- }
- if (fwrite(&offsetMs, 4, 1, _rtpFile) != 1) {
- return;
- }
- if (fwrite(rtpHeader, 12, 1, _rtpFile) != 1) {
- return;
- }
- if (fwrite(payloadData, 1, payloadSize, _rtpFile) != payloadSize) {
- return;
- }
+ EXPECT_EQ(1u, fwrite(&lengthBytes, 2, 1, _rtpFile));
+ EXPECT_EQ(1u, fwrite(&plen, 2, 1, _rtpFile));
+ EXPECT_EQ(1u, fwrite(&offsetMs, 4, 1, _rtpFile));
+ EXPECT_EQ(1u, fwrite(&rtpHeader, 12, 1, _rtpFile));
+ EXPECT_EQ(payloadSize, fwrite(payloadData, 1, payloadSize, _rtpFile));
}
-uint16_t RTPFile::Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
- uint16_t payloadSize, uint32_t* offset) {
+size_t RTPFile::Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
+ size_t payloadSize, uint32_t* offset) {
uint16_t lengthBytes;
uint16_t plen;
uint8_t rtpHeader[12];
@@ -237,7 +219,7 @@
if (lengthBytes < 20) {
return 0;
}
- if (payloadSize < (lengthBytes - 20)) {
+ if (payloadSize < static_cast<size_t>((lengthBytes - 20))) {
return 0;
}
lengthBytes -= 20;
diff --git a/webrtc/modules/audio_coding/main/test/RTPFile.h b/webrtc/modules/audio_coding/main/test/RTPFile.h
index 460553b..9a2d43a 100644
--- a/webrtc/modules/audio_coding/main/test/RTPFile.h
+++ b/webrtc/modules/audio_coding/main/test/RTPFile.h
@@ -28,12 +28,12 @@
virtual void Write(const uint8_t payloadType, const uint32_t timeStamp,
const int16_t seqNo, const uint8_t* payloadData,
- const uint16_t payloadSize, uint32_t frequency) = 0;
+ const size_t payloadSize, uint32_t frequency) = 0;
// Returns the packet's payload size. Zero should be treated as an
// end-of-stream (in the case that EndOfFile() is true) or an error.
- virtual uint16_t Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
- uint16_t payloadSize, uint32_t* offset) = 0;
+ virtual size_t Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
+ size_t payloadSize, uint32_t* offset) = 0;
virtual bool EndOfFile() const = 0;
protected:
@@ -46,7 +46,7 @@
class RTPPacket {
public:
RTPPacket(uint8_t payloadType, uint32_t timeStamp, int16_t seqNo,
- const uint8_t* payloadData, uint16_t payloadSize,
+ const uint8_t* payloadData, size_t payloadSize,
uint32_t frequency);
~RTPPacket();
@@ -55,7 +55,7 @@
uint32_t timeStamp;
int16_t seqNo;
uint8_t* payloadData;
- uint16_t payloadSize;
+ size_t payloadSize;
uint32_t frequency;
};
@@ -67,10 +67,10 @@
virtual void Write(const uint8_t payloadType, const uint32_t timeStamp,
const int16_t seqNo, const uint8_t* payloadData,
- const uint16_t payloadSize, uint32_t frequency) OVERRIDE;
+ const size_t payloadSize, uint32_t frequency) OVERRIDE;
- virtual uint16_t Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
- uint16_t payloadSize, uint32_t* offset) OVERRIDE;
+ virtual size_t Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
+ size_t payloadSize, uint32_t* offset) OVERRIDE;
virtual bool EndOfFile() const OVERRIDE;
@@ -99,10 +99,10 @@
virtual void Write(const uint8_t payloadType, const uint32_t timeStamp,
const int16_t seqNo, const uint8_t* payloadData,
- const uint16_t payloadSize, uint32_t frequency) OVERRIDE;
+ const size_t payloadSize, uint32_t frequency) OVERRIDE;
- virtual uint16_t Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
- uint16_t payloadSize, uint32_t* offset) OVERRIDE;
+ virtual size_t Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
+ size_t payloadSize, uint32_t* offset) OVERRIDE;
virtual bool EndOfFile() const OVERRIDE {
return _rtpEOF;
diff --git a/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc b/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc
index 9b7667b..cd5a94e 100644
--- a/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc
+++ b/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc
@@ -10,7 +10,8 @@
#include "webrtc/modules/audio_coding/main/test/TestAllCodecs.h"
-#include <stdio.h>
+#include <cstdio>
+#include <limits>
#include <string>
#include "testing/gtest/include/gtest/gtest.h"
@@ -32,6 +33,10 @@
// The test loops through all available mono codecs, encode at "a" sends over
// the channel, and decodes at "b".
+namespace {
+const size_t kVariableSize = std::numeric_limits<size_t>::max();
+}
+
namespace webrtc {
// Class for simulating packet handling.
@@ -54,7 +59,7 @@
int32_t TestPack::SendData(FrameType frame_type, uint8_t payload_type,
uint32_t timestamp, const uint8_t* payload_data,
- uint16_t payload_size,
+ size_t payload_size,
const RTPFragmentationHeader* fragmentation) {
WebRtcRTPHeader rtp_info;
int32_t status;
@@ -87,7 +92,7 @@
return status;
}
-uint16_t TestPack::payload_size() {
+size_t TestPack::payload_size() {
return payload_size_;
}
@@ -459,13 +464,13 @@
test_count_++;
OpenOutFile(test_count_);
char codec_isac[] = "ISAC";
- RegisterSendCodec('A', codec_isac, 16000, -1, 480, -1);
+ RegisterSendCodec('A', codec_isac, 16000, -1, 480, kVariableSize);
Run(channel_a_to_b_);
- RegisterSendCodec('A', codec_isac, 16000, -1, 960, -1);
+ RegisterSendCodec('A', codec_isac, 16000, -1, 960, kVariableSize);
Run(channel_a_to_b_);
- RegisterSendCodec('A', codec_isac, 16000, 15000, 480, -1);
+ RegisterSendCodec('A', codec_isac, 16000, 15000, 480, kVariableSize);
Run(channel_a_to_b_);
- RegisterSendCodec('A', codec_isac, 16000, 32000, 960, -1);
+ RegisterSendCodec('A', codec_isac, 16000, 32000, 960, kVariableSize);
Run(channel_a_to_b_);
outfile_b_.Close();
#endif
@@ -475,13 +480,13 @@
}
test_count_++;
OpenOutFile(test_count_);
- RegisterSendCodec('A', codec_isac, 32000, -1, 960, -1);
+ RegisterSendCodec('A', codec_isac, 32000, -1, 960, kVariableSize);
Run(channel_a_to_b_);
- RegisterSendCodec('A', codec_isac, 32000, 56000, 960, -1);
+ RegisterSendCodec('A', codec_isac, 32000, 56000, 960, kVariableSize);
Run(channel_a_to_b_);
- RegisterSendCodec('A', codec_isac, 32000, 37000, 960, -1);
+ RegisterSendCodec('A', codec_isac, 32000, 37000, 960, kVariableSize);
Run(channel_a_to_b_);
- RegisterSendCodec('A', codec_isac, 32000, 32000, 960, -1);
+ RegisterSendCodec('A', codec_isac, 32000, 32000, 960, kVariableSize);
Run(channel_a_to_b_);
outfile_b_.Close();
#endif
@@ -611,19 +616,19 @@
test_count_++;
OpenOutFile(test_count_);
char codec_opus[] = "OPUS";
- RegisterSendCodec('A', codec_opus, 48000, 6000, 480, -1);
+ RegisterSendCodec('A', codec_opus, 48000, 6000, 480, kVariableSize);
Run(channel_a_to_b_);
- RegisterSendCodec('A', codec_opus, 48000, 20000, 480*2, -1);
+ RegisterSendCodec('A', codec_opus, 48000, 20000, 480*2, kVariableSize);
Run(channel_a_to_b_);
- RegisterSendCodec('A', codec_opus, 48000, 32000, 480*4, -1);
+ RegisterSendCodec('A', codec_opus, 48000, 32000, 480*4, kVariableSize);
Run(channel_a_to_b_);
- RegisterSendCodec('A', codec_opus, 48000, 48000, 480, -1);
+ RegisterSendCodec('A', codec_opus, 48000, 48000, 480, kVariableSize);
Run(channel_a_to_b_);
- RegisterSendCodec('A', codec_opus, 48000, 64000, 480*4, -1);
+ RegisterSendCodec('A', codec_opus, 48000, 64000, 480*4, kVariableSize);
Run(channel_a_to_b_);
- RegisterSendCodec('A', codec_opus, 48000, 96000, 480*6, -1);
+ RegisterSendCodec('A', codec_opus, 48000, 96000, 480*6, kVariableSize);
Run(channel_a_to_b_);
- RegisterSendCodec('A', codec_opus, 48000, 500000, 480*2, -1);
+ RegisterSendCodec('A', codec_opus, 48000, 500000, 480*2, kVariableSize);
Run(channel_a_to_b_);
outfile_b_.Close();
#endif
@@ -686,10 +691,11 @@
// packet_size - packet size in samples
// extra_byte - if extra bytes needed compared to the bitrate
// used when registering, can be an internal header
-// set to -1 if the codec is a variable rate codec
+// set to kVariableSize if the codec is a variable
+// rate codec
void TestAllCodecs::RegisterSendCodec(char side, char* codec_name,
int32_t sampling_freq_hz, int rate,
- int packet_size, int extra_byte) {
+ int packet_size, size_t extra_byte) {
if (test_mode_ != 0) {
// Print out codec and settings.
printf("codec: %s Freq: %d Rate: %d PackSize: %d\n", codec_name,
@@ -711,14 +717,14 @@
// Store the expected packet size in bytes, used to validate the received
// packet. If variable rate codec (extra_byte == -1), set to -1.
- if (extra_byte != -1) {
+ if (extra_byte != kVariableSize) {
// Add 0.875 to always round up to a whole byte
- packet_size_bytes_ = static_cast<int>(static_cast<float>(packet_size
- * rate) / static_cast<float>(sampling_freq_hz * 8) + 0.875)
- + extra_byte;
+ packet_size_bytes_ = static_cast<size_t>(
+ static_cast<float>(packet_size * rate) /
+ static_cast<float>(sampling_freq_hz * 8) + 0.875) + extra_byte;
} else {
// Packets will have a variable size.
- packet_size_bytes_ = -1;
+ packet_size_bytes_ = kVariableSize;
}
// Set pointer to the ACM where to register the codec.
@@ -751,7 +757,7 @@
AudioFrame audio_frame;
int32_t out_freq_hz = outfile_b_.SamplingFrequency();
- uint16_t receive_size;
+ size_t receive_size;
uint32_t timestamp_diff;
channel->reset_payload_size();
int error_count = 0;
@@ -768,8 +774,8 @@
// Verify that the received packet size matches the settings.
receive_size = channel->payload_size();
if (receive_size) {
- if ((static_cast<int>(receive_size) != packet_size_bytes_) &&
- (packet_size_bytes_ > -1)) {
+ if ((receive_size != packet_size_bytes_) &&
+ (packet_size_bytes_ != kVariableSize)) {
error_count++;
}
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
diff --git a/webrtc/modules/audio_coding/main/test/TestStereo.cc b/webrtc/modules/audio_coding/main/test/TestStereo.cc
index 9c22548..86a75e5 100644
--- a/webrtc/modules/audio_coding/main/test/TestStereo.cc
+++ b/webrtc/modules/audio_coding/main/test/TestStereo.cc
@@ -48,7 +48,7 @@
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) {
WebRtcRTPHeader rtp_info;
int32_t status = 0;
@@ -114,18 +114,26 @@
test_cntr_(0),
pack_size_samp_(0),
pack_size_bytes_(0),
- counter_(0),
- g722_pltype_(0),
- l16_8khz_pltype_(-1),
- l16_16khz_pltype_(-1),
- l16_32khz_pltype_(-1),
- pcma_pltype_(-1),
- pcmu_pltype_(-1),
- celt_pltype_(-1),
- opus_pltype_(-1),
- cn_8khz_pltype_(-1),
- cn_16khz_pltype_(-1),
- cn_32khz_pltype_(-1) {
+ counter_(0)
+#ifdef WEBRTC_CODEC_G722
+ , g722_pltype_(0)
+#endif
+#ifdef WEBRTC_CODEC_PCM16
+ , l16_8khz_pltype_(-1)
+ , l16_16khz_pltype_(-1)
+ , l16_32khz_pltype_(-1)
+#endif
+#ifdef PCMA_AND_PCMU
+ , pcma_pltype_(-1)
+ , pcmu_pltype_(-1)
+#endif
+#ifdef WEBRTC_CODEC_CELT
+ , celt_pltype_(-1)
+#endif
+#ifdef WEBRTC_CODEC_OPUS
+ , opus_pltype_(-1)
+#endif
+ {
// test_mode = 0 for silent test (auto test)
test_mode_ = test_mode;
}
@@ -302,7 +310,6 @@
Run(channel_a2b_, audio_channels, codec_channels);
out_file_.Close();
#endif
-#define PCMA_AND_PCMU
#ifdef PCMA_AND_PCMU
if (test_mode_ != 0) {
printf("===========================================================\n");
diff --git a/webrtc/modules/audio_coding/main/test/TestStereo.h b/webrtc/modules/audio_coding/main/test/TestStereo.h
index 8aefa7f..0eb0e52 100644
--- a/webrtc/modules/audio_coding/main/test/TestStereo.h
+++ b/webrtc/modules/audio_coding/main/test/TestStereo.h
@@ -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 @@
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 @@
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 @@
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
diff --git a/webrtc/modules/audio_coding/main/test/dual_stream_unittest.cc b/webrtc/modules/audio_coding/main/test/dual_stream_unittest.cc
index 7cd2466..9b960af 100644
--- a/webrtc/modules/audio_coding/main/test/dual_stream_unittest.cc
+++ b/webrtc/modules/audio_coding/main/test/dual_stream_unittest.cc
@@ -36,9 +36,11 @@
void ApiTest();
virtual int32_t SendData(
- FrameType frameType, uint8_t payload_type,
- uint32_t timestamp, const uint8_t* payload_data,
- uint16_t payload_size,
+ FrameType frameType,
+ uint8_t payload_type,
+ uint32_t timestamp,
+ const uint8_t* payload_data,
+ size_t payload_size,
const RTPFragmentationHeader* fragmentation) OVERRIDE;
void Perform(bool start_in_sync, int num_channels_input);
@@ -49,9 +51,9 @@
void PopulateCodecInstances(int frame_size_primary_ms,
int num_channels_primary, int sampling_rate);
- void Validate(bool start_in_sync, int tolerance);
+ void Validate(bool start_in_sync, size_t tolerance);
bool EqualTimestamp(int stream, int position);
- int EqualPayloadLength(int stream, int position);
+ size_t EqualPayloadLength(int stream, int position);
bool EqualPayloadData(int stream, int position);
static const int kMaxNumStoredPayloads = 2;
@@ -77,8 +79,8 @@
uint32_t timestamp_ref_[kMaxNumStreams][kMaxNumStoredPayloads];
uint32_t timestamp_dual_[kMaxNumStreams][kMaxNumStoredPayloads];
- int payload_len_ref_[kMaxNumStreams][kMaxNumStoredPayloads];
- int payload_len_dual_[kMaxNumStreams][kMaxNumStoredPayloads];
+ size_t payload_len_ref_[kMaxNumStreams][kMaxNumStoredPayloads];
+ size_t payload_len_dual_[kMaxNumStreams][kMaxNumStoredPayloads];
uint8_t payload_data_ref_[kMaxNumStreams][MAX_PAYLOAD_SIZE_BYTE
* kMaxNumStoredPayloads];
@@ -174,7 +176,7 @@
pcm_file.ReadStereo(num_channels_input == 2);
AudioFrame audio_frame;
- int tolerance = 0;
+ size_t tolerance = 0;
if (num_channels_input == 2 && primary_encoder_.channels == 2
&& secondary_encoder_.channels == 1) {
tolerance = 12;
@@ -253,10 +255,10 @@
return true;
}
-int DualStreamTest::EqualPayloadLength(int stream_index, int position) {
- return abs(
- payload_len_dual_[stream_index][position]
- - payload_len_ref_[stream_index][position]);
+size_t DualStreamTest::EqualPayloadLength(int stream_index, int position) {
+ size_t dual = payload_len_dual_[stream_index][position];
+ size_t ref = payload_len_ref_[stream_index][position];
+ return (dual > ref) ? (dual - ref) : (ref - dual);
}
bool DualStreamTest::EqualPayloadData(int stream_index, int position) {
@@ -264,7 +266,7 @@
payload_len_dual_[stream_index][position]
== payload_len_ref_[stream_index][position]);
int offset = position * MAX_PAYLOAD_SIZE_BYTE;
- for (int n = 0; n < payload_len_dual_[stream_index][position]; n++) {
+ for (size_t n = 0; n < payload_len_dual_[stream_index][position]; n++) {
if (payload_data_dual_[stream_index][offset + n]
!= payload_data_ref_[stream_index][offset + n]) {
return false;
@@ -273,9 +275,9 @@
return true;
}
-void DualStreamTest::Validate(bool start_in_sync, int tolerance) {
+void DualStreamTest::Validate(bool start_in_sync, size_t tolerance) {
for (int stream_index = 0; stream_index < kMaxNumStreams; stream_index++) {
- int my_tolerance = stream_index == kPrimary ? 0 : tolerance;
+ size_t my_tolerance = stream_index == kPrimary ? 0 : tolerance;
for (int position = 0; position < kMaxNumStoredPayloads; position++) {
if (payload_ref_is_stored_[stream_index][position] == 1
&& payload_dual_is_stored_[stream_index][position] == 1) {
@@ -296,7 +298,7 @@
int32_t DualStreamTest::SendData(FrameType frameType, uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
- uint16_t payload_size,
+ size_t payload_size,
const RTPFragmentationHeader* fragmentation) {
int position;
int stream_index;
diff --git a/webrtc/modules/audio_coding/main/test/target_delay_unittest.cc b/webrtc/modules/audio_coding/main/test/target_delay_unittest.cc
index 02b8467..a902499 100644
--- a/webrtc/modules/audio_coding/main/test/target_delay_unittest.cc
+++ b/webrtc/modules/audio_coding/main/test/target_delay_unittest.cc
@@ -46,7 +46,7 @@
int16_t audio[kFrameSizeSamples];
const int kRange = 0x7FF; // 2047, easy for masking.
- for (int n = 0; n < kFrameSizeSamples; ++n)
+ for (size_t n = 0; n < kFrameSizeSamples; ++n)
audio[n] = (rand() & kRange) - kRange / 2;
WebRtcPcm16b_Encode(audio, kFrameSizeSamples, payload_);
}
@@ -133,7 +133,7 @@
private:
static const int kSampleRateHz = 16000;
static const int kNum10msPerFrame = 2;
- static const int kFrameSizeSamples = 320; // 20 ms @ 16 kHz.
+ static const size_t kFrameSizeSamples = 320; // 20 ms @ 16 kHz.
// payload-len = frame-samples * 2 bytes/sample.
static const int kPayloadLenBytes = 320 * 2;
// Inter-arrival time in number of packets in a jittery channel. One is no