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/codecs/cng/include/webrtc_cng.h b/webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h
index d90a269..b016f40 100644
--- a/webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h
+++ b/webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h
@@ -12,6 +12,7 @@
#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_MAIN_INTERFACE_WEBRTC_CNG_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_MAIN_INTERFACE_WEBRTC_CNG_H_
+#include <stddef.h>
#include "webrtc/typedefs.h"
#ifdef __cplusplus
@@ -120,7 +121,7 @@
* -1 - Error
*/
int16_t WebRtcCng_UpdateSid(CNG_dec_inst* cng_inst, uint8_t* SID,
- int16_t length);
+ size_t length);
/****************************************************************************
* WebRtcCng_Generate(...)
diff --git a/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.c b/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.c
index 28bfaae..614a3df 100644
--- a/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.c
+++ b/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.c
@@ -411,7 +411,7 @@
* -1 - Error
*/
int16_t WebRtcCng_UpdateSid(CNG_dec_inst* cng_inst, uint8_t* SID,
- int16_t length) {
+ size_t length) {
WebRtcCngDecInst_t* inst = (WebRtcCngDecInst_t*) cng_inst;
int16_t refCs[WEBRTC_CNG_MAX_LPC_ORDER];
@@ -427,7 +427,7 @@
if (length > (WEBRTC_CNG_MAX_LPC_ORDER + 1))
length = WEBRTC_CNG_MAX_LPC_ORDER + 1;
- inst->dec_order = length - 1;
+ inst->dec_order = (int16_t)length - 1;
if (SID[0] > 93)
SID[0] = 93;
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
diff --git a/webrtc/modules/audio_coding/neteq/dtmf_buffer.cc b/webrtc/modules/audio_coding/neteq/dtmf_buffer.cc
index 91debee..b07d561 100644
--- a/webrtc/modules/audio_coding/neteq/dtmf_buffer.cc
+++ b/webrtc/modules/audio_coding/neteq/dtmf_buffer.cc
@@ -55,7 +55,7 @@
//
int DtmfBuffer::ParseEvent(uint32_t rtp_timestamp,
const uint8_t* payload,
- int payload_length_bytes,
+ size_t payload_length_bytes,
DtmfEvent* event) {
if (!payload || !event) {
return kInvalidPointer;
diff --git a/webrtc/modules/audio_coding/neteq/dtmf_buffer.h b/webrtc/modules/audio_coding/neteq/dtmf_buffer.h
index 5dd31c2..5da3a16 100644
--- a/webrtc/modules/audio_coding/neteq/dtmf_buffer.h
+++ b/webrtc/modules/audio_coding/neteq/dtmf_buffer.h
@@ -69,7 +69,7 @@
// |rtp_timestamp| is simply copied into the struct.
static int ParseEvent(uint32_t rtp_timestamp,
const uint8_t* payload,
- int payload_length_bytes,
+ size_t payload_length_bytes,
DtmfEvent* event);
// Inserts |event| into the buffer. The method looks for a matching event and
diff --git a/webrtc/modules/audio_coding/neteq/interface/neteq.h b/webrtc/modules/audio_coding/neteq/interface/neteq.h
index 560e77b..b630e86 100644
--- a/webrtc/modules/audio_coding/neteq/interface/neteq.h
+++ b/webrtc/modules/audio_coding/neteq/interface/neteq.h
@@ -132,7 +132,7 @@
// Returns 0 on success, -1 on failure.
virtual int InsertPacket(const WebRtcRTPHeader& rtp_header,
const uint8_t* payload,
- int length_bytes,
+ size_t length_bytes,
uint32_t receive_timestamp) = 0;
// Inserts a sync-packet into packet queue. Sync-packets are decoded to
diff --git a/webrtc/modules/audio_coding/neteq/mock/mock_payload_splitter.h b/webrtc/modules/audio_coding/neteq/mock/mock_payload_splitter.h
index 09fa4e1..9fa05e9 100644
--- a/webrtc/modules/audio_coding/neteq/mock/mock_payload_splitter.h
+++ b/webrtc/modules/audio_coding/neteq/mock/mock_payload_splitter.h
@@ -28,11 +28,11 @@
MOCK_METHOD2(SplitAudio,
int(PacketList* packet_list, const DecoderDatabase& decoder_database));
MOCK_METHOD4(SplitBySamples,
- void(const Packet* packet, int bytes_per_ms, int timestamps_per_ms,
- PacketList* new_packets));
+ void(const Packet* packet, size_t bytes_per_ms,
+ uint32_t timestamps_per_ms, PacketList* new_packets));
MOCK_METHOD4(SplitByFrames,
- int(const Packet* packet, int bytes_per_frame, int timestamps_per_frame,
- PacketList* new_packets));
+ int(const Packet* packet, size_t bytes_per_frame,
+ uint32_t timestamps_per_frame, PacketList* new_packets));
};
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
index d41bc54..ae2d1ae 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
@@ -203,7 +203,7 @@
int sample_rate_hz_;
int samples_per_ms_;
const int frame_size_ms_;
- int frame_size_samples_;
+ size_t frame_size_samples_;
int output_size_samples_;
NetEq* neteq_external_;
NetEq* neteq_;
@@ -214,7 +214,7 @@
int16_t output_[kMaxBlockSize];
int16_t output_external_[kMaxBlockSize];
WebRtcRTPHeader rtp_header_;
- int payload_size_bytes_;
+ size_t payload_size_bytes_;
int last_send_time_;
int last_arrival_time_;
scoped_ptr<test::InputAudioFile> input_file_;
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.cc b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
index 7e8af3c..958eb76 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
@@ -117,7 +117,7 @@
int NetEqImpl::InsertPacket(const WebRtcRTPHeader& rtp_header,
const uint8_t* payload,
- int length_bytes,
+ size_t length_bytes,
uint32_t receive_timestamp) {
CriticalSectionScoped lock(crit_sect_.get());
LOG(LS_VERBOSE) << "InsertPacket: ts=" << rtp_header.header.timestamp <<
@@ -399,7 +399,7 @@
int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
const uint8_t* payload,
- int length_bytes,
+ size_t length_bytes,
uint32_t receive_timestamp,
bool is_sync_packet) {
if (!payload) {
@@ -1241,7 +1241,7 @@
assert(*operation == kNormal || *operation == kAccelerate ||
*operation == kMerge || *operation == kPreemptiveExpand);
packet_list->pop_front();
- int payload_length = packet->payload_length;
+ size_t payload_length = packet->payload_length;
int16_t decode_length;
if (packet->sync_packet) {
// Decode to silence with the same frame size as the last decode.
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.h b/webrtc/modules/audio_coding/neteq/neteq_impl.h
index 348f483..fa96512 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.h
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.h
@@ -81,7 +81,7 @@
// Returns 0 on success, -1 on failure.
virtual int InsertPacket(const WebRtcRTPHeader& rtp_header,
const uint8_t* payload,
- int length_bytes,
+ size_t length_bytes,
uint32_t receive_timestamp) OVERRIDE;
// Inserts a sync-packet into packet queue. Sync-packets are decoded to
@@ -210,7 +210,7 @@
// TODO(hlundin): Merge this with InsertPacket above?
int InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
const uint8_t* payload,
- int length_bytes,
+ size_t length_bytes,
uint32_t receive_timestamp,
bool is_sync_packet)
EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
index 56ea425..89a4d42 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
@@ -253,7 +253,7 @@
TEST_F(NetEqImplTest, InsertPacket) {
CreateInstance();
- const int kPayloadLength = 100;
+ const size_t kPayloadLength = 100;
const uint8_t kPayloadType = 0;
const uint16_t kFirstSequenceNumber = 0x1234;
const uint32_t kFirstTimestamp = 0x12345678;
diff --git a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
index 7ed9a87..0ee1d06 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
@@ -192,7 +192,7 @@
static const int kBlockSize8kHz = kTimeStepMs * 8;
static const int kBlockSize16kHz = kTimeStepMs * 16;
static const int kBlockSize32kHz = kTimeStepMs * 32;
- static const int kMaxBlockSize = kBlockSize32kHz;
+ static const size_t kMaxBlockSize = kBlockSize32kHz;
static const int kInitSampleRateHz = 8000;
NetEqDecodingTest();
@@ -213,7 +213,7 @@
int timestamp,
WebRtcRTPHeader* rtp_info,
uint8_t* payload,
- int* payload_len);
+ size_t* payload_len);
void WrapTest(uint16_t start_seq_no, uint32_t start_timestamp,
const std::set<uint16_t>& drop_seq_numbers,
@@ -244,7 +244,7 @@
const int NetEqDecodingTest::kBlockSize8kHz;
const int NetEqDecodingTest::kBlockSize16kHz;
const int NetEqDecodingTest::kBlockSize32kHz;
-const int NetEqDecodingTest::kMaxBlockSize;
+const size_t NetEqDecodingTest::kMaxBlockSize;
const int NetEqDecodingTest::kInitSampleRateHz;
NetEqDecodingTest::NetEqDecodingTest()
@@ -396,7 +396,7 @@
int timestamp,
WebRtcRTPHeader* rtp_info,
uint8_t* payload,
- int* payload_len) {
+ size_t* payload_len) {
rtp_info->header.sequenceNumber = frame_index;
rtp_info->header.timestamp = timestamp;
rtp_info->header.ssrc = 0x1234; // Just an arbitrary SSRC.
@@ -448,8 +448,8 @@
TEST_F(NetEqDecodingTestFaxMode, TestFrameWaitingTimeStatistics) {
// Insert 30 dummy packets at once. Each packet contains 10 ms 16 kHz audio.
size_t num_frames = 30;
- const int kSamples = 10 * 16;
- const int kPayloadBytes = kSamples * 2;
+ const size_t kSamples = 10 * 16;
+ const size_t kPayloadBytes = kSamples * 2;
for (size_t i = 0; i < num_frames; ++i) {
uint16_t payload[kSamples] = {0};
WebRtcRTPHeader rtp_info;
@@ -518,8 +518,8 @@
TEST_F(NetEqDecodingTest, TestAverageInterArrivalTimeNegative) {
const int kNumFrames = 3000; // Needed for convergence.
int frame_index = 0;
- const int kSamples = 10 * 16;
- const int kPayloadBytes = kSamples * 2;
+ const size_t kSamples = 10 * 16;
+ const size_t kPayloadBytes = kSamples * 2;
while (frame_index < kNumFrames) {
// Insert one packet each time, except every 10th time where we insert two
// packets at once. This will create a negative clock-drift of approx. 10%.
@@ -549,8 +549,8 @@
TEST_F(NetEqDecodingTest, TestAverageInterArrivalTimePositive) {
const int kNumFrames = 5000; // Needed for convergence.
int frame_index = 0;
- const int kSamples = 10 * 16;
- const int kPayloadBytes = kSamples * 2;
+ const size_t kSamples = 10 * 16;
+ const size_t kPayloadBytes = kSamples * 2;
for (int i = 0; i < kNumFrames; ++i) {
// Insert one packet each time, except every 10th time where we don't insert
// any packet. This will create a positive clock-drift of approx. 11%.
@@ -585,8 +585,8 @@
uint16_t seq_no = 0;
uint32_t timestamp = 0;
const int kFrameSizeMs = 30;
- const int kSamples = kFrameSizeMs * 16;
- const int kPayloadBytes = kSamples * 2;
+ const size_t kSamples = kFrameSizeMs * 16;
+ const size_t kPayloadBytes = kSamples * 2;
double next_input_time_ms = 0.0;
double t_ms;
int out_len;
@@ -625,7 +625,7 @@
while (next_input_time_ms <= t_ms) {
// Insert one CNG frame each 100 ms.
uint8_t payload[kPayloadBytes];
- int payload_len;
+ size_t payload_len;
WebRtcRTPHeader rtp_info;
PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, payload_len, 0));
@@ -672,7 +672,7 @@
}
// Insert one CNG frame each 100 ms.
uint8_t payload[kPayloadBytes];
- int payload_len;
+ size_t payload_len;
WebRtcRTPHeader rtp_info;
PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, payload_len, 0));
@@ -797,7 +797,7 @@
}
TEST_F(NetEqDecodingTest, UnknownPayloadType) {
- const int kPayloadBytes = 100;
+ const size_t kPayloadBytes = 100;
uint8_t payload[kPayloadBytes] = {0};
WebRtcRTPHeader rtp_info;
PopulateRtpInfo(0, 0, &rtp_info);
@@ -808,7 +808,7 @@
}
TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(DecoderError)) {
- const int kPayloadBytes = 100;
+ const size_t kPayloadBytes = 100;
uint8_t payload[kPayloadBytes] = {0};
WebRtcRTPHeader rtp_info;
PopulateRtpInfo(0, 0, &rtp_info);
@@ -817,7 +817,7 @@
NetEqOutputType type;
// Set all of |out_data_| to 1, and verify that it was set to 0 by the call
// to GetAudio.
- for (int i = 0; i < kMaxBlockSize; ++i) {
+ for (size_t i = 0; i < kMaxBlockSize; ++i) {
out_data_[i] = 1;
}
int num_channels;
@@ -838,7 +838,7 @@
SCOPED_TRACE(ss.str()); // Print out the parameter values on failure.
EXPECT_EQ(0, out_data_[i]);
}
- for (int i = kExpectedOutputLength; i < kMaxBlockSize; ++i) {
+ for (size_t i = kExpectedOutputLength; i < kMaxBlockSize; ++i) {
std::ostringstream ss;
ss << "i = " << i;
SCOPED_TRACE(ss.str()); // Print out the parameter values on failure.
@@ -850,7 +850,7 @@
NetEqOutputType type;
// Set all of |out_data_| to 1, and verify that it was set to 0 by the call
// to GetAudio.
- for (int i = 0; i < kMaxBlockSize; ++i) {
+ for (size_t i = 0; i < kMaxBlockSize; ++i) {
out_data_[i] = 1;
}
int num_channels;
@@ -875,7 +875,7 @@
bool should_be_faded) = 0;
void CheckBgn(int sampling_rate_hz) {
- int expected_samples_per_channel = 0;
+ int16_t expected_samples_per_channel = 0;
uint8_t payload_type = 0xFF; // Invalid.
if (sampling_rate_hz == 8000) {
expected_samples_per_channel = kBlockSize8kHz;
@@ -899,7 +899,7 @@
ASSERT_TRUE(input.Init(
webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"),
10 * sampling_rate_hz, // Max 10 seconds loop length.
- expected_samples_per_channel));
+ static_cast<size_t>(expected_samples_per_channel)));
// Payload of 10 ms of PCM16 32 kHz.
uint8_t payload[kBlockSize32kHz * sizeof(int16_t)];
@@ -912,7 +912,7 @@
uint32_t receive_timestamp = 0;
for (int n = 0; n < 10; ++n) { // Insert few packets and get audio.
- int enc_len_bytes =
+ int16_t enc_len_bytes =
WebRtcPcm16b_EncodeW16(input.GetNextBlock(),
expected_samples_per_channel,
reinterpret_cast<int16_t*>(payload));
@@ -921,8 +921,9 @@
number_channels = 0;
samples_per_channel = 0;
ASSERT_EQ(0,
- neteq_->InsertPacket(
- rtp_info, payload, enc_len_bytes, receive_timestamp));
+ neteq_->InsertPacket(rtp_info, payload,
+ static_cast<size_t>(enc_len_bytes),
+ receive_timestamp));
ASSERT_EQ(0,
neteq_->GetAudio(kBlockSize32kHz,
output,
@@ -1074,7 +1075,7 @@
EXPECT_EQ(-1, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
// Payload length of 10 ms PCM16 16 kHz.
- const int kPayloadBytes = kBlockSize16kHz * sizeof(int16_t);
+ const size_t kPayloadBytes = kBlockSize16kHz * sizeof(int16_t);
uint8_t payload[kPayloadBytes] = {0};
ASSERT_EQ(0, neteq_->InsertPacket(
rtp_info, payload, kPayloadBytes, receive_timestamp));
@@ -1125,11 +1126,11 @@
TEST_F(NetEqDecodingTest, SyncPacketDecode) {
WebRtcRTPHeader rtp_info;
PopulateRtpInfo(0, 0, &rtp_info);
- const int kPayloadBytes = kBlockSize16kHz * sizeof(int16_t);
+ const size_t kPayloadBytes = kBlockSize16kHz * sizeof(int16_t);
uint8_t payload[kPayloadBytes];
int16_t decoded[kBlockSize16kHz];
int algorithmic_frame_delay = algorithmic_delay_ms_ / 10 + 1;
- for (int n = 0; n < kPayloadBytes; ++n) {
+ for (size_t n = 0; n < kPayloadBytes; ++n) {
payload[n] = (rand() & 0xF0) + 1; // Non-zero random sequence.
}
// Insert some packets which decode to noise. We are not interested in
@@ -1204,10 +1205,10 @@
TEST_F(NetEqDecodingTest, SyncPacketBufferSizeAndOverridenByNetworkPackets) {
WebRtcRTPHeader rtp_info;
PopulateRtpInfo(0, 0, &rtp_info);
- const int kPayloadBytes = kBlockSize16kHz * sizeof(int16_t);
+ const size_t kPayloadBytes = kBlockSize16kHz * sizeof(int16_t);
uint8_t payload[kPayloadBytes];
int16_t decoded[kBlockSize16kHz];
- for (int n = 0; n < kPayloadBytes; ++n) {
+ for (size_t n = 0; n < kPayloadBytes; ++n) {
payload[n] = (rand() & 0xF0) + 1; // Non-zero random sequence.
}
// Insert some packets which decode to noise. We are not interested in
@@ -1279,7 +1280,7 @@
const int kBlocksPerFrame = 3; // Number of 10 ms blocks per frame.
const int kFrameSizeMs = kBlocksPerFrame * kTimeStepMs;
const int kSamples = kBlockSize16kHz * kBlocksPerFrame;
- const int kPayloadBytes = kSamples * sizeof(int16_t);
+ const size_t kPayloadBytes = kSamples * sizeof(int16_t);
double next_input_time_ms = 0.0;
int16_t decoded[kBlockSize16kHz];
int num_channels;
@@ -1380,7 +1381,7 @@
const int kFrameSizeMs = 10;
const int kSampleRateKhz = 16;
const int kSamples = kFrameSizeMs * kSampleRateKhz;
- const int kPayloadBytes = kSamples * 2;
+ const size_t kPayloadBytes = kSamples * 2;
const int algorithmic_delay_samples = std::max(
algorithmic_delay_ms_ * kSampleRateKhz, 5 * kSampleRateKhz / 8);
@@ -1409,7 +1410,7 @@
// Insert same CNG packet twice.
const int kCngPeriodMs = 100;
const int kCngPeriodSamples = kCngPeriodMs * kSampleRateKhz;
- int payload_len;
+ size_t payload_len;
PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
// This is the first time this CNG packet is inserted.
ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, payload_len, 0));
diff --git a/webrtc/modules/audio_coding/neteq/packet.h b/webrtc/modules/audio_coding/neteq/packet.h
index 89ddda7..723ed8b 100644
--- a/webrtc/modules/audio_coding/neteq/packet.h
+++ b/webrtc/modules/audio_coding/neteq/packet.h
@@ -22,7 +22,7 @@
struct Packet {
RTPHeader header;
uint8_t* payload; // Datagram excluding RTP header and header extension.
- int payload_length;
+ size_t payload_length;
bool primary; // Primary, i.e., not redundant payload.
int waiting_time;
bool sync_packet;
diff --git a/webrtc/modules/audio_coding/neteq/payload_splitter.cc b/webrtc/modules/audio_coding/neteq/payload_splitter.cc
index 1d61ef0..118556b 100644
--- a/webrtc/modules/audio_coding/neteq/payload_splitter.cc
+++ b/webrtc/modules/audio_coding/neteq/payload_splitter.cc
@@ -46,7 +46,7 @@
// +-+-+-+-+-+-+-+-+
bool last_block = false;
- int sum_length = 0;
+ size_t sum_length = 0;
while (!last_block) {
Packet* new_packet = new Packet;
new_packet->header = red_packet->header;
@@ -82,7 +82,7 @@
// |payload_ptr| now points at the first payload byte.
PacketList::iterator new_it;
for (new_it = new_packets.begin(); new_it != new_packets.end(); ++new_it) {
- int payload_length = (*new_it)->payload_length;
+ size_t payload_length = (*new_it)->payload_length;
if (payload_ptr + payload_length >
red_packet->payload + red_packet->payload_length) {
// The block lengths in the RED headers do not match the overall packet
@@ -291,11 +291,12 @@
break;
}
case kDecoderILBC: {
- int bytes_per_frame;
+ size_t bytes_per_frame;
int timestamps_per_frame;
if (packet->payload_length >= 950) {
return kTooLargePayload;
- } else if (packet->payload_length % 38 == 0) {
+ }
+ if (packet->payload_length % 38 == 0) {
// 20 ms frames.
bytes_per_frame = 38;
timestamps_per_frame = 160;
@@ -345,28 +346,28 @@
}
void PayloadSplitter::SplitBySamples(const Packet* packet,
- int bytes_per_ms,
- int timestamps_per_ms,
+ size_t bytes_per_ms,
+ uint32_t timestamps_per_ms,
PacketList* new_packets) {
assert(packet);
assert(new_packets);
- int split_size_bytes = packet->payload_length;
+ size_t split_size_bytes = packet->payload_length;
// Find a "chunk size" >= 20 ms and < 40 ms.
- int min_chunk_size = bytes_per_ms * 20;
+ size_t min_chunk_size = bytes_per_ms * 20;
// Reduce the split size by half as long as |split_size_bytes| is at least
// twice the minimum chunk size (so that the resulting size is at least as
// large as the minimum chunk size).
while (split_size_bytes >= 2 * min_chunk_size) {
split_size_bytes >>= 1;
}
- int timestamps_per_chunk =
- split_size_bytes * timestamps_per_ms / bytes_per_ms;
+ uint32_t timestamps_per_chunk = static_cast<uint32_t>(
+ split_size_bytes * timestamps_per_ms / bytes_per_ms);
uint32_t timestamp = packet->header.timestamp;
uint8_t* payload_ptr = packet->payload;
- int len = packet->payload_length;
+ size_t len = packet->payload_length;
while (len >= (2 * split_size_bytes)) {
Packet* new_packet = new Packet;
new_packet->payload_length = split_size_bytes;
@@ -394,22 +395,21 @@
}
int PayloadSplitter::SplitByFrames(const Packet* packet,
- int bytes_per_frame,
- int timestamps_per_frame,
+ size_t bytes_per_frame,
+ uint32_t timestamps_per_frame,
PacketList* new_packets) {
if (packet->payload_length % bytes_per_frame != 0) {
return kFrameSplitError;
}
- int num_frames = packet->payload_length / bytes_per_frame;
- if (num_frames == 1) {
+ if (packet->payload_length == bytes_per_frame) {
// Special case. Do not split the payload.
return kNoSplit;
}
uint32_t timestamp = packet->header.timestamp;
uint8_t* payload_ptr = packet->payload;
- int len = packet->payload_length;
+ size_t len = packet->payload_length;
while (len > 0) {
assert(len >= bytes_per_frame);
Packet* new_packet = new Packet;
diff --git a/webrtc/modules/audio_coding/neteq/payload_splitter.h b/webrtc/modules/audio_coding/neteq/payload_splitter.h
index a3dd77e..6023d4e 100644
--- a/webrtc/modules/audio_coding/neteq/payload_splitter.h
+++ b/webrtc/modules/audio_coding/neteq/payload_splitter.h
@@ -71,16 +71,16 @@
// Splits the payload in |packet|. The payload is assumed to be from a
// sample-based codec.
virtual void SplitBySamples(const Packet* packet,
- int bytes_per_ms,
- int timestamps_per_ms,
+ size_t bytes_per_ms,
+ uint32_t timestamps_per_ms,
PacketList* new_packets);
// Splits the payload in |packet|. The payload will be split into chunks of
// size |bytes_per_frame|, corresponding to a |timestamps_per_frame|
// RTP timestamps.
virtual int SplitByFrames(const Packet* packet,
- int bytes_per_frame,
- int timestamps_per_frame,
+ size_t bytes_per_frame,
+ uint32_t timestamps_per_frame,
PacketList* new_packets);
DISALLOW_COPY_AND_ASSIGN(PayloadSplitter);
diff --git a/webrtc/modules/audio_coding/neteq/payload_splitter_unittest.cc b/webrtc/modules/audio_coding/neteq/payload_splitter_unittest.cc
index cf29581..d397a07 100644
--- a/webrtc/modules/audio_coding/neteq/payload_splitter_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/payload_splitter_unittest.cc
@@ -27,8 +27,8 @@
namespace webrtc {
static const int kRedPayloadType = 100;
-static const int kPayloadLength = 10;
-static const int kRedHeaderLength = 4; // 4 bytes RED header.
+static const size_t kPayloadLength = 10;
+static const size_t kRedHeaderLength = 4; // 4 bytes RED header.
static const uint16_t kSequenceNumber = 0;
static const uint32_t kBaseTimestamp = 0x12345678;
@@ -50,7 +50,7 @@
// by the values in array |payload_types| (which must be of length
// |num_payloads|). Each redundant payload is |timestamp_offset| samples
// "behind" the the previous payload.
-Packet* CreateRedPayload(int num_payloads,
+Packet* CreateRedPayload(size_t num_payloads,
uint8_t* payload_types,
int timestamp_offset) {
Packet* packet = new Packet;
@@ -61,7 +61,7 @@
(num_payloads - 1) * (kPayloadLength + kRedHeaderLength);
uint8_t* payload = new uint8_t[packet->payload_length];
uint8_t* payload_ptr = payload;
- for (int i = 0; i < num_payloads; ++i) {
+ for (size_t i = 0; i < num_payloads; ++i) {
// Write the RED headers.
if (i == num_payloads - 1) {
// Special case for last payload.
@@ -82,9 +82,9 @@
*payload_ptr = kPayloadLength & 0xFF;
++payload_ptr;
}
- for (int i = 0; i < num_payloads; ++i) {
+ for (size_t i = 0; i < num_payloads; ++i) {
// Write |i| to all bytes in each payload.
- memset(payload_ptr, i, kPayloadLength);
+ memset(payload_ptr, static_cast<int>(i), kPayloadLength);
payload_ptr += kPayloadLength;
}
packet->payload = payload;
@@ -104,7 +104,7 @@
// : |
// | |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-Packet* CreateOpusFecPacket(uint8_t payload_type, int payload_length,
+Packet* CreateOpusFecPacket(uint8_t payload_type, size_t payload_length,
uint8_t payload_value) {
Packet* packet = new Packet;
packet->header.payloadType = payload_type;
@@ -120,7 +120,7 @@
}
// Create a packet with all payload bytes set to |payload_value|.
-Packet* CreatePacket(uint8_t payload_type, int payload_length,
+Packet* CreatePacket(uint8_t payload_type, size_t payload_length,
uint8_t payload_value) {
Packet* packet = new Packet;
packet->header.payloadType = payload_type;
@@ -135,7 +135,7 @@
// Checks that |packet| has the attributes given in the remaining parameters.
void VerifyPacket(const Packet* packet,
- int payload_length,
+ size_t payload_length,
uint8_t payload_type,
uint16_t sequence_number,
uint32_t timestamp,
@@ -147,7 +147,7 @@
EXPECT_EQ(timestamp, packet->header.timestamp);
EXPECT_EQ(primary, packet->primary);
ASSERT_FALSE(packet->payload == NULL);
- for (int i = 0; i < packet->payload_length; ++i) {
+ for (size_t i = 0; i < packet->payload_length; ++i) {
EXPECT_EQ(payload_value, packet->payload[i]);
}
}
@@ -295,7 +295,7 @@
// found in the list (which is PCMu).
TEST(RedPayloadSplitter, CheckRedPayloads) {
PacketList packet_list;
- for (int i = 0; i <= 3; ++i) {
+ for (uint8_t i = 0; i <= 3; ++i) {
// Create packet with payload type |i|, payload length 10 bytes, all 0.
Packet* packet = CreatePacket(i, 10, 0);
packet_list.push_back(packet);
@@ -357,7 +357,7 @@
// Set up packets with different RTP payload types. The actual values do not
// matter, since we are mocking the decoder database anyway.
PacketList packet_list;
- for (int i = 0; i < 6; ++i) {
+ for (uint8_t i = 0; i < 6; ++i) {
// Let the payload type be |i|, and the payload value 10 * |i|.
packet_list.push_back(CreatePacket(i, kPayloadLength, 10 * i));
}
@@ -415,7 +415,7 @@
TEST(AudioPayloadSplitter, UnknownPayloadType) {
PacketList packet_list;
static const uint8_t kPayloadType = 17; // Just a random number.
- int kPayloadLengthBytes = 4711; // Random number.
+ size_t kPayloadLengthBytes = 4711; // Random number.
packet_list.push_back(CreatePacket(kPayloadType, kPayloadLengthBytes, 0));
MockDecoderDatabase decoder_database;
@@ -502,7 +502,7 @@
break;
}
}
- int bytes_per_ms_;
+ size_t bytes_per_ms_;
int samples_per_ms_;
NetEqDecoder decoder_type_;
};
@@ -514,7 +514,7 @@
for (int payload_size_ms = 10; payload_size_ms <= 60; payload_size_ms += 10) {
// The payload values are set to be the same as the payload_size, so that
// one can distinguish from which packet the split payloads come from.
- int payload_size_bytes = payload_size_ms * bytes_per_ms_;
+ size_t payload_size_bytes = payload_size_ms * bytes_per_ms_;
packet_list.push_back(CreatePacket(kPayloadType, payload_size_bytes,
payload_size_ms));
}
@@ -548,7 +548,7 @@
PacketList::iterator it = packet_list.begin();
int i = 0;
while (it != packet_list.end()) {
- int length_bytes = expected_size_ms[i] * bytes_per_ms_;
+ size_t length_bytes = expected_size_ms[i] * bytes_per_ms_;
uint32_t expected_timestamp = kBaseTimestamp +
expected_timestamp_offset_ms[i] * samples_per_ms_;
VerifyPacket((*it), length_bytes, kPayloadType, kSequenceNumber,
@@ -583,7 +583,7 @@
}
size_t num_frames_;
int frame_length_ms_;
- int frame_length_bytes_;
+ size_t frame_length_bytes_;
};
// Test splitting sample-based payloads.
@@ -591,10 +591,10 @@
PacketList packet_list;
static const uint8_t kPayloadType = 17; // Just a random number.
const int frame_length_samples = frame_length_ms_ * 8;
- int payload_length_bytes = frame_length_bytes_ * num_frames_;
+ size_t payload_length_bytes = frame_length_bytes_ * num_frames_;
Packet* packet = CreatePacket(kPayloadType, payload_length_bytes, 0);
// Fill payload with increasing integers {0, 1, 2, ...}.
- for (int i = 0; i < packet->payload_length; ++i) {
+ for (size_t i = 0; i < packet->payload_length; ++i) {
packet->payload[i] = static_cast<uint8_t>(i);
}
packet_list.push_back(packet);
@@ -624,7 +624,7 @@
EXPECT_EQ(kSequenceNumber, packet->header.sequenceNumber);
EXPECT_EQ(true, packet->primary);
ASSERT_FALSE(packet->payload == NULL);
- for (int i = 0; i < packet->payload_length; ++i) {
+ for (size_t i = 0; i < packet->payload_length; ++i) {
EXPECT_EQ(payload_value, packet->payload[i]);
++payload_value;
}
@@ -661,7 +661,7 @@
TEST(IlbcPayloadSplitter, TooLargePayload) {
PacketList packet_list;
static const uint8_t kPayloadType = 17; // Just a random number.
- int kPayloadLengthBytes = 950;
+ size_t kPayloadLengthBytes = 950;
Packet* packet = CreatePacket(kPayloadType, kPayloadLengthBytes, 0);
packet_list.push_back(packet);
@@ -692,7 +692,7 @@
TEST(IlbcPayloadSplitter, UnevenPayload) {
PacketList packet_list;
static const uint8_t kPayloadType = 17; // Just a random number.
- int kPayloadLengthBytes = 39; // Not an even number of frames.
+ size_t kPayloadLengthBytes = 39; // Not an even number of frames.
Packet* packet = CreatePacket(kPayloadType, kPayloadLengthBytes, 0);
packet_list.push_back(packet);
@@ -744,7 +744,7 @@
packet = packet_list.front();
EXPECT_EQ(0, packet->header.payloadType);
EXPECT_EQ(kBaseTimestamp - 20 * 48, packet->header.timestamp);
- EXPECT_EQ(10, packet->payload_length);
+ EXPECT_EQ(10U, packet->payload_length);
EXPECT_FALSE(packet->primary);
delete [] packet->payload;
delete packet;
@@ -754,7 +754,7 @@
packet = packet_list.front();
EXPECT_EQ(0, packet->header.payloadType);
EXPECT_EQ(kBaseTimestamp, packet->header.timestamp);
- EXPECT_EQ(10, packet->payload_length);
+ EXPECT_EQ(10U, packet->payload_length);
EXPECT_TRUE(packet->primary);
delete [] packet->payload;
delete packet;
diff --git a/webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.cc b/webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.cc
index 7f94851..d4c2191 100644
--- a/webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.cc
+++ b/webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.cc
@@ -329,7 +329,7 @@
}
}
-int16_t NETEQTEST_RTPpacket::payloadLen()
+size_t NETEQTEST_RTPpacket::payloadLen()
{
parseHeader();
return _payloadLen;
@@ -752,7 +752,7 @@
int stride)
{
if(!_payloadPtr || !slaveRtp || !slaveRtp->_payloadPtr
- || _payloadLen <= 0 || slaveRtp->_memSize < _memSize)
+ || _payloadLen == 0 || slaveRtp->_memSize < _memSize)
{
return;
}
@@ -761,7 +761,7 @@
uint8_t *writeDataPtr = _payloadPtr;
uint8_t *slaveData = slaveRtp->_payloadPtr;
- while (readDataPtr - _payloadPtr < _payloadLen)
+ while (readDataPtr - _payloadPtr < static_cast<ptrdiff_t>(_payloadLen))
{
// master data
for (int ix = 0; ix < stride; ix++) {
@@ -786,7 +786,7 @@
void NETEQTEST_RTPpacket::splitStereoFrame(NETEQTEST_RTPpacket* slaveRtp)
{
if(!_payloadPtr || !slaveRtp || !slaveRtp->_payloadPtr
- || _payloadLen <= 0 || slaveRtp->_memSize < _memSize)
+ || _payloadLen == 0 || slaveRtp->_memSize < _memSize)
{
return;
}
@@ -799,7 +799,7 @@
void NETEQTEST_RTPpacket::splitStereoDouble(NETEQTEST_RTPpacket* slaveRtp)
{
if(!_payloadPtr || !slaveRtp || !slaveRtp->_payloadPtr
- || _payloadLen <= 0 || slaveRtp->_memSize < _memSize)
+ || _payloadLen == 0 || slaveRtp->_memSize < _memSize)
{
return;
}
@@ -868,7 +868,7 @@
{
parseHeader();
- for (int i = 0; i < _payloadLen; ++i)
+ for (size_t i = 0; i < _payloadLen; ++i)
{
_payloadPtr[i] = static_cast<uint8_t>(rand());
}
diff --git a/webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h b/webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h
index 8a31274..86bf3b0 100644
--- a/webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h
+++ b/webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h
@@ -42,7 +42,7 @@
const webrtc::WebRtcRTPHeader* RTPinfo() const;
uint8_t * datagram() const;
uint8_t * payload() const;
- int16_t payloadLen();
+ size_t payloadLen();
int16_t dataLen() const;
bool isParsed() const;
bool isLost() const;
@@ -73,7 +73,7 @@
uint8_t * _payloadPtr;
int _memSize;
int16_t _datagramLen;
- int16_t _payloadLen;
+ size_t _payloadLen;
webrtc::WebRtcRTPHeader _rtpInfo;
bool _rtpParsed;
uint32_t _receiveTime;
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc b/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
index 433546f..ebe0784 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
@@ -64,9 +64,9 @@
const int16_t* input_samples = audio_loop.GetNextBlock();
if (!input_samples) exit(1);
uint8_t input_payload[kInputBlockSizeSamples * sizeof(int16_t)];
- int payload_len = WebRtcPcm16b_Encode(const_cast<int16_t*>(input_samples),
- kInputBlockSizeSamples,
- input_payload);
+ size_t payload_len = WebRtcPcm16b_Encode(const_cast<int16_t*>(input_samples),
+ kInputBlockSizeSamples,
+ input_payload);
assert(payload_len == kInputBlockSizeSamples * sizeof(int16_t));
// Main loop.
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h b/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h
index e0a43b6..00a2499 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h
@@ -118,7 +118,7 @@
// Expected output number of samples per channel in a frame.
const int out_size_samples_;
- int payload_size_bytes_;
+ size_t payload_size_bytes_;
int max_payload_bytes_;
scoped_ptr<InputAudioFile> in_file_;
@@ -134,7 +134,7 @@
scoped_ptr<int16_t[]> out_data_;
WebRtcRTPHeader rtp_header_;
- long total_payload_size_bytes_;
+ size_t total_payload_size_bytes_;
};
} // namespace test
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc b/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
index ef2c0b6..4247807 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
@@ -286,7 +286,7 @@
int error =
neteq->InsertPacket(rtp_header,
payload_ptr,
- static_cast<int>(payload_len),
+ payload_len,
packet->time_ms() * sample_rate_hz / 1000);
if (error != NetEq::kOK) {
if (neteq->LastError() == NetEq::kUnknownRtpPayloadType) {