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/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));