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/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) {