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Use size_t more consistently for packet/payload lengths.

Browse Source 
See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information.

This CL was reviewed and approved in pieces in the following CLs:
https://webrtc-codereview.appspot.com/24209004/
https://webrtc-codereview.appspot.com/24229004/
https://webrtc-codereview.appspot.com/24259004/
https://webrtc-codereview.appspot.com/25109004/
https://webrtc-codereview.appspot.com/26099004/
https://webrtc-codereview.appspot.com/27069004/
https://webrtc-codereview.appspot.com/27969004/
https://webrtc-codereview.appspot.com/27989004/
https://webrtc-codereview.appspot.com/29009004/
https://webrtc-codereview.appspot.com/30929004/
https://webrtc-codereview.appspot.com/30939004/
https://webrtc-codereview.appspot.com/31999004/
Committing as TBR to the original reviewers.

BUG=chromium:81439
TEST=none
TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom

Review URL: https://webrtc-codereview.appspot.com/23129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pkasting@chromium.org
2014-11-20 22:28:14 +00:00
parent edc6e57a92
commit 4591fbd09f
341 changed files with 2610 additions and 2613 deletions
Download Patch File Download Diff File Expand all files Collapse all files

2
webrtc/modules/audio_coding/neteq/dtmf_buffer.cc
View File

@ -55,7 +55,7 @@ namespace webrtc {
//
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;

2
webrtc/modules/audio_coding/neteq/dtmf_buffer.h
View File

@ -69,7 +69,7 @@ class DtmfBuffer {
// |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

2
webrtc/modules/audio_coding/neteq/interface/neteq.h
View File

@ -132,7 +132,7 @@ class NetEq {
// 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

8
webrtc/modules/audio_coding/neteq/mock/mock_payload_splitter.h
View File

@ -28,11 +28,11 @@ class MockPayloadSplitter : public PayloadSplitter {
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

4
webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
View File

@ -203,7 +203,7 @@ class NetEqExternalDecoderTest : public ::testing::Test {
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 @@ class NetEqExternalDecoderTest : public ::testing::Test {
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_;

6
webrtc/modules/audio_coding/neteq/neteq_impl.cc
View File

@ -117,7 +117,7 @@ NetEqImpl::~NetEqImpl() {
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 @@ const SyncBuffer* NetEqImpl::sync_buffer_for_test() const {
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 @@ int NetEqImpl::DecodeLoop(PacketList* packet_list, Operations* operation,
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.

4
webrtc/modules/audio_coding/neteq/neteq_impl.h
View File

@ -81,7 +81,7 @@ class NetEqImpl : public webrtc::NetEq {
// 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 @@ class NetEqImpl : public webrtc::NetEq {
// 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_);

2
webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
View File

@ -253,7 +253,7 @@ TEST_F(NetEqImplTest, RemovePayloadType) {
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;

65
webrtc/modules/audio_coding/neteq/neteq_unittest.cc
View File

@ -192,7 +192,7 @@ class NetEqDecodingTest : public ::testing::Test {
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 @@ class NetEqDecodingTest : public ::testing::Test {
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::kTimeStepMs;
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 @@ void NetEqDecodingTest::PopulateCng(int frame_index,
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 @@ class NetEqDecodingTestFaxMode : public NetEqDecodingTest {
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(NetEqDecodingTestFaxMode, TestFrameWaitingTimeStatistics) {
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, TestAverageInterArrivalTimeNegative) {
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 @@ void NetEqDecodingTest::LongCngWithClockDrift(double drift_factor,
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 @@ void NetEqDecodingTest::LongCngWithClockDrift(double drift_factor,
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 @@ void NetEqDecodingTest::LongCngWithClockDrift(double drift_factor,
}
// 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, LongCngWithoutClockDrift) {
}
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, UnknownPayloadType) {
}
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 @@ TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(DecoderError)) {
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 @@ TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(DecoderError)) {
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 @@ TEST_F(NetEqDecodingTest, GetAudioBeforeInsertPacket) {
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 @@ class NetEqBgnTest : public NetEqDecodingTest {
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 @@ class NetEqBgnTest : public NetEqDecodingTest {
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 @@ class NetEqBgnTest : public NetEqDecodingTest {
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 @@ class NetEqBgnTest : public NetEqDecodingTest {
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 @@ TEST_F(NetEqDecodingTest, SyncPacketInsert) {
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, SyncPacketInsert) {
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, SyncPacketDecode) {
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 @@ void NetEqDecodingTest::WrapTest(uint16_t start_seq_no,
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 @@ void NetEqDecodingTest::DuplicateCng() {
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 @@ void NetEqDecodingTest::DuplicateCng() {
// 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));

2
webrtc/modules/audio_coding/neteq/packet.h
View File

@ -22,7 +22,7 @@ namespace webrtc {
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;

32
webrtc/modules/audio_coding/neteq/payload_splitter.cc
View File

@ -46,7 +46,7 @@ int PayloadSplitter::SplitRed(PacketList* packet_list) {
// +-+-+-+-+-+-+-+-+
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 @@ int PayloadSplitter::SplitRed(PacketList* packet_list) {
// |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 @@ int PayloadSplitter::SplitAudio(PacketList* packet_list,
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 @@ int PayloadSplitter::SplitAudio(PacketList* packet_list,
}
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 @@ void PayloadSplitter::SplitBySamples(const Packet* packet,
}
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;

8
webrtc/modules/audio_coding/neteq/payload_splitter.h
View File

@ -71,16 +71,16 @@ class PayloadSplitter {
// 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);

48
webrtc/modules/audio_coding/neteq/payload_splitter_unittest.cc
View File

@ -27,8 +27,8 @@ using ::testing::ReturnNull;
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 @@ static const uint32_t kBaseTimestamp = 0x12345678;
// 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 @@ Packet* CreateRedPayload(int num_payloads,
(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 @@ Packet* CreateRedPayload(int num_payloads,
*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* CreateRedPayload(int num_payloads,
// : |
// | |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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 @@ Packet* CreateOpusFecPacket(uint8_t payload_type, int payload_length,
}
// 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 @@ Packet* CreatePacket(uint8_t payload_type, int payload_length,
// 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 @@ void VerifyPacket(const Packet* packet,
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 @@ TEST(RedPayloadSplitter, TwoPacketsThreePayloads) {
// 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 @@ TEST(AudioPayloadSplitter, NonSplittable) {
// 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, NonSplittable) {
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 @@ class SplitBySamplesTest : public ::testing::TestWithParam<NetEqDecoder> {
break;
}
}
int bytes_per_ms_;
size_t bytes_per_ms_;
int samples_per_ms_;
NetEqDecoder decoder_type_;
};
@ -514,7 +514,7 @@ TEST_P(SplitBySamplesTest, PayloadSizes) {
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 @@ TEST_P(SplitBySamplesTest, PayloadSizes) {
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 @@ class SplitIlbcTest : public ::testing::TestWithParam<std::pair<int, int> > {
}
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 @@ TEST_P(SplitIlbcTest, NumFrames) {
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 @@ TEST_P(SplitIlbcTest, NumFrames) {
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 @@ INSTANTIATE_TEST_CASE_P(
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, TooLargePayload) {
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 @@ TEST(FecPayloadSplitter, MixedPayload) {
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 @@ TEST(FecPayloadSplitter, MixedPayload) {
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;

12
webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.cc
View File

@ -329,7 +329,7 @@ uint8_t * NETEQTEST_RTPpacket::payload() const
}
}
int16_t NETEQTEST_RTPpacket::payloadLen()
size_t NETEQTEST_RTPpacket::payloadLen()
{
parseHeader();
return _payloadLen;
@ -752,7 +752,7 @@ void NETEQTEST_RTPpacket::splitStereoSample(NETEQTEST_RTPpacket* slaveRtp,
int stride)
{
if(!_payloadPtr || !slaveRtp || !slaveRtp->_payloadPtr
|| _payloadLen <= 0 || slaveRtp->_memSize < _memSize)
|| _payloadLen == 0 || slaveRtp->_memSize < _memSize)
{
return;
}
@ -761,7 +761,7 @@ void NETEQTEST_RTPpacket::splitStereoSample(NETEQTEST_RTPpacket* slaveRtp,
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::splitStereoSample(NETEQTEST_RTPpacket* slaveRtp,
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::splitStereoFrame(NETEQTEST_RTPpacket* slaveRtp)
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 @@ void NETEQTEST_RTPpacket::scramblePayload(void)
{
parseHeader();
for (int i = 0; i < _payloadLen; ++i)
for (size_t i = 0; i < _payloadLen; ++i)
{
_payloadPtr[i] = static_cast<uint8_t>(rand());
}

4
webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h
View File

@ -42,7 +42,7 @@ public:
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 @@ public:
uint8_t * _payloadPtr;
int _memSize;
int16_t _datagramLen;
int16_t _payloadLen;
size_t _payloadLen;
webrtc::WebRtcRTPHeader _rtpInfo;
bool _rtpParsed;
uint32_t _receiveTime;

6
webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
View File

@ -64,9 +64,9 @@ int64_t NetEqPerformanceTest::Run(int runtime_ms,
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.

4
webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h
View File

@ -118,7 +118,7 @@ class NetEqQualityTest : public ::testing::Test {
// 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 @@ class NetEqQualityTest : public ::testing::Test {
scoped_ptr<int16_t[]> out_data_;
WebRtcRTPHeader rtp_header_;
long total_payload_size_bytes_;
size_t total_payload_size_bytes_;
};
} // namespace test

2
webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
View File

@ -286,7 +286,7 @@ int main(int argc, char* argv[]) {
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) {