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Update a ton of audio code to use size_t more correctly and in general reduce

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use of int16_t/uint16_t.

This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.

This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002

The change is being landed as TBR to all the folks who reviewed the above.

BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher

Review URL: https://codereview.webrtc.org/1230503003 .

Cr-Commit-Position: refs/heads/master@{#9768}
This commit is contained in:
Peter Kasting
2015-08-24 14:52:23 -07:00
parent b594041ec8
commit dce40cf804
471 changed files with 3716 additions and 3499 deletions
Download Patch File Download Diff File Expand all files Collapse all files

68
webrtc/modules/audio_coding/neteq/neteq_unittest.cc
View File

@ -37,16 +37,16 @@ DEFINE_bool(gen_ref, false, "Generate reference files.");
namespace webrtc {
static bool IsAllZero(const int16_t* buf, int buf_length) {
static bool IsAllZero(const int16_t* buf, size_t buf_length) {
bool all_zero = true;
for (int n = 0; n < buf_length && all_zero; ++n)
for (size_t n = 0; n < buf_length && all_zero; ++n)
all_zero = buf[n] == 0;
return all_zero;
}
static bool IsAllNonZero(const int16_t* buf, int buf_length) {
static bool IsAllNonZero(const int16_t* buf, size_t buf_length) {
bool all_non_zero = true;
for (int n = 0; n < buf_length && all_non_zero; ++n)
for (size_t n = 0; n < buf_length && all_non_zero; ++n)
all_non_zero = buf[n] != 0;
return all_non_zero;
}
@ -172,7 +172,8 @@ void RefFiles::ReadFromFileAndCompare(
ASSERT_EQ(stats.preemptive_rate, ref_stats.preemptive_rate);
ASSERT_EQ(stats.accelerate_rate, ref_stats.accelerate_rate);
ASSERT_EQ(stats.clockdrift_ppm, ref_stats.clockdrift_ppm);
ASSERT_EQ(stats.added_zero_samples, ref_stats.added_zero_samples);
ASSERT_EQ(stats.added_zero_samples,
static_cast<size_t>(ref_stats.added_zero_samples));
ASSERT_EQ(stats.secondary_decoded_rate, 0);
ASSERT_LE(stats.speech_expand_rate, ref_stats.expand_rate);
}
@ -220,9 +221,9 @@ class NetEqDecodingTest : public ::testing::Test {
// NetEQ must be polled for data once every 10 ms. Thus, neither of the
// constants below can be changed.
static const int kTimeStepMs = 10;
static const int kBlockSize8kHz = kTimeStepMs * 8;
static const int kBlockSize16kHz = kTimeStepMs * 16;
static const int kBlockSize32kHz = kTimeStepMs * 32;
static const size_t kBlockSize8kHz = kTimeStepMs * 8;
static const size_t kBlockSize16kHz = kTimeStepMs * 16;
static const size_t kBlockSize32kHz = kTimeStepMs * 32;
static const size_t kMaxBlockSize = kBlockSize32kHz;
static const int kInitSampleRateHz = 8000;
@ -232,7 +233,7 @@ class NetEqDecodingTest : public ::testing::Test {
void SelectDecoders(NetEqDecoder* used_codec);
void LoadDecoders();
void OpenInputFile(const std::string &rtp_file);
void Process(int* out_len);
void Process(size_t* out_len);
void DecodeAndCompare(const std::string& rtp_file,
const std::string& ref_file,
const std::string& stat_ref_file,
@ -272,9 +273,9 @@ class NetEqDecodingTest : public ::testing::Test {
// Allocating the static const so that it can be passed by reference.
const int NetEqDecodingTest::kTimeStepMs;
const int NetEqDecodingTest::kBlockSize8kHz;
const int NetEqDecodingTest::kBlockSize16kHz;
const int NetEqDecodingTest::kBlockSize32kHz;
const size_t NetEqDecodingTest::kBlockSize8kHz;
const size_t NetEqDecodingTest::kBlockSize16kHz;
const size_t NetEqDecodingTest::kBlockSize32kHz;
const size_t NetEqDecodingTest::kMaxBlockSize;
const int NetEqDecodingTest::kInitSampleRateHz;
@ -334,7 +335,7 @@ void NetEqDecodingTest::OpenInputFile(const std::string &rtp_file) {
rtp_source_.reset(test::RtpFileSource::Create(rtp_file));
}
void NetEqDecodingTest::Process(int* out_len) {
void NetEqDecodingTest::Process(size_t* out_len) {
// Check if time to receive.
while (packet_ && sim_clock_ >= packet_->time_ms()) {
if (packet_->payload_length_bytes() > 0) {
@ -358,7 +359,7 @@ void NetEqDecodingTest::Process(int* out_len) {
ASSERT_TRUE((*out_len == kBlockSize8kHz) ||
(*out_len == kBlockSize16kHz) ||
(*out_len == kBlockSize32kHz));
output_sample_rate_ = *out_len / 10 * 1000;
output_sample_rate_ = static_cast<int>(*out_len / 10 * 1000);
// Increase time.
sim_clock_ += kTimeStepMs;
@ -394,7 +395,7 @@ void NetEqDecodingTest::DecodeAndCompare(const std::string& rtp_file,
std::ostringstream ss;
ss << "Lap number " << i++ << " in DecodeAndCompare while loop";
SCOPED_TRACE(ss.str()); // Print out the parameter values on failure.
int out_len = 0;
size_t out_len = 0;
ASSERT_NO_FATAL_FAILURE(Process(&out_len));
ASSERT_NO_FATAL_FAILURE(ref_files.ProcessReference(out_data_, out_len));
@ -498,7 +499,7 @@ TEST_F(NetEqDecodingTestFaxMode, TestFrameWaitingTimeStatistics) {
}
// Pull out all data.
for (size_t i = 0; i < num_frames; ++i) {
int out_len;
size_t out_len;
int num_channels;
NetEqOutputType type;
ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
@ -536,7 +537,7 @@ TEST_F(NetEqDecodingTestFaxMode, TestFrameWaitingTimeStatistics) {
rtp_info,
reinterpret_cast<uint8_t*>(payload),
kPayloadBytes, 0));
int out_len;
size_t out_len;
int num_channels;
NetEqOutputType type;
ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
@ -566,7 +567,7 @@ TEST_F(NetEqDecodingTest, TestAverageInterArrivalTimeNegative) {
}
// Pull out data once.
int out_len;
size_t out_len;
int num_channels;
NetEqOutputType type;
ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
@ -597,7 +598,7 @@ TEST_F(NetEqDecodingTest, TestAverageInterArrivalTimePositive) {
}
// Pull out data once.
int out_len;
size_t out_len;
int num_channels;
NetEqOutputType type;
ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
@ -622,7 +623,7 @@ void NetEqDecodingTest::LongCngWithClockDrift(double drift_factor,
const size_t kPayloadBytes = kSamples * 2;
double next_input_time_ms = 0.0;
double t_ms;
int out_len;
size_t out_len;
int num_channels;
NetEqOutputType type;
@ -854,7 +855,7 @@ TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(DecoderError)) {
out_data_[i] = 1;
}
int num_channels;
int samples_per_channel;
size_t samples_per_channel;
EXPECT_EQ(NetEq::kFail,
neteq_->GetAudio(kMaxBlockSize, out_data_,
&samples_per_channel, &num_channels, &type));
@ -887,7 +888,7 @@ TEST_F(NetEqDecodingTest, GetAudioBeforeInsertPacket) {
out_data_[i] = 1;
}
int num_channels;
int samples_per_channel;
size_t samples_per_channel;
EXPECT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_,
&samples_per_channel,
&num_channels, &type));
@ -908,7 +909,7 @@ class NetEqBgnTest : public NetEqDecodingTest {
bool should_be_faded) = 0;
void CheckBgn(int sampling_rate_hz) {
int16_t expected_samples_per_channel = 0;
size_t expected_samples_per_channel = 0;
uint8_t payload_type = 0xFF; // Invalid.
if (sampling_rate_hz == 8000) {
expected_samples_per_channel = kBlockSize8kHz;
@ -932,7 +933,7 @@ class NetEqBgnTest : public NetEqDecodingTest {
ASSERT_TRUE(input.Init(
webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"),
10 * sampling_rate_hz, // Max 10 seconds loop length.
static_cast<size_t>(expected_samples_per_channel)));
expected_samples_per_channel));
// Payload of 10 ms of PCM16 32 kHz.
uint8_t payload[kBlockSize32kHz * sizeof(int16_t)];
@ -941,19 +942,18 @@ class NetEqBgnTest : public NetEqDecodingTest {
rtp_info.header.payloadType = payload_type;
int number_channels = 0;
int samples_per_channel = 0;
size_t samples_per_channel = 0;
uint32_t receive_timestamp = 0;
for (int n = 0; n < 10; ++n) { // Insert few packets and get audio.
int16_t enc_len_bytes = WebRtcPcm16b_Encode(
size_t enc_len_bytes = WebRtcPcm16b_Encode(
input.GetNextBlock(), expected_samples_per_channel, payload);
ASSERT_EQ(enc_len_bytes, expected_samples_per_channel * 2);
number_channels = 0;
samples_per_channel = 0;
ASSERT_EQ(0,
neteq_->InsertPacket(rtp_info, payload,
static_cast<size_t>(enc_len_bytes),
neteq_->InsertPacket(rtp_info, payload, enc_len_bytes,
receive_timestamp));
ASSERT_EQ(0,
neteq_->GetAudio(kBlockSize32kHz,
@ -1009,7 +1009,7 @@ class NetEqBgnTest : public NetEqDecodingTest {
if (type == kOutputPLCtoCNG) {
plc_to_cng = true;
double sum_squared = 0;
for (int k = 0; k < number_channels * samples_per_channel; ++k)
for (size_t k = 0; k < number_channels * samples_per_channel; ++k)
sum_squared += output[k] * output[k];
TestCondition(sum_squared, n > kFadingThreshold);
} else {
@ -1168,7 +1168,7 @@ TEST_F(NetEqDecodingTest, SyncPacketDecode) {
// actual decoded values.
NetEqOutputType output_type;
int num_channels;
int samples_per_channel;
size_t samples_per_channel;
uint32_t receive_timestamp = 0;
for (int n = 0; n < 100; ++n) {
ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes,
@ -1246,7 +1246,7 @@ TEST_F(NetEqDecodingTest, SyncPacketBufferSizeAndOverridenByNetworkPackets) {
// actual decoded values.
NetEqOutputType output_type;
int num_channels;
int samples_per_channel;
size_t samples_per_channel;
uint32_t receive_timestamp = 0;
int algorithmic_frame_delay = algorithmic_delay_ms_ / 10 + 1;
for (int n = 0; n < algorithmic_frame_delay; ++n) {
@ -1315,7 +1315,7 @@ void NetEqDecodingTest::WrapTest(uint16_t start_seq_no,
double next_input_time_ms = 0.0;
int16_t decoded[kBlockSize16kHz];
int num_channels;
int samples_per_channel;
size_t samples_per_channel;
NetEqOutputType output_type;
uint32_t receive_timestamp = 0;
@ -1418,7 +1418,7 @@ void NetEqDecodingTest::DuplicateCng() {
algorithmic_delay_ms_ * kSampleRateKhz, 5 * kSampleRateKhz / 8);
// Insert three speech packets. Three are needed to get the frame length
// correct.
int out_len;
size_t out_len;
int num_channels;
NetEqOutputType type;
uint8_t payload[kPayloadBytes] = {0};
@ -1515,7 +1515,7 @@ TEST_F(NetEqDecodingTest, CngFirst) {
timestamp += kCngPeriodSamples;
// Pull audio once and make sure CNG is played.
int out_len;
size_t out_len;
int num_channels;
NetEqOutputType type;
ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,