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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

58
webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
View File

@ -384,7 +384,7 @@ TEST_F(NetEqImplTest, InsertPacketsUntilBufferIsFull) {
neteq_->RegisterPayloadType(kDecoderPCM16B, kPayloadType));
// Insert packets. The buffer should not flush.
for (int i = 1; i <= config_.max_packets_in_buffer; ++i) {
for (size_t i = 1; i <= config_.max_packets_in_buffer; ++i) {
EXPECT_EQ(NetEq::kOK,
neteq_->InsertPacket(
rtp_header, payload, kPayloadLengthBytes, kReceiveTime));
@ -398,7 +398,7 @@ TEST_F(NetEqImplTest, InsertPacketsUntilBufferIsFull) {
EXPECT_EQ(NetEq::kOK,
neteq_->InsertPacket(
rtp_header, payload, kPayloadLengthBytes, kReceiveTime));
EXPECT_EQ(1, packet_buffer_->NumPacketsInBuffer());
EXPECT_EQ(1u, packet_buffer_->NumPacketsInBuffer());
const RTPHeader* test_header = packet_buffer_->NextRtpHeader();
EXPECT_EQ(rtp_header.header.timestamp, test_header->timestamp);
EXPECT_EQ(rtp_header.header.sequenceNumber, test_header->sequenceNumber);
@ -413,7 +413,8 @@ TEST_F(NetEqImplTest, VerifyTimestampPropagation) {
const uint8_t kPayloadType = 17; // Just an arbitrary number.
const uint32_t kReceiveTime = 17; // Value doesn't matter for this test.
const int kSampleRateHz = 8000;
const int kPayloadLengthSamples = 10 * kSampleRateHz / 1000; // 10 ms.
const size_t kPayloadLengthSamples =
static_cast<size_t>(10 * kSampleRateHz / 1000); // 10 ms.
const size_t kPayloadLengthBytes = kPayloadLengthSamples;
uint8_t payload[kPayloadLengthBytes] = {0};
WebRtcRTPHeader rtp_header;
@ -466,9 +467,9 @@ TEST_F(NetEqImplTest, VerifyTimestampPropagation) {
rtp_header, payload, kPayloadLengthBytes, kReceiveTime));
// Pull audio once.
const int kMaxOutputSize = 10 * kSampleRateHz / 1000;
const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
int16_t output[kMaxOutputSize];
int samples_per_channel;
size_t samples_per_channel;
int num_channels;
NetEqOutputType type;
EXPECT_EQ(
@ -480,7 +481,8 @@ TEST_F(NetEqImplTest, VerifyTimestampPropagation) {
EXPECT_EQ(kOutputNormal, type);
// Start with a simple check that the fake decoder is behaving as expected.
EXPECT_EQ(kPayloadLengthSamples, decoder_.next_value() - 1);
EXPECT_EQ(kPayloadLengthSamples,
static_cast<size_t>(decoder_.next_value() - 1));
// The value of the last of the output samples is the same as the number of
// samples played from the decoded packet. Thus, this number + the RTP
@ -500,7 +502,7 @@ TEST_F(NetEqImplTest, VerifyTimestampPropagation) {
// Check that the number of samples still to play from the sync buffer add
// up with what was already played out.
EXPECT_EQ(kPayloadLengthSamples - output[samples_per_channel - 1],
static_cast<int>(sync_buffer->FutureLength()));
sync_buffer->FutureLength());
}
TEST_F(NetEqImplTest, ReorderedPacket) {
@ -510,7 +512,8 @@ TEST_F(NetEqImplTest, ReorderedPacket) {
const uint8_t kPayloadType = 17; // Just an arbitrary number.
const uint32_t kReceiveTime = 17; // Value doesn't matter for this test.
const int kSampleRateHz = 8000;
const int kPayloadLengthSamples = 10 * kSampleRateHz / 1000; // 10 ms.
const size_t kPayloadLengthSamples =
static_cast<size_t>(10 * kSampleRateHz / 1000); // 10 ms.
const size_t kPayloadLengthBytes = kPayloadLengthSamples;
uint8_t payload[kPayloadLengthBytes] = {0};
WebRtcRTPHeader rtp_header;
@ -544,9 +547,9 @@ TEST_F(NetEqImplTest, ReorderedPacket) {
rtp_header, payload, kPayloadLengthBytes, kReceiveTime));
// Pull audio once.
const int kMaxOutputSize = 10 * kSampleRateHz / 1000;
const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
int16_t output[kMaxOutputSize];
int samples_per_channel;
size_t samples_per_channel;
int num_channels;
NetEqOutputType type;
EXPECT_EQ(
@ -606,7 +609,8 @@ TEST_F(NetEqImplTest, FirstPacketUnknown) {
const uint8_t kPayloadType = 17; // Just an arbitrary number.
const uint32_t kReceiveTime = 17; // Value doesn't matter for this test.
const int kSampleRateHz = 8000;
const int kPayloadLengthSamples = 10 * kSampleRateHz / 1000; // 10 ms.
const size_t kPayloadLengthSamples =
static_cast<size_t>(10 * kSampleRateHz / 1000); // 10 ms.
const size_t kPayloadLengthBytes = kPayloadLengthSamples;
uint8_t payload[kPayloadLengthBytes] = {0};
WebRtcRTPHeader rtp_header;
@ -623,9 +627,9 @@ TEST_F(NetEqImplTest, FirstPacketUnknown) {
EXPECT_EQ(NetEq::kUnknownRtpPayloadType, neteq_->LastError());
// Pull audio once.
const int kMaxOutputSize = 10 * kSampleRateHz / 1000;
const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
int16_t output[kMaxOutputSize];
int samples_per_channel;
size_t samples_per_channel;
int num_channels;
NetEqOutputType type;
EXPECT_EQ(NetEq::kOK,
@ -641,7 +645,7 @@ TEST_F(NetEqImplTest, FirstPacketUnknown) {
neteq_->RegisterPayloadType(kDecoderPCM16B, kPayloadType));
// Insert 10 packets.
for (int i = 0; i < 10; ++i) {
for (size_t i = 0; i < 10; ++i) {
rtp_header.header.sequenceNumber++;
rtp_header.header.timestamp += kPayloadLengthSamples;
EXPECT_EQ(NetEq::kOK,
@ -651,7 +655,7 @@ TEST_F(NetEqImplTest, FirstPacketUnknown) {
}
// Pull audio repeatedly and make sure we get normal output, that is not PLC.
for (int i = 0; i < 3; ++i) {
for (size_t i = 0; i < 3; ++i) {
EXPECT_EQ(NetEq::kOK,
neteq_->GetAudio(kMaxOutputSize, output, &samples_per_channel,
&num_channels, &type));
@ -672,8 +676,9 @@ TEST_F(NetEqImplTest, CodecInternalCng) {
const uint8_t kPayloadType = 17; // Just an arbitrary number.
const uint32_t kReceiveTime = 17; // Value doesn't matter for this test.
const int kSampleRateKhz = 48;
const int kPayloadLengthSamples = 20 * kSampleRateKhz; // 20 ms.
const int kPayloadLengthBytes = 10;
const size_t kPayloadLengthSamples =
static_cast<size_t>(20 * kSampleRateKhz); // 20 ms.
const size_t kPayloadLengthBytes = 10;
uint8_t payload[kPayloadLengthBytes] = {0};
int16_t dummy_output[kPayloadLengthSamples] = {0};
@ -736,9 +741,9 @@ TEST_F(NetEqImplTest, CodecInternalCng) {
neteq_->InsertPacket(
rtp_header, payload, kPayloadLengthBytes, kReceiveTime));
const int kMaxOutputSize = 10 * kSampleRateKhz;
const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateKhz);
int16_t output[kMaxOutputSize];
int samples_per_channel;
size_t samples_per_channel;
int num_channels;
uint32_t timestamp;
uint32_t last_timestamp;
@ -762,7 +767,7 @@ TEST_F(NetEqImplTest, CodecInternalCng) {
&num_channels, &type));
EXPECT_TRUE(neteq_->GetPlayoutTimestamp(&last_timestamp));
for (int i = 1; i < 6; ++i) {
for (size_t i = 1; i < 6; ++i) {
ASSERT_EQ(kMaxOutputSize, samples_per_channel);
EXPECT_EQ(1, num_channels);
EXPECT_EQ(expected_type[i - 1], type);
@ -783,7 +788,7 @@ TEST_F(NetEqImplTest, CodecInternalCng) {
neteq_->InsertPacket(
rtp_header, payload, kPayloadLengthBytes, kReceiveTime));
for (int i = 6; i < 8; ++i) {
for (size_t i = 6; i < 8; ++i) {
ASSERT_EQ(kMaxOutputSize, samples_per_channel);
EXPECT_EQ(1, num_channels);
EXPECT_EQ(expected_type[i - 1], type);
@ -811,7 +816,8 @@ TEST_F(NetEqImplTest, UnsupportedDecoder) {
const uint32_t kReceiveTime = 17; // Value doesn't matter for this test.
const int kSampleRateHz = 8000;
const int kPayloadLengthSamples = 10 * kSampleRateHz / 1000; // 10 ms.
const size_t kPayloadLengthSamples =
static_cast<size_t>(10 * kSampleRateHz / 1000); // 10 ms.
const size_t kPayloadLengthBytes = 1;
uint8_t payload[kPayloadLengthBytes]= {0};
int16_t dummy_output[kPayloadLengthSamples * kChannels] = {0};
@ -852,7 +858,8 @@ TEST_F(NetEqImplTest, UnsupportedDecoder) {
dummy_output +
kPayloadLengthSamples * kChannels),
SetArgPointee<4>(AudioDecoder::kSpeech),
Return(kPayloadLengthSamples * kChannels)));
Return(static_cast<int>(
kPayloadLengthSamples * kChannels))));
EXPECT_CALL(decoder_, PacketDuration(Pointee(kSecondPayloadValue),
kPayloadLengthBytes))
@ -879,9 +886,10 @@ TEST_F(NetEqImplTest, UnsupportedDecoder) {
neteq_->InsertPacket(
rtp_header, payload, kPayloadLengthBytes, kReceiveTime));
const int kMaxOutputSize = 10 * kSampleRateHz / 1000 * kChannels;
const size_t kMaxOutputSize =
static_cast<size_t>(10 * kSampleRateHz / 1000 * kChannels);
int16_t output[kMaxOutputSize];
int samples_per_channel;
size_t samples_per_channel;
int num_channels;
NetEqOutputType type;