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Remove the type parameter to NetEq::GetAudio

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The type is included in the AudioFrame output parameter.

Rename the type NetEqOutputType to just OutputType, since it is now
internal to NetEq.

BUG=webrtc:5607

Review URL: https://codereview.webrtc.org/1769883002

Cr-Commit-Position: refs/heads/master@{#11903}
This commit is contained in:
henrik.lundin
2016-03-08 02:37:57 -08:00
committed by Commit bot
parent 500c04bc86
commit 55480f5efa
14 changed files with 131 additions and 165 deletions
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93
webrtc/modules/audio_coding/neteq/neteq_unittest.cc
View File

@ -426,8 +426,7 @@ void NetEqDecodingTest::Process() {
}
// Get audio from NetEq.
NetEqOutputType type;
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &type));
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
ASSERT_TRUE((out_frame_.samples_per_channel_ == kBlockSize8kHz) ||
(out_frame_.samples_per_channel_ == kBlockSize16kHz) ||
(out_frame_.samples_per_channel_ == kBlockSize32kHz) ||
@ -611,8 +610,7 @@ TEST_F(NetEqDecodingTestFaxMode, TestFrameWaitingTimeStatistics) {
}
// Pull out all data.
for (size_t i = 0; i < num_frames; ++i) {
NetEqOutputType type;
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &type));
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
}
@ -653,8 +651,7 @@ TEST_F(NetEqDecodingTest, TestAverageInterArrivalTimeNegative) {
}
// Pull out data once.
NetEqOutputType type;
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &type));
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
}
@ -681,8 +678,7 @@ TEST_F(NetEqDecodingTest, TestAverageInterArrivalTimePositive) {
}
// Pull out data once.
NetEqOutputType type;
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &type));
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
}
@ -703,7 +699,6 @@ void NetEqDecodingTest::LongCngWithClockDrift(double drift_factor,
const size_t kPayloadBytes = kSamples * 2;
double next_input_time_ms = 0.0;
double t_ms;
NetEqOutputType type;
// Insert speech for 5 seconds.
const int kSpeechDurationMs = 5000;
@ -720,11 +715,11 @@ void NetEqDecodingTest::LongCngWithClockDrift(double drift_factor,
next_input_time_ms += static_cast<double>(kFrameSizeMs) * drift_factor;
}
// Pull out data once.
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &type));
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
}
EXPECT_EQ(kOutputNormal, type);
EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
int32_t delay_before = timestamp - PlayoutTimestamp();
// Insert CNG for 1 minute (= 60000 ms).
@ -747,11 +742,11 @@ void NetEqDecodingTest::LongCngWithClockDrift(double drift_factor,
next_input_time_ms += static_cast<double>(kCngPeriodMs) * drift_factor;
}
// Pull out data once.
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &type));
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
}
EXPECT_EQ(kOutputCNG, type);
EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
if (network_freeze_ms > 0) {
// First keep pulling audio for |network_freeze_ms| without inserting
@ -760,9 +755,9 @@ void NetEqDecodingTest::LongCngWithClockDrift(double drift_factor,
const double loop_end_time = t_ms + network_freeze_ms;
for (; t_ms < loop_end_time; t_ms += 10) {
// Pull out data once.
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &type));
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
EXPECT_EQ(kOutputCNG, type);
EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
}
bool pull_once = pull_audio_during_freeze;
// If |pull_once| is true, GetAudio will be called once half-way through
@ -772,9 +767,9 @@ void NetEqDecodingTest::LongCngWithClockDrift(double drift_factor,
if (pull_once && next_input_time_ms >= pull_time_ms) {
pull_once = false;
// Pull out data once.
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &type));
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
EXPECT_EQ(kOutputCNG, type);
EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
t_ms += 10;
}
// Insert one CNG frame each 100 ms.
@ -793,7 +788,7 @@ void NetEqDecodingTest::LongCngWithClockDrift(double drift_factor,
// Insert speech again until output type is speech.
double speech_restart_time_ms = t_ms;
while (type != kOutputNormal) {
while (out_frame_.speech_type_ != AudioFrame::kNormalSpeech) {
// Each turn in this for loop is 10 ms.
while (next_input_time_ms <= t_ms) {
// Insert one 30 ms speech frame.
@ -806,7 +801,7 @@ void NetEqDecodingTest::LongCngWithClockDrift(double drift_factor,
next_input_time_ms += kFrameSizeMs * drift_factor;
}
// Pull out data once.
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &type));
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
// Increase clock.
t_ms += 10;
@ -927,13 +922,12 @@ TEST_F(NetEqDecodingTest, MAYBE_DecoderError) {
PopulateRtpInfo(0, 0, &rtp_info);
rtp_info.header.payloadType = 103; // iSAC, but the payload is invalid.
EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
NetEqOutputType type;
// Set all of |out_data_| to 1, and verify that it was set to 0 by the call
// to GetAudio.
for (size_t i = 0; i < AudioFrame::kMaxDataSizeSamples; ++i) {
out_frame_.data_[i] = 1;
}
EXPECT_EQ(NetEq::kFail, neteq_->GetAudio(&out_frame_, &type));
EXPECT_EQ(NetEq::kFail, neteq_->GetAudio(&out_frame_));
// Verify that there is a decoder error to check.
EXPECT_EQ(NetEq::kDecoderErrorCode, neteq_->LastError());
@ -965,13 +959,12 @@ TEST_F(NetEqDecodingTest, MAYBE_DecoderError) {
}
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 (size_t i = 0; i < AudioFrame::kMaxDataSizeSamples; ++i) {
out_frame_.data_[i] = 1;
}
EXPECT_EQ(0, neteq_->GetAudio(&out_frame_, &type));
EXPECT_EQ(0, neteq_->GetAudio(&out_frame_));
// Verify that the first block of samples is set to 0.
static const int kExpectedOutputLength =
kInitSampleRateHz / 100; // 10 ms at initial sample rate.
@ -1006,7 +999,6 @@ class NetEqBgnTest : public NetEqDecodingTest {
ASSERT_TRUE(false); // Unsupported test case.
}
NetEqOutputType type;
AudioFrame output;
test::AudioLoop input;
// We are using the same 32 kHz input file for all tests, regardless of
@ -1035,10 +1027,10 @@ class NetEqBgnTest : public NetEqDecodingTest {
payload, enc_len_bytes),
receive_timestamp));
output.Reset();
ASSERT_EQ(0, neteq_->GetAudio(&output, &type));
ASSERT_EQ(0, neteq_->GetAudio(&output));
ASSERT_EQ(1u, output.num_channels_);
ASSERT_EQ(expected_samples_per_channel, output.samples_per_channel_);
ASSERT_EQ(kOutputNormal, type);
ASSERT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
// Next packet.
rtp_info.header.timestamp += expected_samples_per_channel;
@ -1051,7 +1043,7 @@ class NetEqBgnTest : public NetEqDecodingTest {
// Get audio without inserting packets, expecting PLC and PLC-to-CNG. Pull
// one frame without checking speech-type. This is the first frame pulled
// without inserting any packet, and might not be labeled as PLC.
ASSERT_EQ(0, neteq_->GetAudio(&output, &type));
ASSERT_EQ(0, neteq_->GetAudio(&output));
ASSERT_EQ(1u, output.num_channels_);
ASSERT_EQ(expected_samples_per_channel, output.samples_per_channel_);
@ -1066,10 +1058,10 @@ class NetEqBgnTest : public NetEqDecodingTest {
for (int n = 0; n < kFadingThreshold + kNumPlcToCngTestFrames; ++n) {
output.Reset();
memset(output.data_, 1, sizeof(output.data_)); // Set to non-zero.
ASSERT_EQ(0, neteq_->GetAudio(&output, &type));
ASSERT_EQ(0, neteq_->GetAudio(&output));
ASSERT_EQ(1u, output.num_channels_);
ASSERT_EQ(expected_samples_per_channel, output.samples_per_channel_);
if (type == kOutputPLCtoCNG) {
if (output.speech_type_ == AudioFrame::kPLCCNG) {
plc_to_cng = true;
double sum_squared = 0;
for (size_t k = 0;
@ -1077,7 +1069,7 @@ class NetEqBgnTest : public NetEqDecodingTest {
sum_squared += output.data_[k] * output.data_[k];
TestCondition(sum_squared, n > kFadingThreshold);
} else {
EXPECT_EQ(kOutputPLC, type);
EXPECT_EQ(AudioFrame::kPLC, output.speech_type_);
}
}
EXPECT_TRUE(plc_to_cng); // Just to be sure that PLC-to-CNG has occurred.
@ -1239,11 +1231,10 @@ TEST_F(NetEqDecodingTest, SyncPacketDecode) {
}
// Insert some packets which decode to noise. We are not interested in
// actual decoded values.
NetEqOutputType output_type;
uint32_t receive_timestamp = 0;
for (int n = 0; n < 100; ++n) {
ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, receive_timestamp));
ASSERT_EQ(0, neteq_->GetAudio(&output, &output_type));
ASSERT_EQ(0, neteq_->GetAudio(&output));
ASSERT_EQ(kBlockSize16kHz, output.samples_per_channel_);
ASSERT_EQ(1u, output.num_channels_);
@ -1259,7 +1250,7 @@ TEST_F(NetEqDecodingTest, SyncPacketDecode) {
// Insert sync-packets, the decoded sequence should be all-zero.
for (int n = 0; n < kNumSyncPackets; ++n) {
ASSERT_EQ(0, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
ASSERT_EQ(0, neteq_->GetAudio(&output, &output_type));
ASSERT_EQ(0, neteq_->GetAudio(&output));
ASSERT_EQ(kBlockSize16kHz, output.samples_per_channel_);
ASSERT_EQ(1u, output.num_channels_);
if (n > algorithmic_frame_delay) {
@ -1275,7 +1266,7 @@ TEST_F(NetEqDecodingTest, SyncPacketDecode) {
// network statistics would show some packet loss.
for (int n = 0; n <= algorithmic_frame_delay + 10; ++n) {
ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, receive_timestamp));
ASSERT_EQ(0, neteq_->GetAudio(&output, &output_type));
ASSERT_EQ(0, neteq_->GetAudio(&output));
if (n >= algorithmic_frame_delay + 1) {
// Expect that this frame contain samples from regular RTP.
EXPECT_TRUE(IsAllNonZero(
@ -1309,12 +1300,11 @@ TEST_F(NetEqDecodingTest, SyncPacketBufferSizeAndOverridenByNetworkPackets) {
}
// Insert some packets which decode to noise. We are not interested in
// actual decoded values.
NetEqOutputType output_type;
uint32_t receive_timestamp = 0;
int algorithmic_frame_delay = algorithmic_delay_ms_ / 10 + 1;
for (int n = 0; n < algorithmic_frame_delay; ++n) {
ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, receive_timestamp));
ASSERT_EQ(0, neteq_->GetAudio(&output, &output_type));
ASSERT_EQ(0, neteq_->GetAudio(&output));
ASSERT_EQ(kBlockSize16kHz, output.samples_per_channel_);
ASSERT_EQ(1u, output.num_channels_);
rtp_info.header.sequenceNumber++;
@ -1351,7 +1341,7 @@ TEST_F(NetEqDecodingTest, SyncPacketBufferSizeAndOverridenByNetworkPackets) {
// Decode.
for (int n = 0; n < kNumSyncPackets; ++n) {
ASSERT_EQ(0, neteq_->GetAudio(&output, &output_type));
ASSERT_EQ(0, neteq_->GetAudio(&output));
ASSERT_EQ(kBlockSize16kHz, output.samples_per_channel_);
ASSERT_EQ(1u, output.num_channels_);
EXPECT_TRUE(IsAllNonZero(
@ -1418,8 +1408,7 @@ void NetEqDecodingTest::WrapTest(uint16_t start_seq_no,
}
// Pull out data once.
AudioFrame output;
NetEqOutputType output_type;
ASSERT_EQ(0, neteq_->GetAudio(&output, &output_type));
ASSERT_EQ(0, neteq_->GetAudio(&output));
ASSERT_EQ(kBlockSize16kHz, output.samples_per_channel_);
ASSERT_EQ(1u, output.num_channels_);
@ -1471,7 +1460,6 @@ void NetEqDecodingTest::DuplicateCng() {
algorithmic_delay_ms_ * kSampleRateKhz, 5 * kSampleRateKhz / 8);
// Insert three speech packets. Three are needed to get the frame length
// correct.
NetEqOutputType type;
uint8_t payload[kPayloadBytes] = {0};
WebRtcRTPHeader rtp_info;
for (int i = 0; i < 3; ++i) {
@ -1481,11 +1469,11 @@ void NetEqDecodingTest::DuplicateCng() {
timestamp += kSamples;
// Pull audio once.
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &type));
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
}
// Verify speech output.
EXPECT_EQ(kOutputNormal, type);
EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
// Insert same CNG packet twice.
const int kCngPeriodMs = 100;
@ -1498,9 +1486,9 @@ void NetEqDecodingTest::DuplicateCng() {
rtp_info, rtc::ArrayView<const uint8_t>(payload, payload_len), 0));
// Pull audio once and make sure CNG is played.
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &type));
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
EXPECT_EQ(kOutputCNG, type);
EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
EXPECT_EQ(timestamp - algorithmic_delay_samples, PlayoutTimestamp());
// Insert the same CNG packet again. Note that at this point it is old, since
@ -1512,9 +1500,9 @@ void NetEqDecodingTest::DuplicateCng() {
// Pull audio until we have played |kCngPeriodMs| of CNG. Start at 10 ms since
// we have already pulled out CNG once.
for (int cng_time_ms = 10; cng_time_ms < kCngPeriodMs; cng_time_ms += 10) {
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &type));
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
EXPECT_EQ(kOutputCNG, type);
EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
EXPECT_EQ(timestamp - algorithmic_delay_samples,
PlayoutTimestamp());
}
@ -1526,9 +1514,9 @@ void NetEqDecodingTest::DuplicateCng() {
ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
// Pull audio once and verify that the output is speech again.
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &type));
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
EXPECT_EQ(kOutputNormal, type);
EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
EXPECT_EQ(timestamp + kSamples - algorithmic_delay_samples,
PlayoutTimestamp());
}
@ -1564,10 +1552,9 @@ TEST_F(NetEqDecodingTest, CngFirst) {
timestamp += kCngPeriodSamples;
// Pull audio once and make sure CNG is played.
NetEqOutputType type;
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &type));
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
EXPECT_EQ(kOutputCNG, type);
EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
// Insert some speech packets.
for (int i = 0; i < 3; ++i) {
@ -1577,11 +1564,11 @@ TEST_F(NetEqDecodingTest, CngFirst) {
timestamp += kSamples;
// Pull audio once.
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &type));
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
}
// Verify speech output.
EXPECT_EQ(kOutputNormal, type);
EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
}
} // namespace webrtc