webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc - Issue 1725143003: Changed AudioEncoder::Encode to take an rtc::Buffer* instead of uint8_t* and a maximum size.

Unified Diff: webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc

Issue 1725143003: Changed AudioEncoder::Encode to take an rtc::Buffer* instead of uint8_t* and a maximum size. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: Added more fixes for override hiding in AudioEncoder implementations. Created 4 years, 10 months ago

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Index: webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc

diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc

index f169d0500872c18a868d2a74d803548b9f4af03a..00c1f04f77bf89f381313dfa931595b549bebc0f 100644

--- a/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc

+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc

@@ -774,8 +774,7 @@ class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi {

bool CbReceiveImpl() {

SleepMs(1);

- const size_t max_encoded_bytes = isac_encoder_->MaxEncodedBytes();

- std::unique_ptr<uint8_t[]> encoded(new uint8_t[max_encoded_bytes]);

+ rtc::Buffer encoded;

AudioEncoder::EncodedInfo info;

{

rtc::CritScope lock(&crit_sect_);

@@ -790,7 +789,7 @@ class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi {

while (info.encoded_bytes == 0) {

info =

isac_encoder_->Encode(input_timestamp, audio_loop_.GetNextBlock(),

- max_encoded_bytes, encoded.get());

+ &encoded);

input_timestamp += 160; // 10 ms at 16 kHz.

}

EXPECT_EQ(rtp_header_.header.timestamp + kPacketSizeSamples,

@@ -801,7 +800,7 @@ class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi {

// Now we're not holding the crit sect when calling ACM.

// Insert into ACM.

- EXPECT_EQ(0, acm_->IncomingPacket(encoded.get(), info.encoded_bytes,

+ EXPECT_EQ(0, acm_->IncomingPacket(encoded.data(), info.encoded_bytes,

rtp_header_));

// Pull audio.

@@ -1633,9 +1632,6 @@ TEST_F(AcmSenderBitExactnessOldApi, External_Pcmu_20ms) {

MockAudioEncoder mock_encoder;

// Set expectations on the mock encoder and also delegate the calls to the

// real encoder.

- EXPECT_CALL(mock_encoder, MaxEncodedBytes())

- .Times(AtLeast(1))

- .WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::MaxEncodedBytes));

EXPECT_CALL(mock_encoder, SampleRateHz())

.Times(AtLeast(1))

.WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::SampleRateHz));

@@ -1652,9 +1648,14 @@ TEST_F(AcmSenderBitExactnessOldApi, External_Pcmu_20ms) {

EXPECT_CALL(mock_encoder, GetTargetBitrate())

.Times(AtLeast(1))

.WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::GetTargetBitrate));

- EXPECT_CALL(mock_encoder, EncodeInternal(_, _, _, _))

+ EXPECT_CALL(mock_encoder, EncodeInternal(_, _, _))

.Times(AtLeast(1))

- .WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::EncodeInternal));

+ .WillRepeatedly(Invoke(&encoder,

+ static_cast<

+ AudioEncoder::EncodedInfo(AudioEncoder::*)(

+ uint32_t,

+ rtc::ArrayView<const int16_t>,

+ rtc::Buffer*)>(&AudioEncoderPcmU::Encode)));

EXPECT_CALL(mock_encoder, SetFec(_))

.Times(AtLeast(1))

.WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::SetFec));