zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Revert "Remove legacy delay manger field trial and update default config."

Browse Source 
This reverts commit 93849d4b2a976b0a46059d6f74d9efd8f12eab92.

Reason for revert: AcmReceiverBitExactnessOldApi tests failing on MacARM64; first failing build https://ci.chromium.org/ui/p/webrtc/builders/ci/MacARM64%20M1%20Release/1038/overview
Example faliure
[ RUN      ] AcmReceiverBitExactnessOldApi.8kHzOutput
...
(rtp_file_reader.cc:165): Failed to read
../../modules/audio_coding/acm2/audio_coding_module_unittest.cc:912: Failure
Expected equality of these values:
  checksum_ref
    Which is: "636efe6d0a148f22c5383f356da3deac"
  checksum_string
    Which is: "6a288942d67e82076b38b17777cdaee4"

Original change's description:
> Remove legacy delay manger field trial and update default config.
>
> Bug: webrtc:10333
> Change-Id: I20e55d8d111d93657d1afe556fe3a325337c074c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232820
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35321}

TBR=ivoc@webrtc.org,jakobi@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com

Change-Id: I0bd3832aacba8dcd8e836650786cea20b4c083be
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10333
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237441
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35324}
This commit is contained in:
Olga Sharonova
2021-11-08 19:09:28 +00:00
committed by WebRTC LUCI CQ
parent 42a850d250
commit 46814941f2
11 changed files with 294 additions and 145 deletions
Download Patch File Download Diff File Expand all files Collapse all files

132
modules/audio_coding/neteq/neteq_impl_unittest.cc
View File

@ -1026,37 +1026,22 @@ TEST_F(NetEqImplTest, CodecInternalCng) {
EXPECT_CALL(mock_decoder, PacketDuration(nullptr, 0))
.WillRepeatedly(Return(rtc::checked_cast<int>(kPayloadLengthSamples)));
EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType,
SdpAudioFormat("opus", 48000, 2)));
// Pointee(x) verifies that first byte of the payload equals x, this makes it
// possible to verify that the correct payload is fed to Decode().
EXPECT_CALL(mock_decoder, DecodeInternal(Pointee(0), kPayloadLengthBytes,
kSampleRateKhz * 1000, _, _))
.WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
dummy_output + kPayloadLengthSamples),
SetArgPointee<4>(AudioDecoder::kSpeech),
Return(rtc::checked_cast<int>(kPayloadLengthSamples))));
struct Packet {
int sequence_number_delta;
int timestamp_delta;
AudioDecoder::SpeechType decoder_output_type;
};
std::vector<Packet> packets = {
{0, 0, AudioDecoder::kSpeech},
{1, kPayloadLengthSamples, AudioDecoder::kComfortNoise},
{2, 2 * kPayloadLengthSamples, AudioDecoder::kSpeech},
{1, kPayloadLengthSamples, AudioDecoder::kSpeech}};
EXPECT_CALL(mock_decoder, DecodeInternal(Pointee(1), kPayloadLengthBytes,
kSampleRateKhz * 1000, _, _))
.WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
dummy_output + kPayloadLengthSamples),
SetArgPointee<4>(AudioDecoder::kComfortNoise),
Return(rtc::checked_cast<int>(kPayloadLengthSamples))));
for (size_t i = 0; i < packets.size(); ++i) {
rtp_header.sequenceNumber += packets[i].sequence_number_delta;
rtp_header.timestamp += packets[i].timestamp_delta;
payload[0] = i;
EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
// Pointee(x) verifies that first byte of the payload equals x, this makes
// it possible to verify that the correct payload is fed to Decode().
EXPECT_CALL(mock_decoder, DecodeInternal(Pointee(i), kPayloadLengthBytes,
kSampleRateKhz * 1000, _, _))
.WillOnce(DoAll(SetArrayArgument<3>(
dummy_output, dummy_output + kPayloadLengthSamples),
SetArgPointee<4>(packets[i].decoder_output_type),
Return(rtc::checked_cast<int>(kPayloadLengthSamples))));
}
// Expect comfort noise to be returned by the decoder.
EXPECT_CALL(mock_decoder,
DecodeInternal(IsNull(), 0, kSampleRateKhz * 1000, _, _))
.WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
@ -1064,24 +1049,87 @@ TEST_F(NetEqImplTest, CodecInternalCng) {
SetArgPointee<4>(AudioDecoder::kComfortNoise),
Return(rtc::checked_cast<int>(kPayloadLengthSamples))));
std::vector<AudioFrame::SpeechType> expected_output = {
AudioFrame::kNormalSpeech, AudioFrame::kCNG, AudioFrame::kNormalSpeech};
size_t output_index = 0;
EXPECT_CALL(mock_decoder, DecodeInternal(Pointee(2), kPayloadLengthBytes,
kSampleRateKhz * 1000, _, _))
.WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
dummy_output + kPayloadLengthSamples),
SetArgPointee<4>(AudioDecoder::kSpeech),
Return(rtc::checked_cast<int>(kPayloadLengthSamples))));
int timeout_counter = 0;
while (!packet_buffer_->Empty()) {
ASSERT_LT(timeout_counter++, 20) << "Test timed out";
AudioFrame output;
bool muted;
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
if (output_index + 1 < expected_output.size() &&
output.speech_type_ == expected_output[output_index + 1]) {
++output_index;
EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType,
SdpAudioFormat("opus", 48000, 2)));
const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateKhz);
AudioFrame output;
AudioFrame::SpeechType expected_type[8] = {
AudioFrame::kNormalSpeech, AudioFrame::kNormalSpeech, AudioFrame::kCNG,
AudioFrame::kCNG, AudioFrame::kCNG, AudioFrame::kCNG,
AudioFrame::kNormalSpeech, AudioFrame::kNormalSpeech};
int expected_timestamp_increment[8] = {
-1, // will not be used.
10 * kSampleRateKhz,
-1,
-1, // timestamp will be empty during CNG mode; indicated by -1 here.
-1,
-1,
50 * kSampleRateKhz,
10 * kSampleRateKhz};
// Insert one packet (decoder will return speech).
EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
bool muted;
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
absl::optional<uint32_t> last_timestamp = neteq_->GetPlayoutTimestamp();
ASSERT_TRUE(last_timestamp);
// Insert second packet (decoder will return CNG).
payload[0] = 1;
rtp_header.sequenceNumber++;
rtp_header.timestamp += kPayloadLengthSamples;
EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
// Lambda for verifying the timestamps.
auto verify_timestamp = [&last_timestamp, &expected_timestamp_increment](
absl::optional<uint32_t> ts, size_t i) {
if (expected_timestamp_increment[i] == -1) {
// Expect to get an empty timestamp value during CNG and PLC.
EXPECT_FALSE(ts) << "i = " << i;
} else {
EXPECT_EQ(output.speech_type_, expected_output[output_index]);
ASSERT_TRUE(ts) << "i = " << i;
EXPECT_EQ(*ts, *last_timestamp + expected_timestamp_increment[i])
<< "i = " << i;
last_timestamp = ts;
}
};
for (size_t i = 1; i < 6; ++i) {
ASSERT_EQ(kMaxOutputSize, output.samples_per_channel_);
EXPECT_EQ(1u, output.num_channels_);
EXPECT_EQ(expected_type[i - 1], output.speech_type_);
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
SCOPED_TRACE("");
verify_timestamp(neteq_->GetPlayoutTimestamp(), i);
}
// Insert third packet, which leaves a gap from last packet.
payload[0] = 2;
rtp_header.sequenceNumber += 2;
rtp_header.timestamp += 2 * kPayloadLengthSamples;
EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
for (size_t i = 6; i < 8; ++i) {
ASSERT_EQ(kMaxOutputSize, output.samples_per_channel_);
EXPECT_EQ(1u, output.num_channels_);
EXPECT_EQ(expected_type[i - 1], output.speech_type_);
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
SCOPED_TRACE("");
verify_timestamp(neteq_->GetPlayoutTimestamp(), i);
}
// Now check the packet buffer, and make sure it is empty.
EXPECT_TRUE(packet_buffer_->Empty());
EXPECT_CALL(mock_decoder, Die());
}