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red: handle opus dtx 400ms timestamp gap

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by not encoding redundancy. The timestamp gap of 400ms means a
rtp timestamp difference of 19200 which would overflow the 14 bit
RED timestamp difference field.

To test in Chrome, go to
  https://webrtc.github.io/samples/src/content/peerconnection/audio/
set `useDtx = true` in the console and be very quiet.

BUG=webrtc:13182,webrtc:11640

Change-Id: I1cedc7d846ac7ae821bb7cb5c0f37a17511ac727
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231940
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35005}
This commit is contained in:
Philipp Hancke
2021-09-15 13:10:38 +02:00
committed by WebRTC LUCI CQ
parent b01e6457fe
commit 1061686107
2 changed files with 32 additions and 2 deletions
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11
modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
View File

@ -24,6 +24,8 @@ namespace webrtc {
static constexpr const int kRedMaxPacketSize =
1 << 10; // RED packets must be less than 1024 bytes to fit the 10 bit
// block length.
static constexpr const size_t kRedMaxTimestampDelta =
1 << 14; // RED packets can encode a timestamp delta of 14 bits.
static constexpr const size_t kAudioMaxRtpPacketLen =
1200; // The typical MTU is 1200 bytes.
@ -100,7 +102,7 @@ AudioEncoder::EncodedInfo AudioEncoderCopyRed::EncodeImpl(
RTC_CHECK(info.redundant.empty()) << "Cannot use nested redundant encoders.";
RTC_DCHECK_EQ(primary_encoded_.size(), info.encoded_bytes);
if (info.encoded_bytes == 0 || info.encoded_bytes > kRedMaxPacketSize) {
if (info.encoded_bytes == 0 || info.encoded_bytes >= kRedMaxPacketSize) {
return info;
}
RTC_DCHECK_GT(max_packet_length_, info.encoded_bytes);
@ -110,7 +112,9 @@ AudioEncoder::EncodedInfo AudioEncoderCopyRed::EncodeImpl(
auto it = redundant_encodings_.begin();
// Determine how much redundancy we can fit into our packet by
// iterating forward.
// iterating forward. This is determined both by the length as well
// as the timestamp difference. The latter can occur with opus DTX which
// has timestamp gaps of 400ms which exceeds REDs timestamp delta field size.
for (; it != redundant_encodings_.end(); it++) {
if (bytes_available < kRedHeaderLength + it->first.encoded_bytes) {
break;
@ -118,6 +122,9 @@ AudioEncoder::EncodedInfo AudioEncoderCopyRed::EncodeImpl(
if (it->first.encoded_bytes == 0) {
break;
}
if (rtp_timestamp - it->first.encoded_timestamp >= kRedMaxTimestampDelta) {
break;
}
bytes_available -= kRedHeaderLength + it->first.encoded_bytes;
header_length_bytes += kRedHeaderLength;
}

23
modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
View File

@ -583,6 +583,29 @@ TEST_F(AudioEncoderCopyRedTest, RespectsPayloadMTU) {
EXPECT_EQ(encoded_.size(), 5u + 500u + 400u);
}
TEST_F(AudioEncoderCopyRedTest, LargeTimestampGap) {
const int primary_payload_type = red_payload_type_ + 1;
AudioEncoder::EncodedInfo info;
info.encoded_bytes = 100;
info.encoded_timestamp = timestamp_;
info.payload_type = primary_payload_type;
EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
.WillOnce(Invoke(MockAudioEncoder::FakeEncoding(info)));
Encode();
// Update timestamp to simulate a 400ms gap like the one
// opus DTX causes.
timestamp_ += 19200;
info.encoded_timestamp = timestamp_; // update timestamp.
info.encoded_bytes = 200;
EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
.WillOnce(Invoke(MockAudioEncoder::FakeEncoding(info)));
Encode();
// The old packet will be dropped.
EXPECT_EQ(encoded_.size(), 1u + 200u);
}
#if GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
// This test fixture tests various error conditions that makes the