With stereo decoding and mono packets produce mono after CN/PLC

The workaround in https://webrtc-review.googlesource.com/c/src/+/367740
is incomplete because it does not fix the issue for the first decoded
mono packet after CN/PLC. This CL extends the workaround to such a case
and adds a unit test for it.

Note: it was verified that the 2nd packet after CN/PLC is trivial
stereo.

Credits: jakobi@webrtc.org for raising the concern

Bug: webrtc:376493209
Change-Id: Ide27e411781693f14629cf9db8b6c0c0fc762a17
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368160
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43393}
This commit is contained in:
Alessio Bazzica 2024-11-13 13:49:30 +01:00 committed by WebRTC LUCI CQ
parent b7f5e7fb29
commit ebb11c4c87
3 changed files with 89 additions and 27 deletions

View File

@ -1505,6 +1505,7 @@ if (rtc_include_tests) {
"../../test:test_support",
"codecs/opus/test",
"codecs/opus/test:test_unittest",
"//testing/gmock",
"//testing/gtest",
"//third_party/abseil-cpp/absl/flags:flag",
"//third_party/abseil-cpp/absl/memory",

View File

@ -28,6 +28,7 @@
#include "rtc_base/checks.h"
#include "rtc_base/random.h"
#include "test/explicit_key_value_config.h"
#include "test/gmock.h"
#include "test/gtest.h"
#include "test/testsupport/file_utils.h"
@ -35,6 +36,7 @@ namespace webrtc {
namespace {
using test::ExplicitKeyValueConfig;
using testing::SizeIs;
using DecodeResult = ::webrtc::AudioDecoder::EncodedAudioFrame::DecodeResult;
using ParseResult = ::webrtc::AudioDecoder::ParseResult;
@ -158,7 +160,6 @@ void EncodeDecodeNoiseUntilDecoderInDtxMode(AudioEncoderOpusImpl& encoder,
std::vector<int16_t> decoded_frame(kEncoderFrameLength *
decoder_num_channels);
bool dtx_packet_found = false;
for (int i = 0; i < 50; ++i) {
generator.GenerateNextFrame(input_frame);
rtc::Buffer payload;
@ -170,8 +171,9 @@ void EncodeDecodeNoiseUntilDecoderInDtxMode(AudioEncoderOpusImpl& encoder,
continue;
}
// Decode `payload`. If not a DTX packet, decoding it may update the
// internal decoder parameters for comfort noise generation.
// Decode `payload`. If it encodes a DTX packet (i.e., 1 byte payload), the
// decoder will switch to DTX mode. Otherwise, it may update the internal
// decoder parameters for comfort noise generation.
std::vector<ParseResult> parse_results =
decoder.ParsePayload(std::move(payload), timestamp++);
RTC_CHECK_EQ(parse_results.size(), 1);
@ -179,14 +181,62 @@ void EncodeDecodeNoiseUntilDecoderInDtxMode(AudioEncoderOpusImpl& encoder,
parse_results[0].frame->Decode(decoded_frame);
RTC_CHECK(decode_results);
RTC_CHECK_EQ(decode_results->num_decoded_samples, decoded_frame.size());
if (parse_results[0].frame->IsDtxPacket()) {
// The decoder is now in DTX mode.
dtx_packet_found = true;
break;
return;
}
}
RTC_CHECK(dtx_packet_found);
RTC_CHECK_NOTREACHED();
}
// Generates packets by encoding speech frames and decodes them until a non-DTX
// packet is generated and, when that condition is met, returns the decoded
// audio samples.
std::vector<int16_t> EncodeDecodeSpeechUntilOneFrameIsDecoded(
AudioEncoderOpusImpl& encoder,
AudioDecoderOpusImpl& decoder,
uint32_t& rtp_timestamp,
uint32_t& timestamp) {
RTC_CHECK(encoder.NumChannels() == 1 || encoder.NumChannels() == 2);
const bool stereo_encoding = encoder.NumChannels() == 2;
const size_t decoder_num_channels = decoder.Channels();
std::vector<int16_t> decoded_frame(kEncoderFrameLength *
decoder_num_channels);
PCMFile pcm_file;
pcm_file.Open(test::ResourcePath(
stereo_encoding ? "near48_stereo" : "near48_mono", "pcm"),
kSampleRateHz, "rb");
pcm_file.ReadStereo(stereo_encoding);
AudioFrame audio_frame;
while (true) {
if (pcm_file.EndOfFile()) {
break;
}
pcm_file.Read10MsData(audio_frame);
rtc::Buffer payload;
encoder.Encode(rtp_timestamp++, audio_frame.data_view().data(), &payload);
// Ignore empty payloads: the encoder needs more audio to produce a packet.
if (payload.size() == 0) {
continue;
}
// Decode `payload`.
std::vector<ParseResult> parse_results =
decoder.ParsePayload(std::move(payload), timestamp++);
RTC_CHECK_EQ(parse_results.size(), 1);
std::optional<DecodeResult> decode_results =
parse_results[0].frame->Decode(decoded_frame);
RTC_CHECK(decode_results);
if (parse_results[0].frame->IsDtxPacket()) {
continue;
}
RTC_CHECK_EQ(decode_results->num_decoded_samples, decoded_frame.size());
return decoded_frame;
}
RTC_CHECK_NOTREACHED();
}
} // namespace
@ -239,6 +289,7 @@ TEST(AudioDecoderOpusTest,
constexpr size_t kDecoderNumChannels = 2;
AudioDecoderOpusImpl decoder(env.field_trials(), kDecoderNumChannels,
kSampleRateHz);
std::vector<int16_t> decoded_frame;
uint32_t rtp_timestamp = 0xFFFu;
uint32_t timestamp = 0;
@ -250,7 +301,7 @@ TEST(AudioDecoderOpusTest,
timestamp);
// Decode an empty packet so that Opus generates comfort noise.
std::array<int16_t, kEncoderFrameLength * kDecoderNumChannels> decoded_frame;
decoded_frame.resize(kEncoderFrameLength * kDecoderNumChannels);
AudioDecoder::SpeechType speech_type;
const int num_decoded_samples =
decoder.Decode(/*encoded=*/nullptr, /*encoded_len=*/0, kSampleRateHz,
@ -262,8 +313,14 @@ TEST(AudioDecoderOpusTest,
num_decoded_samples);
// Make sure that comfort noise is not a muted frame.
ASSERT_FALSE(IsZeroedFrame(decoded_view));
EXPECT_TRUE(IsTrivialStereo(decoded_view));
// Also check the first decoded audio frame after comfort noise.
decoded_frame = EncodeDecodeSpeechUntilOneFrameIsDecoded(
encoder, decoder, rtp_timestamp, timestamp);
ASSERT_THAT(decoded_frame, SizeIs(kDecoderNumChannels * kEncoderFrameLength));
ASSERT_FALSE(IsZeroedFrame(decoded_frame));
EXPECT_TRUE(IsTrivialStereo(decoded_frame));
}
TEST(AudioDecoderOpusTest, MonoEncoderStereoDecoderOutputsTrivialStereoPlc) {
@ -296,8 +353,14 @@ TEST(AudioDecoderOpusTest, MonoEncoderStereoDecoderOutputsTrivialStereoPlc) {
concealment_audio.size());
// Make sure that packet loss concealment is not a muted frame.
ASSERT_FALSE(IsZeroedFrame(decoded_view));
EXPECT_TRUE(IsTrivialStereo(decoded_view));
// Also check the first decoded audio frame after packet loss concealment.
std::vector<int16_t> decoded_frame = EncodeDecodeSpeechUntilOneFrameIsDecoded(
encoder, decoder, rtp_timestamp, timestamp);
ASSERT_THAT(decoded_frame, SizeIs(kDecoderNumChannels * kEncoderFrameLength));
ASSERT_FALSE(IsZeroedFrame(decoded_frame));
EXPECT_TRUE(IsTrivialStereo(decoded_frame));
}
TEST(AudioDecoderOpusTest,

View File

@ -547,22 +547,9 @@ int WebRtcOpus_Decode(OpusDecInst* inst,
int16_t* decoded,
int16_t* audio_type) {
int decoded_samples_per_channel;
if (encoded_bytes == 0) {
*audio_type = DetermineAudioType(inst, encoded_bytes);
decoded_samples_per_channel = DecodePlc(inst, decoded);
// TODO: https://issues.webrtc.org/376493209 - When fixed, remove block
// below.
if (inst->channels == 2 && inst->last_packet_num_channels == 1) {
// Stereo decoding is enabled and the last observed packet to decode
// encoded mono audio. In this case, Opus generates non-trivial stereo
// audio. Since this is unwanted, copy the left channel into the right
// one.
for (int i = 0; i < decoded_samples_per_channel << 1; i += 2) {
decoded[i + 1] = decoded[i];
}
}
} else {
decoded_samples_per_channel = DecodeNative(
inst, encoded, encoded_bytes,
@ -570,14 +557,25 @@ int WebRtcOpus_Decode(OpusDecInst* inst,
// TODO: https://issues.webrtc.org/376493209 - When fixed, remove block
// below.
const int num_channels = opus_packet_get_nb_channels(encoded);
RTC_DCHECK(num_channels == 1 || num_channels == 2);
inst->last_packet_num_channels = num_channels;
inst->last_packet_num_channels = opus_packet_get_nb_channels(encoded);
RTC_DCHECK(inst->last_packet_num_channels == 1 ||
inst->last_packet_num_channels == 2);
}
if (decoded_samples_per_channel < 0) {
return -1;
}
// TODO: https://issues.webrtc.org/376493209 - When fixed, remove block below.
// When stereo decoding is enabled and the last observed non-empty packet
// encoded mono audio, the Opus decoder may generate non-trivial stereo audio.
// As that is undesired, in that case make sure that `decoded` contains
// trivial stereo audio by copying the left channel into the right one.
if (inst->channels == 2 && inst->last_packet_num_channels == 1) {
for (int i = 0; i < decoded_samples_per_channel << 1; i += 2) {
decoded[i + 1] = decoded[i];
}
}
return decoded_samples_per_channel;
}