Bug 1934695 - Vendor libwebrtc from 463ce639a6

Upstream commit: https://webrtc.googlesource.com/src/+/463ce639a6175abb69dff6650e213cfd2ed89955
    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: [email protected] for raising the concern

    (cherry picked from commit ebb11c4c87e4c0288d1ac18f38b4efa4254b3c6a)

    Bug: chromium:378896415
    Change-Id: Ide27e411781693f14629cf9db8b6c0c0fc762a17
    Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368160
    Reviewed-by: Jakob Ivarsson‎ <[email protected]>
    Commit-Queue: Alessio Bazzica <[email protected]>
    Cr-Original-Commit-Position: refs/heads/main@{#43393}
    Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368560
    Reviewed-by: Henrik Lundin <[email protected]>
    Cr-Commit-Position: refs/branch-heads/6834@{#1}
    Cr-Branched-From: a5d71009ac1dce7da23813dc9413c03073cfa8ca-refs/heads/main@{#43387}

third_party/libwebrtc/README.moz-ff-commit

@@ -33966,3 +33966,6 @@ badfd6347e
 # MOZ_LIBWEBRTC_SRC=/Users/mfroman/no_tm_backup/elm/.moz-fast-forward/moz-libwebrtc MOZ_LIBWEBRTC_BRANCH=mozpatches bash dom/media/webrtc/third_party_build/fast-forward-libwebrtc.sh
 # base of lastest vendoring
 a5d71009ac
+# MOZ_LIBWEBRTC_SRC=/Users/mfroman/no_tm_backup/elm/.moz-fast-forward/moz-libwebrtc MOZ_LIBWEBRTC_BRANCH=mozpatches bash dom/media/webrtc/third_party_build/fast-forward-libwebrtc.sh
+# base of lastest vendoring
+463ce639a6

third_party/libwebrtc/README.mozilla

@@ -22670,3 +22670,5 @@ libwebrtc updated from /Users/mfroman/no_tm_backup/elm/.moz-fast-forward/moz-lib
 libwebrtc updated from /Users/mfroman/no_tm_backup/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2024-12-05T22:18:19.097360+00:00.
 # ./mach python dom/media/webrtc/third_party_build/vendor-libwebrtc.py --from-local /Users/mfroman/no_tm_backup/elm/.moz-fast-forward/moz-libwebrtc --commit mozpatches libwebrtc
 libwebrtc updated from /Users/mfroman/no_tm_backup/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2024-12-05T22:19:55.432580+00:00.
+# ./mach python dom/media/webrtc/third_party_build/vendor-libwebrtc.py --from-local /Users/mfroman/no_tm_backup/elm/.moz-fast-forward/moz-libwebrtc --commit mozpatches libwebrtc
+libwebrtc updated from /Users/mfroman/no_tm_backup/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2024-12-05T22:21:31.486441+00:00.

third_party/libwebrtc/modules/audio_coding/BUILD.gn

@@ -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",

third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_opus_unittest.cc

@@ -28,13 +28,15 @@
 #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"
 
 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;
@@ -170,23 +171,72 @@ 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);
     std::optional<DecodeResult> decode_results =
         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;
+      return;
+    }
+  }
+  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(dtx_packet_found);
+  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,

third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_interface.cc

@@ -547,37 +547,35 @@ 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,
         MaxFrameSizePerChannel(inst->sample_rate_hz), decoded, audio_type, 0);
 
     // 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;
 }
 

third_party/libwebrtc/moz-patch-stack/0030.patch

@@ -753,7 +753,7 @@ index e354db54a0..d2855cbfb0 100644
  using webrtc::FrameDecryptorInterface;
  using webrtc::FrameEncryptorInterface;
 diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
-index a2a1ca8328..05150752be 100644
+index 78bdccfefe..1df856e2e9 100644
 --- a/modules/audio_coding/BUILD.gn
 +++ b/modules/audio_coding/BUILD.gn
 @@ -358,7 +358,7 @@ rtc_library("webrtc_opus_wrapper") {

third_party/libwebrtc/moz-patch-stack/0124.patch

@@ -16,7 +16,7 @@ Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/bbec1b95ddb5e0096
  5 files changed, 26 insertions(+), 6 deletions(-)
 
 diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
-index 05150752be..8a552ed917 100644
+index 1df856e2e9..e547f76cff 100644
 --- a/modules/audio_coding/BUILD.gn
 +++ b/modules/audio_coding/BUILD.gn
 @@ -358,7 +358,7 @@ rtc_library("webrtc_opus_wrapper") {