Updates webrtc to m138 HEAD. (#11492)

Updates webrtc from e4445e46a910eb407571ec0b0b8b7043562678cf to
7ad61a0a06fac695468b843e17b81a6ab19d0feb.

Bug: 537464074

Author: briantting

third_party/webrtc/docs/monorail-bug-tracker-migration/monorail-issue-tracker-map.csv

@@ -2598,9 +2598,14 @@ bool WebRtcVoiceReceiveChannel::MaybeCreateDefaultReceiveStream(
   // it up to the *latest* unsignaled stream we've seen, in order to support
   // the case where the SSRC of one unsignaled stream changes.
   if (default_sink_) {
-    for (uint32_t drop_ssrc : unsignaled_recv_ssrcs_) {
-      auto it = recv_streams_.find(drop_ssrc);
-      it->second->SetRawAudioSink(nullptr);
+    // The new ssrc has already been appended to `unsignaled_recv_ssrcs_`.
+    // If there are 2 or more streams, the stream at `size - 2` is the previous
+    // latest stream which currently possesses the default sink.
+    if (unsignaled_recv_ssrcs_.size() >= 2) {
+      // Detach the default sink from the previous latest stream.
+      uint32_t prev_ssrc =
+          unsignaled_recv_ssrcs_[unsignaled_recv_ssrcs_.size() - 2];
+      SetRawAudioSink(prev_ssrc, nullptr);
     }
     std::unique_ptr<AudioSinkInterface> proxy_sink(
         new ProxySink(default_sink_.get()));
@@ -2799,7 +2804,22 @@ bool WebRtcVoiceReceiveChannel::MaybeDeregisterUnsignaledRecvStream(
   RTC_DCHECK_RUN_ON(worker_thread_);
   auto it = absl::c_find(unsignaled_recv_ssrcs_, ssrc);
   if (it != unsignaled_recv_ssrcs_.end()) {
+    bool is_latest_unsignaled = (it == unsignaled_recv_ssrcs_.end() - 1);
     unsignaled_recv_ssrcs_.erase(it);
+    if (default_sink_) {
+      // Detach the default sink from the deregistered stream. This is needed
+      // to prevent the ProxySink from holding a dangling pointer to the
+      // default_sink_.
+      SetRawAudioSink(ssrc, nullptr);
+      if (is_latest_unsignaled && !unsignaled_recv_ssrcs_.empty()) {
+        // The deregistered stream was the latest unsignaled stream, so it held
+        // the default sink. Since it was removed, we must pass the default sink
+        // to the *new* latest unsignaled stream via a new ProxySink.
+        std::unique_ptr<AudioSinkInterface> proxy_sink(
+            new ProxySink(default_sink_.get()));
+        SetRawAudioSink(unsignaled_recv_ssrcs_.back(), std::move(proxy_sink));
+      }
+    }
     return true;
   }
   return false;

third_party/webrtc/media/engine/webrtc_voice_engine.cc

@@ -47,6 +47,7 @@ rtc_library("audio_coding") {
     "../../api/neteq:default_neteq_factory",
     "../../api/neteq:neteq_api",
     "../../api/units:timestamp",
+    "../../audio/utility:audio_frame_operations",
     "../../common_audio",
     "../../common_audio:common_audio_c",
     "../../rtc_base:buffer",
@@ -819,7 +820,6 @@ if (rtc_include_tests) {
         ":acm_send_test",
         ":audio_codec_speed_tests",
         ":audio_decoder_unittests",
-        ":audio_decoder_unittests",
         ":g711_test",
         ":g722_test",
         ":neteq_opus_quality_test",
@@ -1063,7 +1063,7 @@ if (rtc_include_tests) {
         "../../api:scoped_refptr",
         "../../api/audio_codecs:audio_codecs_api",
         "../../api/audio_codecs:builtin_audio_decoder_factory",
-        "../../api/environment:environment",
+        "../../api/environment",
         "../../api/neteq:neteq_api",
         "../../logging:rtc_event_log_parser",
         "../../rtc_base:checks",

third_party/webrtc/modules/audio_coding/BUILD.gn

@@ -1,4 +1,5 @@
 include_rules = [
+  "+audio/utility",
   "+call",
   "+common_audio",
   "+logging/rtc_event_log",

third_party/webrtc/modules/audio_coding/DEPS

@@ -11,11 +11,14 @@
 #include "modules/audio_coding/acm2/acm_resampler.h"
 
 #include <array>
+#include <cstddef>
 #include <cstdint>
 
 #include "api/audio/audio_frame.h"
 #include "api/audio/audio_view.h"
+#include "audio/utility/audio_frame_operations.h"
 #include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
 
 namespace webrtc {
 namespace acm2 {
@@ -37,6 +40,18 @@ bool ResamplerHelper::MaybeResample(int desired_sample_rate_hz,
       (desired_sample_rate_hz != -1) &&
       (current_sample_rate_hz != desired_sample_rate_hz);
 
+  if (need_resampling) {
+    const size_t target_size =
+        audio_frame->num_channels_ *
+        SampleRateToDefaultChannelSize(desired_sample_rate_hz);
+    if (target_size > AudioFrame::kMaxDataSizeSamples) {
+      RTC_LOG(LS_ERROR) << "AudioFrame cannot hold resampled data.";
+      AudioFrameOperations::Mute(audio_frame);
+      audio_frame->SetSampleRateAndChannelSize(desired_sample_rate_hz);
+      return false;
+    }
+  }
+
   if (need_resampling && !resampled_last_output_frame_) {
     // Prime the resampler with the last frame.
     InterleavedView<const int16_t> src(last_audio_buffer_.data(),

third_party/webrtc/modules/audio_coding/acm2/acm_resampler.cc

@@ -81,9 +81,6 @@ class ScopedDav1dPicture : public RefCountedNonVirtual<ScopedDav1dPicture> {
 
 constexpr char kDav1dName[] = "dav1d";
 
-// Calling `dav1d_data_wrap` requires a `free_callback` to be registered.
-void NullFreeCallback(const uint8_t* /* buffer */, void* /* opaque */) {}
-
 Dav1dDecoder::Dav1dDecoder() = default;
 
 Dav1dDecoder::Dav1dDecoder(const Environment& env)
@@ -141,9 +138,25 @@ int32_t Dav1dDecoder::Decode(const EncodedImage& encoded_image,
 
   ScopedDav1dData scoped_dav1d_data;
   Dav1dData& dav1d_data = scoped_dav1d_data.Data();
-  dav1d_data_wrap(&dav1d_data, encoded_image.data(), encoded_image.size(),
-                  /*free_callback=*/&NullFreeCallback,
-                  /*user_data=*/nullptr);
+
+  // Calling GetEncodedData will create a new `scoped_refptr` and increment the
+  // ref count. By simply releasing we are now responsible for decrementing
+  // the ref count when appropriate, which is when dav1d calls the
+  // `free_callback` to indicate that the buffer is no longer needed.
+  EncodedImageBufferInterface* bitstream_buffer =
+      encoded_image.GetEncodedData().release();
+
+  // Note that the `bitstream_buffer` can have a higher capacity than what is
+  // actually being used, so `encoded_image.size()` should be used to get the
+  // actual size of the bitstream.
+  dav1d_data_wrap(
+      &dav1d_data, encoded_image.data(), encoded_image.size(),
+      /*free_callback=*/
+      [](const uint8_t* /* buffer */, void* user_data) {
+        auto* bb = static_cast<EncodedImageBufferInterface*>(user_data);
+        bb->Release();
+      },
+      /*user_data=*/bitstream_buffer);
 
   if (int decode_res = dav1d_send_data(context_, &dav1d_data)) {
     RTC_LOG(LS_WARNING)

third_party/webrtc/modules/video_coding/codecs/av1/dav1d_decoder.cc

@@ -61,13 +61,9 @@ class TestAv1Decoder : public DecodedImageCallback {
   TestAv1Decoder(const TestAv1Decoder&) = delete;
   TestAv1Decoder& operator=(const TestAv1Decoder&) = delete;
 
-  void Decode(const EncodedImage& image) {
-    ASSERT_THAT(decoder_, NotNull());
+  int32_t Decode(const EncodedImage& image) {
     decoded_frame_ = std::nullopt;
-    int32_t error =
-        decoder_->Decode(image, /*render_time_ms=*/image.capture_time_ms_);
-    ASSERT_EQ(error, WEBRTC_VIDEO_CODEC_OK);
-    ASSERT_THAT(decoded_frame_, Not(Eq(std::nullopt)));
+    return decoder_->Decode(image, /*render_time_ms=*/image.capture_time_ms_);
   }
 
   VideoFrame& decoded_frame() { return *decoded_frame_; }
@@ -89,8 +85,9 @@ class TestAv1Decoder : public DecodedImageCallback {
 
 TEST(Dav1dDecoderTest, KeepsDecodedResolutionByDefault) {
   TestAv1Decoder decoder(CreateEnvironment());
-  decoder.Decode(
-      CreateEncodedImage(kAv1FrameWith36x20EncodededAnd32x16RenderResolution));
+  EXPECT_EQ(decoder.Decode(CreateEncodedImage(
+                kAv1FrameWith36x20EncodededAnd32x16RenderResolution)),
+            WEBRTC_VIDEO_CODEC_OK);
   EXPECT_EQ(decoder.decoded_frame().width(), 36);
   EXPECT_EQ(decoder.decoded_frame().height(), 20);
 }
@@ -99,8 +96,9 @@ TEST(Dav1dDecoderTest, CropsToRenderResolutionWhenCropIsEnabled) {
   TestAv1Decoder decoder(
       CreateEnvironment(std::make_unique<ExplicitKeyValueConfig>(
           "WebRTC-Dav1dDecoder-CropToRenderResolution/Enabled/")));
-  decoder.Decode(
-      CreateEncodedImage(kAv1FrameWith36x20EncodededAnd32x16RenderResolution));
+  EXPECT_EQ(decoder.Decode(CreateEncodedImage(
+                kAv1FrameWith36x20EncodededAnd32x16RenderResolution)),
+            WEBRTC_VIDEO_CODEC_OK);
   EXPECT_EQ(decoder.decoded_frame().width(), 32);
   EXPECT_EQ(decoder.decoded_frame().height(), 16);
 }
@@ -109,12 +107,44 @@ TEST(Dav1dDecoderTest, DoesNotCropToRenderResolutionWhenCropIsDisabled) {
   TestAv1Decoder decoder(
       CreateEnvironment(std::make_unique<ExplicitKeyValueConfig>(
           "WebRTC-Dav1dDecoder-CropToRenderResolution/Disabled/")));
-  decoder.Decode(
-      CreateEncodedImage(kAv1FrameWith36x20EncodededAnd32x16RenderResolution));
+  EXPECT_EQ(decoder.Decode(CreateEncodedImage(
+                kAv1FrameWith36x20EncodededAnd32x16RenderResolution)),
+            WEBRTC_VIDEO_CODEC_OK);
   EXPECT_EQ(decoder.decoded_frame().width(), 36);
   EXPECT_EQ(decoder.decoded_frame().height(), 20);
 }
 
+TEST(Dav1dDecoderTest, FailDecoderOnTiles) {
+  // This test tests buffer management, any failures would be detected
+  // by running this test under ASAN.
+  static constexpr uint8_t kTile0[] = {
+      0x0a, 0x0a, 0x00, 0x00, 0x00, 0x04, 0x3c, 0xff, 0xbc, 0x01, 0xa0, 0x08,
+      0x32, 0xbd, 0x01, 0x10, 0x60, 0xb0, 0x01, 0xa6, 0x9a, 0x68, 0x50, 0x91,
+      0xb0, 0x80, 0xb0, 0xd9, 0x8a, 0x40, 0xe4, 0x3e, 0xfb, 0x06, 0x1f, 0x0e,
+      0xcc, 0xdc, 0xcb, 0x8c, 0x2d, 0x5c, 0xb0, 0x52, 0xe0, 0xb3, 0x89, 0x2b,
+      0x6b, 0x8b, 0x56, 0xd3, 0x0a, 0x70, 0xad, 0x26, 0xf0, 0xca, 0xfa, 0xad,
+      0x27, 0xcc, 0x5b, 0x8c, 0x0c, 0x72, 0x76, 0xa7, 0x42, 0x69, 0x4c, 0x47,
+      0xf4, 0x2b, 0xa9, 0x5e, 0x13, 0x81, 0x69, 0x47, 0x40, 0x13, 0x3e, 0xac,
+      0xef, 0x0f, 0x74, 0x02, 0x79, 0xa7, 0x50, 0xf4, 0x50, 0x82, 0x4d, 0x4f,
+      0x70, 0x02, 0x9c, 0x0f, 0x61, 0xbc, 0x1c, 0x22, 0xc2, 0xac, 0x9f, 0x88,
+      0xbd, 0x2a, 0xdc, 0xdb, 0xf0, 0xbe, 0x95, 0x54, 0x6e, 0xda, 0x08, 0x76,
+      0x2c, 0x86, 0xd2, 0x0f, 0x86, 0xc1, 0x86, 0xcb, 0x98, 0x35, 0xfc, 0x2c,
+      0xe7, 0x51, 0x44, 0x62, 0xfe, 0xf1, 0x97, 0x1f, 0xb0, 0x7f, 0x14, 0xc4,
+      0xef, 0xb0, 0x01, 0x7c, 0x07, 0x22, 0x3e, 0xda, 0x2b, 0xef, 0x59, 0xcc,
+      0x57, 0x65, 0xc5, 0x22, 0xe9, 0x61, 0x99, 0x0c, 0xd6, 0x07, 0x68, 0x1a,
+      0x26, 0xad, 0x9b, 0x1e, 0x91, 0x2c, 0xee, 0xe2, 0xe4, 0x7c, 0x4d, 0x07,
+      0x51, 0x72, 0x89, 0x51, 0xbf, 0x17, 0x67, 0xc0, 0x6d, 0x02, 0x69, 0x92,
+      0x20, 0x7b, 0xeb, 0x11, 0xea, 0xc1, 0x43, 0x59, 0x4f, 0x75, 0xbb, 0x20};
+  static constexpr uint8_t kTile1[] = {
+      0x22, 0x1e, 0xe0, 0xc1, 0xb6, 0x2f, 0x07, 0xd0, 0x0b, 0xaf, 0x14,
+      0xbf, 0x51, 0xa3, 0x0d, 0xf4, 0xb6, 0x45, 0xd8, 0x16, 0xda, 0x8c,
+      0xb0, 0xbf, 0x29, 0x39, 0xdf, 0x9a, 0x9e, 0x9c, 0x69, 0x80};
+
+  TestAv1Decoder decoder(CreateEnvironment());
+  decoder.Decode(CreateEncodedImage(kTile0));
+  decoder.Decode(CreateEncodedImage(kTile1));
+}
+
 }  // namespace
 }  // namespace test
 }  // namespace webrtc

third_party/webrtc/modules/video_coding/codecs/av1/dav1d_decoder_unittest.cc

@@ -429,6 +429,11 @@ int32_t H264EncoderImpl::Encode(
                       << " image to I420. Can't encode frame.";
     return WEBRTC_VIDEO_CODEC_ENCODER_FAILURE;
   }
+  if (frame_buffer->StrideU() != frame_buffer->StrideV()) {
+    // TODO: crbug.com/chromium:491655161 - Remove once the root cause is fixed.
+    RTC_LOG(LS_ERROR) << "OpenH264 requires the U and V strides to be equal.";
+    return WEBRTC_VIDEO_CODEC_ENCODER_FAILURE;
+  }
   RTC_CHECK(frame_buffer->type() == VideoFrameBuffer::Type::kI420 ||
             frame_buffer->type() == VideoFrameBuffer::Type::kI420A);
 

third_party/webrtc/modules/video_coding/codecs/h264/h264_encoder_impl.cc

@@ -12,7 +12,13 @@
 #include "modules/video_coding/codecs/h264/h264_encoder_impl.h"
 
 #include "api/environment/environment_factory.h"
+#include "api/test/mock_video_encoder.h"
+#include "api/video/i420_buffer.h"
+#include "api/video/video_codec_type.h"
+#include "api/video/video_frame.h"
+#include "api/video_codecs/video_codec.h"
 #include "api/video_codecs/video_encoder.h"
+#include "modules/video_coding/codecs/h264/include/h264_globals.h"
 #include "modules/video_coding/include/video_error_codes.h"
 #include "test/gtest.h"
 
@@ -74,6 +80,31 @@ TEST(H264EncoderImplTest, CanInitializeWithSingleNalUnitModeExplicitly) {
             encoder.PacketizationModeForTesting());
 }
 
+TEST(H264EncoderImplTest, RejectsFramesWithUnequalChromaStrides) {
+  H264EncoderImpl encoder(CreateEnvironment(), {});
+  VideoCodec codec_settings;
+  SetDefaultSettings(&codec_settings);
+  MockEncodedImageCallback callback;
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder.InitEncode(&codec_settings, kSettings));
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder.RegisterEncodeCompleteCallback(&callback));
+  // Create a VideoFrame where the U and V strides are different.
+  auto buffer = I420Buffer::Create(
+      /*width=*/codec_settings.width,
+      /*height=*/codec_settings.height,
+      /*stride_y=*/codec_settings.width,
+      /*stride_u=*/(codec_settings.width + 1) / 2,
+      /*stride_v=*/(codec_settings.width + 1) / 2 + 1);
+
+  VideoFrame frame = VideoFrame::Builder()
+                         .set_video_frame_buffer(buffer)
+                         .set_rtp_timestamp(0)
+                         .build();
+
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_ENCODER_FAILURE, encoder.Encode(frame, nullptr));
+}
+
 }  // anonymous namespace
 
 }  // namespace webrtc