Snap for 11518112 from 29601f2cae48d941fa1bdaf4f87ede9820f0779a to emu-34-3-releaseemu-34-3-release

Change-Id: I6c0b7c62d56e0363586d16d285ce773e4156bc87

6 files changed, 14 insertions, 12 deletions

diff --git a/host/commands/assemble_cvd/graphics_flags.cc b/host/commands/assemble_cvd/graphics_flags.cc
index 2a0f7c78d..19aebc519 100644
--- a/host/commands/assemble_cvd/graphics_flags.cc
+++ b/host/commands/assemble_cvd/graphics_flags.cc
@@ -71,8 +71,9 @@ Result<RenderingMode> GetRenderingMode(const std::string& mode) {
 
 struct AngleFeatures {
   // Prefer linear filtering for YUV AHBs to pass
-  // android.media.decoder.cts.DecodeAccuracyTest.
-  bool prefer_linear_filtering_for_yuv = true;
+  // android.media.decoder.cts.DecodeAccuracyTest on older branches.
+  // Generally not needed after b/315387961.
+  bool prefer_linear_filtering_for_yuv = false;
 
   // Map unspecified color spaces to PASS_THROUGH to pass
   // android.media.codec.cts.DecodeEditEncodeTest and
diff --git a/host/commands/process_sandboxer/main.cpp b/host/commands/process_sandboxer/main.cpp
index 3c5af47d5..2ddcfb5f3 100644
--- a/host/commands/process_sandboxer/main.cpp
+++ b/host/commands/process_sandboxer/main.cpp
@@ -24,6 +24,7 @@
 #include "absl/flags/parse.h"
 #include "absl/log/check.h"
 #include "absl/log/initialize.h"
+#include "absl/strings/numbers.h"
 #pragma clang diagnostic push
 #pragma clang diagnostic ignored "-Wunused-parameter"
 #include "sandboxed_api/sandbox2/executor.h"
diff --git a/host/frontend/webrtc/audio_handler.cpp b/host/frontend/webrtc/audio_handler.cpp
index 52a2294d7..be15ed7c2 100644
--- a/host/frontend/webrtc/audio_handler.cpp
+++ b/host/frontend/webrtc/audio_handler.cpp
@@ -443,7 +443,7 @@ void AudioHandler::OnPlaybackBuffer(TxBuffer buffer) {
         // webrtc api doesn't expect volatile memory. This should be safe though
         // because webrtc will use the contents of the buffer before returning
         // and only then we release it.
-        auto audio_frame_buffer = std::make_shared<CvdAudioFrameBuffer>(
+        CvdAudioFrameBuffer audio_frame_buffer(
             const_cast<const uint8_t*>(&buffer.get()[pos]),
             stream_desc.bits_per_sample, stream_desc.sample_rate,
             stream_desc.channels, frames);
@@ -453,9 +453,9 @@ void AudioHandler::OnPlaybackBuffer(TxBuffer buffer) {
         pos += holding_buffer.Add(buffer.get() + pos, buffer.len() - pos);
         if (holding_buffer.full()) {
           auto buffer_ptr = const_cast<const uint8_t*>(holding_buffer.data());
-          auto audio_frame_buffer = std::make_shared<CvdAudioFrameBuffer>(
-              buffer_ptr, stream_desc.bits_per_sample,
-              stream_desc.sample_rate, stream_desc.channels, frames);
+          CvdAudioFrameBuffer audio_frame_buffer(
+              buffer_ptr, stream_desc.bits_per_sample, stream_desc.sample_rate,
+              stream_desc.channels, frames);
           audio_sink_->OnFrame(audio_frame_buffer, base_time);
           holding_buffer.count = 0;
         }
diff --git a/host/frontend/webrtc/libdevice/audio_sink.h b/host/frontend/webrtc/libdevice/audio_sink.h
index 1baa88115..614aada91 100644
--- a/host/frontend/webrtc/libdevice/audio_sink.h
+++ b/host/frontend/webrtc/libdevice/audio_sink.h
@@ -26,7 +26,7 @@ namespace webrtc_streaming {
 class AudioSink {
  public:
   virtual ~AudioSink() = default;
-  virtual void OnFrame(std::shared_ptr<AudioFrameBuffer> frame,
+  virtual void OnFrame(const AudioFrameBuffer& frame,
                        int64_t timestamp_us) = 0;
 };
 
diff --git a/host/frontend/webrtc/libdevice/audio_track_source_impl.cpp b/host/frontend/webrtc/libdevice/audio_track_source_impl.cpp
index 72756e195..c10e2630b 100644
--- a/host/frontend/webrtc/libdevice/audio_track_source_impl.cpp
+++ b/host/frontend/webrtc/libdevice/audio_track_source_impl.cpp
@@ -51,12 +51,12 @@ const cricket::AudioOptions AudioTrackSourceImpl::options() const {
   return cricket::AudioOptions();
 }
 
-void AudioTrackSourceImpl::OnFrame(std::shared_ptr<AudioFrameBuffer> frame,
+void AudioTrackSourceImpl::OnFrame(const AudioFrameBuffer& frame,
                                    int64_t timestamp_ms) {
     std::lock_guard<std::mutex> lock(sinks_mutex_);
     for (auto sink : sinks_) {
-      sink->OnData(frame->data(), frame->bits_per_sample(),
-                   frame->sample_rate(), frame->channels(), frame->frames(),
+      sink->OnData(frame.data(), frame.bits_per_sample(),
+                   frame.sample_rate(), frame.channels(), frame.frames(),
                    timestamp_ms);
     }
 }
diff --git a/host/frontend/webrtc/libdevice/audio_track_source_impl.h b/host/frontend/webrtc/libdevice/audio_track_source_impl.h
index 53dffeacf..61f01cff4 100644
--- a/host/frontend/webrtc/libdevice/audio_track_source_impl.h
+++ b/host/frontend/webrtc/libdevice/audio_track_source_impl.h
@@ -44,7 +44,7 @@ class AudioTrackSourceImpl : public webrtc::AudioSourceInterface {
   // audio network adaptation on the source is the wrong layer of abstraction).
   virtual const cricket::AudioOptions options() const;
 
-  void OnFrame(std::shared_ptr<AudioFrameBuffer> frame, int64_t timestamp_ms);
+  void OnFrame(const AudioFrameBuffer& frame, int64_t timestamp_ms);
 
   // MediaSourceInterface implementation
   SourceState state() const override;
@@ -74,7 +74,7 @@ class AudioTrackSourceImplSinkWrapper : public AudioSink {
   AudioTrackSourceImplSinkWrapper(rtc::scoped_refptr<AudioTrackSourceImpl> obj)
       : track_source_impl_(obj) {}
 
-  void OnFrame(std::shared_ptr<AudioFrameBuffer> frame,
+  void OnFrame(const AudioFrameBuffer& frame,
                int64_t timestamp_ms) override {
     track_source_impl_->OnFrame(frame, timestamp_ms);
   }