fixup!^2 [M120 Migration][WebRTC] Port of Audio Bypass feature from Tizen 8.0

* Add missing guards
* Add missing thread boosting for audio bypass

Bug: https://jira-eu.sec.samsung.net/browse/VDGAME-494
Change-Id: I6f93567c8f325131ecfd25016815f8db7630f2be

diff --git a/third_party/webrtc/api/audio_codecs/audio_decoder.cc b/third_party/webrtc/api/audio_codecs/audio_decoder.cc
index ef8a5562f45e150b10be40bea1cc7f8d57af6222..818567cc13ac95e738af72728d77a4e50081d6b9 100644 (file)
--- a/third_party/webrtc/api/audio_codecs/audio_decoder.cc
+++ b/third_party/webrtc/api/audio_codecs/audio_decoder.cc
@@ -43,7 +43,9 @@ class OldStyleEncodedFrame final : public AudioDecoder::EncodedAudioFrame {
                          {static_cast<size_t>(ret), speech_type});
   }
 
+#if defined(WEBRTC_TIZEN_TV)
   const rtc::Buffer& GetPayload() const override { return payload_; }
+#endif
 
  private:
   AudioDecoder* const decoder_;

diff --git a/third_party/webrtc/api/audio_codecs/audio_decoder.h b/third_party/webrtc/api/audio_codecs/audio_decoder.h
index f647fff4ca75264af4cb350248dbb7e165db7671..6c260f95ceac8e311d781a3196df8b6454cb8aa2 100644 (file)
--- a/third_party/webrtc/api/audio_codecs/audio_decoder.h
+++ b/third_party/webrtc/api/audio_codecs/audio_decoder.h
@@ -64,7 +64,9 @@ class AudioDecoder {
     virtual absl::optional<DecodeResult> Decode(
         rtc::ArrayView<int16_t> decoded) const = 0;
 
+#if defined(WEBRTC_TIZEN_TV)
     virtual const rtc::Buffer& GetPayload() const = 0;
+#endif
   };
 
   struct ParseResult {

diff --git a/third_party/webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h b/third_party/webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h
index cffe7fb402ac6832fbbf5e105b025eaefdbfb9d9..2164e072de7810ff454c046c1738f650cee61c96 100644 (file)
--- a/third_party/webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h
+++ b/third_party/webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h
@@ -37,7 +37,9 @@ class LegacyEncodedAudioFrame final : public AudioDecoder::EncodedAudioFrame {
 
   size_t Duration() const override;
 
+#if defined(WEBRTC_TIZEN_TV)
   const rtc::Buffer& GetPayload() const override { return payload_; }
+#endif
 
   absl::optional<DecodeResult> Decode(
       rtc::ArrayView<int16_t> decoded) const override;

diff --git a/third_party/webrtc/modules/audio_coding/codecs/opus/audio_coder_opus_common.h b/third_party/webrtc/modules/audio_coding/codecs/opus/audio_coder_opus_common.h
index a3d9553e1eb85a6eda60ce20f3ced22d669867e7..867823baa8bdf12fc08b200faa6c87e0e67bfde0 100644 (file)
--- a/third_party/webrtc/modules/audio_coding/codecs/opus/audio_coder_opus_common.h
+++ b/third_party/webrtc/modules/audio_coding/codecs/opus/audio_coder_opus_common.h
@@ -58,7 +58,9 @@ class OpusFrame : public AudioDecoder::EncodedAudioFrame {
 
   bool IsDtxPacket() const override { return payload_.size() <= 2; }
 
+#if defined(WEBRTC_TIZEN_TV)
   const rtc::Buffer& GetPayload() const override { return payload_; }
+#endif
 
   absl::optional<DecodeResult> Decode(
       rtc::ArrayView<int16_t> decoded) const override {

diff --git a/third_party/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc b/third_party/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
index a669ad727e1237fda94c84005e262c6827a40a04..c61f09b3e0f9c1fd15f11493be2ad86f0b3b87fc 100644 (file)
--- a/third_party/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
+++ b/third_party/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
@@ -73,6 +73,13 @@ class MockAudioDecoder final : public AudioDecoder {
       }
     }
 
+#if defined(WEBRTC_TIZEN_TV)
+    const rtc::Buffer& GetPayload() const override {
+      static rtc::Buffer* buffer = new rtc::Buffer();
+      return *buffer;
+    }
+#endif
+
    private:
     const size_t num_channels_;
   };

diff --git a/third_party/webrtc/modules/audio_coding/neteq/packet_buffer_unittest.cc b/third_party/webrtc/modules/audio_coding/neteq/packet_buffer_unittest.cc
index b0079645ff1175afd8d52075265ec15c89f64588..28a976683185deb8e9524d29f0750ad73ccbb17d 100644 (file)
--- a/third_party/webrtc/modules/audio_coding/neteq/packet_buffer_unittest.cc
+++ b/third_party/webrtc/modules/audio_coding/neteq/packet_buffer_unittest.cc
@@ -40,6 +40,10 @@ class MockEncodedAudioFrame : public webrtc::AudioDecoder::EncodedAudioFrame {
               Decode,
               (rtc::ArrayView<int16_t> decoded),
               (const, override));
+
+#if defined(WEBRTC_TIZEN_TV)
+  MOCK_METHOD(const rtc::Buffer&, GetPayload, (), (const, override));
+#endif
 };
 
 // Helper class to generate packets. Packets must be deleted by the user.

diff --git a/third_party/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc b/third_party/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc
index ad52239ae366a65f04d7ffe15e3a8beca30114e5..53d2ee7f0cf956137f3ac4b868e795a4fed6e22f 100644 (file)
--- a/third_party/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc
+++ b/third_party/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc
@@ -45,6 +45,13 @@ class FakeEncodedFrame : public AudioDecoder::EncodedAudioFrame {
 
   bool IsDtxPacket() const override { return is_dtx_; }
 
+#if defined(WEBRTC_TIZEN_TV)
+  const rtc::Buffer& GetPayload() const override {
+    static rtc::Buffer* buffer = new rtc::Buffer();
+    return *buffer;
+  }
+#endif
+
  private:
   FakeDecodeFromFile* const decoder_;
   const uint32_t timestamp_;

diff --git a/tizen_src/chromium_impl/third_party/blink/renderer/platform/peerconnection/BUILD.gn b/tizen_src/chromium_impl/third_party/blink/renderer/platform/peerconnection/BUILD.gn
index 99e100ad13e1290da16d21c8513566575f2d6493..dcc68fab4b1896320230535ea2d3e8f786769fda 100755 (executable)
--- a/tizen_src/chromium_impl/third_party/blink/renderer/platform/peerconnection/BUILD.gn
+++ b/tizen_src/chromium_impl/third_party/blink/renderer/platform/peerconnection/BUILD.gn
@@ -3,6 +3,11 @@
 # found in the LICENSE file.
 
 import("//tizen_src/build/config/tizen_features.gni")
+import("//tizen_src/chromium_impl/media/base/tizen/logger/logger.gni")
+
+tizen_media_logger_config("tizen_media_coding_logger_config") {
+  logger_tag = "tizen_media_coding"
+}
 
 source_set("tizen_media_coding") {
   cflags = [
@@ -41,6 +46,8 @@ source_set("tizen_media_coding") {
   # TODO(m.napiorkows): This should be in public_configs
   # of //media/base
   configs += [
+    ":tizen_media_coding_logger_config",
+
     # Needed for including renderer_media_player_manager_efl.h
     "//tizen_src/build:capi-media-player",
     "//tizen_src/build:libcapi-media-player",
@@ -54,7 +61,13 @@ source_set("tizen_media_coding") {
     "video_coding_utils.h",
   ]
 
-  deps += [ "//tizen_src/chromium_impl/media/filters/tizen:stream_player_renderer_part" ]
+  if (tizen_thread_booster_service) {
+    deps += [ "//tizen_src/chromium_impl/services/thread_booster/public/cpp" ]
+  }
+
+  deps += [
+    "//tizen_src/chromium_impl/media/filters/tizen:stream_player_renderer_part",
+  ]
 
   if (tizen_multimedia && tizen_video_hole) {
     sources += [