Removed blocking get_state calls from aml codec switch (#480)

Summary: Removed blocking get_state calls from aml codec switch
Type: Fix
Test Plan: UT/CT, Fullstack
Jira: RDKEMW-13606, DELIA-70194

Co-authored-by: Adam Czynszak <92790185+aczs@users.noreply.github.com>

Author: smudri85

media/server/gstplayer/include/GstGenericPlayer.h

@@ -321,6 +321,61 @@ class GstGenericPlayer : public IGstGenericPlayer, public IGstGenericPlayerPriva
     std::optional<firebolt::rialto::wrappers::AudioAttributesPrivate>
     createAudioAttributes(const std::unique_ptr<IMediaPipeline::MediaSource> &source) const;
 
+    /**
+     * @brief Configures audio caps based on audio attributes.
+     *        Called by worker thread only!
+     *
+     * @param[in]     pAttrib    : The audio attributes.
+     * @param[out]    audioaac   : Set to true if AAC, false otherwise.
+     * @param[in]     svpenabled : Whether SVP is enabled.
+     * @param[in,out] appsrcCaps : The caps to configure.
+     */
+    void configAudioCap(firebolt::rialto::wrappers::AudioAttributesPrivate *pAttrib, bool *audioaac, bool svpenabled,
+                        GstCaps **appsrcCaps);
+
+    /**
+     * @brief Halts audio playback by setting playsink to READY and decodebin to PAUSED.
+     *        Called by worker thread only!
+     */
+    void haltAudioPlayback();
+
+    /**
+     * @brief Resumes audio playback by syncing playsink and decodebin with parent.
+     *        Called by worker thread only!
+     */
+    void resumeAudioPlayback();
+
+    /**
+     * @brief First-time codec switch from AC3 to AAC when no decoder exists yet.
+     *        Called by worker thread only!
+     *
+     * @param[in] newAudioCaps : The new audio caps to apply.
+     */
+    void firstTimeSwitchFromAC3toAAC(GstCaps *newAudioCaps);
+
+    /**
+     * @brief Switches the audio codec by unlinking old parser/decoder and linking new ones.
+     *        Called by worker thread only!
+     *
+     * @param[in] isAudioAAC   : Whether the new codec is AAC.
+     * @param[in] newAudioCaps : The new audio caps to apply.
+     *
+     * @retval true if codec was switched, false if same codec.
+     */
+    bool switchAudioCodec(bool isAudioAAC, GstCaps *newAudioCaps);
+
+    /**
+     * @brief Top-level audio track codec channel switch, ported from rdk_gstreamer_utils_soc.
+     *        Called by worker thread only!
+     */
+    bool performAudioTrackCodecChannelSwitch(const void *pSampleAttr,
+                                             firebolt::rialto::wrappers::AudioAttributesPrivate *pAudioAttr,
+                                             uint32_t *pStatus, unsigned int *pui32Delay,
+                                             long long *pAudioChangeTargetPts, // NOLINT(runtime/int)
+                                             const long long *pcurrentDispPts, // NOLINT(runtime/int)
+                                             unsigned int *audioChangeStage, GstCaps **appsrcCaps, bool *audioaac,
+                                             bool svpenabled, GstElement *aSrc, bool *ret);
+
     /**
      * @brief Sets text track position before pushing data
      *

media/server/gstplayer/source/GstGenericPlayer.cpp

@@ -220,6 +220,13 @@ class GstWrapperMock : public IGstWrapper
     MOCK_METHOD(gboolean, gstIsBaseParse, (GstElement * element), (const, override));
     MOCK_METHOD(void, gstBaseParseSetPtsInterpolation, (GstBaseParse * parse, gboolean ptsInterpolate),
                 (const, override));
+    MOCK_METHOD(GstStateChangeReturn, gstElementGetState,
+                (GstElement * element, GstState *state, GstState *pending, GstClockTime timeout), (override));
+    MOCK_METHOD(GstPad *, gstPadGetPeer, (GstPad * pad), (override));
+    MOCK_METHOD(gboolean, gstPadUnlink, (GstPad * srcpad, GstPad *sinkpad), (override));
+    MOCK_METHOD(GstPadLinkReturn, gstPadLink, (GstPad * srcpad, GstPad *sinkpad), (override));
+    MOCK_METHOD(gboolean, gstBinRemove, (GstBin * bin, GstElement *element), (override));
+    MOCK_METHOD(GstObject *, gstPadGetParent, (GstPad * pad), (override));
 
     GstCaps *gstCapsNewSimple(const char *media_type, const char *fieldname, ...) const override
     {

tests/common/externalLibraryMocks/GstWrapperMock.h

@@ -26,6 +26,8 @@
 #include "MessageBuilders.h"
 
 using testing::_;
+using testing::AtLeast;
+using testing::Invoke;
 using testing::Return;
 using testing::StrEq;
 
@@ -39,11 +41,20 @@ namespace firebolt::rialto::server::ct
 class AudioSourceSwitchTest : public MediaPipelineTest
 {
 public:
-    AudioSourceSwitchTest() = default;
-    ~AudioSourceSwitchTest() = default;
+    AudioSourceSwitchTest() { m_audioSink = gst_element_factory_make("fakesrc", nullptr); }
+    ~AudioSourceSwitchTest() override { gst_object_unref(m_audioSink); }
 
     void willSwitchAudioSource()
     {
+        EXPECT_CALL(*m_glibWrapperMock, gObjectGetStub(_, StrEq("audio-sink"), _))
+            .WillOnce(Invoke(
+                [this](gpointer object, const gchar *first_property_name, void *element)
+                {
+                    GstElement **elementPtr = reinterpret_cast<GstElement **>(element);
+                    *elementPtr = m_audioSink;
+                }));
+        EXPECT_CALL(*m_gstWrapperMock, gstObjectUnref(m_audioSink)).Times(AtLeast(0));
+        EXPECT_CALL(*m_glibWrapperMock, gTypeName(G_OBJECT_TYPE(m_audioSink))).WillRepeatedly(Return("audio_sink"));
         EXPECT_CALL(*m_gstWrapperMock, gstCapsNewEmptySimple(StrEq("audio/mpeg"))).WillOnce(Return(&m_audioCaps));
         EXPECT_CALL(*m_gstWrapperMock,
                     gstCapsSetSimpleStringStub(&m_audioCaps, StrEq("alignment"), G_TYPE_STRING, StrEq("nal")));
@@ -61,6 +72,7 @@ class AudioSourceSwitchTest : public MediaPipelineTest
         EXPECT_CALL(*m_gstWrapperMock, gstCapsIsEqual(&m_audioCaps, &m_oldCaps)).WillOnce(Return(FALSE));
         EXPECT_CALL(*m_gstWrapperMock, gstCapsToString(&m_oldCaps)).WillOnce(Return(&m_oldCapsStr));
         EXPECT_CALL(*m_glibWrapperMock, gFree(&m_oldCapsStr));
+        EXPECT_CALL(*m_glibWrapperMock, gStrHasPrefix(_, StrEq("amlhalasink"))).WillOnce(Return(FALSE));
         EXPECT_CALL(*m_rdkGstreamerUtilsWrapperMock,
                     performAudioTrackCodecChannelSwitch(_, _, _, _, _, _, _, _, _, _, kSvpEnabled,
                                                         GST_ELEMENT(&m_audioAppSrc), _))
@@ -80,6 +92,7 @@ class AudioSourceSwitchTest : public MediaPipelineTest
 private:
     GstCaps m_oldCaps{};
     gchar m_oldCapsStr{};
+    GstElement *m_audioSink{};
 };
 /*
  * Component Test: Switch audio source procedure test