Remove AAC related changes

niyatim23 · niyatim23 · commit 17a2d028b5d9 · 2024-05-02T12:33:21.000-07:00
diff --git a/CMakeLists.txt b/CMakeLists.txt
@@ -440,7 +440,6 @@ if (BUILD_SAMPLE)
   # copy sample frames to build folder, in case developer runs sample program with command `samples/kvsWebrtcClientMaster` from `build` dir.
   file(COPY "${CMAKE_CURRENT_SOURCE_DIR}/samples/opusSampleFrames" DESTINATION .)
   file(COPY "${CMAKE_CURRENT_SOURCE_DIR}/samples/h264SampleFrames" DESTINATION .)
-  file(COPY "${CMAKE_CURRENT_SOURCE_DIR}/samples/aacSampleFrames" DESTINATION .)
   file(COPY "${CMAKE_CURRENT_SOURCE_DIR}/samples/h265SampleFrames" DESTINATION .)
 
   add_subdirectory(samples)
diff --git a/README.md b/README.md
@@ -277,7 +277,7 @@ To use the **Storage for WebRTC** feature, run the same command as above but wit
 ./samples/kvsWebrtcClientMaster <channelName> 1 <audio-codec> <video-codec>
 ```
 
-Allowed audio-codec: opus (default codec if nothing is specified), aac
+Allowed audio-codec: opus (default codec if nothing is specified)
 Allowed video-codec: h264 (default codec if nothing is specified), h265
 
 #### Sample: kvsWebrtcClientMasterGstSample
@@ -301,7 +301,7 @@ Example:
 ./samples/kvsWebrtcClientMasterGstSample <channelName> audio-video testsrc opus h264
 ```
 
-Allowed audio-codec: opus (default codec if nothing is specified), aac
+Allowed audio-codec: opus (default codec if nothing is specified)
 Allowed video-codec: h264 (default codec if nothing is specified), h265
 
 #### Sample: kvsWebrtcClientViewer
@@ -312,7 +312,7 @@ To run:
 ./samples/kvsWebrtcClientViewer <channelName> <audio-codec> <video-codec>
 ```
 
-Allowed audio-codec: opus (default codec if nothing is specified), aac
+Allowed audio-codec: opus (default codec if nothing is specified)
 Allowed video-codec: h264 (default codec if nothing is specified), h265
 
 #### Sample: kvsWebrtcClientViewerGstSample
@@ -323,7 +323,7 @@ To run:
 ./samples/kvsWebrtcClientViewerGstSample <channelName> <mediaType> <audio-codec> <video-codec>
 ```
 
-Allowed audio-codec: opus (default codec if nothing is specified), aac
+Allowed audio-codec: opus (default codec if nothing is specified)
 Allowed video-codec: h264 (default codec if nothing is specified), h265
 
 ##### Known issues:
@@ -346,18 +346,6 @@ gst-launch-1.0 videotestsrc pattern=ball num-buffers=1500 ! timeoverlay ! videoc
 gst-launch-1.0 videotestsrc pattern=ball num-buffers=1500 ! timeoverlay ! videoconvert ! video/x-raw,format=I420,width=1280,height=720,framerate=25/1 ! queue ! x265enc speed-preset=veryfast bitrate=512 tune=zerolatency ! video/x-h265,stream-format=byte-stream,alignment=au,profile=main ! multifilesink location="frame-%04d.h265" index=1
 ```
 
-##### AAC
-
-###### ADTS LC
-```shell
-gst-launch-1.0 audiotestsrc num-buffers=1500 ! audioconvert ! audioresample ! faac ! capsfilter caps=audio/mpeg,mpegversion=4,stream-format=adts,base-profile=lc,channels=2,rate=16000 ! multifilesink location="sample-%03d.aac" index=1
-```
-
-###### RAW LC
-```shell
-gst-launch-1.0 audiotestsrc num-buffers=1500 ! audioconvert ! audioresample ! faac ! capsfilter caps=audio/mpeg,mpegversion=4,stream-format=raw,base-profile=lc,channels=2,rate=16000 ! multifilesink location="sample-%03d.aac" index=1
-```
-
 ### Viewing Master Samples
 
 After running one of the master samples, when the command line application prints "Signaling client connection to socket established", indicating that your signaling channel is created and the connected master is streaming media to it, you can view the stream. To do so, check the media playback viewer on the KVS Signaling Channels console or open the [WebRTC SDK Test Page](https://awslabs.github.io/amazon-kinesis-video-streams-webrtc-sdk-js/examples/index.html).
diff --git a/samples/CMakeLists.txt b/samples/CMakeLists.txt
@@ -56,7 +56,6 @@ link_directories(${OPEN_SRC_INSTALL_PREFIX}/lib)
 # copy sample frames to this subproject build folder, in case developer runs sample program with command `kvsWebrtcClientMaster` from `build/samples` dir.
 file(COPY "${CMAKE_CURRENT_SOURCE_DIR}/opusSampleFrames" DESTINATION .)
 file(COPY "${CMAKE_CURRENT_SOURCE_DIR}/h264SampleFrames" DESTINATION .)
-file(COPY "${CMAKE_CURRENT_SOURCE_DIR}/aacSampleFrames" DESTINATION .)
 file(COPY "${CMAKE_CURRENT_SOURCE_DIR}/h265SampleFrames" DESTINATION .)
 
 add_executable(
diff --git a/samples/GstAudioVideoReceiver.c b/samples/GstAudioVideoReceiver.c
@@ -5,7 +5,6 @@
 
 static UINT64 presentationTsIncrement = 0;
 static BOOL eos = FALSE;
-static RTC_CODEC audioCodec = RTC_CODEC_OPUS;
 
 // This function is a callback for the transceiver for every single video frame it receives
 // It writes these frames to a buffer and pushes it to the `appsrcVideo` element of the
@@ -68,12 +67,8 @@ VOID onGstAudioFrameReady(UINT64 customData, PFrame pFrame)
 
         GST_BUFFER_DTS(buffer) = presentationTsIncrement;
         GST_BUFFER_PTS(buffer) = presentationTsIncrement;
-
-        if (audioCodec == RTC_CODEC_AAC) {
-            GST_BUFFER_DURATION(buffer) = gst_util_uint64_scale(pFrame->size, GST_SECOND, DEFAULT_AUDIO_AAC_BYTE_RATE);
-        } else {
-            GST_BUFFER_DURATION(buffer) = gst_util_uint64_scale(pFrame->size, GST_SECOND, DEFAULT_AUDIO_OPUS_BYTE_RATE);
-        }
+        GST_BUFFER_DURATION(buffer) = gst_util_uint64_scale(pFrame->size, GST_SECOND, DEFAULT_AUDIO_OPUS_BYTE_RATE);
+        
         // TODO: check for other codecs once the pipelines are added
 
         if (gst_buffer_fill(buffer, 0, pFrame->frameData, pFrame->size) != pFrame->size) {
@@ -118,8 +113,6 @@ PVOID receiveGstreamerAudioVideo(PVOID args)
         roleType = "Master";
     }
 
-    audioCodec = pSampleConfiguration->audioCodec;
-
     CHK_ERR(gst_init_check(NULL, NULL, &error), STATUS_INTERNAL_ERROR, "[KVS GStreamer %s] GStreamer initialization failed");
 
     CHK_ERR(pSampleStreamingSession != NULL, STATUS_NULL_ARG, "[KVS Gstreamer %s] Sample streaming session is NULL", roleType);
@@ -156,13 +149,6 @@ PVOID receiveGstreamerAudioVideo(PVOID args)
                                                 G_TYPE_INT, 1, NULL);
                 break;
 
-            case RTC_CODEC_AAC:
-                audioDescription = "appsrc name=appsrc-audio ! queue ! aacparse ! mux.";
-                audiocaps = gst_caps_new_simple("audio/mpeg", "mpegversion", G_TYPE_INT, 4, "rate", G_TYPE_INT, DEFAULT_AUDIO_AAC_SAMPLE_RATE_HZ,
-                                                "channels", G_TYPE_INT, DEFAULT_AUDIO_AAC_CHANNELS, "stream-format", G_TYPE_STRING, "adts",
-                                                "base-profile", G_TYPE_STRING, "lc", NULL);
-                break;
-
             // TODO: make sure this pipeline works. Figure out the caps for this
             case RTC_CODEC_MULAW:
             case RTC_CODEC_ALAW:
diff --git a/samples/Samples.h b/samples/Samples.h
@@ -19,17 +19,13 @@ extern "C" {
 #define DEFAULT_VIDEO_HEIGHT_PIXELS              720
 #define DEFAULT_VIDEO_WIDTH_PIXELS               1280
 #define DEFAULT_AUDIO_OPUS_CHANNELS              2
-#define DEFAULT_AUDIO_AAC_CHANNELS               2
 #define DEFAULT_AUDIO_OPUS_SAMPLE_RATE_HZ        48000
-#define DEFAULT_AUDIO_AAC_SAMPLE_RATE_HZ         16000
 #define DEFAULT_AUDIO_OPUS_BITS_PER_SAMPLE       16
-#define DEFAULT_AUDIO_AAC_BITS_PER_SAMPLE        16
 #define DEFAULT_MAX_CONCURRENT_STREAMING_SESSION 10
 
 #define AUDIO_CODEC_NAME_ALAW  "alaw"
 #define AUDIO_CODEC_NAME_MULAW "mulaw"
 #define AUDIO_CODEC_NAME_OPUS  "opus"
-#define AUDIO_CODEC_NAME_AAC   "aac"
 #define VIDEO_CODEC_NAME_H264  "h264"
 #define VIDEO_CODEC_NAME_H265  "h265"
 #define VIDEO_CODEC_NAME_VP8   "vp8"
@@ -39,7 +35,6 @@ extern "C" {
 #define SAMPLE_CHANNEL_NAME     (PCHAR) "ScaryTestChannel"
 
 #define DEFAULT_AUDIO_OPUS_BYTE_RATE (DEFAULT_AUDIO_OPUS_SAMPLE_RATE_HZ * DEFAULT_AUDIO_OPUS_CHANNELS * DEFAULT_AUDIO_OPUS_BITS_PER_SAMPLE) / 8
-#define DEFAULT_AUDIO_AAC_BYTE_RATE  (DEFAULT_AUDIO_AAC_SAMPLE_RATE_HZ * DEFAULT_AUDIO_AAC_CHANNELS * DEFAULT_AUDIO_AAC_BITS_PER_SAMPLE) / 8
 
 #define SAMPLE_AUDIO_FRAME_DURATION (20 * HUNDREDS_OF_NANOS_IN_A_MILLISECOND)
 #define SAMPLE_STATS_DURATION       (60 * HUNDREDS_OF_NANOS_IN_A_SECOND)
diff --git a/samples/kvsWebRTCClientMaster.c b/samples/kvsWebRTCClientMaster.c
@@ -30,10 +30,8 @@ INT32 main(INT32 argc, CHAR* argv[])
     CHK_STATUS(createSampleConfiguration(pChannelName, SIGNALING_CHANNEL_ROLE_TYPE_MASTER, TRUE, TRUE, logLevel, &pSampleConfiguration));
 
     if (argc > 3) {
-        if (!STRCMP(argv[3], AUDIO_CODEC_NAME_AAC)) {
-            audioCodec = RTC_CODEC_AAC;
-        } else {
-            DLOGI("[KVS Master] Defaulting to opus as the specified codec's sample frames may not be available");
+        if (!STRCMP(argv[3], AUDIO_CODEC_NAME_OPUS)) {
+            audioCodec = RTC_CODEC_OPUS;
         }
     }
 
@@ -75,9 +73,6 @@ INT32 main(INT32 argc, CHAR* argv[])
     if (audioCodec == RTC_CODEC_OPUS) {
         CHK_STATUS(readFrameFromDisk(NULL, &frameSize, "./opusSampleFrames/sample-001.opus"));
         DLOGI("[KVS Master] Checked Opus sample audio frame availability....available");
-    } else if (audioCodec == RTC_CODEC_AAC) {
-        CHK_STATUS(readFrameFromDisk(NULL, &frameSize, "./aacSampleFrames/sample-001.aac"));
-        DLOGI("[KVS Master] Checked AAC sample audio frame availability....available");
     }
 
     // Initialize KVS WebRTC. This must be done before anything else, and must only be done once.
@@ -253,9 +248,7 @@ PVOID sendAudioPackets(PVOID args)
     while (!ATOMIC_LOAD_BOOL(&pSampleConfiguration->appTerminateFlag)) {
         fileIndex = fileIndex % NUMBER_OF_OPUS_FRAME_FILES + 1;
 
-        if (pSampleConfiguration->audioCodec == RTC_CODEC_AAC) {
-            SNPRINTF(filePath, MAX_PATH_LEN, "./aacSampleFrames/sample-%03d.aac", fileIndex);
-        } else if (pSampleConfiguration->audioCodec == RTC_CODEC_OPUS) {
+        if (pSampleConfiguration->audioCodec == RTC_CODEC_OPUS) {
             SNPRINTF(filePath, MAX_PATH_LEN, "./opusSampleFrames/sample-%03d.opus", fileIndex);
         }
 
diff --git a/samples/kvsWebRTCClientViewer.c b/samples/kvsWebRTCClientViewer.c
@@ -60,8 +60,8 @@ INT32 main(INT32 argc, CHAR* argv[])
 #endif
 
     if (argc > 2) {
-        if (!STRCMP(argv[2], AUDIO_CODEC_NAME_AAC)) {
-            audioCodec = RTC_CODEC_AAC;
+        if (!STRCMP(argv[2], AUDIO_CODEC_NAME_OPUS)) {
+            audioCodec = RTC_CODEC_OPUS;
         } else if (!STRCMP(argv[2], AUDIO_CODEC_NAME_ALAW)) {
             audioCodec = RTC_CODEC_ALAW;
         } else if (!STRCMP(argv[2], AUDIO_CODEC_NAME_MULAW)) {
diff --git a/samples/kvsWebRTCClientViewerGstSample.c b/samples/kvsWebRTCClientViewerGstSample.c
@@ -60,8 +60,8 @@ INT32 main(INT32 argc, CHAR* argv[])
 #endif
 
     if (argc > 3) {
-        if (!STRCMP(argv[3], AUDIO_CODEC_NAME_AAC)) {
-            audioCodec = RTC_CODEC_AAC;
+        if (!STRCMP(argv[3], AUDIO_CODEC_NAME_OPUS)) {
+            audioCodec = RTC_CODEC_OPUS;
         } else {
             DLOGI("[KVS Gstreamer Viewer] Defaulting to Opus audio codec");
         }
diff --git a/samples/kvsWebrtcClientMasterGstSample.c b/samples/kvsWebrtcClientMasterGstSample.c
@@ -171,17 +171,6 @@ PVOID sendGstreamerAudioVideo(PVOID args)
      * vp8enc error-resilient=partitions keyframe-max-dist=10 auto-alt-ref=true cpu-used=5 deadline=1 !
      * appsink sync=TRUE emit-signals=TRUE name=appsink-video
      *
-     * For H265/AAC
-     * pipeline =
-     *          gst_parse_launch("videotestsrc pattern=ball is-live=TRUE ! timeoverlay ! queue ! videoconvert !
-     *                                     "video/x-raw,format=I420,width=1280,height=720,framerate=25/1 ! " "queue ! "
-     *                                     "x265enc speed-preset=veryfast bitrate=512 tune=zerolatency ! "
-     *                                     "video/x-h265,stream-format=byte-stream,alignment=au,profile=main ! appsink sync=TRUE "
-     *                                     "emit-signals=TRUE name=appsink-video audiotestsrc wave=triangle is-live=TRUE ! "
-     *                                     "queue leaky=2 max-size-buffers=400 ! audioconvert ! audioresample ! faac ! "
-     *                                     "capsfilter caps=audio/mpeg,mpegversion=4,stream-format=adts,base-profile=lc,channels=2,rate=48000 ! "
-     *                                     "appsink sync=TRUE emit-signals=TRUE name=appsink-audio",
-     *                                     &error);
      *
      * Raspberry Pi Hardware Encode Example
      * "v4l2src device=\"/dev/video0\" ! queue ! v4l2convert ! "
@@ -275,17 +264,6 @@ PVOID sendGstreamerAudioVideo(PVOID args)
                                              "audio/x-opus,rate=48000,channels=2 ! appsink sync=TRUE emit-signals=TRUE name=appsink-audio",
                                              &error);
 
-                    } else if (pSampleConfiguration->videoCodec == RTC_CODEC_H265 && pSampleConfiguration->audioCodec == RTC_CODEC_AAC) {
-                        senderPipeline =
-                            gst_parse_launch("videotestsrc pattern=ball is-live=TRUE ! timeoverlay ! queue ! videoconvert ! "
-                                             "video/x-raw,format=I420,width=1280,height=720,framerate=25/1 ! queue ! "
-                                             "x265enc speed-preset=veryfast bitrate=512 tune=zerolatency ! "
-                                             "video/x-h265,stream-format=byte-stream,alignment=au,profile=main ! appsink sync=TRUE "
-                                             "emit-signals=TRUE name=appsink-video audiotestsrc wave=triangle is-live=TRUE ! "
-                                             "queue leaky=2 max-size-buffers=400 ! audioconvert ! audioresample ! faac ! "
-                                             "capsfilter caps=audio/mpeg,mpegversion=4,stream-format=adts,base-profile=lc,channels=2,rate=48000 ! "
-                                             "appsink sync=TRUE emit-signals=TRUE name=appsink-audio",
-                                             &error);
                     }
                     // TODO: test and add more such combinations
                     break;
@@ -399,8 +377,8 @@ INT32 main(INT32 argc, CHAR* argv[])
 
     if (argc > 3 && STRCMP(argv[3], "testsrc") == 0) {
         if (argc > 4) {
-            if (!STRCMP(argv[4], AUDIO_CODEC_NAME_AAC)) {
-                audioCodec = RTC_CODEC_AAC;
+            if (!STRCMP(argv[4], AUDIO_CODEC_NAME_OPUS)) {
+                audioCodec = RTC_CODEC_OPUS;
             }
         }
 
diff --git a/src/include/com/amazonaws/kinesis/video/webrtcclient/Include.h b/src/include/com/amazonaws/kinesis/video/webrtcclient/Include.h
@@ -836,7 +836,6 @@ typedef enum {
     RTC_CODEC_ALAW = 5,                                                           //!< ALAW audio codec
     RTC_CODEC_UNKNOWN = 6,
     RTC_CODEC_H265 = 7, //!< H265 video codec
-    RTC_CODEC_AAC = 8,  //!< AAC audio codec
 
     // RTC_CODEC_MAX **MUST** be the last enum in the list **ALWAYS** and not assigned a value
     RTC_CODEC_MAX //!< Placeholder for max number of supported codecs
