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| Original file line number | Diff line number | Diff line change |
|---|---|---|
| @@ -0,0 +1,224 @@ | ||
| #include "Samples.h" | ||
| #include <gst/gst.h> | ||
| #include <gst/app/app.h> | ||
| #include <gst/app/gstappsink.h> | ||
|
|
||
| // 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 | ||
| // GStreamer pipeline created in `receiveGstreamerAudioVideo`. Any logic to modify / discard the frames would go here | ||
| VOID onGstVideoFrameReady(UINT64 customData, PFrame pFrame) | ||
| { | ||
| STATUS retStatus = STATUS_SUCCESS; | ||
| GstFlowReturn ret; | ||
| GstBuffer* buffer; | ||
| GstElement* appsrcVideo = (GstElement*) customData; | ||
|
|
||
| CHK_ERR(appsrcVideo != NULL, STATUS_NULL_ARG, "appsrcVideo is null"); | ||
| CHK_ERR(pFrame != NULL, STATUS_NULL_ARG, "Video frame is null"); | ||
|
|
||
| buffer = gst_buffer_new_allocate(NULL, pFrame->size, NULL); | ||
| CHK_ERR(buffer != NULL, STATUS_NULL_ARG, "Buffer allocation failed"); | ||
|
|
||
| DLOGV("Frame size: %d, presentationTs: %llu", pFrame->size, pFrame->presentationTs); | ||
| GST_BUFFER_PTS(buffer) = pFrame->presentationTs; | ||
| GST_BUFFER_DURATION(buffer) = gst_util_uint64_scale(1, GST_SECOND, DEFAULT_FPS_VALUE); | ||
| if (gst_buffer_fill(buffer, 0, pFrame->frameData, pFrame->size) != pFrame->size) { | ||
| DLOGE("Buffer fill did not complete correctly"); | ||
| gst_buffer_unref(buffer); | ||
| return; | ||
| } | ||
| g_signal_emit_by_name(appsrcVideo, "push-buffer", buffer, &ret); | ||
| if (ret != GST_FLOW_OK) { | ||
| DLOGE("Error pushing buffer: %s", gst_flow_get_name(ret)); | ||
| } | ||
| gst_buffer_unref(buffer); | ||
|
|
||
| CleanUp: | ||
| return; | ||
| } | ||
|
|
||
| // This function is a callback for the transceiver for every single audio frame it receives | ||
| // It writes these frames to a buffer and pushes it to the `appsrcAudio` element of the | ||
| // GStreamer pipeline created in `receiveGstreamerAudioVideo`. Any logic to modify / discard the frames would go here | ||
| VOID onGstAudioFrameReady(UINT64 customData, PFrame pFrame) | ||
| { | ||
| STATUS retStatus = STATUS_SUCCESS; | ||
| GstFlowReturn ret; | ||
| GstBuffer* buffer; | ||
| GstElement* appsrcAudio = (GstElement*) customData; | ||
|
|
||
| CHK_ERR(appsrcAudio != NULL, STATUS_NULL_ARG, "appsrcAudio is null"); | ||
| CHK_ERR(pFrame != NULL, STATUS_NULL_ARG, "Audio frame is null"); | ||
|
|
||
| buffer = gst_buffer_new_allocate(NULL, pFrame->size, NULL); | ||
| CHK_ERR(buffer != NULL, STATUS_NULL_ARG, "Buffer allocation failed"); | ||
|
|
||
| DLOGV("Audio Frame size: %d, presentationTs: %llu", pFrame->size, pFrame->presentationTs); | ||
| GST_BUFFER_PTS(buffer) = pFrame->presentationTs; | ||
|
|
||
| // Recalculate the byte-rate if not using the default values | ||
| GST_BUFFER_DURATION(buffer) = gst_util_uint64_scale(pFrame->size, GST_SECOND, DEFAULT_AUDIO_OPUS_BYTE_RATE); | ||
| if (gst_buffer_fill(buffer, 0, pFrame->frameData, pFrame->size) != pFrame->size) { | ||
| DLOGE("Buffer fill did not complete correctly"); | ||
| gst_buffer_unref(buffer); | ||
| return; | ||
| } | ||
| g_signal_emit_by_name(appsrcAudio, "push-buffer", buffer, &ret); | ||
| if (ret != GST_FLOW_OK) { | ||
| DLOGE("Error pushing buffer: %s", gst_flow_get_name(ret)); | ||
| } | ||
| gst_buffer_unref(buffer); | ||
|
|
||
| CleanUp: | ||
| return; | ||
| } | ||
|
|
||
| // This function is a callback for the streaming session shutdown event. We send an eos to the pipeline to exit the | ||
| // application using this. | ||
| VOID onSampleStreamingSessionShutdown(UINT64 customData, PSampleStreamingSession pSampleStreamingSession) | ||
| { | ||
| (void) (pSampleStreamingSession); | ||
| GstElement* pipeline = (GstElement*) customData; | ||
| gst_element_send_event(pipeline, gst_event_new_eos()); | ||
| } | ||
|
|
||
| PVOID receiveGstreamerAudioVideo(PVOID args) | ||
| { | ||
| STATUS retStatus = STATUS_SUCCESS; | ||
| GstElement *pipeline = NULL, *appsrcAudio = NULL, *appsrcVideo = NULL; | ||
| GstBus* bus; | ||
| GstMessage* msg; | ||
| GError* error = NULL; | ||
| GstCaps *audiocaps, *videocaps; | ||
| PSampleStreamingSession pSampleStreamingSession = (PSampleStreamingSession) args; | ||
| PSampleConfiguration pSampleConfiguration = pSampleStreamingSession->pSampleConfiguration; | ||
| PCHAR roleType = "Viewer"; | ||
| gchar *videoDescription = "", *audioDescription = "", *audioVideoDescription; | ||
|
|
||
| if (pSampleConfiguration->channelInfo.channelRoleType == SIGNALING_CHANNEL_ROLE_TYPE_MASTER) { | ||
| roleType = "Master"; | ||
| } | ||
|
|
||
| CHK_ERR(gst_init_check(NULL, NULL, &error), STATUS_INTERNAL_ERROR, "[KVS %s] GStreamer initialization failed"); | ||
|
|
||
| CHK_ERR(pSampleStreamingSession != NULL, STATUS_NULL_ARG, "[KVS %s] Sample streaming session is NULL", roleType); | ||
|
|
||
| // It is advised to modify the pipeline and the caps as per the source of the media. Customers can also modify this pipeline to | ||
| // use any other sinks instead of `filesink` like `autovideosink` and `autoaudiosink`. The existing pipelines are not complex enough to | ||
| // change caps and properties dynamically, more complex logic may be needed to support the same. | ||
| switch (pSampleStreamingSession->pVideoRtcRtpTransceiver->receiver.track.codec) { | ||
| case RTC_CODEC_H264_PROFILE_42E01F_LEVEL_ASYMMETRY_ALLOWED_PACKETIZATION_MODE: | ||
| videoDescription = "appsrc name=appsrc-video ! queue ! h264parse ! queue ! matroskamux name=mux ! queue ! filesink location=video.mkv"; | ||
| videocaps = gst_caps_new_simple("video/x-h264", "stream-format", G_TYPE_STRING, "byte-stream", "alignment", G_TYPE_STRING, "au", | ||
| "profile", G_TYPE_STRING, "baseline", "height", G_TYPE_INT, DEFAULT_VIDEO_HEIGHT_PIXELS, "width", | ||
| G_TYPE_INT, DEFAULT_VIDEO_WIDTH_PIXELS, NULL); | ||
| break; | ||
|
|
||
| case RTC_CODEC_H265: | ||
| videoDescription = "appsrc name=appsrc-video ! queue ! h265parse ! queue ! matroskamux name=mux ! queue ! filesink location=video.mkv "; | ||
| videocaps = gst_caps_new_simple("video/x-h265", "stream-format", G_TYPE_STRING, "byte-stream", "alignment", G_TYPE_STRING, "au", | ||
| "profile", G_TYPE_STRING, "main", "height", G_TYPE_INT, DEFAULT_VIDEO_HEIGHT_PIXELS, "width", G_TYPE_INT, | ||
| DEFAULT_VIDEO_WIDTH_PIXELS, NULL); | ||
| break; | ||
|
|
||
| // TODO: add a similar pipeline for VP8 | ||
|
|
||
| default: | ||
| break; | ||
| } | ||
|
|
||
| if (pSampleConfiguration->mediaType == SAMPLE_STREAMING_AUDIO_VIDEO) { | ||
| switch (pSampleStreamingSession->pAudioRtcRtpTransceiver->receiver.track.codec) { | ||
| case RTC_CODEC_OPUS: | ||
| audioDescription = "appsrc name=appsrc-audio ! queue ! opusparse ! queue ! mux."; | ||
| audiocaps = gst_caps_new_simple("audio/x-opus", "rate", G_TYPE_INT, DEFAULT_AUDIO_OPUS_SAMPLE_RATE_HZ, "channel-mapping-family", | ||
| 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: | ||
| audioDescription = "appsrc name=appsrc-audio ! rawaudioparse ! decodebin ! autoaudiosink"; | ||
| break; | ||
|
|
||
| default: | ||
| break; | ||
| } | ||
| } | ||
|
|
||
| audioVideoDescription = g_strjoin(" ", videoDescription, audioDescription, NULL); | ||
|
|
||
| pipeline = gst_parse_launch(audioVideoDescription, &error); | ||
| CHK_ERR(pipeline != NULL, STATUS_INTERNAL_ERROR, "[KVS %s] Pipeline is NULL", roleType); | ||
|
|
||
| appsrcVideo = gst_bin_get_by_name(GST_BIN(pipeline), "appsrc-video"); | ||
| CHK_ERR(appsrcVideo != NULL, STATUS_INTERNAL_ERROR, "[KVS %s] Cannot find appsrc video", roleType); | ||
| CHK_STATUS(transceiverOnFrame(pSampleStreamingSession->pVideoRtcRtpTransceiver, (UINT64) appsrcVideo, onGstVideoFrameReady)); | ||
| g_object_set(G_OBJECT(appsrcVideo), "caps", videocaps, NULL); | ||
| gst_caps_unref(videocaps); | ||
|
|
||
| if (pSampleConfiguration->mediaType == SAMPLE_STREAMING_AUDIO_VIDEO) { | ||
| appsrcAudio = gst_bin_get_by_name(GST_BIN(pipeline), "appsrc-audio"); | ||
| CHK_ERR(appsrcAudio != NULL, STATUS_INTERNAL_ERROR, "[KVS %s] Cannot find appsrc audio", roleType); | ||
| CHK_STATUS(transceiverOnFrame(pSampleStreamingSession->pAudioRtcRtpTransceiver, (UINT64) appsrcAudio, onGstAudioFrameReady)); | ||
| g_object_set(G_OBJECT(appsrcAudio), "caps", audiocaps, NULL); | ||
| gst_caps_unref(audiocaps); | ||
| } | ||
|
|
||
| CHK_STATUS(streamingSessionOnShutdown(pSampleStreamingSession, (UINT64) pipeline, onSampleStreamingSessionShutdown)); | ||
| g_free(audioVideoDescription); | ||
|
|
||
| gst_element_set_state(pipeline, GST_STATE_PLAYING); | ||
|
|
||
| /* block until error or EOS */ | ||
| bus = gst_element_get_bus(pipeline); | ||
| CHK_ERR(bus != NULL, STATUS_INTERNAL_ERROR, "[KVS %s] Bus is NULL", roleType); | ||
| msg = gst_bus_timed_pop_filtered(bus, GST_CLOCK_TIME_NONE, GST_MESSAGE_ERROR | GST_MESSAGE_EOS); | ||
|
|
||
| /* Free resources */ | ||
| if (msg != NULL) { | ||
| switch (GST_MESSAGE_TYPE(msg)) { | ||
| case GST_MESSAGE_ERROR: | ||
| gst_message_parse_error(msg, &error, NULL); | ||
| DLOGE("Error received: %s", error->message); | ||
| g_error_free(error); | ||
| break; | ||
| case GST_MESSAGE_EOS: | ||
| DLOGI("End of stream"); | ||
| break; | ||
| default: | ||
| break; | ||
| } | ||
| gst_message_unref(msg); | ||
| } | ||
| if (bus != NULL) { | ||
| gst_object_unref(bus); | ||
| } | ||
| if (pipeline != NULL) { | ||
| gst_element_set_state(pipeline, GST_STATE_NULL); | ||
| gst_object_unref(pipeline); | ||
| } | ||
| if (appsrcAudio != NULL) { | ||
| gst_object_unref(appsrcAudio); | ||
| } | ||
| if (appsrcVideo != NULL) { | ||
| gst_object_unref(appsrcVideo); | ||
| } | ||
|
|
||
| CleanUp: | ||
| if (error != NULL) { | ||
| DLOGE("[KVS %s] %s", roleType, error->message); | ||
| g_clear_error(&error); | ||
| } | ||
|
|
||
| gst_deinit(); | ||
|
|
||
| return (PVOID) (ULONG_PTR) retStatus; | ||
| } |
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| Original file line number | Diff line number | Diff line change |
|---|---|---|
| @@ -0,0 +1 @@ | ||
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