Modified: trunk/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoEncoderFactory.cpp (243195 => 243196)
--- trunk/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoEncoderFactory.cpp 2019-03-20 07:32:40 UTC (rev 243195)
+++ trunk/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoEncoderFactory.cpp 2019-03-20 09:47:26 UTC (rev 243196)
@@ -58,12 +58,6 @@
namespace WebCore {
-typedef struct {
- uint64_t rtpTimestamp;
- int64_t captureTimeMs;
- webrtc::CodecSpecificInfo codecInfo;
-} FrameData;
-
class GStreamerVideoEncoder : public webrtc::VideoEncoder {
public:
GStreamerVideoEncoder(const webrtc::SdpVideoFormat&)
@@ -139,10 +133,8 @@
g_object_set(m_src, "is-live", true, "format", GST_FORMAT_TIME, nullptr);
auto videoconvert = makeElement("videoconvert");
- auto sink = makeElement("appsink");
- gst_app_sink_set_emit_signals(GST_APP_SINK(sink), TRUE);
- g_signal_connect(sink, "new-sample", G_CALLBACK(newSampleCallbackTramp), this);
- g_object_set(sink, "sync", FALSE, nullptr);
+ m_sink = makeElement("appsink");
+ g_object_set(m_sink, "sync", FALSE, nullptr);
auto name = makeString(Name(), "_enc_rawcapsfilter_0x", hex(reinterpret_cast<uintptr_t>(this)));
m_capsFilter = gst_element_factory_make("capsfilter", name.utf8().data());
@@ -149,9 +141,12 @@
if (m_restrictionCaps)
g_object_set(m_capsFilter, "caps", m_restrictionCaps.get(), nullptr);
- gst_bin_add_many(GST_BIN(m_pipeline.get()), m_src, videoconvert, m_capsFilter, encoder.leakRef(), sink, nullptr);
- if (!gst_element_link_many(m_src, videoconvert, m_capsFilter, m_encoder, sink, nullptr))
+ gst_bin_add_many(GST_BIN(m_pipeline.get()), m_src, videoconvert, m_capsFilter, encoder.leakRef(), m_sink, nullptr);
+ if (!gst_element_link_many(m_src, videoconvert, m_capsFilter, m_encoder, m_sink, nullptr)) {
+ GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS(GST_BIN(m_pipeline.get()), GST_DEBUG_GRAPH_SHOW_VERBOSE, "webkit-webrtc-encoder.error");
+
ASSERT_NOT_REACHED();
+ }
gst_element_set_state(m_pipeline.get(), GST_STATE_PLAYING);
@@ -182,6 +177,7 @@
m_src = nullptr;
m_encoder = nullptr;
m_capsFilter = nullptr;
+ m_sink = nullptr;
m_pipeline = nullptr;
}
@@ -188,10 +184,25 @@
return WEBRTC_VIDEO_CODEC_OK;
}
+ int32_t returnFromFlowReturn(GstFlowReturn flow)
+ {
+ switch (flow) {
+ case GST_FLOW_OK:
+ return WEBRTC_VIDEO_CODEC_OK;
+ case GST_FLOW_FLUSHING:
+ return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+ default:
+ return WEBRTC_VIDEO_CODEC_ERROR;
+ }
+ }
+
+
int32_t Encode(const webrtc::VideoFrame& frame,
- const webrtc::CodecSpecificInfo* codecInfo,
+ const webrtc::CodecSpecificInfo*,
const std::vector<webrtc::FrameType>* frameTypes) final
{
+ int32_t res;
+
if (!m_imageReadyCb) {
GST_INFO_OBJECT(m_pipeline.get(), "No encoded callback set yet!");
@@ -213,13 +224,6 @@
gst_pad_set_offset(pad.get(), -m_firstFramePts);
}
- webrtc::CodecSpecificInfo localCodecInfo;
- FrameData frameData = { frame.timestamp(), frame.render_time_ms(), codecInfo ? *codecInfo : localCodecInfo };
- {
- auto locker = holdLock(m_bufferMapLock);
- m_framesData.append(frameData);
- }
-
for (auto frame_type : *frameTypes) {
if (frame_type == webrtc::kVideoFrameKey) {
auto pad = adoptGRef(gst_element_get_static_pad(m_src, "src"));
@@ -234,70 +238,45 @@
}
}
- switch (gst_app_src_push_sample(GST_APP_SRC(m_src), sample.get())) {
- case GST_FLOW_OK:
- return WEBRTC_VIDEO_CODEC_OK;
- case GST_FLOW_FLUSHING:
- return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
- default:
+ res = returnFromFlowReturn(gst_app_src_push_sample(GST_APP_SRC(m_src), sample.get()));
+ if (res != WEBRTC_VIDEO_CODEC_OK)
+ return res;
+
+ auto encodedSample = adoptGRef(gst_app_sink_try_pull_sample(GST_APP_SINK(m_sink), 5 * GST_SECOND));
+ if (!encodedSample) {
+ GST_ERROR("Didn't get any encodedSample");
return WEBRTC_VIDEO_CODEC_ERROR;
}
- }
- GstFlowReturn newSampleCallback(GstElement* sink)
- {
- auto sample = adoptGRef(gst_app_sink_pull_sample(GST_APP_SINK(sink)));
- auto buffer = gst_sample_get_buffer(sample.get());
- auto caps = gst_sample_get_caps(sample.get());
+ auto encodedBuffer = gst_sample_get_buffer(encodedSample.get());
+ auto encodedCaps = gst_sample_get_caps(encodedSample.get());
- webrtc::CodecSpecificInfo localCodecInfo;
- FrameData frameData = { 0, 0, localCodecInfo};
- {
- auto locker = holdLock(m_bufferMapLock);
- if (!m_framesData.size()) {
- gst_adapter_push(m_adapter.get(), gst_buffer_ref(buffer));
+ webrtc::RTPFragmentationHeader fragmentationInfo;
- return GST_FLOW_OK;
- }
-
- if (gst_adapter_available(m_adapter.get()) > 0) {
- uint flags = GST_BUFFER_FLAGS(buffer);
-
- GST_INFO_OBJECT(m_pipeline.get(), "Got more buffer than pushed frame, trying to deal with it.");
- gst_adapter_push(m_adapter.get(), gst_buffer_ref(buffer));
-
- buffer = gst_adapter_take_buffer(m_adapter.get(), gst_adapter_available(m_adapter.get()));
- GST_BUFFER_FLAGS(buffer) = flags;
- }
- frameData = m_framesData[0];
- m_framesData.remove(static_cast<size_t>(0));
- }
-
- webrtc::RTPFragmentationHeader fragmentationInfo;
- Fragmentize(&m_encodedFrame, &m_encodedImageBuffer, &m_encodedImageBufferSize, buffer, &fragmentationInfo);
+ Fragmentize(&m_encodedFrame, &m_encodedImageBuffer, &m_encodedImageBufferSize, encodedBuffer, &fragmentationInfo);
if (!m_encodedFrame._size)
- return GST_FLOW_OK;
+ return WEBRTC_VIDEO_CODEC_OK;
- gst_structure_get(gst_caps_get_structure(caps, 0),
+ gst_structure_get(gst_caps_get_structure(encodedCaps, 0),
"width", G_TYPE_INT, &m_encodedFrame._encodedWidth,
"height", G_TYPE_INT, &m_encodedFrame._encodedHeight,
nullptr);
- m_encodedFrame._frameType = GST_BUFFER_FLAG_IS_SET(buffer, GST_BUFFER_FLAG_DELTA_UNIT) ? webrtc::kVideoFrameDelta : webrtc::kVideoFrameKey;
- m_encodedFrame._completeFrame = false;
- m_encodedFrame.capture_time_ms_ = frameData.captureTimeMs;
- m_encodedFrame.SetTimestamp(frameData.rtpTimestamp);
+ m_encodedFrame._frameType = GST_BUFFER_FLAG_IS_SET(encodedBuffer, GST_BUFFER_FLAG_DELTA_UNIT) ? webrtc::kVideoFrameDelta : webrtc::kVideoFrameKey;
+ m_encodedFrame._completeFrame = true;
+ m_encodedFrame.capture_time_ms_ = frame.render_time_ms();
+ m_encodedFrame.SetTimestamp(frame.timestamp());
GST_LOG_OBJECT(m_pipeline.get(), "Got buffer capture_time_ms: %ld _timestamp: %u",
m_encodedFrame.capture_time_ms_, m_encodedFrame.Timestamp());
- PopulateCodecSpecific(&frameData.codecInfo, buffer);
-
- webrtc::EncodedImageCallback::Result result = m_imageReadyCb->OnEncodedImage(m_encodedFrame, &frameData.codecInfo, &fragmentationInfo);
+ webrtc::CodecSpecificInfo codecInfo;
+ PopulateCodecSpecific(&codecInfo, encodedBuffer);
+ webrtc::EncodedImageCallback::Result result = m_imageReadyCb->OnEncodedImage(m_encodedFrame, &codecInfo, &fragmentationInfo);
if (result.error != webrtc::EncodedImageCallback::Result::OK)
GST_ERROR_OBJECT(m_pipeline.get(), "Encode callback failed: %d", result.error);
- return GST_FLOW_OK;
+ return WEBRTC_VIDEO_CODEC_OK;
}
GRefPtr<GstElement> createEncoder(void)
@@ -386,11 +365,6 @@
}
private:
- static GstFlowReturn newSampleCallbackTramp(GstElement* sink, GStreamerVideoEncoder* enc)
- {
- return enc->newSampleCallback(sink);
- }
-
GRefPtr<GstElement> m_pipeline;
GstElement* m_src;
GstElement* m_encoder;
@@ -405,7 +379,7 @@
Lock m_bufferMapLock;
GRefPtr<GstAdapter> m_adapter;
- Vector<FrameData> m_framesData;
+ GstElement* m_sink;
};
class GStreamerH264Encoder : public GStreamerVideoEncoder {
@@ -495,11 +469,11 @@
GstH264NalParser* m_parser;
webrtc::VideoCodecType CodecType() final { return webrtc::kVideoCodecH264; }
- void PopulateCodecSpecific(webrtc::CodecSpecificInfo* codecSpecifiInfos, GstBuffer*) final
+ void PopulateCodecSpecific(webrtc::CodecSpecificInfo* codecSpecificInfos, GstBuffer*) final
{
- codecSpecifiInfos->codecType = CodecType();
- codecSpecifiInfos->codec_name = ImplementationName();
- webrtc::CodecSpecificInfoH264* h264Info = &(codecSpecifiInfos->codecSpecific.H264);
+ codecSpecificInfos->codecType = CodecType();
+ codecSpecificInfos->codec_name = ImplementationName();
+ webrtc::CodecSpecificInfoH264* h264Info = &(codecSpecificInfos->codecSpecific.H264);
h264Info->packetization_mode = packetizationMode;
}
@@ -519,11 +493,11 @@
return codecSettings->VP8().keyFrameInterval;
}
- void PopulateCodecSpecific(webrtc::CodecSpecificInfo* codecSpecifiInfos, GstBuffer* buffer) final
+ void PopulateCodecSpecific(webrtc::CodecSpecificInfo* codecSpecificInfos, GstBuffer* buffer) final
{
- codecSpecifiInfos->codecType = webrtc::kVideoCodecVP8;
- codecSpecifiInfos->codec_name = ImplementationName();
- webrtc::CodecSpecificInfoVP8* vp8Info = &(codecSpecifiInfos->codecSpecific.VP8);
+ codecSpecificInfos->codecType = webrtc::kVideoCodecVP8;
+ codecSpecificInfos->codec_name = ImplementationName();
+ webrtc::CodecSpecificInfoVP8* vp8Info = &(codecSpecificInfos->codecSpecific.VP8);
vp8Info->temporalIdx = 0;
vp8Info->keyIdx = webrtc::kNoKeyIdx;
