Modified: trunk/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoDecoderFactory.cpp (236396 => 236397)
--- trunk/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoDecoderFactory.cpp 2018-09-24 10:20:38 UTC (rev 236396)
+++ trunk/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoDecoderFactory.cpp 2018-09-24 11:16:58 UTC (rev 236397)
@@ -159,15 +159,15 @@
}
// FIXME- Use a GstBufferPool.
- auto buffer = gst_buffer_new_wrapped(g_memdup(inputImage._buffer, inputImage._size),
- inputImage._size);
- GST_BUFFER_DTS(buffer) = (static_cast<guint64>(inputImage._timeStamp) * GST_MSECOND) - m_firstBufferDts;
- GST_BUFFER_PTS(buffer) = (static_cast<guint64>(renderTimeMs) * GST_MSECOND) - m_firstBufferPts;
- m_dtsPtsMap[GST_BUFFER_PTS(buffer)] = inputImage._timeStamp;
+ auto buffer = adoptGRef(gst_buffer_new_wrapped(g_memdup(inputImage._buffer, inputImage._size),
+ inputImage._size));
+ GST_BUFFER_DTS(buffer.get()) = (static_cast<guint64>(inputImage._timeStamp) * GST_MSECOND) - m_firstBufferDts;
+ GST_BUFFER_PTS(buffer.get()) = (static_cast<guint64>(renderTimeMs) * GST_MSECOND) - m_firstBufferPts;
+ m_dtsPtsMap[GST_BUFFER_PTS(buffer.get())] = inputImage._timeStamp;
GST_LOG_OBJECT(pipeline(), "%ld Decoding: %" GST_PTR_FORMAT, renderTimeMs, buffer);
- switch (gst_app_src_push_sample(GST_APP_SRC(m_src),
- gst_sample_new(buffer, GetCapsForFrame(inputImage), nullptr, nullptr))) {
+ auto sample = adoptGRef(gst_sample_new(buffer.get(), GetCapsForFrame(inputImage), nullptr, nullptr));
+ 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:
Modified: trunk/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoEncoderFactory.cpp (236396 => 236397)
--- trunk/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoEncoderFactory.cpp 2018-09-24 10:20:38 UTC (rev 236396)
+++ trunk/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoEncoderFactory.cpp 2018-09-24 11:16:58 UTC (rev 236397)
@@ -78,8 +78,8 @@
GST_INFO_OBJECT(m_pipeline.get(), "New bitrate: %d - framerate is %d",
newBitrate, frameRate);
- auto caps = gst_caps_make_writable(m_restrictionCaps.get());
- gst_caps_set_simple(caps, "framerate", GST_TYPE_FRACTION, frameRate, 1, nullptr);
+ auto caps = adoptGRef(gst_caps_copy(m_restrictionCaps.get()));
+ gst_caps_set_simple(caps.get(), "framerate", GST_TYPE_FRACTION, frameRate, 1, nullptr);
SetRestrictionCaps(caps);
@@ -107,6 +107,14 @@
g_return_val_if_fail(codecSettings, WEBRTC_VIDEO_CODEC_ERR_PARAMETER);
g_return_val_if_fail(codecSettings->codecType == CodecType(), WEBRTC_VIDEO_CODEC_ERR_PARAMETER);
+ m_encodedFrame._size = codecSettings->width * codecSettings->height * 3;
+ m_encodedFrame._buffer = new uint8_t[m_encodedFrame._size];
+ encoded_image_buffer_.reset(m_encodedFrame._buffer);
+ m_encodedFrame._completeFrame = true;
+ m_encodedFrame._encodedWidth = 0;
+ m_encodedFrame._encodedHeight = 0;
+ m_encodedFrame._length = 0;
+
m_pipeline = makeElement("pipeline");
connectSimpleBusMessageCallback(m_pipeline.get());
@@ -150,6 +158,8 @@
int32_t Release() final
{
+ m_encodedFrame._buffer = nullptr;
+ encoded_image_buffer_.reset();
GRefPtr<GstBus> bus = adoptGRef(gst_pipeline_get_bus(GST_PIPELINE(m_pipeline.get())));
gst_bus_set_sync_handler(bus.get(), nullptr, nullptr, nullptr);
@@ -219,26 +229,26 @@
auto buffer = gst_sample_get_buffer(sample.get());
auto caps = gst_sample_get_caps(sample.get());
- webrtc::RTPFragmentationHeader* fragmentationInfo;
- auto frame = Fragmentize(buffer, &fragmentationInfo);
- if (!frame._size)
+ webrtc::RTPFragmentationHeader fragmentationInfo;
+ Fragmentize(&m_encodedFrame, &encoded_image_buffer_, buffer, &fragmentationInfo);
+ if (!m_encodedFrame._size)
return GST_FLOW_OK;
gst_structure_get(gst_caps_get_structure(caps, 0),
- "width", G_TYPE_INT, &frame._encodedWidth,
- "height", G_TYPE_INT, &frame._encodedHeight,
+ "width", G_TYPE_INT, &m_encodedFrame._encodedWidth,
+ "height", G_TYPE_INT, &m_encodedFrame._encodedHeight,
nullptr);
- frame._frameType = GST_BUFFER_FLAG_IS_SET(buffer, GST_BUFFER_FLAG_DELTA_UNIT) ? webrtc::kVideoFrameDelta : webrtc::kVideoFrameKey;
- frame._completeFrame = true;
- frame.capture_time_ms_ = GST_TIME_AS_MSECONDS(GST_BUFFER_PTS(buffer));
- frame._timeStamp = GST_TIME_AS_MSECONDS(GST_BUFFER_DTS(buffer));
+ m_encodedFrame._frameType = GST_BUFFER_FLAG_IS_SET(buffer, GST_BUFFER_FLAG_DELTA_UNIT) ? webrtc::kVideoFrameDelta : webrtc::kVideoFrameKey;
+ m_encodedFrame._completeFrame = true;
+ m_encodedFrame.capture_time_ms_ = GST_TIME_AS_MSECONDS(GST_BUFFER_PTS(buffer));
+ m_encodedFrame._timeStamp = GST_TIME_AS_MSECONDS(GST_BUFFER_DTS(buffer));
GST_LOG_OBJECT(m_pipeline.get(), "Got buffer TS: %" GST_TIME_FORMAT, GST_TIME_ARGS(GST_BUFFER_PTS(buffer)));
webrtc::CodecSpecificInfo codecSpecifiInfos;
PopulateCodecSpecific(&codecSpecifiInfos, buffer);
- webrtc::EncodedImageCallback::Result result = m_imageReadyCb->OnEncodedImage(frame, &codecSpecifiInfos, fragmentationInfo);
+ webrtc::EncodedImageCallback::Result result = m_imageReadyCb->OnEncodedImage(m_encodedFrame, &codecSpecifiInfos, &fragmentationInfo);
if (result.error != webrtc::EncodedImageCallback::Result::OK) {
GST_ELEMENT_ERROR(m_pipeline.get(), LIBRARY, FAILED, (nullptr),
("Encode callback failed: %d", result.error));
@@ -349,26 +359,25 @@
virtual void PopulateCodecSpecific(webrtc::CodecSpecificInfo*, GstBuffer*) = 0;
- virtual webrtc::EncodedImage Fragmentize(GstBuffer* buffer, webrtc::RTPFragmentationHeader** outFragmentationInfo)
+ virtual void Fragmentize(webrtc::EncodedImage* encodedImage, std::unique_ptr<uint8_t[]>* encoded_image_buffer, GstBuffer* buffer,
+ webrtc::RTPFragmentationHeader* fragmentationInfo)
{
GstMapInfo map;
gst_buffer_map(buffer, &map, GST_MAP_READ);
- webrtc::EncodedImage frame(map.data, map.size, map.size);
+ if (encodedImage->_size < map.size) {
+ encodedImage->_size = map.size;
+ encodedImage->_buffer = new uint8_t[encodedImage->_size];
+ encoded_image_buffer->reset(encodedImage->_buffer);
+ memcpy(encodedImage->_buffer, map.data, map.size);
+ }
gst_buffer_unmap(buffer, &map);
- // No fragmentation by default.
- webrtc::RTPFragmentationHeader* fragmentationInfo = new webrtc::RTPFragmentationHeader();
-
fragmentationInfo->VerifyAndAllocateFragmentationHeader(1);
fragmentationInfo->fragmentationOffset[0] = 0;
fragmentationInfo->fragmentationLength[0] = gst_buffer_get_size(buffer);
fragmentationInfo->fragmentationPlType[0] = 0;
fragmentationInfo->fragmentationTimeDiff[0] = 0;
-
- *outFragmentationInfo = fragmentationInfo;
-
- return frame;
}
const char* ImplementationName() const
@@ -380,9 +389,9 @@
virtual const gchar* Name() = 0;
- void SetRestrictionCaps(GstCaps* caps)
+ void SetRestrictionCaps(GRefPtr<GstCaps> caps)
{
- if (caps && m_profile.get() && gst_caps_is_equal(m_restrictionCaps.get(), caps))
+ if (caps && m_profile.get() && gst_caps_is_equal(m_restrictionCaps.get(), caps.get()))
g_object_set(m_profile.get(), "restriction-caps", caps, nullptr);
m_restrictionCaps = caps;
@@ -403,6 +412,8 @@
GRefPtr<GstCaps> m_restrictionCaps;
GRefPtr<GstEncodingProfile> m_profile;
BitrateSetter m_bitrateSetter;
+ webrtc::EncodedImage m_encodedFrame;
+ std::unique_ptr<uint8_t[]> encoded_image_buffer_;
};
class H264Encoder : public GStreamerVideoEncoder {
@@ -420,7 +431,8 @@
}
// FIXME - MT. safety!
- webrtc::EncodedImage Fragmentize(GstBuffer* gstbuffer, webrtc::RTPFragmentationHeader** outFragmentationInfo) final
+ void Fragmentize(webrtc::EncodedImage* encodedImage, std::unique_ptr<uint8_t[]>* encoded_image_buffer,
+ GstBuffer* gstbuffer, webrtc::RTPFragmentationHeader* fragmentationHeader) final
{
GstMapInfo map;
GstH264NalUnit nalu;
@@ -430,7 +442,6 @@
size_t requiredSize = 0;
std::vector<GstH264NalUnit> nals;
- webrtc::EncodedImage encodedImage;
const uint8_t startCode[4] = { 0, 0, 0, 1 };
gst_buffer_map(gstbuffer, &map, GST_MAP_READ);
@@ -447,13 +458,16 @@
offset = nalu.offset + nalu.size;
}
- encodedImage._size = requiredSize;
- encodedImage._buffer = new uint8_t[encodedImage._size];
+ if (encodedImage->_size < requiredSize) {
+ encodedImage->_size = requiredSize;
+ encodedImage->_buffer = new uint8_t[encodedImage->_size];
+ encoded_image_buffer->reset(encodedImage->_buffer);
+ }
+
// Iterate nal units and fill the Fragmentation info.
- webrtc::RTPFragmentationHeader* fragmentationHeader = new webrtc::RTPFragmentationHeader();
fragmentationHeader->VerifyAndAllocateFragmentationHeader(nals.size());
size_t fragmentIndex = 0;
- encodedImage._length = 0;
+ encodedImage->_length = 0;
for (std::vector<GstH264NalUnit>::iterator nal = nals.begin(); nal != nals.end(); ++nal, fragmentIndex++) {
ASSERT(map.data[nal->sc_offset + 0] == startCode[0]);
@@ -464,14 +478,12 @@
fragmentationHeader->fragmentationOffset[fragmentIndex] = nal->offset;
fragmentationHeader->fragmentationLength[fragmentIndex] = nal->size;
- memcpy(encodedImage._buffer + encodedImage._length, &map.data[nal->sc_offset],
+ memcpy(encodedImage->_buffer + encodedImage->_length, &map.data[nal->sc_offset],
sizeof(startCode) + nal->size);
- encodedImage._length += nal->size + sizeof(startCode);
+ encodedImage->_length += nal->size + sizeof(startCode);
}
- *outFragmentationInfo = fragmentationHeader;
gst_buffer_unmap(gstbuffer, &map);
- return encodedImage;
}
GstElement* CreateFilter() final
