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Added RtpFrameObject::SetBitstream so that the frame can be updated with the decrypted payload.

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Bug: webrtc:9361
Change-Id: I5d61219033f7c3ff7e7691b74322bfa44f49e326
Reviewed-on: https://webrtc-review.googlesource.com/103221
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24934}
This commit is contained in:
philipel
2018-10-02 17:01:45 +02:00
committed by Commit Bot
parent d2650d1a28
commit 5fb245498c
2 changed files with 28 additions and 14 deletions
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39
modules/video_coding/frame_object.cc
View File

@ -48,20 +48,7 @@ RtpFrameObject::RtpFrameObject(PacketBuffer* packet_buffer,
// as of the first packet's.
SetPlayoutDelay(first_packet->video_header.playout_delay);
// Since FFmpeg use an optimized bitstream reader that reads in chunks of
// 32/64 bits we have to add at least that much padding to the buffer
// to make sure the decoder doesn't read out of bounds.
// NOTE! EncodedImage::_size is the size of the buffer (think capacity of
// an std::vector) and EncodedImage::_length is the actual size of
// the bitstream (think size of an std::vector).
if (codec_type_ == kVideoCodecH264)
_size = frame_size + EncodedImage::kBufferPaddingBytesH264;
else
_size = frame_size;
_buffer = new uint8_t[_size];
_length = frame_size;
AllocateBitstreamBuffer(frame_size);
bool bitstream_copied = GetBitstream(_buffer);
RTC_DCHECK(bitstream_copied);
_encodedWidth = first_packet->width;
@ -135,6 +122,11 @@ VideoCodecType RtpFrameObject::codec_type() const {
return codec_type_;
}
void RtpFrameObject::SetBitstream(rtc::ArrayView<const uint8_t> bitstream) {
AllocateBitstreamBuffer(bitstream.size());
memcpy(_buffer, bitstream.data(), _length);
}
bool RtpFrameObject::GetBitstream(uint8_t* destination) const {
return packet_buffer_->GetBitstream(*this, destination);
}
@ -176,5 +168,24 @@ absl::optional<FrameMarking> RtpFrameObject::GetFrameMarking() const {
return packet->video_header.frame_marking;
}
void RtpFrameObject::AllocateBitstreamBuffer(size_t frame_size) {
// Since FFmpeg use an optimized bitstream reader that reads in chunks of
// 32/64 bits we have to add at least that much padding to the buffer
// to make sure the decoder doesn't read out of bounds.
// NOTE! EncodedImage::_size is the size of the buffer (think capacity of
// an std::vector) and EncodedImage::_length is the actual size of
// the bitstream (think size of an std::vector).
size_t new_size = frame_size + (codec_type_ == kVideoCodecH264
? EncodedImage::kBufferPaddingBytesH264
: 0);
if (_size < new_size) {
delete[] _buffer;
_buffer = new uint8_t[new_size];
_size = new_size;
}
_length = frame_size;
}
} // namespace video_coding
} // namespace webrtc

3
modules/video_coding/frame_object.h
View File

@ -37,6 +37,7 @@ class RtpFrameObject : public EncodedFrame {
int times_nacked() const;
enum FrameType frame_type() const;
VideoCodecType codec_type() const;
void SetBitstream(rtc::ArrayView<const uint8_t> bitstream);
bool GetBitstream(uint8_t* destination) const override;
int64_t ReceivedTime() const override;
int64_t RenderTime() const override;
@ -46,6 +47,8 @@ class RtpFrameObject : public EncodedFrame {
absl::optional<FrameMarking> GetFrameMarking() const;
private:
void AllocateBitstreamBuffer(size_t frame_size);
rtc::scoped_refptr<PacketBuffer> packet_buffer_;
enum FrameType frame_type_;
VideoCodecType codec_type_;