webrtc/modules/rtp_rtcp/source/rtp_format_h264.h - Issue 2337453002: H.264 packetization mode 0 (try 2)

Unified Diff: webrtc/modules/rtp_rtcp/source/rtp_format_h264.h

Issue 2337453002: H.264 packetization mode 0 (try 2) (Closed)

Patch Set: Working H.264 test where packetization mode 0 is set Created 4 years, 2 months ago

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Index: webrtc/modules/rtp_rtcp/source/rtp_format_h264.h

diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_h264.h b/webrtc/modules/rtp_rtcp/source/rtp_format_h264.h

index 9cf3150dfa25231fa80ec79456fe6e83ad4381df..4e7772d8ea5b13eb24d86d9eae2bcecc6e4fe60a 100644

--- a/webrtc/modules/rtp_rtcp/source/rtp_format_h264.h

+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_h264.h

@@ -12,6 +12,7 @@

#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_

#include <deque>

+#include <memory>

#include <queue>

#include <string>

@@ -25,7 +26,8 @@ class RtpPacketizerH264 : public RtpPacketizer {

public:

// Initialize with payload from encoder.

// The payload_data must be exactly one encoded H264 frame.

- RtpPacketizerH264(FrameType frame_type, size_t max_payload_len);

+ RtpPacketizerH264(const RTPVideoHeaderH264& parameters,

+ size_t max_payload_len);

virtual ~RtpPacketizerH264();

@@ -92,6 +94,7 @@ class RtpPacketizerH264 : public RtpPacketizer {

void NextAggregatePacket(uint8_t* buffer, size_t* bytes_to_send);

void NextFragmentPacket(uint8_t* buffer, size_t* bytes_to_send);

+ H264PacketizationMode packetization_mode_;

hbos 2016/10/31 10:32:26 Would it make more sense to store all parameters a

hta-webrtc 2016/10/31 15:03:04 Absolutely not. webrtc::RTPVideoHeaderH264 from mo

hbos 2016/10/31 16:27:15 Uhm, confused. I'm thinking about this line: https

const size_t max_payload_len_;

std::deque<Fragment> input_fragments_;

std::queue<PacketUnit> packets_;