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Add new fmtp parameter for H.264

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Bug: webrtc:11769, webrtc:8423, webrtc:11376
Change-Id: Ia8f22ff90f817ba46ca03de1e43d3088c05023cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178904
Commit-Queue: Eldar Rello <elrello@microsoft.com>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31878}
This commit is contained in:
Eldar Rello
2020-08-07 12:08:14 +03:00
committed by Commit Bot
parent f7721fb246
commit 2127aaa64e
7 changed files with 94 additions and 22 deletions
Download Patch File Download Diff File Expand all files Collapse all files

8
modules/video_coding/packet_buffer.cc
View File

@ -31,7 +31,6 @@
#include "rtc_base/logging.h"
#include "rtc_base/numerics/mod_ops.h"
#include "system_wrappers/include/clock.h"
#include "system_wrappers/include/field_trial.h"
namespace webrtc {
namespace video_coding {
@ -63,8 +62,7 @@ PacketBuffer::PacketBuffer(Clock* clock,
first_packet_received_(false),
is_cleared_to_first_seq_num_(false),
buffer_(start_buffer_size),
sps_pps_idr_is_h264_keyframe_(
field_trial::IsEnabled("WebRTC-SpsPpsIdrIsH264Keyframe")) {
sps_pps_idr_is_h264_keyframe_(false) {
RTC_DCHECK_LE(start_buffer_size, max_buffer_size);
// Buffer size must always be a power of 2.
RTC_DCHECK((start_buffer_size & (start_buffer_size - 1)) == 0);
@ -194,7 +192,9 @@ absl::optional<int64_t> PacketBuffer::LastReceivedKeyframePacketMs() const {
MutexLock lock(&mutex_);
return last_received_keyframe_packet_ms_;
}
void PacketBuffer::ForceSpsPpsIdrIsH264Keyframe() {
sps_pps_idr_is_h264_keyframe_ = true;
}
void PacketBuffer::ClearInternal() {
for (auto& entry : buffer_) {
entry = nullptr;

3
modules/video_coding/packet_buffer.h
View File

@ -94,6 +94,7 @@ class PacketBuffer {
RTC_LOCKS_EXCLUDED(mutex_);
absl::optional<int64_t> LastReceivedKeyframePacketMs() const
RTC_LOCKS_EXCLUDED(mutex_);
void ForceSpsPpsIdrIsH264Keyframe();
private:
Clock* const clock_;
@ -147,7 +148,7 @@ class PacketBuffer {
// Indicates if we should require SPS, PPS, and IDR for a particular
// RTP timestamp to treat the corresponding frame as a keyframe.
const bool sps_pps_idr_is_h264_keyframe_;
bool sps_pps_idr_is_h264_keyframe_;
};
} // namespace video_coding

15
modules/video_coding/packet_buffer_unittest.cc
View File

@ -100,9 +100,8 @@ void PrintTo(const PacketBufferInsertResult& result, std::ostream* os) {
class PacketBufferTest : public ::testing::Test {
protected:
explicit PacketBufferTest(std::string field_trials = "")
: scoped_field_trials_(field_trials),
rand_(0x7732213),
PacketBufferTest()
: rand_(0x7732213),
clock_(0),
packet_buffer_(&clock_, kStartSize, kMaxSize) {}
@ -133,7 +132,6 @@ class PacketBufferTest : public ::testing::Test {
packet_buffer_.InsertPacket(std::move(packet)));
}
const test::ScopedFieldTrials scoped_field_trials_;
Random rand_;
SimulatedClock clock_;
PacketBuffer packet_buffer_;
@ -391,10 +389,11 @@ TEST_F(PacketBufferTest, InsertPacketAfterSequenceNumberWrapAround) {
class PacketBufferH264Test : public PacketBufferTest {
protected:
explicit PacketBufferH264Test(bool sps_pps_idr_is_keyframe)
: PacketBufferTest(sps_pps_idr_is_keyframe
? "WebRTC-SpsPpsIdrIsH264Keyframe/Enabled/"
: ""),
sps_pps_idr_is_keyframe_(sps_pps_idr_is_keyframe) {}
: PacketBufferTest(), sps_pps_idr_is_keyframe_(sps_pps_idr_is_keyframe) {
if (sps_pps_idr_is_keyframe) {
packet_buffer_.ForceSpsPpsIdrIsH264Keyframe();
}
}
PacketBufferInsertResult InsertH264(
uint16_t seq_num, // packet sequence number