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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
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8
modules/video_coding/packet_buffer.cc
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@ -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;