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[fix] fix compile error on MacOSX

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This commit is contained in:
dijunkun
2025-02-10 15:46:22 +08:00
parent 1f3c93c77a
commit 2c48ce12c5
11 changed files with 23 additions and 24 deletions
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2
src/channel/rtp_channel/rtp_video_receiver.h
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@@ -14,7 +14,7 @@
#include "rtcp_receiver_report.h"
#include "rtp_packet_av1.h"
#include "rtp_packet_h264.h"
#include "rtp_rtcp_defines.h "
#include "rtp_rtcp_defines.h"
#include "rtp_statistics.h"
#include "thread_base.h"
#include "video_frame.h"

2
src/common/copy_on_write_buffer.h
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@@ -38,7 +38,7 @@ class CopyOnWriteBuffer {
private:
void EnsureUnique() {
if (!buffer_.unique()) {
if (buffer_.use_count() != 1) {
buffer_ = std::make_shared<std::vector<uint8_t>>(*buffer_);
}
}

2
src/common/rtc_base/ref_counter.h
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@@ -12,7 +12,7 @@
#include <atomic>
#include "ref_count.h"
#include "api/ref_count.h"
namespace webrtc {
namespace webrtc_impl {

2
src/common/rtc_base/system_time.cc
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@@ -18,10 +18,10 @@
#if defined(__POSIX__)
#include <sys/time.h>
#endif
#if defined(__APPLE__)
#include <mach/mach_time.h>
#endif
#endif
#if defined(_WIN32)
// clang-format off

8
src/common/rtc_base/time_utils.cc
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@@ -15,13 +15,12 @@
#endif
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/system_time.h"
#include "rtc_base/time_utils.h"
#include "system_time.h"
#if defined(_WIN32)
#include "rtc_base/win32.h"
#endif
#if defined(_WIN32)
#include <minwinbase.h>
#include "rtc_base/win32.h"
#endif
namespace rtc {
@@ -228,6 +227,7 @@ int64_t TimeUTCMicros() {
return (li.QuadPart - kFileTimeToUnixTimeEpochOffset) /
kFileTimeToMicroSeconds;
#endif
return 0;
}
int64_t TimeUTCMillis() { return TimeUTCMicros() / kNumMicrosecsPerMillisec; }

2
src/common/rtc_base/time_utils.h
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@@ -14,7 +14,7 @@
#include <stdint.h>
#include <time.h>
#include "system_time.h"
#include "rtc_base/system_time.h"
namespace rtc {

1
src/qos/transport_feedback_adapter.cc
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@@ -96,6 +96,7 @@ void TransportFeedbackAdapter::AddPacket(const RtpPacketToSend& packet_to_send,
feedback.sent.sequence_number = seq_num_unwrapper_.Unwrap(
packet_to_send.transport_sequence_number().value_or(0));
feedback.sent.size = DataSize::Bytes(packet_to_send.size() + overhead_bytes);
LOG_WARN("!!! packet_to_send.size: {}", packet_to_send.size());
feedback.sent.audio =
packet_to_send.packet_type() == RtpPacketMediaType::kAudio;
feedback.network_route = network_route_;

9
src/rtp/rtp_packetizer/rtp_packetizer_generic.cpp
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@@ -24,9 +24,9 @@ std::vector<RtpPacket> RtpPacketizerGeneric::Build(uint8_t* payload,
payload_size / MAX_NALU_LEN + (last_packet_size ? 1 : 0);
// TODO: use frame timestamp
timestamp_ = std::chrono::duration_cast<std::chrono::microseconds>(
std::chrono::system_clock::now().time_since_epoch())
.count();
uint64_t timestamp = std::chrono::duration_cast<std::chrono::microseconds>(
std::chrono::system_clock::now().time_since_epoch())
.count();
std::vector<RtpPacket> rtp_packets;
for (uint32_t index = 0; index < packet_num; index++) {
@@ -37,11 +37,10 @@ std::vector<RtpPacket> RtpPacketizerGeneric::Build(uint8_t* payload,
marker_ = index == packet_num - 1 ? 1 : 0;
payload_type_ = rtp::PAYLOAD_TYPE(payload_type_);
sequence_number_++;
timestamp_ = timestamp_;
timestamp_ = timestamp;
ssrc_ = ssrc_;
if (!csrc_count_) {
csrcs_ = csrcs_;
}
rtp_packet_frame_.clear();

10
src/rtp/rtp_packetizer/rtp_packetizer_h264.cpp
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@@ -81,10 +81,8 @@ std::vector<RtpPacket> RtpPacketizerH264::BuildNalu(uint8_t* payload,
timestamp_ = std::chrono::duration_cast<std::chrono::microseconds>(
std::chrono::system_clock::now().time_since_epoch())
.count();
ssrc_ = ssrc_;
if (!csrc_count_) {
csrcs_ = csrcs_;
}
rtp::FU_INDICATOR fu_indicator;
@@ -141,9 +139,9 @@ std::vector<RtpPacket> RtpPacketizerH264::BuildFua(uint8_t* payload,
payload_size / MAX_NALU_LEN + (last_packet_size ? 1 : 0);
// TODO: use frame timestamp
timestamp_ = std::chrono::duration_cast<std::chrono::microseconds>(
std::chrono::system_clock::now().time_since_epoch())
.count();
uint64_t timestamp = std::chrono::duration_cast<std::chrono::microseconds>(
std::chrono::system_clock::now().time_since_epoch())
.count();
for (uint32_t index = 0; index < packet_num; index++) {
version_ = kRtpVersion;
@@ -153,7 +151,7 @@ std::vector<RtpPacket> RtpPacketizerH264::BuildFua(uint8_t* payload,
marker_ = (index == (packet_num - 1)) ? 1 : 0;
payload_type_ = rtp::PAYLOAD_TYPE(payload_type_);
sequence_number_++;
timestamp_ = timestamp_;
timestamp_ = timestamp;
ssrc_ = ssrc_;
if (!csrc_count_) {

6
src/rtp/rtp_packetizer/rtp_packetizer_h264.h
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@@ -18,11 +18,9 @@ class RtpPacketizerH264 : public RtpPacketizer {
std::vector<RtpPacket> Build(uint8_t* payload,
uint32_t payload_size) override;
std::vector<RtpPacket> RtpPacketizerH264::BuildNalu(uint8_t* payload,
uint32_t payload_size);
std::vector<RtpPacket> BuildNalu(uint8_t* payload, uint32_t payload_size);
std::vector<RtpPacket> RtpPacketizerH264::BuildFua(uint8_t* payload,
uint32_t payload_size);
std::vector<RtpPacket> BuildFua(uint8_t* payload, uint32_t payload_size);
private:
bool EncodeH264Fua(RtpPacket& rtp_packet, uint8_t* payload,

3
xmake.lua
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@@ -48,6 +48,9 @@ target("common")
"src/common/rtc_base/numerics/*.cc",
"src/common/api/units/*.cc",
"src/common/api/transport/*.cc")
if not is_os("windows") then
remove_files("src/common/rtc_base/win32.cc")
end
add_includedirs("src/common", {public = true})
target("inih")