commit 7a2ee012715e41f722b35ca3397d430c839f9c9a
parent 312d76660c8fa4b677b215f1a93a8c8998ca7156
Author: Michael Froman <mfroman@mozilla.com>
Date: Wed, 8 Oct 2025 23:21:25 -0500
Bug 1993083 - Vendor libwebrtc from b01ae85f9a
Upstream commit: https://webrtc.googlesource.com/src/+/b01ae85f9a0465da301222bb599735838ade831e
Use ArrayView in webrtc voice engine unittests
Bug: webrtc:42225170
Change-Id: I6c9bfb7c48fcde7c992841ab8d300bb491b61b11
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/399501
Auto-Submit: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#45113}
Diffstat:
4 files changed, 67 insertions(+), 74 deletions(-)
diff --git a/third_party/libwebrtc/README.mozilla.last-vendor b/third_party/libwebrtc/README.mozilla.last-vendor
@@ -1,4 +1,4 @@
# ./mach python dom/media/webrtc/third_party_build/vendor-libwebrtc.py --from-local /home/mfroman/mozilla/elm/.moz-fast-forward/moz-libwebrtc --commit mozpatches libwebrtc
-libwebrtc updated from /home/mfroman/mozilla/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2025-10-09T04:20:11.801364+00:00.
+libwebrtc updated from /home/mfroman/mozilla/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2025-10-09T04:21:16.462311+00:00.
# base of lastest vendoring
-52b34e1d20
+b01ae85f9a
diff --git a/third_party/libwebrtc/media/engine/fake_webrtc_call.cc b/third_party/libwebrtc/media/engine/fake_webrtc_call.cc
@@ -121,16 +121,15 @@ void FakeAudioReceiveStream::SetStats(
stats_ = stats;
}
-bool FakeAudioReceiveStream::VerifyLastPacket(const uint8_t* data,
- size_t length) const {
- return last_packet_ == Buffer(data, length);
+bool FakeAudioReceiveStream::VerifyLastPacket(
+ ArrayView<const uint8_t> data) const {
+ return last_packet_ == Buffer(data.data(), data.size());
}
-bool FakeAudioReceiveStream::DeliverRtp(const uint8_t* packet,
- size_t length,
+bool FakeAudioReceiveStream::DeliverRtp(ArrayView<const uint8_t> packet,
int64_t /* packet_time_us */) {
++received_packets_;
- last_packet_.SetData(packet, length);
+ last_packet_.SetData(packet);
return true;
}
@@ -690,7 +689,7 @@ bool FakeCall::DeliverPacketInternal(MediaType media_type,
if (media_type == MediaType::AUDIO) {
for (auto receiver : audio_receive_streams_) {
if (receiver->GetConfig().rtp.remote_ssrc == ssrc) {
- receiver->DeliverRtp(packet.cdata(), packet.size(), arrival_time.us());
+ receiver->DeliverRtp(packet, arrival_time.us());
++delivered_packets_by_ssrc_[ssrc];
return true;
}
diff --git a/third_party/libwebrtc/media/engine/fake_webrtc_call.h b/third_party/libwebrtc/media/engine/fake_webrtc_call.h
@@ -124,10 +124,10 @@ class FakeAudioReceiveStream final : public AudioReceiveStreamInterface {
const AudioReceiveStreamInterface::Config& GetConfig() const;
void SetStats(const AudioReceiveStreamInterface::Stats& stats);
int received_packets() const { return received_packets_; }
- bool VerifyLastPacket(const uint8_t* data, size_t length) const;
+ bool VerifyLastPacket(ArrayView<const uint8_t> data) const;
const AudioSinkInterface* sink() const { return sink_; }
float gain() const { return gain_; }
- bool DeliverRtp(const uint8_t* packet, size_t length, int64_t packet_time_us);
+ bool DeliverRtp(ArrayView<const uint8_t> packet, int64_t packet_time_us);
bool started() const { return started_; }
int base_mininum_playout_delay_ms() const {
return base_mininum_playout_delay_ms_;
diff --git a/third_party/libwebrtc/media/engine/webrtc_voice_engine_unittest.cc b/third_party/libwebrtc/media/engine/webrtc_voice_engine_unittest.cc
@@ -22,6 +22,7 @@
#include <vector>
#include "absl/strings/match.h"
+#include "api/array_view.h"
#include "api/audio/audio_processing.h"
#include "api/audio/builtin_audio_processing_builder.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
@@ -79,6 +80,7 @@
#include "test/mock_audio_decoder_factory.h"
#include "test/mock_audio_encoder_factory.h"
+namespace webrtc {
namespace {
using ::testing::_;
using ::testing::ContainerEq;
@@ -211,8 +213,6 @@ std::vector<webrtc::Codec> ReceiveCodecsWithId(
return AddIdToCodecs(pt_mapper, std::move(codecs));
}
-} // namespace
-
// Tests that our stub library "works".
TEST(WebRtcVoiceEngineTestStubLibrary, StartupShutdown) {
Environment env = CreateEnvironment();
@@ -351,9 +351,9 @@ class WebRtcVoiceEngineTestFake : public ::testing::TestWithParam<bool> {
EXPECT_FALSE(call_.GetAudioSendStream(kSsrcX));
}
- void DeliverPacket(const void* data, int len) {
+ void DeliverPacket(ArrayView<const uint8_t> data) {
webrtc::RtpPacketReceived packet;
- packet.Parse(reinterpret_cast<const uint8_t*>(data), len);
+ packet.Parse(data);
receive_channel_->OnPacketReceived(packet);
webrtc::Thread::Current()->ProcessMessages(0);
}
@@ -1601,7 +1601,7 @@ TEST_P(WebRtcVoiceEngineTestFake, GetRtpReceiveParametersWithUnsignaledSsrc) {
EXPECT_FALSE(rtp_parameters.encodings[0].ssrc);
// Receive PCMU packet (SSRC=1).
- DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
+ DeliverPacket(kPcmuFrame);
// The `ssrc` member should still be unset.
rtp_parameters = receive_channel_->GetDefaultRtpReceiveParameters();
@@ -2597,7 +2597,7 @@ TEST_P(WebRtcVoiceEngineTestFake, GetStatsWithMultipleSendStreams) {
{
webrtc::VoiceMediaSendInfo send_info;
webrtc::VoiceMediaReceiveInfo receive_info;
- DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
+ DeliverPacket(kPcmuFrame);
SetAudioReceiveStreamStats();
EXPECT_CALL(*adm_, GetPlayoutUnderrunCount()).WillOnce(Return(0));
EXPECT_EQ(true, send_channel_->GetStats(&send_info));
@@ -2756,7 +2756,7 @@ TEST_P(WebRtcVoiceEngineTestFake, GetStats) {
// Deliver a new packet - a default receive stream should be created and we
// should see stats again.
{
- DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
+ DeliverPacket(kPcmuFrame);
SetAudioReceiveStreamStats();
EXPECT_CALL(*adm_, GetPlayoutUnderrunCount()).WillOnce(Return(0));
webrtc::VoiceMediaSendInfo send_info;
@@ -2795,10 +2795,9 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendSsrcAfterCreatingReceiveChannel) {
TEST_P(WebRtcVoiceEngineTestFake, Recv) {
EXPECT_TRUE(SetupChannel());
EXPECT_TRUE(AddRecvStream(1));
- DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
+ DeliverPacket(kPcmuFrame);
- EXPECT_TRUE(
- GetRecvStream(1).VerifyLastPacket(kPcmuFrame, sizeof(kPcmuFrame)));
+ EXPECT_TRUE(GetRecvStream(1).VerifyLastPacket(kPcmuFrame));
}
// Test that we can properly receive packets on multiple streams.
@@ -2811,7 +2810,7 @@ TEST_P(WebRtcVoiceEngineTestFake, RecvWithMultipleStreams) {
EXPECT_TRUE(AddRecvStream(ssrc2));
EXPECT_TRUE(AddRecvStream(ssrc3));
// Create packets with the right SSRCs.
- unsigned char packets[4][sizeof(kPcmuFrame)];
+ uint8_t packets[4][sizeof(kPcmuFrame)];
for (size_t i = 0; i < std::size(packets); ++i) {
memcpy(packets[i], kPcmuFrame, sizeof(kPcmuFrame));
webrtc::SetBE32(packets[i] + 8, static_cast<uint32_t>(i));
@@ -2825,28 +2824,28 @@ TEST_P(WebRtcVoiceEngineTestFake, RecvWithMultipleStreams) {
EXPECT_EQ(s2.received_packets(), 0);
EXPECT_EQ(s3.received_packets(), 0);
- DeliverPacket(packets[0], sizeof(packets[0]));
+ DeliverPacket(packets[0]);
EXPECT_EQ(s1.received_packets(), 0);
EXPECT_EQ(s2.received_packets(), 0);
EXPECT_EQ(s3.received_packets(), 0);
- DeliverPacket(packets[1], sizeof(packets[1]));
+ DeliverPacket(packets[1]);
EXPECT_EQ(s1.received_packets(), 1);
- EXPECT_TRUE(s1.VerifyLastPacket(packets[1], sizeof(packets[1])));
+ EXPECT_TRUE(s1.VerifyLastPacket(packets[1]));
EXPECT_EQ(s2.received_packets(), 0);
EXPECT_EQ(s3.received_packets(), 0);
- DeliverPacket(packets[2], sizeof(packets[2]));
+ DeliverPacket(packets[2]);
EXPECT_EQ(s1.received_packets(), 1);
EXPECT_EQ(s2.received_packets(), 1);
- EXPECT_TRUE(s2.VerifyLastPacket(packets[2], sizeof(packets[2])));
+ EXPECT_TRUE(s2.VerifyLastPacket(packets[2]));
EXPECT_EQ(s3.received_packets(), 0);
- DeliverPacket(packets[3], sizeof(packets[3]));
+ DeliverPacket(packets[3]);
EXPECT_EQ(s1.received_packets(), 1);
EXPECT_EQ(s2.received_packets(), 1);
EXPECT_EQ(s3.received_packets(), 1);
- EXPECT_TRUE(s3.VerifyLastPacket(packets[3], sizeof(packets[3])));
+ EXPECT_TRUE(s3.VerifyLastPacket(packets[3]));
EXPECT_TRUE(receive_channel_->RemoveRecvStream(ssrc3));
EXPECT_TRUE(receive_channel_->RemoveRecvStream(ssrc2));
@@ -2858,11 +2857,10 @@ TEST_P(WebRtcVoiceEngineTestFake, RecvUnsignaled) {
EXPECT_TRUE(SetupChannel());
EXPECT_EQ(0u, call_.GetAudioReceiveStreams().size());
- DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
+ DeliverPacket(kPcmuFrame);
EXPECT_EQ(1u, call_.GetAudioReceiveStreams().size());
- EXPECT_TRUE(
- GetRecvStream(kSsrc1).VerifyLastPacket(kPcmuFrame, sizeof(kPcmuFrame)));
+ EXPECT_TRUE(GetRecvStream(kSsrc1).VerifyLastPacket(kPcmuFrame));
}
// Tests that when we add a stream without SSRCs, but contains a stream_id
@@ -2877,12 +2875,10 @@ TEST_P(WebRtcVoiceEngineTestFake, RecvUnsignaledSsrcWithSignaledStreamId) {
// The stream shouldn't have been created at this point because it doesn't
// have any SSRCs.
EXPECT_EQ(0u, call_.GetAudioReceiveStreams().size());
-
- DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
+ DeliverPacket(kPcmuFrame);
EXPECT_EQ(1u, call_.GetAudioReceiveStreams().size());
- EXPECT_TRUE(
- GetRecvStream(kSsrc1).VerifyLastPacket(kPcmuFrame, sizeof(kPcmuFrame)));
+ EXPECT_TRUE(GetRecvStream(kSsrc1).VerifyLastPacket(kPcmuFrame));
EXPECT_EQ(kSyncLabel, GetRecvStream(kSsrc1).GetConfig().sync_group);
// Remset the unsignaled stream to clear the cached parameters. If a new
@@ -2890,11 +2886,10 @@ TEST_P(WebRtcVoiceEngineTestFake, RecvUnsignaledSsrcWithSignaledStreamId) {
receive_channel_->ResetUnsignaledRecvStream();
receive_channel_->RemoveRecvStream(kSsrc1);
- DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
+ DeliverPacket(kPcmuFrame);
EXPECT_EQ(1u, call_.GetAudioReceiveStreams().size());
- EXPECT_TRUE(
- GetRecvStream(kSsrc1).VerifyLastPacket(kPcmuFrame, sizeof(kPcmuFrame)));
+ EXPECT_TRUE(GetRecvStream(kSsrc1).VerifyLastPacket(kPcmuFrame));
EXPECT_TRUE(GetRecvStream(kSsrc1).GetConfig().sync_group.empty());
}
@@ -2905,12 +2900,12 @@ TEST_P(WebRtcVoiceEngineTestFake,
ASSERT_TRUE(call_.GetAudioReceiveStreams().empty());
// Deliver a couple packets with unsignaled SSRCs.
- unsigned char packet[sizeof(kPcmuFrame)];
+ uint8_t packet[sizeof(kPcmuFrame)];
memcpy(packet, kPcmuFrame, sizeof(kPcmuFrame));
webrtc::SetBE32(&packet[8], 0x1234);
- DeliverPacket(packet, sizeof(packet));
+ DeliverPacket(packet);
webrtc::SetBE32(&packet[8], 0x5678);
- DeliverPacket(packet, sizeof(packet));
+ DeliverPacket(packet);
// Verify that the receive streams were created.
const auto& receivers1 = call_.GetAudioReceiveStreams();
@@ -2926,34 +2921,34 @@ TEST_P(WebRtcVoiceEngineTestFake,
// that packets are forwarded to them all.
TEST_P(WebRtcVoiceEngineTestFake, RecvMultipleUnsignaled) {
EXPECT_TRUE(SetupChannel());
- unsigned char packet[sizeof(kPcmuFrame)];
+ uint8_t packet[sizeof(kPcmuFrame)];
memcpy(packet, kPcmuFrame, sizeof(kPcmuFrame));
// Note that SSRC = 0 is not supported.
for (uint32_t ssrc = 1; ssrc < (1 + kMaxUnsignaledRecvStreams); ++ssrc) {
webrtc::SetBE32(&packet[8], ssrc);
- DeliverPacket(packet, sizeof(packet));
+ DeliverPacket(packet);
// Verify we have one new stream for each loop iteration.
EXPECT_EQ(ssrc, call_.GetAudioReceiveStreams().size());
EXPECT_EQ(1, GetRecvStream(ssrc).received_packets());
- EXPECT_TRUE(GetRecvStream(ssrc).VerifyLastPacket(packet, sizeof(packet)));
+ EXPECT_TRUE(GetRecvStream(ssrc).VerifyLastPacket(packet));
}
// Sending on the same SSRCs again should not create new streams.
for (uint32_t ssrc = 1; ssrc < (1 + kMaxUnsignaledRecvStreams); ++ssrc) {
webrtc::SetBE32(&packet[8], ssrc);
- DeliverPacket(packet, sizeof(packet));
+ DeliverPacket(packet);
EXPECT_EQ(kMaxUnsignaledRecvStreams, call_.GetAudioReceiveStreams().size());
EXPECT_EQ(2, GetRecvStream(ssrc).received_packets());
- EXPECT_TRUE(GetRecvStream(ssrc).VerifyLastPacket(packet, sizeof(packet)));
+ EXPECT_TRUE(GetRecvStream(ssrc).VerifyLastPacket(packet));
}
// Send on another SSRC, the oldest unsignaled stream (SSRC=1) is replaced.
constexpr uint32_t kAnotherSsrc = 667;
webrtc::SetBE32(&packet[8], kAnotherSsrc);
- DeliverPacket(packet, sizeof(packet));
+ DeliverPacket(packet);
const auto& streams = call_.GetAudioReceiveStreams();
EXPECT_EQ(kMaxUnsignaledRecvStreams, streams.size());
@@ -2972,32 +2967,30 @@ TEST_P(WebRtcVoiceEngineTestFake, RecvMultipleUnsignaled) {
// added, and that this stream will get any packets for unknown SSRCs.
TEST_P(WebRtcVoiceEngineTestFake, RecvUnsignaledAfterSignaled) {
EXPECT_TRUE(SetupChannel());
- unsigned char packet[sizeof(kPcmuFrame)];
+ uint8_t packet[sizeof(kPcmuFrame)];
memcpy(packet, kPcmuFrame, sizeof(kPcmuFrame));
// Add a known stream, send packet and verify we got it.
const uint32_t signaled_ssrc = 1;
webrtc::SetBE32(&packet[8], signaled_ssrc);
EXPECT_TRUE(AddRecvStream(signaled_ssrc));
- DeliverPacket(packet, sizeof(packet));
- EXPECT_TRUE(
- GetRecvStream(signaled_ssrc).VerifyLastPacket(packet, sizeof(packet)));
+ DeliverPacket(packet);
+ EXPECT_TRUE(GetRecvStream(signaled_ssrc).VerifyLastPacket(packet));
EXPECT_EQ(1u, call_.GetAudioReceiveStreams().size());
// Note that the first unknown SSRC cannot be 0, because we only support
// creating receive streams for SSRC!=0.
const uint32_t unsignaled_ssrc = 7011;
webrtc::SetBE32(&packet[8], unsignaled_ssrc);
- DeliverPacket(packet, sizeof(packet));
- EXPECT_TRUE(
- GetRecvStream(unsignaled_ssrc).VerifyLastPacket(packet, sizeof(packet)));
+ DeliverPacket(packet);
+ EXPECT_TRUE(GetRecvStream(unsignaled_ssrc).VerifyLastPacket(packet));
EXPECT_EQ(2u, call_.GetAudioReceiveStreams().size());
- DeliverPacket(packet, sizeof(packet));
+ DeliverPacket(packet);
EXPECT_EQ(2, GetRecvStream(unsignaled_ssrc).received_packets());
webrtc::SetBE32(&packet[8], signaled_ssrc);
- DeliverPacket(packet, sizeof(packet));
+ DeliverPacket(packet);
EXPECT_EQ(2, GetRecvStream(signaled_ssrc).received_packets());
EXPECT_EQ(2u, call_.GetAudioReceiveStreams().size());
}
@@ -3009,10 +3002,9 @@ TEST_P(WebRtcVoiceEngineTestFake, AddRecvStreamAfterUnsignaled_NoRecreate) {
EXPECT_TRUE(SetupChannel());
// Spawn unsignaled stream with SSRC=1.
- DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
+ DeliverPacket(kPcmuFrame);
EXPECT_EQ(1u, call_.GetAudioReceiveStreams().size());
- EXPECT_TRUE(
- GetRecvStream(1).VerifyLastPacket(kPcmuFrame, sizeof(kPcmuFrame)));
+ EXPECT_TRUE(GetRecvStream(1).VerifyLastPacket(kPcmuFrame));
// Verify that the underlying stream object in Call is not recreated when a
// stream with SSRC=1 is added.
@@ -3028,10 +3020,9 @@ TEST_P(WebRtcVoiceEngineTestFake, AddRecvStreamAfterUnsignaled_Updates) {
EXPECT_TRUE(SetupChannel());
// Spawn unsignaled stream with SSRC=1.
- DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
+ DeliverPacket(kPcmuFrame);
EXPECT_EQ(1u, call_.GetAudioReceiveStreams().size());
- EXPECT_TRUE(
- GetRecvStream(1).VerifyLastPacket(kPcmuFrame, sizeof(kPcmuFrame)));
+ EXPECT_TRUE(GetRecvStream(1).VerifyLastPacket(kPcmuFrame));
// Verify that the underlying stream object in Call gets updated when a
// stream with SSRC=1 is added, and which has changed stream parameters.
@@ -3461,7 +3452,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetOutputVolumeUnsignaledRecvStream) {
EXPECT_TRUE(SetupChannel());
// Spawn an unsignaled stream by sending a packet - gain should be 1.
- DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
+ DeliverPacket(kPcmuFrame);
EXPECT_DOUBLE_EQ(1, GetRecvStream(kSsrc1).gain());
// Should remember the volume "2" which will be set on new unsignaled streams,
@@ -3470,10 +3461,10 @@ TEST_P(WebRtcVoiceEngineTestFake, SetOutputVolumeUnsignaledRecvStream) {
EXPECT_DOUBLE_EQ(2, GetRecvStream(kSsrc1).gain());
// Spawn an unsignaled stream by sending a packet - gain should be 2.
- unsigned char pcmuFrame2[sizeof(kPcmuFrame)];
+ uint8_t pcmuFrame2[sizeof(kPcmuFrame)];
memcpy(pcmuFrame2, kPcmuFrame, sizeof(kPcmuFrame));
webrtc::SetBE32(&pcmuFrame2[8], kSsrcX);
- DeliverPacket(pcmuFrame2, sizeof(pcmuFrame2));
+ DeliverPacket(pcmuFrame2);
EXPECT_DOUBLE_EQ(2, GetRecvStream(kSsrcX).gain());
// Setting gain for all unsignaled streams.
@@ -3511,7 +3502,7 @@ TEST_P(WebRtcVoiceEngineTestFake,
EXPECT_TRUE(SetupChannel());
// Spawn an unsignaled stream by sending a packet - delay should be 0.
- DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
+ DeliverPacket(kPcmuFrame);
EXPECT_EQ(
0, receive_channel_->GetBaseMinimumPlayoutDelayMs(kSsrc1).value_or(-1));
// Check that it doesn't provide default values for unknown ssrc.
@@ -3532,10 +3523,10 @@ TEST_P(WebRtcVoiceEngineTestFake,
receive_channel_->GetBaseMinimumPlayoutDelayMs(kSsrcY).has_value());
// Spawn an unsignaled stream by sending a packet - delay should be 100.
- unsigned char pcmuFrame2[sizeof(kPcmuFrame)];
+ uint8_t pcmuFrame2[sizeof(kPcmuFrame)];
memcpy(pcmuFrame2, kPcmuFrame, sizeof(kPcmuFrame));
webrtc::SetBE32(&pcmuFrame2[8], kSsrcX);
- DeliverPacket(pcmuFrame2, sizeof(pcmuFrame2));
+ DeliverPacket(pcmuFrame2);
EXPECT_EQ(
100, receive_channel_->GetBaseMinimumPlayoutDelayMs(kSsrcX).value_or(-1));
@@ -3636,7 +3627,7 @@ TEST_P(WebRtcVoiceEngineTestFake, DeliverAudioPacket_Call) {
// Test that packets are forwarded to the Call when configured accordingly.
const uint32_t kAudioSsrc = 1;
webrtc::CopyOnWriteBuffer kPcmuPacket(kPcmuFrame, sizeof(kPcmuFrame));
- static const unsigned char kRtcp[] = {
+ static const uint8_t kRtcp[] = {
0x80, 0xc9, 0x00, 0x01, 0x00, 0x00, 0x00, 0x02, 0x00, 0x00, 0x00,
0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00};
@@ -3719,7 +3710,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetRawAudioSinkUnsignaledRecvStream) {
// Spawn an unsignaled stream by sending a packet - it should be assigned the
// default sink.
- DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
+ DeliverPacket(kPcmuFrame);
EXPECT_NE(nullptr, GetRecvStream(kSsrc1).sink());
// Try resetting the default sink.
@@ -3732,15 +3723,15 @@ TEST_P(WebRtcVoiceEngineTestFake, SetRawAudioSinkUnsignaledRecvStream) {
// If we remove and add a default stream, it should get the same sink.
EXPECT_TRUE(receive_channel_->RemoveRecvStream(kSsrc1));
- DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
+ DeliverPacket(kPcmuFrame);
EXPECT_NE(nullptr, GetRecvStream(kSsrc1).sink());
// Spawn another unsignaled stream - it should be assigned the default sink
// and the previous unsignaled stream should lose it.
- unsigned char pcmuFrame2[sizeof(kPcmuFrame)];
+ uint8_t pcmuFrame2[sizeof(kPcmuFrame)];
memcpy(pcmuFrame2, kPcmuFrame, sizeof(kPcmuFrame));
webrtc::SetBE32(&pcmuFrame2[8], kSsrcX);
- DeliverPacket(pcmuFrame2, sizeof(pcmuFrame2));
+ DeliverPacket(pcmuFrame2);
if (kMaxUnsignaledRecvStreams > 1) {
EXPECT_EQ(nullptr, GetRecvStream(kSsrc1).sink());
}
@@ -4214,3 +4205,6 @@ TEST(WebRtcVoiceEngineTest, CollectRecvCodecsWithLatePtAssignment) {
EXPECT_GE(find_codec({"telephone-event", 48000, 1}), num_specs);
}
}
+
+} // namespace
+} // namespace webrtc
