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Delete obsolete method AudioReceiveStream::OnRtpPacket

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Bug: webrtc:10198
Change-Id: Ib7746cd9550a35cb64e6c91ce87ea42892592ff7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182842
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32051}
This commit is contained in:
Niels Möller 2020-08-28 16:04:18 +02:00 committed by Commit Bot
parent ec622d051b
commit 9e9c8b7155
3 changed files with 0 additions and 76 deletions
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8
audio/audio_receive_stream.cc
View File

@ -349,14 +349,6 @@ void AudioReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) {
channel_receive_->ReceivedRTCPPacket(packet, length);
}
void AudioReceiveStream::OnRtpPacket(const RtpPacketReceived& packet) {
// TODO(solenberg): Tests call this function on a network thread, libjingle
// calls on the worker thread. We should move towards always using a network
// thread. Then this check can be enabled.
// RTC_DCHECK(!thread_checker_.IsCurrent());
channel_receive_->OnRtpPacket(packet);
}
const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
return config_;

6
audio/audio_receive_stream.h
View File

@ -74,12 +74,6 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream,
int GetBaseMinimumPlayoutDelayMs() const override;
std::vector<webrtc::RtpSource> GetSources() const override;
// TODO(nisse): We don't formally implement RtpPacketSinkInterface, and this
// method shouldn't be needed. But it's currently used by the
// AudioReceiveStreamTest.ReceiveRtpPacket unittest. Figure out if that test
// shuld be refactored or deleted, and then delete this method.
void OnRtpPacket(const RtpPacketReceived& packet);
// AudioMixer::Source
AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz,
AudioFrame* audio_frame) override;

62
audio/audio_receive_stream_unittest.cc
View File

@ -53,8 +53,6 @@ AudioDecodingCallStats MakeAudioDecodeStatsForTest() {
const uint32_t kRemoteSsrc = 1234;
const uint32_t kLocalSsrc = 5678;
const size_t kOneByteExtensionHeaderLength = 4;
const size_t kOneByteExtensionLength = 4;
const int kAudioLevelId = 3;
const int kTransportSequenceNumberId = 4;
const int kJitterBufferDelay = -7;
@ -169,45 +167,6 @@ struct ConfigHelper {
MockTransport rtcp_send_transport_;
};
void BuildOneByteExtension(std::vector<uint8_t>::iterator it,
int id,
uint32_t extension_value,
size_t value_length) {
const uint16_t kRtpOneByteHeaderExtensionId = 0xBEDE;
ByteWriter<uint16_t>::WriteBigEndian(&(*it), kRtpOneByteHeaderExtensionId);
it += 2;
ByteWriter<uint16_t>::WriteBigEndian(&(*it), kOneByteExtensionLength / 4);
it += 2;
const size_t kExtensionDataLength = kOneByteExtensionLength - 1;
uint32_t shifted_value = extension_value
<< (8 * (kExtensionDataLength - value_length));
*it = (id << 4) + (static_cast<uint8_t>(value_length) - 1);
++it;
ByteWriter<uint32_t, kExtensionDataLength>::WriteBigEndian(&(*it),
shifted_value);
}
const std::vector<uint8_t> CreateRtpHeaderWithOneByteExtension(
int extension_id,
uint32_t extension_value,
size_t value_length) {
std::vector<uint8_t> header;
header.resize(webrtc::kRtpHeaderSize + kOneByteExtensionHeaderLength +
kOneByteExtensionLength);
header[0] = 0x80; // Version 2.
header[0] |= 0x10; // Set extension bit.
header[1] = 100; // Payload type.
header[1] |= 0x80; // Marker bit is set.
ByteWriter<uint16_t>::WriteBigEndian(&header[2], 0x1234); // Sequence number.
ByteWriter<uint32_t>::WriteBigEndian(&header[4], 0x5678); // Timestamp.
ByteWriter<uint32_t>::WriteBigEndian(&header[8], 0x4321); // SSRC.
BuildOneByteExtension(header.begin() + webrtc::kRtpHeaderSize, extension_id,
extension_value, value_length);
return header;
}
const std::vector<uint8_t> CreateRtcpSenderReport() {
std::vector<uint8_t> packet;
const size_t kRtcpSrLength = 28; // In bytes.
@ -242,27 +201,6 @@ TEST(AudioReceiveStreamTest, ConstructDestruct) {
}
}
TEST(AudioReceiveStreamTest, ReceiveRtpPacket) {
for (bool use_null_audio_processing : {false, true}) {
ConfigHelper helper(use_null_audio_processing);
helper.config().rtp.transport_cc = true;
auto recv_stream = helper.CreateAudioReceiveStream();
const int kTransportSequenceNumberValue = 1234;
std::vector<uint8_t> rtp_packet = CreateRtpHeaderWithOneByteExtension(
kTransportSequenceNumberId, kTransportSequenceNumberValue, 2);
constexpr int64_t packet_time_us = 5678000;
RtpPacketReceived parsed_packet;
ASSERT_TRUE(parsed_packet.Parse(&rtp_packet[0], rtp_packet.size()));
parsed_packet.set_arrival_time_ms((packet_time_us + 500) / 1000);
EXPECT_CALL(*helper.channel_receive(),
OnRtpPacket(::testing::Ref(parsed_packet)));
recv_stream->OnRtpPacket(parsed_packet);
}
}
TEST(AudioReceiveStreamTest, ReceiveRtcpPacket) {
for (bool use_null_audio_processing : {false, true}) {
ConfigHelper helper(use_null_audio_processing);