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Set local ssrc at construction of Rtp module

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The SetSSRC() method is slated for removal, make sure we set the local
SSRC at construction time.

Bug: webrtc:10774
Change-Id: I431e828caf60c5e0134adbe82d1d3345745cc6ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149827
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28926}
This commit is contained in:
Erik Språng 2019-08-21 13:36:20 +02:00 committed by Commit Bot
parent 21e99dac24
commit 70efddeced
10 changed files with 23 additions and 48 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

12
audio/audio_receive_stream.cc
View File

@ -78,8 +78,9 @@ std::unique_ptr<voe::ChannelReceiveInterface> CreateChannelReceive(
return voe::CreateChannelReceive(
clock, module_process_thread, internal_audio_state->audio_device_module(),
config.media_transport_config, config.rtcp_send_transport, event_log,
config.rtp.remote_ssrc, config.jitter_buffer_max_packets,
config.jitter_buffer_fast_accelerate, config.jitter_buffer_min_delay_ms,
config.rtp.local_ssrc, config.rtp.remote_ssrc,
config.jitter_buffer_max_packets, config.jitter_buffer_fast_accelerate,
config.jitter_buffer_min_delay_ms,
config.jitter_buffer_enable_rtx_handling, config.decoder_factory,
config.codec_pair_id, config.frame_decryptor, config.crypto_options);
}
@ -381,12 +382,9 @@ void AudioReceiveStream::ConfigureStream(AudioReceiveStream* stream,
RTC_DCHECK(first_time ||
old_config.decoder_factory == new_config.decoder_factory);
if (first_time || old_config.rtp.local_ssrc != new_config.rtp.local_ssrc) {
channel_receive->SetLocalSSRC(new_config.rtp.local_ssrc);
}
if (!first_time) {
// Remote ssrc can't be changed mid-stream.
// SSRC can't be changed mid-stream.
RTC_DCHECK_EQ(old_config.rtp.local_ssrc, new_config.rtp.local_ssrc);
RTC_DCHECK_EQ(old_config.rtp.remote_ssrc, new_config.rtp.remote_ssrc);
}

3
audio/audio_receive_stream_unittest.cc
View File

@ -88,7 +88,6 @@ struct ConfigHelper {
audio_state_ = AudioState::Create(config);
channel_receive_ = new ::testing::StrictMock<MockChannelReceive>();
EXPECT_CALL(*channel_receive_, SetLocalSSRC(kLocalSsrc)).Times(1);
EXPECT_CALL(*channel_receive_, SetNACKStatus(true, 15)).Times(1);
EXPECT_CALL(*channel_receive_,
RegisterReceiverCongestionControlObjects(&packet_router_))
@ -365,7 +364,6 @@ TEST(AudioReceiveStreamTest, ReconfigureWithUpdatedConfig) {
auto recv_stream = helper.CreateAudioReceiveStream();
auto new_config = helper.config();
new_config.rtp.local_ssrc = kLocalSsrc + 1;
new_config.rtp.nack.rtp_history_ms = 300 + 20;
new_config.rtp.extensions.clear();
new_config.rtp.extensions.push_back(
@ -376,7 +374,6 @@ TEST(AudioReceiveStreamTest, ReconfigureWithUpdatedConfig) {
new_config.decoder_map.emplace(1, SdpAudioFormat("foo", 8000, 1));
MockChannelReceive& channel_receive = *helper.channel_receive();
EXPECT_CALL(channel_receive, SetLocalSSRC(kLocalSsrc + 1)).Times(1);
EXPECT_CALL(channel_receive, SetNACKStatus(true, 15 + 1)).Times(1);
EXPECT_CALL(channel_receive, SetReceiveCodecs(new_config.decoder_map));

8
audio/audio_send_stream.cc
View File

@ -243,12 +243,8 @@ void AudioSendStream::ConfigureStream(
// Configuration parameters which cannot be changed.
RTC_DCHECK(first_time ||
old_config.send_transport == new_config.send_transport);
if (old_config.rtp.ssrc != new_config.rtp.ssrc) {
channel_send->SetLocalSSRC(new_config.rtp.ssrc);
}
if (stream->suspended_rtp_state_ &&
(first_time || old_config.rtp.ssrc != new_config.rtp.ssrc)) {
RTC_DCHECK(first_time || old_config.rtp.ssrc == new_config.rtp.ssrc);
if (stream->suspended_rtp_state_ && first_time) {
stream->rtp_rtcp_module_->SetRtpState(*stream->suspended_rtp_state_);
}
if (first_time || old_config.rtp.c_name != new_config.rtp.c_name) {

2
audio/audio_send_stream_unittest.cc
View File

@ -208,7 +208,7 @@ struct ConfigHelper {
EXPECT_CALL(*channel_send_, GetRtpRtcp()).WillRepeatedly(Invoke([this]() {
return &this->rtp_rtcp_;
}));
EXPECT_CALL(*channel_send_, SetLocalSSRC(kSsrc)).Times(1);
EXPECT_CALL(rtp_rtcp_, SSRC).WillRepeatedly(Return(kSsrc));
EXPECT_CALL(*channel_send_, SetRTCP_CNAME(StrEq(kCName))).Times(1);
EXPECT_CALL(*channel_send_, SetFrameEncryptor(_)).Times(1);
EXPECT_CALL(*channel_send_, SetExtmapAllowMixed(false)).Times(1);

15
audio/channel_receive.cc
View File

@ -102,6 +102,7 @@ class ChannelReceive : public ChannelReceiveInterface,
const MediaTransportConfig& media_transport_config,
Transport* rtcp_send_transport,
RtcEventLog* rtc_event_log,
uint32_t local_ssrc,
uint32_t remote_ssrc,
size_t jitter_buffer_max_packets,
bool jitter_buffer_fast_playout,
@ -155,9 +156,6 @@ class ChannelReceive : public ChannelReceiveInterface,
// Produces the transport-related timestamps; current_delay_ms is left unset.
absl::optional<Syncable::Info> GetSyncInfo() const override;
// RTP+RTCP
void SetLocalSSRC(unsigned int ssrc) override;
void RegisterReceiverCongestionControlObjects(
PacketRouter* packet_router) override;
void ResetReceiverCongestionControlObjects() override;
@ -456,6 +454,7 @@ ChannelReceive::ChannelReceive(
const MediaTransportConfig& media_transport_config,
Transport* rtcp_send_transport,
RtcEventLog* rtc_event_log,
uint32_t local_ssrc,
uint32_t remote_ssrc,
size_t jitter_buffer_max_packets,
bool jitter_buffer_fast_playout,
@ -508,8 +507,8 @@ ChannelReceive::ChannelReceive(
configuration.receiver_only = true;
configuration.outgoing_transport = rtcp_send_transport;
configuration.receive_statistics = rtp_receive_statistics_.get();
configuration.event_log = event_log_;
configuration.local_media_ssrc = local_ssrc;
_rtpRtcpModule = RtpRtcp::Create(configuration);
_rtpRtcpModule->SetSendingMediaStatus(false);
@ -701,11 +700,6 @@ void ChannelReceive::SetChannelOutputVolumeScaling(float scaling) {
_outputGain = scaling;
}
void ChannelReceive::SetLocalSSRC(uint32_t ssrc) {
RTC_DCHECK(worker_thread_checker_.IsCurrent());
_rtpRtcpModule->SetSSRC(ssrc);
}
void ChannelReceive::RegisterReceiverCongestionControlObjects(
PacketRouter* packet_router) {
RTC_DCHECK(worker_thread_checker_.IsCurrent());
@ -959,6 +953,7 @@ std::unique_ptr<ChannelReceiveInterface> CreateChannelReceive(
const MediaTransportConfig& media_transport_config,
Transport* rtcp_send_transport,
RtcEventLog* rtc_event_log,
uint32_t local_ssrc,
uint32_t remote_ssrc,
size_t jitter_buffer_max_packets,
bool jitter_buffer_fast_playout,
@ -970,7 +965,7 @@ std::unique_ptr<ChannelReceiveInterface> CreateChannelReceive(
const webrtc::CryptoOptions& crypto_options) {
return absl::make_unique<ChannelReceive>(
clock, module_process_thread, audio_device_module, media_transport_config,
rtcp_send_transport, rtc_event_log, remote_ssrc,
rtcp_send_transport, rtc_event_log, local_ssrc, remote_ssrc,
jitter_buffer_max_packets, jitter_buffer_fast_playout,
jitter_buffer_min_delay_ms, jitter_buffer_enable_rtx_handling,
decoder_factory, codec_pair_id, frame_decryptor, crypto_options);

4
audio/channel_receive.h
View File

@ -116,9 +116,6 @@ class ChannelReceiveInterface : public RtpPacketSinkInterface {
// Produces the transport-related timestamps; current_delay_ms is left unset.
virtual absl::optional<Syncable::Info> GetSyncInfo() const = 0;
// RTP+RTCP
virtual void SetLocalSSRC(uint32_t ssrc) = 0;
virtual void RegisterReceiverCongestionControlObjects(
PacketRouter* packet_router) = 0;
virtual void ResetReceiverCongestionControlObjects() = 0;
@ -145,6 +142,7 @@ std::unique_ptr<ChannelReceiveInterface> CreateChannelReceive(
const MediaTransportConfig& media_transport_config,
Transport* rtcp_send_transport,
RtcEventLog* rtc_event_log,
uint32_t local_ssrc,
uint32_t remote_ssrc,
size_t jitter_buffer_max_packets,
bool jitter_buffer_fast_playout,

18
audio/channel_send.cc
View File

@ -142,7 +142,6 @@ class ChannelSend : public ChannelSendInterface,
int payload_frequency) override;
// RTP+RTCP
void SetLocalSSRC(uint32_t ssrc) override;
void SetRid(const std::string& rid,
int extension_id,
int repaired_extension_id) override;
@ -279,7 +278,7 @@ class ChannelSend : public ChannelSendInterface,
int media_transport_sequence_number_ RTC_GUARDED_BY(encoder_queue_) = 0;
rtc::CriticalSection media_transport_lock_;
// Currently set by SetLocalSSRC.
// Currently set to local SSRC at construction.
uint64_t media_transport_channel_id_ RTC_GUARDED_BY(&media_transport_lock_) =
0;
// Cache payload type and sampling frequency from most recent call to
@ -702,6 +701,10 @@ ChannelSend::ChannelSend(Clock* clock,
configuration.rtcp_report_interval_ms = rtcp_report_interval_ms;
configuration.local_media_ssrc = ssrc;
if (media_transport_config_.media_transport) {
rtc::CritScope cs(&media_transport_lock_);
media_transport_channel_id_ = ssrc;
}
_rtpRtcpModule = RtpRtcp::Create(configuration);
_rtpRtcpModule->SetSendingMediaStatus(false);
@ -951,17 +954,6 @@ void ChannelSend::SetSendTelephoneEventPayloadType(int payload_type,
payload_frequency, 0, 0);
}
void ChannelSend::SetLocalSSRC(uint32_t ssrc) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
RTC_DCHECK(!sending_);
if (media_transport_config_.media_transport) {
rtc::CritScope cs(&media_transport_lock_);
media_transport_channel_id_ = ssrc;
}
_rtpRtcpModule->SetSSRC(ssrc);
}
void ChannelSend::SetRid(const std::string& rid,
int extension_id,
int repaired_extension_id) {

1
audio/channel_send.h
View File

@ -77,7 +77,6 @@ class ChannelSendInterface {
rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier) = 0;
virtual void CallEncoder(rtc::FunctionView<void(AudioEncoder*)> modifier) = 0;
virtual void SetLocalSSRC(uint32_t ssrc) = 0;
// Use 0 to indicate that the extension should not be registered.
virtual void SetRid(const std::string& rid,
int extension_id,

2
audio/mock_voe_channel_proxy.h
View File

@ -28,7 +28,6 @@ namespace test {
class MockChannelReceive : public voe::ChannelReceiveInterface {
public:
MOCK_METHOD1(SetLocalSSRC, void(uint32_t ssrc));
MOCK_METHOD2(SetNACKStatus, void(bool enable, int max_packets));
MOCK_METHOD1(RegisterReceiverCongestionControlObjects,
void(PacketRouter* packet_router));
@ -83,7 +82,6 @@ class MockChannelSend : public voe::ChannelSendInterface {
int extension_id,
int repaired_extension_id));
MOCK_METHOD2(SetMid, void(const std::string& mid, int extension_id));
MOCK_METHOD1(SetLocalSSRC, void(uint32_t ssrc));
MOCK_METHOD1(SetRTCP_CNAME, void(absl::string_view c_name));
MOCK_METHOD1(SetExtmapAllowMixed, void(bool extmap_allow_mixed));
MOCK_METHOD2(SetSendAudioLevelIndicationStatus, void(bool enable, int id));

6
media/engine/webrtc_voice_engine.cc
View File

@ -1065,8 +1065,10 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
void SetLocalSsrc(uint32_t local_ssrc) {
RTC_DCHECK(worker_thread_checker_.IsCurrent());
config_.rtp.local_ssrc = local_ssrc;
ReconfigureAudioReceiveStream();
if (local_ssrc != config_.rtp.local_ssrc) {
config_.rtp.local_ssrc = local_ssrc;
RecreateAudioReceiveStream();
}
}
void SetUseTransportCcAndRecreateStream(bool use_transport_cc,