Remove VoEVideoSync interface.
The removed tests are covered by cases in call_perf_tests.
BUG=webrtc:4690
Review-Url: https://codereview.webrtc.org/2672583002
Cr-Commit-Position: refs/heads/master@{#16621}
diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc
index 05d6edf..2fe199f 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -260,11 +260,7 @@
return rtc::Optional<Syncable::Info>();
}
- int jitter_buffer_delay_ms = 0;
- int playout_buffer_delay_ms = 0;
- channel_proxy_->GetDelayEstimate(&jitter_buffer_delay_ms,
- &playout_buffer_delay_ms);
- info.current_delay_ms = jitter_buffer_delay_ms + playout_buffer_delay_ms;
+ info.current_delay_ms = channel_proxy_->GetDelayEstimate();
return rtc::Optional<Syncable::Info>(info);
}
diff --git a/webrtc/modules/audio_coding/include/audio_coding_module.h b/webrtc/modules/audio_coding/include/audio_coding_module.h
index cd99579..7843fb8 100644
--- a/webrtc/modules/audio_coding/include/audio_coding_module.h
+++ b/webrtc/modules/audio_coding/include/audio_coding_module.h
@@ -637,7 +637,8 @@
//
virtual int SetMaximumPlayoutDelay(int time_ms) = 0;
- //
+ // TODO(kwiberg): Consider if this is needed anymore, now that voe::Channel
+ // doesn't use it.
// The shortest latency, in milliseconds, required by jitter buffer. This
// is computed based on inter-arrival times and playout mode of NetEq. The
// actual delay is the maximum of least-required-delay and the minimum-delay
diff --git a/webrtc/test/mock_voe_channel_proxy.h b/webrtc/test/mock_voe_channel_proxy.h
index ea51154..7932473 100644
--- a/webrtc/test/mock_voe_channel_proxy.h
+++ b/webrtc/test/mock_voe_channel_proxy.h
@@ -76,8 +76,6 @@
MOCK_METHOD0(DisassociateSendChannel, void());
MOCK_CONST_METHOD2(GetRtpRtcp, void(RtpRtcp** rtp_rtcp,
RtpReceiver** rtp_receiver));
- MOCK_CONST_METHOD2(GetDelayEstimate, void(int* jitter_buffer_delay_ms,
- int* playout_buffer_delay_ms));
MOCK_CONST_METHOD0(GetPlayoutTimestamp, uint32_t());
MOCK_METHOD1(SetMinimumPlayoutDelay, void(int delay_ms));
MOCK_CONST_METHOD1(GetRecCodec, bool(CodecInst* codec_inst));
diff --git a/webrtc/test/mock_voice_engine.h b/webrtc/test/mock_voice_engine.h
index cd0b7d2..6177304 100644
--- a/webrtc/test/mock_voice_engine.h
+++ b/webrtc/test/mock_voice_engine.h
@@ -289,22 +289,6 @@
int(int channel, bool& enable, int& redPayloadtype));
MOCK_METHOD3(SetNACKStatus, int(int channel, bool enable, int maxNoPackets));
- // VoEVideoSync
- MOCK_METHOD1(GetPlayoutBufferSize, int(int& buffer_ms));
- MOCK_METHOD2(SetMinimumPlayoutDelay, int(int channel, int delay_ms));
- MOCK_METHOD3(GetDelayEstimate,
- int(int channel,
- int* jitter_buffer_delay_ms,
- int* playout_buffer_delay_ms));
- MOCK_CONST_METHOD1(GetLeastRequiredDelayMs, int(int channel));
- MOCK_METHOD2(SetInitTimestamp, int(int channel, unsigned int timestamp));
- MOCK_METHOD2(SetInitSequenceNumber, int(int channel, short sequenceNumber));
- MOCK_METHOD2(GetPlayoutTimestamp, int(int channel, unsigned int& timestamp));
- MOCK_METHOD3(GetRtpRtcp,
- int(int channel,
- RtpRtcp** rtpRtcpModule,
- RtpReceiver** rtp_receiver));
-
// VoEVolumeControl
MOCK_METHOD1(SetSpeakerVolume, int(unsigned int volume));
MOCK_METHOD1(GetSpeakerVolume, int(unsigned int& volume));
diff --git a/webrtc/voice_engine/BUILD.gn b/webrtc/voice_engine/BUILD.gn
index 028e4d1..e2755f3 100644
--- a/webrtc/voice_engine/BUILD.gn
+++ b/webrtc/voice_engine/BUILD.gn
@@ -83,7 +83,6 @@
"include/voe_neteq_stats.h",
"include/voe_network.h",
"include/voe_rtp_rtcp.h",
- "include/voe_video_sync.h",
"include/voe_volume_control.h",
"monitor_module.cc",
"monitor_module.h",
@@ -115,8 +114,6 @@
"voe_network_impl.h",
"voe_rtp_rtcp_impl.cc",
"voe_rtp_rtcp_impl.h",
- "voe_video_sync_impl.cc",
- "voe_video_sync_impl.h",
"voe_volume_control_impl.cc",
"voe_volume_control_impl.h",
"voice_engine_defines.h",
@@ -376,7 +373,6 @@
"test/auto_test/standard/rtp_rtcp_before_streaming_test.cc",
"test/auto_test/standard/rtp_rtcp_extensions.cc",
"test/auto_test/standard/rtp_rtcp_test.cc",
- "test/auto_test/standard/video_sync_test.cc",
"test/auto_test/standard/voe_base_misc_test.cc",
"test/auto_test/standard/volume_test.cc",
"test/auto_test/voe_conference_test.cc",
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index d69654f..006e3db 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -1150,16 +1150,16 @@
int32_t Channel::StartSend() {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::StartSend()");
- // Resume the previous sequence number which was reset by StopSend().
- // This needs to be done before |sending| is set to true.
- if (send_sequence_number_)
- SetInitSequenceNumber(send_sequence_number_);
-
if (channel_state_.Get().sending) {
return 0;
}
channel_state_.SetSending(true);
+ // Resume the previous sequence number which was reset by StopSend(). This
+ // needs to be done before |sending| is set to true on the RTP/RTCP module.
+ if (send_sequence_number_) {
+ _rtpRtcpModule->SetSequenceNumber(send_sequence_number_);
+ }
_rtpRtcpModule->SetSendingMediaStatus(true);
if (_rtpRtcpModule->SetSendingStatus(true) != 0) {
_engineStatisticsPtr->SetLastError(
@@ -2740,23 +2740,9 @@
audio_coding_->GetDecodingCallStatistics(stats);
}
-bool Channel::GetDelayEstimate(int* jitter_buffer_delay_ms,
- int* playout_buffer_delay_ms) const {
- rtc::CritScope lock(&video_sync_lock_);
- *jitter_buffer_delay_ms = audio_coding_->FilteredCurrentDelayMs();
- *playout_buffer_delay_ms = playout_delay_ms_;
- return true;
-}
-
uint32_t Channel::GetDelayEstimate() const {
- int jitter_buffer_delay_ms = 0;
- int playout_buffer_delay_ms = 0;
- GetDelayEstimate(&jitter_buffer_delay_ms, &playout_buffer_delay_ms);
- return jitter_buffer_delay_ms + playout_buffer_delay_ms;
-}
-
-int Channel::LeastRequiredDelayMs() const {
- return audio_coding_->LeastRequiredDelayMs();
+ rtc::CritScope lock(&video_sync_lock_);
+ return audio_coding_->FilteredCurrentDelayMs() + playout_delay_ms_;
}
int Channel::SetMinimumPlayoutDelay(int delayMs) {
@@ -2794,30 +2780,6 @@
return 0;
}
-int Channel::SetInitTimestamp(unsigned int timestamp) {
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::SetInitTimestamp()");
- if (channel_state_.Get().sending) {
- _engineStatisticsPtr->SetLastError(VE_SENDING, kTraceError,
- "SetInitTimestamp() already sending");
- return -1;
- }
- _rtpRtcpModule->SetStartTimestamp(timestamp);
- return 0;
-}
-
-int Channel::SetInitSequenceNumber(short sequenceNumber) {
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::SetInitSequenceNumber()");
- if (channel_state_.Get().sending) {
- _engineStatisticsPtr->SetLastError(
- VE_SENDING, kTraceError, "SetInitSequenceNumber() already sending");
- return -1;
- }
- _rtpRtcpModule->SetSequenceNumber(sequenceNumber);
- return 0;
-}
-
int Channel::GetRtpRtcp(RtpRtcp** rtpRtcpModule,
RtpReceiver** rtp_receiver) const {
*rtpRtcpModule = _rtpRtcpModule.get();
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
index d53cc06..03b19cc 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -258,17 +258,10 @@
int GetNetworkStatistics(NetworkStatistics& stats);
void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const;
- // VoEVideoSync
- bool GetDelayEstimate(int* jitter_buffer_delay_ms,
- int* playout_buffer_delay_ms) const;
+ // Audio+Video Sync
uint32_t GetDelayEstimate() const;
- int LeastRequiredDelayMs() const;
int SetMinimumPlayoutDelay(int delayMs);
int GetPlayoutTimestamp(unsigned int& timestamp);
- int SetInitTimestamp(unsigned int timestamp);
- int SetInitSequenceNumber(short sequenceNumber);
-
- // VoEVideoSyncExtended
int GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const;
// DTMF
diff --git a/webrtc/voice_engine/channel_proxy.cc b/webrtc/voice_engine/channel_proxy.cc
index a58b2e1..4e690fb 100644
--- a/webrtc/voice_engine/channel_proxy.cc
+++ b/webrtc/voice_engine/channel_proxy.cc
@@ -23,22 +23,24 @@
ChannelProxy::ChannelProxy(const ChannelOwner& channel_owner) :
channel_owner_(channel_owner) {
RTC_CHECK(channel_owner_.channel());
+ module_process_thread_checker_.DetachFromThread();
}
ChannelProxy::~ChannelProxy() {}
void ChannelProxy::SetRTCPStatus(bool enable) {
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
channel()->SetRTCPStatus(enable);
}
void ChannelProxy::SetLocalSSRC(uint32_t ssrc) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
int error = channel()->SetLocalSSRC(ssrc);
RTC_DCHECK_EQ(0, error);
}
void ChannelProxy::SetRTCP_CNAME(const std::string& c_name) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
// Note: VoERTP_RTCP::SetRTCP_CNAME() accepts a char[256] array.
std::string c_name_limited = c_name.substr(0, 255);
int error = channel()->SetRTCP_CNAME(c_name_limited.c_str());
@@ -46,29 +48,29 @@
}
void ChannelProxy::SetNACKStatus(bool enable, int max_packets) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
channel()->SetNACKStatus(enable, max_packets);
}
void ChannelProxy::SetSendAudioLevelIndicationStatus(bool enable, int id) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
int error = channel()->SetSendAudioLevelIndicationStatus(enable, id);
RTC_DCHECK_EQ(0, error);
}
void ChannelProxy::SetReceiveAudioLevelIndicationStatus(bool enable, int id) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
int error = channel()->SetReceiveAudioLevelIndicationStatus(enable, id);
RTC_DCHECK_EQ(0, error);
}
void ChannelProxy::EnableSendTransportSequenceNumber(int id) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
channel()->EnableSendTransportSequenceNumber(id);
}
void ChannelProxy::EnableReceiveTransportSequenceNumber(int id) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
channel()->EnableReceiveTransportSequenceNumber(id);
}
@@ -77,7 +79,7 @@
TransportFeedbackObserver* transport_feedback_observer,
PacketRouter* packet_router,
RtcpBandwidthObserver* bandwidth_observer) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
channel()->RegisterSenderCongestionControlObjects(
rtp_packet_sender, transport_feedback_observer, packet_router,
bandwidth_observer);
@@ -85,17 +87,17 @@
void ChannelProxy::RegisterReceiverCongestionControlObjects(
PacketRouter* packet_router) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
channel()->RegisterReceiverCongestionControlObjects(packet_router);
}
void ChannelProxy::ResetCongestionControlObjects() {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
channel()->ResetCongestionControlObjects();
}
CallStatistics ChannelProxy::GetRTCPStatistics() const {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
CallStatistics stats = {0};
int error = channel()->GetRTPStatistics(stats);
RTC_DCHECK_EQ(0, error);
@@ -103,7 +105,7 @@
}
std::vector<ReportBlock> ChannelProxy::GetRemoteRTCPReportBlocks() const {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
std::vector<webrtc::ReportBlock> blocks;
int error = channel()->GetRemoteRTCPReportBlocks(&blocks);
RTC_DCHECK_EQ(0, error);
@@ -111,7 +113,7 @@
}
NetworkStatistics ChannelProxy::GetNetworkStatistics() const {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
NetworkStatistics stats = {0};
int error = channel()->GetNetworkStatistics(stats);
RTC_DCHECK_EQ(0, error);
@@ -119,14 +121,14 @@
}
AudioDecodingCallStats ChannelProxy::GetDecodingCallStatistics() const {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
AudioDecodingCallStats stats;
channel()->GetDecodingCallStatistics(&stats);
return stats;
}
int32_t ChannelProxy::GetSpeechOutputLevelFullRange() const {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
uint32_t level = 0;
int error = channel()->GetSpeechOutputLevelFullRange(level);
RTC_DCHECK_EQ(0, error);
@@ -134,53 +136,55 @@
}
uint32_t ChannelProxy::GetDelayEstimate() const {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread() ||
+ module_process_thread_checker_.CalledOnValidThread());
return channel()->GetDelayEstimate();
}
bool ChannelProxy::SetSendTelephoneEventPayloadType(int payload_type,
int payload_frequency) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
return channel()->SetSendTelephoneEventPayloadType(payload_type,
payload_frequency) == 0;
}
bool ChannelProxy::SendTelephoneEventOutband(int event, int duration_ms) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
return channel()->SendTelephoneEventOutband(event, duration_ms) == 0;
}
void ChannelProxy::SetBitrate(int bitrate_bps, int64_t probing_interval_ms) {
- // May be called on different threads and needs to be handled by the channel.
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread() ||
+ module_process_thread_checker_.CalledOnValidThread());
channel()->SetBitRate(bitrate_bps, probing_interval_ms);
}
void ChannelProxy::SetRecPayloadType(int payload_type,
const SdpAudioFormat& format) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
const int result = channel()->SetRecPayloadType(payload_type, format);
RTC_DCHECK_EQ(0, result);
}
void ChannelProxy::SetSink(std::unique_ptr<AudioSinkInterface> sink) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
channel()->SetSink(std::move(sink));
}
void ChannelProxy::SetInputMute(bool muted) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
int error = channel()->SetInputMute(muted);
RTC_DCHECK_EQ(0, error);
}
void ChannelProxy::RegisterExternalTransport(Transport* transport) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
int error = channel()->RegisterExternalTransport(transport);
RTC_DCHECK_EQ(0, error);
}
void ChannelProxy::DeRegisterExternalTransport() {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
channel()->DeRegisterExternalTransport();
}
@@ -198,35 +202,35 @@
const rtc::scoped_refptr<AudioDecoderFactory>&
ChannelProxy::GetAudioDecoderFactory() const {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
return channel()->GetAudioDecoderFactory();
}
void ChannelProxy::SetChannelOutputVolumeScaling(float scaling) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
int error = channel()->SetChannelOutputVolumeScaling(scaling);
RTC_DCHECK_EQ(0, error);
}
void ChannelProxy::SetRtcEventLog(RtcEventLog* event_log) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
channel()->SetRtcEventLog(event_log);
}
void ChannelProxy::EnableAudioNetworkAdaptor(const std::string& config_string) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
bool ret = channel()->EnableAudioNetworkAdaptor(config_string);
RTC_DCHECK(ret);
;}
void ChannelProxy::DisableAudioNetworkAdaptor() {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
channel()->DisableAudioNetworkAdaptor();
}
void ChannelProxy::SetReceiverFrameLengthRange(int min_frame_length_ms,
int max_frame_length_ms) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
channel()->SetReceiverFrameLengthRange(min_frame_length_ms,
max_frame_length_ms);
}
@@ -234,51 +238,42 @@
AudioMixer::Source::AudioFrameInfo ChannelProxy::GetAudioFrameWithInfo(
int sample_rate_hz,
AudioFrame* audio_frame) {
- RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
+ RTC_DCHECK_RUNS_SERIALIZED(&audio_thread_race_checker_);
return channel()->GetAudioFrameWithInfo(sample_rate_hz, audio_frame);
}
int ChannelProxy::NeededFrequency() const {
+ RTC_DCHECK_RUNS_SERIALIZED(&audio_thread_race_checker_);
return static_cast<int>(channel()->NeededFrequency(-1));
}
void ChannelProxy::SetTransportOverhead(int transport_overhead_per_packet) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
channel()->SetTransportOverhead(transport_overhead_per_packet);
}
void ChannelProxy::AssociateSendChannel(
const ChannelProxy& send_channel_proxy) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
channel()->set_associate_send_channel(send_channel_proxy.channel_owner_);
}
void ChannelProxy::DisassociateSendChannel() {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
channel()->set_associate_send_channel(ChannelOwner(nullptr));
}
void ChannelProxy::GetRtpRtcp(RtpRtcp** rtp_rtcp,
RtpReceiver** rtp_receiver) const {
- // Called on Call's module_process_thread_.
+ RTC_DCHECK(module_process_thread_checker_.CalledOnValidThread());
RTC_DCHECK(rtp_rtcp);
RTC_DCHECK(rtp_receiver);
int error = channel()->GetRtpRtcp(rtp_rtcp, rtp_receiver);
RTC_DCHECK_EQ(0, error);
}
-void ChannelProxy::GetDelayEstimate(int* jitter_buffer_delay_ms,
- int* playout_buffer_delay_ms) const {
- // Called on Call's module_process_thread_.
- RTC_DCHECK(jitter_buffer_delay_ms);
- RTC_DCHECK(playout_buffer_delay_ms);
- bool error = channel()->GetDelayEstimate(jitter_buffer_delay_ms,
- playout_buffer_delay_ms);
- RTC_DCHECK(error);
-}
-
uint32_t ChannelProxy::GetPlayoutTimestamp() const {
- // Called on video capture thread.
+ RTC_DCHECK_RUNS_SERIALIZED(&video_capture_thread_race_checker_);
unsigned int timestamp = 0;
int error = channel()->GetPlayoutTimestamp(timestamp);
RTC_DCHECK(!error || timestamp == 0);
@@ -286,7 +281,7 @@
}
void ChannelProxy::SetMinimumPlayoutDelay(int delay_ms) {
- // Called on Call's module_process_thread_.
+ RTC_DCHECK(module_process_thread_checker_.CalledOnValidThread());
// Limit to range accepted by both VoE and ACM, so we're at least getting as
// close as possible, instead of failing.
delay_ms = std::max(0, std::min(delay_ms, 10000));
@@ -295,42 +290,42 @@
}
void ChannelProxy::SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
channel()->SetRtcpRttStats(rtcp_rtt_stats);
}
bool ChannelProxy::GetRecCodec(CodecInst* codec_inst) const {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
return channel()->GetRecCodec(*codec_inst) == 0;
}
bool ChannelProxy::GetSendCodec(CodecInst* codec_inst) const {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
return channel()->GetSendCodec(*codec_inst) == 0;
}
bool ChannelProxy::SetVADStatus(bool enable) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
return channel()->SetVADStatus(enable, VADNormal, false) == 0;
}
bool ChannelProxy::SetCodecFECStatus(bool enable) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
return channel()->SetCodecFECStatus(enable) == 0;
}
bool ChannelProxy::SetOpusDtx(bool enable) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
return channel()->SetOpusDtx(enable) == 0;
}
bool ChannelProxy::SetOpusMaxPlaybackRate(int frequency_hz) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
return channel()->SetOpusMaxPlaybackRate(frequency_hz) == 0;
}
bool ChannelProxy::SetSendCodec(const CodecInst& codec_inst) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
// Validation code copied from VoECodecImpl::SetSendCodec().
if ((STR_CASE_CMP(codec_inst.plname, "L16") == 0) &&
(codec_inst.pacsize >= 960)) {
@@ -352,7 +347,7 @@
bool ChannelProxy::SetSendCNPayloadType(int type,
PayloadFrequencies frequency) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
// Validation code copied from VoECodecImpl::SetSendCNPayloadType().
if (type < 96 || type > 127) {
// Only allow dynamic range: 96 to 127
diff --git a/webrtc/voice_engine/channel_proxy.h b/webrtc/voice_engine/channel_proxy.h
index 966abc4..b70dcc4 100644
--- a/webrtc/voice_engine/channel_proxy.h
+++ b/webrtc/voice_engine/channel_proxy.h
@@ -105,8 +105,6 @@
virtual void DisassociateSendChannel();
virtual void GetRtpRtcp(RtpRtcp** rtp_rtcp,
RtpReceiver** rtp_receiver) const;
- virtual void GetDelayEstimate(int* jitter_buffer_delay_ms,
- int* playout_buffer_delay_ms) const;
virtual uint32_t GetPlayoutTimestamp() const;
virtual void SetMinimumPlayoutDelay(int delay_ms);
virtual void SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats);
@@ -122,8 +120,18 @@
private:
Channel* channel() const;
- rtc::ThreadChecker thread_checker_;
- rtc::RaceChecker race_checker_;
+ // Thread checkers document and lock usage of some methods on voe::Channel to
+ // specific threads we know about. The goal is to eventually split up
+ // voe::Channel into parts with single-threaded semantics, and thereby reduce
+ // the need for locks.
+ rtc::ThreadChecker worker_thread_checker_;
+ rtc::ThreadChecker module_process_thread_checker_;
+ // Methods accessed from audio and video threads are checked for sequential-
+ // only access. We don't necessarily own and control these threads, so thread
+ // checkers cannot be used. E.g. Chromium may transfer "ownership" from one
+ // audio thread to another, but access is still sequential.
+ rtc::RaceChecker audio_thread_race_checker_;
+ rtc::RaceChecker video_capture_thread_race_checker_;
ChannelOwner channel_owner_;
RTC_DISALLOW_COPY_AND_ASSIGN(ChannelProxy);
diff --git a/webrtc/voice_engine/include/voe_video_sync.h b/webrtc/voice_engine/include/voe_video_sync.h
deleted file mode 100644
index 655ba63..0000000
--- a/webrtc/voice_engine/include/voe_video_sync.h
+++ /dev/null
@@ -1,99 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// This sub-API supports the following functionalities:
-//
-// - RTP header modification (time stamp and sequence number fields).
-// - Playout delay tuning to synchronize the voice with video.
-// - Playout delay monitoring.
-//
-// Usage example, omitting error checking:
-//
-// using namespace webrtc;
-// VoiceEngine* voe = VoiceEngine::Create();
-// VoEBase* base = VoEBase::GetInterface(voe);
-// VoEVideoSync* vsync = VoEVideoSync::GetInterface(voe);
-// base->Init();
-// ...
-// int buffer_ms(0);
-// vsync->GetPlayoutBufferSize(buffer_ms);
-// ...
-// base->Terminate();
-// base->Release();
-// vsync->Release();
-// VoiceEngine::Delete(voe);
-//
-#ifndef WEBRTC_VOICE_ENGINE_VOE_VIDEO_SYNC_H
-#define WEBRTC_VOICE_ENGINE_VOE_VIDEO_SYNC_H
-
-#include "webrtc/common_types.h"
-
-namespace webrtc {
-
-class RtpReceiver;
-class RtpRtcp;
-class VoiceEngine;
-
-class WEBRTC_DLLEXPORT VoEVideoSync {
- public:
- // Factory for the VoEVideoSync sub-API. Increases an internal
- // reference counter if successful. Returns NULL if the API is not
- // supported or if construction fails.
- static VoEVideoSync* GetInterface(VoiceEngine* voiceEngine);
-
- // Releases the VoEVideoSync sub-API and decreases an internal
- // reference counter. Returns the new reference count. This value should
- // be zero for all sub-API:s before the VoiceEngine object can be safely
- // deleted.
- virtual int Release() = 0;
-
- // Gets the current sound card buffer size (playout delay).
- virtual int GetPlayoutBufferSize(int& buffer_ms) = 0;
-
- // Sets a minimum target delay for the jitter buffer. This delay is
- // maintained by the jitter buffer, unless channel condition (jitter in
- // inter-arrival times) dictates a higher required delay. The overall
- // jitter buffer delay is max of |delay_ms| and the latency that NetEq
- // computes based on inter-arrival times and its playout mode.
- virtual int SetMinimumPlayoutDelay(int channel, int delay_ms) = 0;
-
- // Gets the |jitter_buffer_delay_ms| (including the algorithmic delay), and
- // the |playout_buffer_delay_ms| for a specified |channel|.
- virtual int GetDelayEstimate(int channel,
- int* jitter_buffer_delay_ms,
- int* playout_buffer_delay_ms) = 0;
-
- // Returns the least required jitter buffer delay. This is computed by the
- // the jitter buffer based on the inter-arrival time of RTP packets and
- // playout mode. NetEq maintains this latency unless a higher value is
- // requested by calling SetMinimumPlayoutDelay().
- virtual int GetLeastRequiredDelayMs(int channel) const = 0;
-
- // Manual initialization of the RTP timestamp.
- virtual int SetInitTimestamp(int channel, unsigned int timestamp) = 0;
-
- // Manual initialization of the RTP sequence number.
- virtual int SetInitSequenceNumber(int channel, short sequenceNumber) = 0;
-
- // Get the received RTP timestamp
- virtual int GetPlayoutTimestamp(int channel, unsigned int& timestamp) = 0;
-
- virtual int GetRtpRtcp(int channel,
- RtpRtcp** rtpRtcpModule,
- RtpReceiver** rtp_receiver) = 0;
-
- protected:
- VoEVideoSync() {}
- virtual ~VoEVideoSync() {}
-};
-
-} // namespace webrtc
-
-#endif // #ifndef WEBRTC_VOICE_ENGINE_VOE_VIDEO_SYNC_H
diff --git a/webrtc/voice_engine/test/auto_test/fixtures/before_initialization_fixture.cc b/webrtc/voice_engine/test/auto_test/fixtures/before_initialization_fixture.cc
index 3c55830..79474d7 100644
--- a/webrtc/voice_engine/test/auto_test/fixtures/before_initialization_fixture.cc
+++ b/webrtc/voice_engine/test/auto_test/fixtures/before_initialization_fixture.cc
@@ -23,7 +23,6 @@
voe_apm_ = webrtc::VoEAudioProcessing::GetInterface(voice_engine_);
voe_network_ = webrtc::VoENetwork::GetInterface(voice_engine_);
voe_file_ = webrtc::VoEFile::GetInterface(voice_engine_);
- voe_vsync_ = webrtc::VoEVideoSync::GetInterface(voice_engine_);
voe_hardware_ = webrtc::VoEHardware::GetInterface(voice_engine_);
voe_neteq_stats_ = webrtc::VoENetEqStats::GetInterface(voice_engine_);
}
@@ -36,7 +35,6 @@
voe_apm_->Release();
voe_network_->Release();
voe_file_->Release();
- voe_vsync_->Release();
voe_hardware_->Release();
voe_neteq_stats_->Release();
diff --git a/webrtc/voice_engine/test/auto_test/fixtures/before_initialization_fixture.h b/webrtc/voice_engine/test/auto_test/fixtures/before_initialization_fixture.h
index 63bb6c0..e9f5a50 100644
--- a/webrtc/voice_engine/test/auto_test/fixtures/before_initialization_fixture.h
+++ b/webrtc/voice_engine/test/auto_test/fixtures/before_initialization_fixture.h
@@ -24,7 +24,6 @@
#include "webrtc/voice_engine/include/voe_neteq_stats.h"
#include "webrtc/voice_engine/include/voe_network.h"
#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
-#include "webrtc/voice_engine/include/voe_video_sync.h"
#include "webrtc/voice_engine/include/voe_volume_control.h"
#include "webrtc/voice_engine/test/auto_test/voe_test_common.h"
@@ -56,7 +55,6 @@
webrtc::VoEAudioProcessing* voe_apm_;
webrtc::VoENetwork* voe_network_;
webrtc::VoEFile* voe_file_;
- webrtc::VoEVideoSync* voe_vsync_;
webrtc::VoEHardware* voe_hardware_;
webrtc::VoENetEqStats* voe_neteq_stats_;
};
diff --git a/webrtc/voice_engine/test/auto_test/standard/video_sync_test.cc b/webrtc/voice_engine/test/auto_test/standard/video_sync_test.cc
deleted file mode 100644
index 0aacd53..0000000
--- a/webrtc/voice_engine/test/auto_test/standard/video_sync_test.cc
+++ /dev/null
@@ -1,129 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <math.h>
-
-#include <numeric>
-#include <vector>
-
-#include "webrtc/voice_engine/test/auto_test/fixtures/after_streaming_fixture.h"
-
-#ifdef WEBRTC_IOS
- const int kMinimumReasonableDelayEstimateMs = 30;
-#else
- const int kMinimumReasonableDelayEstimateMs = 45;
-#endif // !WEBRTC_IOS
-
-class VideoSyncTest : public AfterStreamingFixture {
- protected:
- // This test will verify that delay estimates converge (e.g. the standard
- // deviation for the last five seconds' estimates is less than 20) without
- // manual observation. The test runs for 15 seconds, sampling once per second.
- // All samples are checked so they are greater than |min_estimate|.
- int CollectEstimatesDuring15Seconds(int min_estimate) {
- Sleep(1000);
-
- std::vector<int> all_delay_estimates;
- for (int second = 0; second < 15; second++) {
- int jitter_buffer_delay_ms = 0;
- int playout_buffer_delay_ms = 0;
- EXPECT_EQ(0, voe_vsync_->GetDelayEstimate(channel_,
- &jitter_buffer_delay_ms,
- &playout_buffer_delay_ms));
-
- EXPECT_GT(jitter_buffer_delay_ms, min_estimate) <<
- "The delay estimate can not conceivably get lower than " <<
- min_estimate << " ms, it's unrealistic.";
-
- all_delay_estimates.push_back(jitter_buffer_delay_ms);
- Sleep(1000);
- }
-
- return ComputeStandardDeviation(
- all_delay_estimates.begin() + 10, all_delay_estimates.end());
- }
-
- void CheckEstimatesConvergeReasonablyWell(int min_estimate) {
- float standard_deviation = CollectEstimatesDuring15Seconds(min_estimate);
- EXPECT_LT(standard_deviation, 30.0f);
- }
-
- // Computes the standard deviation by first estimating the sample variance
- // with an unbiased estimator.
- float ComputeStandardDeviation(std::vector<int>::const_iterator start,
- std::vector<int>::const_iterator end) const {
- int num_elements = end - start;
- int mean = std::accumulate(start, end, 0) / num_elements;
- assert(num_elements > 1);
-
- float variance = 0;
- for (; start != end; ++start) {
- variance += (*start - mean) * (*start - mean) / (num_elements - 1);
- }
- return sqrt(variance);
- }
-};
-
-TEST_F(VideoSyncTest,
- CanNotGetPlayoutTimestampWhilePlayingWithoutSettingItFirst) {
- unsigned int ignored;
- EXPECT_EQ(-1, voe_vsync_->GetPlayoutTimestamp(channel_, ignored));
-}
-
-TEST_F(VideoSyncTest, CannotSetInitTimestampWhilePlaying) {
- EXPECT_EQ(-1, voe_vsync_->SetInitTimestamp(channel_, 12345));
-}
-
-TEST_F(VideoSyncTest, CannotSetInitSequenceNumberWhilePlaying) {
- EXPECT_EQ(-1, voe_vsync_->SetInitSequenceNumber(channel_, 123));
-}
-
-TEST_F(VideoSyncTest, CanSetInitTimestampWhileStopped) {
- EXPECT_EQ(0, voe_base_->StopSend(channel_));
- EXPECT_EQ(0, voe_vsync_->SetInitTimestamp(channel_, 12345));
-}
-
-TEST_F(VideoSyncTest, CanSetInitSequenceNumberWhileStopped) {
- EXPECT_EQ(0, voe_base_->StopSend(channel_));
- EXPECT_EQ(0, voe_vsync_->SetInitSequenceNumber(channel_, 123));
-}
-
-// TODO(phoglund): pending investigation in
-// http://code.google.com/p/webrtc/issues/detail?id=438
-TEST_F(VideoSyncTest,
- DISABLED_DelayEstimatesStabilizeDuring15sAndAreNotTooLow) {
- EXPECT_EQ(0, voe_base_->StopSend(channel_));
- EXPECT_EQ(0, voe_vsync_->SetInitTimestamp(channel_, 12345));
- EXPECT_EQ(0, voe_vsync_->SetInitSequenceNumber(channel_, 123));
- EXPECT_EQ(0, voe_base_->StartSend(channel_));
-
- CheckEstimatesConvergeReasonablyWell(kMinimumReasonableDelayEstimateMs);
-}
-
-// TODO(phoglund): pending investigation in
-// http://code.google.com/p/webrtc/issues/detail?id=438
-TEST_F(VideoSyncTest,
- DISABLED_DelayEstimatesStabilizeAfterNetEqMinDelayChanges45s) {
- EXPECT_EQ(0, voe_base_->StopSend(channel_));
- EXPECT_EQ(0, voe_vsync_->SetInitTimestamp(channel_, 12345));
- EXPECT_EQ(0, voe_vsync_->SetInitSequenceNumber(channel_, 123));
- EXPECT_EQ(0, voe_base_->StartSend(channel_));
-
- CheckEstimatesConvergeReasonablyWell(kMinimumReasonableDelayEstimateMs);
- EXPECT_EQ(0, voe_vsync_->SetMinimumPlayoutDelay(channel_, 200));
- CheckEstimatesConvergeReasonablyWell(kMinimumReasonableDelayEstimateMs);
- EXPECT_EQ(0, voe_vsync_->SetMinimumPlayoutDelay(channel_, 0));
- CheckEstimatesConvergeReasonablyWell(kMinimumReasonableDelayEstimateMs);
-}
-
-TEST_F(VideoSyncTest, CanGetPlayoutBufferSize) {
- int ignored;
- EXPECT_EQ(0, voe_vsync_->GetPlayoutBufferSize(ignored));
-}
diff --git a/webrtc/voice_engine/test/auto_test/voe_standard_test.cc b/webrtc/voice_engine/test/auto_test/voe_standard_test.cc
index 451082c..1c3a492 100644
--- a/webrtc/voice_engine/test/auto_test/voe_standard_test.cc
+++ b/webrtc/voice_engine/test/auto_test/voe_standard_test.cc
@@ -50,8 +50,6 @@
TEST_LOG(" Network\n");
if (_rtp_rtcp)
TEST_LOG(" RTP_RTCP\n");
- if (_videoSync)
- TEST_LOG(" VideoSync\n");
if (_volumeControl)
TEST_LOG(" VolumeControl\n");
if (_apm)
@@ -72,8 +70,6 @@
TEST_LOG(" Network\n");
if (!_rtp_rtcp)
TEST_LOG(" RTP_RTCP\n");
- if (!_videoSync)
- TEST_LOG(" VideoSync\n");
if (!_volumeControl)
TEST_LOG(" VolumeControl\n");
if (!_apm)
diff --git a/webrtc/voice_engine/test/auto_test/voe_standard_test.h b/webrtc/voice_engine/test/auto_test/voe_standard_test.h
index 6109603..5370ba4 100644
--- a/webrtc/voice_engine/test/auto_test/voe_standard_test.h
+++ b/webrtc/voice_engine/test/auto_test/voe_standard_test.h
@@ -36,7 +36,6 @@
_netEqStats(false),
_network(false),
_rtp_rtcp(false),
- _videoSync(false),
_volumeControl(false),
_apm(false) {
_codec = true;
@@ -45,7 +44,6 @@
_netEqStats = true;
_network = true;
_rtp_rtcp = true;
- _videoSync = true;
_volumeControl = true;
_apm = true;
}
@@ -55,7 +53,7 @@
private:
bool _base, _codec;
bool _file, _hardware;
- bool _netEqStats, _network, _rtp_rtcp, _videoSync, _volumeControl, _apm;
+ bool _netEqStats, _network, _rtp_rtcp, _volumeControl, _apm;
};
} // namespace voetest
diff --git a/webrtc/voice_engine/test/auto_test/voe_test_defines.h b/webrtc/voice_engine/test/auto_test/voe_test_defines.h
index 5d433d6..1db5ee1 100644
--- a/webrtc/voice_engine/test/auto_test/voe_test_defines.h
+++ b/webrtc/voice_engine/test/auto_test/voe_test_defines.h
@@ -24,7 +24,6 @@
#define _TEST_AUDIO_PROCESSING_
#define _TEST_FILE_
#define _TEST_NETWORK_
-#define _TEST_VIDEO_SYNC_
#define _TEST_NETEQ_STATS_
#define TESTED_AUDIO_LAYER kAudioPlatformDefault
diff --git a/webrtc/voice_engine/test/cmd_test/voe_cmd_test.cc b/webrtc/voice_engine/test/cmd_test/voe_cmd_test.cc
index 72d53b9..8f933f4 100644
--- a/webrtc/voice_engine/test/cmd_test/voe_cmd_test.cc
+++ b/webrtc/voice_engine/test/cmd_test/voe_cmd_test.cc
@@ -34,7 +34,6 @@
#include "webrtc/voice_engine/include/voe_neteq_stats.h"
#include "webrtc/voice_engine/include/voe_network.h"
#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
-#include "webrtc/voice_engine/include/voe_video_sync.h"
#include "webrtc/voice_engine/include/voe_volume_control.h"
#include "webrtc/voice_engine/test/channel_transport/channel_transport.h"
@@ -58,7 +57,6 @@
VoEAudioProcessing* apm = NULL;
VoENetwork* netw = NULL;
VoEFile* file = NULL;
-VoEVideoSync* vsync = NULL;
VoEHardware* hardware = NULL;
VoENetEqStats* neteqst = NULL;
@@ -129,7 +127,6 @@
rtp_rtcp = VoERTP_RTCP::GetInterface(m_voe);
netw = VoENetwork::GetInterface(m_voe);
file = VoEFile::GetInterface(m_voe);
- vsync = VoEVideoSync::GetInterface(m_voe);
hardware = VoEHardware::GetInterface(m_voe);
neteqst = VoENetEqStats::GetInterface(m_voe);
@@ -195,9 +192,6 @@
if (file)
file->Release();
- if (vsync)
- vsync->Release();
-
if (hardware)
hardware->Release();
diff --git a/webrtc/voice_engine/voe_video_sync_impl.cc b/webrtc/voice_engine/voe_video_sync_impl.cc
deleted file mode 100644
index e82cbb3..0000000
--- a/webrtc/voice_engine/voe_video_sync_impl.cc
+++ /dev/null
@@ -1,181 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/voice_engine/voe_video_sync_impl.h"
-
-#include "webrtc/system_wrappers/include/trace.h"
-#include "webrtc/voice_engine/channel.h"
-#include "webrtc/voice_engine/include/voe_errors.h"
-#include "webrtc/voice_engine/voice_engine_impl.h"
-
-namespace webrtc {
-
-VoEVideoSync* VoEVideoSync::GetInterface(VoiceEngine* voiceEngine) {
- if (NULL == voiceEngine) {
- return NULL;
- }
- VoiceEngineImpl* s = static_cast<VoiceEngineImpl*>(voiceEngine);
- s->AddRef();
- return s;
-}
-
-VoEVideoSyncImpl::VoEVideoSyncImpl(voe::SharedData* shared) : _shared(shared) {
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1),
- "VoEVideoSyncImpl::VoEVideoSyncImpl() - ctor");
-}
-
-VoEVideoSyncImpl::~VoEVideoSyncImpl() {
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1),
- "VoEVideoSyncImpl::~VoEVideoSyncImpl() - dtor");
-}
-
-int VoEVideoSyncImpl::GetPlayoutTimestamp(int channel,
- unsigned int& timestamp) {
- if (!_shared->statistics().Initialized()) {
- _shared->SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
- voe::Channel* channel_ptr = ch.channel();
- if (channel_ptr == NULL) {
- _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "GetPlayoutTimestamp() failed to locate channel");
- return -1;
- }
- return channel_ptr->GetPlayoutTimestamp(timestamp);
-}
-
-int VoEVideoSyncImpl::SetInitTimestamp(int channel, unsigned int timestamp) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
- "SetInitTimestamp(channel=%d, timestamp=%lu)", channel,
- timestamp);
-
- if (!_shared->statistics().Initialized()) {
- _shared->SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
- voe::Channel* channelPtr = ch.channel();
- if (channelPtr == NULL) {
- _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "SetInitTimestamp() failed to locate channel");
- return -1;
- }
- return channelPtr->SetInitTimestamp(timestamp);
-}
-
-int VoEVideoSyncImpl::SetInitSequenceNumber(int channel, short sequenceNumber) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
- "SetInitSequenceNumber(channel=%d, sequenceNumber=%hd)", channel,
- sequenceNumber);
-
- if (!_shared->statistics().Initialized()) {
- _shared->SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
- voe::Channel* channelPtr = ch.channel();
- if (channelPtr == NULL) {
- _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "SetInitSequenceNumber() failed to locate channel");
- return -1;
- }
- return channelPtr->SetInitSequenceNumber(sequenceNumber);
-}
-
-int VoEVideoSyncImpl::SetMinimumPlayoutDelay(int channel, int delayMs) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
- "SetMinimumPlayoutDelay(channel=%d, delayMs=%d)", channel,
- delayMs);
-
- if (!_shared->statistics().Initialized()) {
- _shared->SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
- voe::Channel* channelPtr = ch.channel();
- if (channelPtr == NULL) {
- _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "SetMinimumPlayoutDelay() failed to locate channel");
- return -1;
- }
- return channelPtr->SetMinimumPlayoutDelay(delayMs);
-}
-
-int VoEVideoSyncImpl::GetDelayEstimate(int channel,
- int* jitter_buffer_delay_ms,
- int* playout_buffer_delay_ms) {
- if (!_shared->statistics().Initialized()) {
- _shared->SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
- voe::Channel* channelPtr = ch.channel();
- if (channelPtr == NULL) {
- _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "GetDelayEstimate() failed to locate channel");
- return -1;
- }
- if (!channelPtr->GetDelayEstimate(jitter_buffer_delay_ms,
- playout_buffer_delay_ms)) {
- return -1;
- }
- return 0;
-}
-
-int VoEVideoSyncImpl::GetPlayoutBufferSize(int& bufferMs) {
- if (!_shared->statistics().Initialized()) {
- _shared->SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- AudioDeviceModule::BufferType type(AudioDeviceModule::kFixedBufferSize);
- uint16_t sizeMS(0);
- if (_shared->audio_device()->PlayoutBuffer(&type, &sizeMS) != 0) {
- _shared->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceError,
- "GetPlayoutBufferSize() failed to read buffer size");
- return -1;
- }
- bufferMs = sizeMS;
- return 0;
-}
-
-int VoEVideoSyncImpl::GetRtpRtcp(int channel,
- RtpRtcp** rtpRtcpModule,
- RtpReceiver** rtp_receiver) {
- if (!_shared->statistics().Initialized()) {
- _shared->SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
- voe::Channel* channelPtr = ch.channel();
- if (channelPtr == NULL) {
- _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "GetPlayoutTimestamp() failed to locate channel");
- return -1;
- }
- return channelPtr->GetRtpRtcp(rtpRtcpModule, rtp_receiver);
-}
-
-int VoEVideoSyncImpl::GetLeastRequiredDelayMs(int channel) const {
- if (!_shared->statistics().Initialized()) {
- _shared->SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
- voe::Channel* channel_ptr = ch.channel();
- if (channel_ptr == NULL) {
- _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "GetLeastRequiredDelayMs() failed to locate channel");
- return -1;
- }
- return channel_ptr->LeastRequiredDelayMs();
-}
-
-} // namespace webrtc
diff --git a/webrtc/voice_engine/voe_video_sync_impl.h b/webrtc/voice_engine/voe_video_sync_impl.h
deleted file mode 100644
index 8b367ee..0000000
--- a/webrtc/voice_engine/voe_video_sync_impl.h
+++ /dev/null
@@ -1,52 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_VOE_VIDEO_SYNC_IMPL_H
-#define WEBRTC_VOICE_ENGINE_VOE_VIDEO_SYNC_IMPL_H
-
-#include "webrtc/voice_engine/include/voe_video_sync.h"
-
-#include "webrtc/voice_engine/shared_data.h"
-
-namespace webrtc {
-
-class VoEVideoSyncImpl : public VoEVideoSync {
- public:
- int GetPlayoutBufferSize(int& bufferMs) override;
-
- int SetMinimumPlayoutDelay(int channel, int delayMs) override;
-
- int GetDelayEstimate(int channel,
- int* jitter_buffer_delay_ms,
- int* playout_buffer_delay_ms) override;
-
- int GetLeastRequiredDelayMs(int channel) const override;
-
- int SetInitTimestamp(int channel, unsigned int timestamp) override;
-
- int SetInitSequenceNumber(int channel, short sequenceNumber) override;
-
- int GetPlayoutTimestamp(int channel, unsigned int& timestamp) override;
-
- int GetRtpRtcp(int channel,
- RtpRtcp** rtpRtcpModule,
- RtpReceiver** rtp_receiver) override;
-
- protected:
- VoEVideoSyncImpl(voe::SharedData* shared);
- ~VoEVideoSyncImpl() override;
-
- private:
- voe::SharedData* _shared;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VOICE_ENGINE_VOE_VIDEO_SYNC_IMPL_H
diff --git a/webrtc/voice_engine/voice_engine_impl.h b/webrtc/voice_engine/voice_engine_impl.h
index ae1edcb..36ea439 100644
--- a/webrtc/voice_engine/voice_engine_impl.h
+++ b/webrtc/voice_engine/voice_engine_impl.h
@@ -23,7 +23,6 @@
#include "webrtc/voice_engine/voe_neteq_stats_impl.h"
#include "webrtc/voice_engine/voe_network_impl.h"
#include "webrtc/voice_engine/voe_rtp_rtcp_impl.h"
-#include "webrtc/voice_engine/voe_video_sync_impl.h"
#include "webrtc/voice_engine/voe_volume_control_impl.h"
namespace webrtc {
@@ -40,7 +39,6 @@
public VoENetEqStatsImpl,
public VoENetworkImpl,
public VoERTP_RTCPImpl,
- public VoEVideoSyncImpl,
public VoEVolumeControlImpl,
public VoEBaseImpl {
public:
@@ -53,7 +51,6 @@
VoENetEqStatsImpl(this),
VoENetworkImpl(this),
VoERTP_RTCPImpl(this),
- VoEVideoSyncImpl(this),
VoEVolumeControlImpl(this),
VoEBaseImpl(this),
_ref_count(0) {}
