ef92755780
This makes it easier to disable RTX by filtering out the RTX codec during call setup/signaling, and won't require that also the SSRCs are filtered out. BUG=1811 R=mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/15629005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6335 4adac7df-926f-26a2-2b94-8c16560cd09d
4565 lines
148 KiB
C++
4565 lines
148 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/voice_engine/channel.h"
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#include "webrtc/common.h"
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#include "webrtc/modules/audio_device/include/audio_device.h"
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#include "webrtc/modules/audio_processing/include/audio_processing.h"
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#include "webrtc/modules/interface/module_common_types.h"
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#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
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#include "webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h"
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#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
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#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
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#include "webrtc/modules/utility/interface/audio_frame_operations.h"
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#include "webrtc/modules/utility/interface/process_thread.h"
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#include "webrtc/modules/utility/interface/rtp_dump.h"
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#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
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#include "webrtc/system_wrappers/interface/logging.h"
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#include "webrtc/system_wrappers/interface/trace.h"
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#include "webrtc/video_engine/include/vie_network.h"
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#include "webrtc/voice_engine/include/voe_base.h"
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#include "webrtc/voice_engine/include/voe_external_media.h"
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#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
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#include "webrtc/voice_engine/output_mixer.h"
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#include "webrtc/voice_engine/statistics.h"
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#include "webrtc/voice_engine/transmit_mixer.h"
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#include "webrtc/voice_engine/utility.h"
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#if defined(_WIN32)
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#include <Qos.h>
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#endif
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namespace webrtc {
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namespace voe {
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// Extend the default RTCP statistics struct with max_jitter, defined as the
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// maximum jitter value seen in an RTCP report block.
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struct ChannelStatistics : public RtcpStatistics {
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ChannelStatistics() : rtcp(), max_jitter(0) {}
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RtcpStatistics rtcp;
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uint32_t max_jitter;
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};
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// Statistics callback, called at each generation of a new RTCP report block.
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class StatisticsProxy : public RtcpStatisticsCallback {
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public:
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StatisticsProxy(uint32_t ssrc)
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: stats_lock_(CriticalSectionWrapper::CreateCriticalSection()),
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ssrc_(ssrc) {}
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virtual ~StatisticsProxy() {}
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virtual void StatisticsUpdated(const RtcpStatistics& statistics,
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uint32_t ssrc) OVERRIDE {
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if (ssrc != ssrc_)
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return;
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CriticalSectionScoped cs(stats_lock_.get());
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stats_.rtcp = statistics;
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if (statistics.jitter > stats_.max_jitter) {
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stats_.max_jitter = statistics.jitter;
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}
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}
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void ResetStatistics() {
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CriticalSectionScoped cs(stats_lock_.get());
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stats_ = ChannelStatistics();
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}
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ChannelStatistics GetStats() {
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CriticalSectionScoped cs(stats_lock_.get());
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return stats_;
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}
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private:
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// StatisticsUpdated calls are triggered from threads in the RTP module,
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// while GetStats calls can be triggered from the public voice engine API,
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// hence synchronization is needed.
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scoped_ptr<CriticalSectionWrapper> stats_lock_;
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const uint32_t ssrc_;
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ChannelStatistics stats_;
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};
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class VoEBitrateObserver : public BitrateObserver {
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public:
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explicit VoEBitrateObserver(Channel* owner)
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: owner_(owner) {}
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virtual ~VoEBitrateObserver() {}
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// Implements BitrateObserver.
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virtual void OnNetworkChanged(const uint32_t bitrate_bps,
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const uint8_t fraction_lost,
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const uint32_t rtt) OVERRIDE {
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// |fraction_lost| has a scale of 0 - 255.
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owner_->OnNetworkChanged(bitrate_bps, fraction_lost, rtt);
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}
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private:
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Channel* owner_;
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};
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int32_t
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Channel::SendData(FrameType frameType,
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uint8_t payloadType,
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uint32_t timeStamp,
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const uint8_t* payloadData,
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uint16_t payloadSize,
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const RTPFragmentationHeader* fragmentation)
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{
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WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
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"Channel::SendData(frameType=%u, payloadType=%u, timeStamp=%u,"
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" payloadSize=%u, fragmentation=0x%x)",
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frameType, payloadType, timeStamp, payloadSize, fragmentation);
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if (_includeAudioLevelIndication)
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{
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// Store current audio level in the RTP/RTCP module.
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// The level will be used in combination with voice-activity state
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// (frameType) to add an RTP header extension
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_rtpRtcpModule->SetAudioLevel(rms_level_.RMS());
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}
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// Push data from ACM to RTP/RTCP-module to deliver audio frame for
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// packetization.
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// This call will trigger Transport::SendPacket() from the RTP/RTCP module.
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if (_rtpRtcpModule->SendOutgoingData((FrameType&)frameType,
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payloadType,
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timeStamp,
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// Leaving the time when this frame was
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// received from the capture device as
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// undefined for voice for now.
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-1,
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payloadData,
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payloadSize,
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fragmentation) == -1)
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{
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_engineStatisticsPtr->SetLastError(
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VE_RTP_RTCP_MODULE_ERROR, kTraceWarning,
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"Channel::SendData() failed to send data to RTP/RTCP module");
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return -1;
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}
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_lastLocalTimeStamp = timeStamp;
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_lastPayloadType = payloadType;
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return 0;
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}
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int32_t
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Channel::InFrameType(int16_t frameType)
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{
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WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
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"Channel::InFrameType(frameType=%d)", frameType);
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CriticalSectionScoped cs(&_callbackCritSect);
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// 1 indicates speech
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_sendFrameType = (frameType == 1) ? 1 : 0;
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return 0;
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}
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int32_t
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Channel::OnRxVadDetected(int vadDecision)
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{
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WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
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"Channel::OnRxVadDetected(vadDecision=%d)", vadDecision);
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CriticalSectionScoped cs(&_callbackCritSect);
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if (_rxVadObserverPtr)
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{
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_rxVadObserverPtr->OnRxVad(_channelId, vadDecision);
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}
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return 0;
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}
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int
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Channel::SendPacket(int channel, const void *data, int len)
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{
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channel = VoEChannelId(channel);
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assert(channel == _channelId);
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WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
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"Channel::SendPacket(channel=%d, len=%d)", channel, len);
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CriticalSectionScoped cs(&_callbackCritSect);
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if (_transportPtr == NULL)
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{
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WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,_channelId),
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"Channel::SendPacket() failed to send RTP packet due to"
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" invalid transport object");
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return -1;
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}
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uint8_t* bufferToSendPtr = (uint8_t*)data;
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int32_t bufferLength = len;
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// Dump the RTP packet to a file (if RTP dump is enabled).
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if (_rtpDumpOut.DumpPacket((const uint8_t*)data, len) == -1)
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{
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WEBRTC_TRACE(kTraceWarning, kTraceVoice,
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VoEId(_instanceId,_channelId),
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"Channel::SendPacket() RTP dump to output file failed");
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}
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int n = _transportPtr->SendPacket(channel, bufferToSendPtr,
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bufferLength);
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if (n < 0) {
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std::string transport_name =
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_externalTransport ? "external transport" : "WebRtc sockets";
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WEBRTC_TRACE(kTraceError, kTraceVoice,
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VoEId(_instanceId,_channelId),
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"Channel::SendPacket() RTP transmission using %s failed",
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transport_name.c_str());
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return -1;
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}
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return n;
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}
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int
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Channel::SendRTCPPacket(int channel, const void *data, int len)
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{
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channel = VoEChannelId(channel);
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assert(channel == _channelId);
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WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
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"Channel::SendRTCPPacket(channel=%d, len=%d)", channel, len);
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CriticalSectionScoped cs(&_callbackCritSect);
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if (_transportPtr == NULL)
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{
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WEBRTC_TRACE(kTraceError, kTraceVoice,
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VoEId(_instanceId,_channelId),
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"Channel::SendRTCPPacket() failed to send RTCP packet"
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" due to invalid transport object");
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return -1;
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}
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uint8_t* bufferToSendPtr = (uint8_t*)data;
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int32_t bufferLength = len;
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// Dump the RTCP packet to a file (if RTP dump is enabled).
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if (_rtpDumpOut.DumpPacket((const uint8_t*)data, len) == -1)
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{
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WEBRTC_TRACE(kTraceWarning, kTraceVoice,
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VoEId(_instanceId,_channelId),
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"Channel::SendPacket() RTCP dump to output file failed");
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}
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int n = _transportPtr->SendRTCPPacket(channel,
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bufferToSendPtr,
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bufferLength);
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if (n < 0) {
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std::string transport_name =
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_externalTransport ? "external transport" : "WebRtc sockets";
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WEBRTC_TRACE(kTraceInfo, kTraceVoice,
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VoEId(_instanceId,_channelId),
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"Channel::SendRTCPPacket() transmission using %s failed",
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transport_name.c_str());
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return -1;
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}
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return n;
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}
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void
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Channel::OnPlayTelephoneEvent(int32_t id,
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uint8_t event,
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uint16_t lengthMs,
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uint8_t volume)
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{
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WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
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"Channel::OnPlayTelephoneEvent(id=%d, event=%u, lengthMs=%u,"
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" volume=%u)", id, event, lengthMs, volume);
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if (!_playOutbandDtmfEvent || (event > 15))
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{
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// Ignore callback since feedback is disabled or event is not a
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// Dtmf tone event.
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return;
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}
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assert(_outputMixerPtr != NULL);
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// Start playing out the Dtmf tone (if playout is enabled).
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// Reduce length of tone with 80ms to the reduce risk of echo.
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_outputMixerPtr->PlayDtmfTone(event, lengthMs - 80, volume);
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}
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void
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Channel::OnIncomingSSRCChanged(int32_t id, uint32_t ssrc)
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{
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WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
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"Channel::OnIncomingSSRCChanged(id=%d, SSRC=%d)",
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id, ssrc);
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// Update ssrc so that NTP for AV sync can be updated.
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_rtpRtcpModule->SetRemoteSSRC(ssrc);
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}
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void Channel::OnIncomingCSRCChanged(int32_t id,
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uint32_t CSRC,
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bool added)
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{
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WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
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"Channel::OnIncomingCSRCChanged(id=%d, CSRC=%d, added=%d)",
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id, CSRC, added);
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}
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void Channel::ResetStatistics(uint32_t ssrc) {
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StreamStatistician* statistician =
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rtp_receive_statistics_->GetStatistician(ssrc);
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if (statistician) {
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statistician->ResetStatistics();
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}
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statistics_proxy_->ResetStatistics();
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}
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void
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Channel::OnApplicationDataReceived(int32_t id,
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uint8_t subType,
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uint32_t name,
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uint16_t length,
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const uint8_t* data)
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{
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WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
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"Channel::OnApplicationDataReceived(id=%d, subType=%u,"
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" name=%u, length=%u)",
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id, subType, name, length);
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int32_t channel = VoEChannelId(id);
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assert(channel == _channelId);
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if (_rtcpObserver)
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{
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CriticalSectionScoped cs(&_callbackCritSect);
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if (_rtcpObserverPtr)
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{
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_rtcpObserverPtr->OnApplicationDataReceived(channel,
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subType,
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name,
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data,
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length);
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}
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}
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}
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int32_t
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Channel::OnInitializeDecoder(
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int32_t id,
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int8_t payloadType,
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const char payloadName[RTP_PAYLOAD_NAME_SIZE],
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int frequency,
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uint8_t channels,
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uint32_t rate)
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{
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WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
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"Channel::OnInitializeDecoder(id=%d, payloadType=%d, "
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"payloadName=%s, frequency=%u, channels=%u, rate=%u)",
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id, payloadType, payloadName, frequency, channels, rate);
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assert(VoEChannelId(id) == _channelId);
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CodecInst receiveCodec = {0};
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CodecInst dummyCodec = {0};
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receiveCodec.pltype = payloadType;
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receiveCodec.plfreq = frequency;
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receiveCodec.channels = channels;
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receiveCodec.rate = rate;
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strncpy(receiveCodec.plname, payloadName, RTP_PAYLOAD_NAME_SIZE - 1);
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audio_coding_->Codec(payloadName, &dummyCodec, frequency, channels);
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receiveCodec.pacsize = dummyCodec.pacsize;
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// Register the new codec to the ACM
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if (audio_coding_->RegisterReceiveCodec(receiveCodec) == -1)
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{
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WEBRTC_TRACE(kTraceWarning, kTraceVoice,
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VoEId(_instanceId, _channelId),
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"Channel::OnInitializeDecoder() invalid codec ("
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"pt=%d, name=%s) received - 1", payloadType, payloadName);
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_engineStatisticsPtr->SetLastError(VE_AUDIO_CODING_MODULE_ERROR);
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return -1;
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}
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return 0;
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}
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void
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Channel::OnPacketTimeout(int32_t id)
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{
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WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
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"Channel::OnPacketTimeout(id=%d)", id);
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CriticalSectionScoped cs(_callbackCritSectPtr);
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if (_voiceEngineObserverPtr)
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{
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if (channel_state_.Get().receiving || _externalTransport)
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{
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int32_t channel = VoEChannelId(id);
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assert(channel == _channelId);
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// Ensure that next OnReceivedPacket() callback will trigger
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// a VE_PACKET_RECEIPT_RESTARTED callback.
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_rtpPacketTimedOut = true;
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// Deliver callback to the observer
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WEBRTC_TRACE(kTraceInfo, kTraceVoice,
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VoEId(_instanceId,_channelId),
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"Channel::OnPacketTimeout() => "
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"CallbackOnError(VE_RECEIVE_PACKET_TIMEOUT)");
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_voiceEngineObserverPtr->CallbackOnError(channel,
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VE_RECEIVE_PACKET_TIMEOUT);
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}
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}
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}
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|
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void
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Channel::OnReceivedPacket(int32_t id,
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RtpRtcpPacketType packetType)
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{
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WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
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"Channel::OnReceivedPacket(id=%d, packetType=%d)",
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id, packetType);
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assert(VoEChannelId(id) == _channelId);
|
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|
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// Notify only for the case when we have restarted an RTP session.
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if (_rtpPacketTimedOut && (kPacketRtp == packetType))
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{
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CriticalSectionScoped cs(_callbackCritSectPtr);
|
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if (_voiceEngineObserverPtr)
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{
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int32_t channel = VoEChannelId(id);
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assert(channel == _channelId);
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// Reset timeout mechanism
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_rtpPacketTimedOut = false;
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// Deliver callback to the observer
|
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WEBRTC_TRACE(kTraceInfo, kTraceVoice,
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VoEId(_instanceId,_channelId),
|
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"Channel::OnPacketTimeout() =>"
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" CallbackOnError(VE_PACKET_RECEIPT_RESTARTED)");
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_voiceEngineObserverPtr->CallbackOnError(
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channel,
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VE_PACKET_RECEIPT_RESTARTED);
|
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}
|
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}
|
|
}
|
|
|
|
void
|
|
Channel::OnPeriodicDeadOrAlive(int32_t id,
|
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RTPAliveType alive)
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::OnPeriodicDeadOrAlive(id=%d, alive=%d)", id, alive);
|
|
|
|
{
|
|
CriticalSectionScoped cs(&_callbackCritSect);
|
|
if (!_connectionObserver)
|
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return;
|
|
}
|
|
|
|
int32_t channel = VoEChannelId(id);
|
|
assert(channel == _channelId);
|
|
|
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// Use Alive as default to limit risk of false Dead detections
|
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bool isAlive(true);
|
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|
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// Always mark the connection as Dead when the module reports kRtpDead
|
|
if (kRtpDead == alive)
|
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{
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isAlive = false;
|
|
}
|
|
|
|
// It is possible that the connection is alive even if no RTP packet has
|
|
// been received for a long time since the other side might use VAD/DTX
|
|
// and a low SID-packet update rate.
|
|
if ((kRtpNoRtp == alive) && channel_state_.Get().playing)
|
|
{
|
|
// Detect Alive for all NetEQ states except for the case when we are
|
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// in PLC_CNG state.
|
|
// PLC_CNG <=> background noise only due to long expand or error.
|
|
// Note that, the case where the other side stops sending during CNG
|
|
// state will be detected as Alive. Dead is is not set until after
|
|
// missing RTCP packets for at least twelve seconds (handled
|
|
// internally by the RTP/RTCP module).
|
|
isAlive = (_outputSpeechType != AudioFrame::kPLCCNG);
|
|
}
|
|
|
|
// Send callback to the registered observer
|
|
if (_connectionObserver)
|
|
{
|
|
CriticalSectionScoped cs(&_callbackCritSect);
|
|
if (_connectionObserverPtr)
|
|
{
|
|
_connectionObserverPtr->OnPeriodicDeadOrAlive(channel, isAlive);
|
|
}
|
|
}
|
|
}
|
|
|
|
int32_t
|
|
Channel::OnReceivedPayloadData(const uint8_t* payloadData,
|
|
uint16_t payloadSize,
|
|
const WebRtcRTPHeader* rtpHeader)
|
|
{
|
|
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::OnReceivedPayloadData(payloadSize=%d,"
|
|
" payloadType=%u, audioChannel=%u)",
|
|
payloadSize,
|
|
rtpHeader->header.payloadType,
|
|
rtpHeader->type.Audio.channel);
|
|
|
|
if (!channel_state_.Get().playing)
|
|
{
|
|
// Avoid inserting into NetEQ when we are not playing. Count the
|
|
// packet as discarded.
|
|
WEBRTC_TRACE(kTraceStream, kTraceVoice,
|
|
VoEId(_instanceId, _channelId),
|
|
"received packet is discarded since playing is not"
|
|
" activated");
|
|
_numberOfDiscardedPackets++;
|
|
return 0;
|
|
}
|
|
|
|
// Push the incoming payload (parsed and ready for decoding) into the ACM
|
|
if (audio_coding_->IncomingPacket(payloadData,
|
|
payloadSize,
|
|
*rtpHeader) != 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_AUDIO_CODING_MODULE_ERROR, kTraceWarning,
|
|
"Channel::OnReceivedPayloadData() unable to push data to the ACM");
|
|
return -1;
|
|
}
|
|
|
|
// Update the packet delay.
|
|
UpdatePacketDelay(rtpHeader->header.timestamp,
|
|
rtpHeader->header.sequenceNumber);
|
|
|
|
uint16_t round_trip_time = 0;
|
|
_rtpRtcpModule->RTT(rtp_receiver_->SSRC(), &round_trip_time,
|
|
NULL, NULL, NULL);
|
|
|
|
std::vector<uint16_t> nack_list = audio_coding_->GetNackList(
|
|
round_trip_time);
|
|
if (!nack_list.empty()) {
|
|
// Can't use nack_list.data() since it's not supported by all
|
|
// compilers.
|
|
ResendPackets(&(nack_list[0]), static_cast<int>(nack_list.size()));
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
bool Channel::OnRecoveredPacket(const uint8_t* rtp_packet,
|
|
int rtp_packet_length) {
|
|
RTPHeader header;
|
|
if (!rtp_header_parser_->Parse(rtp_packet, rtp_packet_length, &header)) {
|
|
WEBRTC_TRACE(kTraceDebug, webrtc::kTraceVoice, _channelId,
|
|
"IncomingPacket invalid RTP header");
|
|
return false;
|
|
}
|
|
header.payload_type_frequency =
|
|
rtp_payload_registry_->GetPayloadTypeFrequency(header.payloadType);
|
|
if (header.payload_type_frequency < 0)
|
|
return false;
|
|
return ReceivePacket(rtp_packet, rtp_packet_length, header, false);
|
|
}
|
|
|
|
int32_t Channel::GetAudioFrame(int32_t id, AudioFrame& audioFrame)
|
|
{
|
|
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::GetAudioFrame(id=%d)", id);
|
|
|
|
// Get 10ms raw PCM data from the ACM (mixer limits output frequency)
|
|
if (audio_coding_->PlayoutData10Ms(audioFrame.sample_rate_hz_,
|
|
&audioFrame) == -1)
|
|
{
|
|
WEBRTC_TRACE(kTraceError, kTraceVoice,
|
|
VoEId(_instanceId,_channelId),
|
|
"Channel::GetAudioFrame() PlayoutData10Ms() failed!");
|
|
// In all likelihood, the audio in this frame is garbage. We return an
|
|
// error so that the audio mixer module doesn't add it to the mix. As
|
|
// a result, it won't be played out and the actions skipped here are
|
|
// irrelevant.
|
|
return -1;
|
|
}
|
|
|
|
if (_RxVadDetection)
|
|
{
|
|
UpdateRxVadDetection(audioFrame);
|
|
}
|
|
|
|
// Convert module ID to internal VoE channel ID
|
|
audioFrame.id_ = VoEChannelId(audioFrame.id_);
|
|
// Store speech type for dead-or-alive detection
|
|
_outputSpeechType = audioFrame.speech_type_;
|
|
|
|
ChannelState::State state = channel_state_.Get();
|
|
|
|
if (state.rx_apm_is_enabled) {
|
|
int err = rx_audioproc_->ProcessStream(&audioFrame);
|
|
if (err) {
|
|
LOG(LS_ERROR) << "ProcessStream() error: " << err;
|
|
assert(false);
|
|
}
|
|
}
|
|
|
|
float output_gain = 1.0f;
|
|
float left_pan = 1.0f;
|
|
float right_pan = 1.0f;
|
|
{
|
|
CriticalSectionScoped cs(&volume_settings_critsect_);
|
|
output_gain = _outputGain;
|
|
left_pan = _panLeft;
|
|
right_pan= _panRight;
|
|
}
|
|
|
|
// Output volume scaling
|
|
if (output_gain < 0.99f || output_gain > 1.01f)
|
|
{
|
|
AudioFrameOperations::ScaleWithSat(output_gain, audioFrame);
|
|
}
|
|
|
|
// Scale left and/or right channel(s) if stereo and master balance is
|
|
// active
|
|
|
|
if (left_pan != 1.0f || right_pan != 1.0f)
|
|
{
|
|
if (audioFrame.num_channels_ == 1)
|
|
{
|
|
// Emulate stereo mode since panning is active.
|
|
// The mono signal is copied to both left and right channels here.
|
|
AudioFrameOperations::MonoToStereo(&audioFrame);
|
|
}
|
|
// For true stereo mode (when we are receiving a stereo signal), no
|
|
// action is needed.
|
|
|
|
// Do the panning operation (the audio frame contains stereo at this
|
|
// stage)
|
|
AudioFrameOperations::Scale(left_pan, right_pan, audioFrame);
|
|
}
|
|
|
|
// Mix decoded PCM output with file if file mixing is enabled
|
|
if (state.output_file_playing)
|
|
{
|
|
MixAudioWithFile(audioFrame, audioFrame.sample_rate_hz_);
|
|
}
|
|
|
|
// External media
|
|
if (_outputExternalMedia)
|
|
{
|
|
CriticalSectionScoped cs(&_callbackCritSect);
|
|
const bool isStereo = (audioFrame.num_channels_ == 2);
|
|
if (_outputExternalMediaCallbackPtr)
|
|
{
|
|
_outputExternalMediaCallbackPtr->Process(
|
|
_channelId,
|
|
kPlaybackPerChannel,
|
|
(int16_t*)audioFrame.data_,
|
|
audioFrame.samples_per_channel_,
|
|
audioFrame.sample_rate_hz_,
|
|
isStereo);
|
|
}
|
|
}
|
|
|
|
// Record playout if enabled
|
|
{
|
|
CriticalSectionScoped cs(&_fileCritSect);
|
|
|
|
if (_outputFileRecording && _outputFileRecorderPtr)
|
|
{
|
|
_outputFileRecorderPtr->RecordAudioToFile(audioFrame);
|
|
}
|
|
}
|
|
|
|
// Measure audio level (0-9)
|
|
_outputAudioLevel.ComputeLevel(audioFrame);
|
|
|
|
audioFrame.ntp_time_ms_ = ntp_estimator_->Estimate(audioFrame.timestamp_);
|
|
|
|
if (!first_frame_arrived_) {
|
|
first_frame_arrived_ = true;
|
|
capture_start_rtp_time_stamp_ = audioFrame.timestamp_;
|
|
} else {
|
|
// |ntp_time_ms_| won't be valid until at least 2 RTCP SRs are received.
|
|
if (audioFrame.ntp_time_ms_ > 0) {
|
|
// Compute |capture_start_ntp_time_ms_| so that
|
|
// |capture_start_ntp_time_ms_| + |elapsed_time_ms| == |ntp_time_ms_|
|
|
CriticalSectionScoped lock(ts_stats_lock_.get());
|
|
uint32_t elapsed_time_ms =
|
|
(audioFrame.timestamp_ - capture_start_rtp_time_stamp_) /
|
|
(audioFrame.sample_rate_hz_ * 1000);
|
|
capture_start_ntp_time_ms_ = audioFrame.ntp_time_ms_ - elapsed_time_ms;
|
|
}
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
int32_t
|
|
Channel::NeededFrequency(int32_t id)
|
|
{
|
|
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::NeededFrequency(id=%d)", id);
|
|
|
|
int highestNeeded = 0;
|
|
|
|
// Determine highest needed receive frequency
|
|
int32_t receiveFrequency = audio_coding_->ReceiveFrequency();
|
|
|
|
// Return the bigger of playout and receive frequency in the ACM.
|
|
if (audio_coding_->PlayoutFrequency() > receiveFrequency)
|
|
{
|
|
highestNeeded = audio_coding_->PlayoutFrequency();
|
|
}
|
|
else
|
|
{
|
|
highestNeeded = receiveFrequency;
|
|
}
|
|
|
|
// Special case, if we're playing a file on the playout side
|
|
// we take that frequency into consideration as well
|
|
// This is not needed on sending side, since the codec will
|
|
// limit the spectrum anyway.
|
|
if (channel_state_.Get().output_file_playing)
|
|
{
|
|
CriticalSectionScoped cs(&_fileCritSect);
|
|
if (_outputFilePlayerPtr)
|
|
{
|
|
if(_outputFilePlayerPtr->Frequency()>highestNeeded)
|
|
{
|
|
highestNeeded=_outputFilePlayerPtr->Frequency();
|
|
}
|
|
}
|
|
}
|
|
|
|
return(highestNeeded);
|
|
}
|
|
|
|
int32_t
|
|
Channel::CreateChannel(Channel*& channel,
|
|
int32_t channelId,
|
|
uint32_t instanceId,
|
|
const Config& config)
|
|
{
|
|
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId,channelId),
|
|
"Channel::CreateChannel(channelId=%d, instanceId=%d)",
|
|
channelId, instanceId);
|
|
|
|
channel = new Channel(channelId, instanceId, config);
|
|
if (channel == NULL)
|
|
{
|
|
WEBRTC_TRACE(kTraceMemory, kTraceVoice,
|
|
VoEId(instanceId,channelId),
|
|
"Channel::CreateChannel() unable to allocate memory for"
|
|
" channel");
|
|
return -1;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
void
|
|
Channel::PlayNotification(int32_t id, uint32_t durationMs)
|
|
{
|
|
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::PlayNotification(id=%d, durationMs=%d)",
|
|
id, durationMs);
|
|
|
|
// Not implement yet
|
|
}
|
|
|
|
void
|
|
Channel::RecordNotification(int32_t id, uint32_t durationMs)
|
|
{
|
|
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::RecordNotification(id=%d, durationMs=%d)",
|
|
id, durationMs);
|
|
|
|
// Not implement yet
|
|
}
|
|
|
|
void
|
|
Channel::PlayFileEnded(int32_t id)
|
|
{
|
|
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::PlayFileEnded(id=%d)", id);
|
|
|
|
if (id == _inputFilePlayerId)
|
|
{
|
|
channel_state_.SetInputFilePlaying(false);
|
|
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
|
|
VoEId(_instanceId,_channelId),
|
|
"Channel::PlayFileEnded() => input file player module is"
|
|
" shutdown");
|
|
}
|
|
else if (id == _outputFilePlayerId)
|
|
{
|
|
channel_state_.SetOutputFilePlaying(false);
|
|
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
|
|
VoEId(_instanceId,_channelId),
|
|
"Channel::PlayFileEnded() => output file player module is"
|
|
" shutdown");
|
|
}
|
|
}
|
|
|
|
void
|
|
Channel::RecordFileEnded(int32_t id)
|
|
{
|
|
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::RecordFileEnded(id=%d)", id);
|
|
|
|
assert(id == _outputFileRecorderId);
|
|
|
|
CriticalSectionScoped cs(&_fileCritSect);
|
|
|
|
_outputFileRecording = false;
|
|
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
|
|
VoEId(_instanceId,_channelId),
|
|
"Channel::RecordFileEnded() => output file recorder module is"
|
|
" shutdown");
|
|
}
|
|
|
|
Channel::Channel(int32_t channelId,
|
|
uint32_t instanceId,
|
|
const Config& config) :
|
|
_fileCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
|
|
_callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
|
|
volume_settings_critsect_(*CriticalSectionWrapper::CreateCriticalSection()),
|
|
_instanceId(instanceId),
|
|
_channelId(channelId),
|
|
rtp_header_parser_(RtpHeaderParser::Create()),
|
|
rtp_payload_registry_(
|
|
new RTPPayloadRegistry(RTPPayloadStrategy::CreateStrategy(true))),
|
|
rtp_receive_statistics_(ReceiveStatistics::Create(
|
|
Clock::GetRealTimeClock())),
|
|
rtp_receiver_(RtpReceiver::CreateAudioReceiver(
|
|
VoEModuleId(instanceId, channelId), Clock::GetRealTimeClock(), this,
|
|
this, this, rtp_payload_registry_.get())),
|
|
telephone_event_handler_(rtp_receiver_->GetTelephoneEventHandler()),
|
|
audio_coding_(AudioCodingModule::Create(
|
|
VoEModuleId(instanceId, channelId))),
|
|
_rtpDumpIn(*RtpDump::CreateRtpDump()),
|
|
_rtpDumpOut(*RtpDump::CreateRtpDump()),
|
|
_outputAudioLevel(),
|
|
_externalTransport(false),
|
|
_audioLevel_dBov(0),
|
|
_inputFilePlayerPtr(NULL),
|
|
_outputFilePlayerPtr(NULL),
|
|
_outputFileRecorderPtr(NULL),
|
|
// Avoid conflict with other channels by adding 1024 - 1026,
|
|
// won't use as much as 1024 channels.
|
|
_inputFilePlayerId(VoEModuleId(instanceId, channelId) + 1024),
|
|
_outputFilePlayerId(VoEModuleId(instanceId, channelId) + 1025),
|
|
_outputFileRecorderId(VoEModuleId(instanceId, channelId) + 1026),
|
|
_outputFileRecording(false),
|
|
_inbandDtmfQueue(VoEModuleId(instanceId, channelId)),
|
|
_inbandDtmfGenerator(VoEModuleId(instanceId, channelId)),
|
|
_outputExternalMedia(false),
|
|
_inputExternalMediaCallbackPtr(NULL),
|
|
_outputExternalMediaCallbackPtr(NULL),
|
|
_timeStamp(0), // This is just an offset, RTP module will add it's own random offset
|
|
_sendTelephoneEventPayloadType(106),
|
|
ntp_estimator_(new RemoteNtpTimeEstimator(Clock::GetRealTimeClock())),
|
|
jitter_buffer_playout_timestamp_(0),
|
|
playout_timestamp_rtp_(0),
|
|
playout_timestamp_rtcp_(0),
|
|
playout_delay_ms_(0),
|
|
_numberOfDiscardedPackets(0),
|
|
send_sequence_number_(0),
|
|
ts_stats_lock_(CriticalSectionWrapper::CreateCriticalSection()),
|
|
first_frame_arrived_(false),
|
|
capture_start_rtp_time_stamp_(0),
|
|
capture_start_ntp_time_ms_(-1),
|
|
_engineStatisticsPtr(NULL),
|
|
_outputMixerPtr(NULL),
|
|
_transmitMixerPtr(NULL),
|
|
_moduleProcessThreadPtr(NULL),
|
|
_audioDeviceModulePtr(NULL),
|
|
_voiceEngineObserverPtr(NULL),
|
|
_callbackCritSectPtr(NULL),
|
|
_transportPtr(NULL),
|
|
_rxVadObserverPtr(NULL),
|
|
_oldVadDecision(-1),
|
|
_sendFrameType(0),
|
|
_rtcpObserverPtr(NULL),
|
|
_externalPlayout(false),
|
|
_externalMixing(false),
|
|
_mixFileWithMicrophone(false),
|
|
_rtcpObserver(false),
|
|
_mute(false),
|
|
_panLeft(1.0f),
|
|
_panRight(1.0f),
|
|
_outputGain(1.0f),
|
|
_playOutbandDtmfEvent(false),
|
|
_playInbandDtmfEvent(false),
|
|
_lastLocalTimeStamp(0),
|
|
_lastPayloadType(0),
|
|
_includeAudioLevelIndication(false),
|
|
_rtpPacketTimedOut(false),
|
|
_rtpPacketTimeOutIsEnabled(false),
|
|
_rtpTimeOutSeconds(0),
|
|
_connectionObserver(false),
|
|
_connectionObserverPtr(NULL),
|
|
_outputSpeechType(AudioFrame::kNormalSpeech),
|
|
vie_network_(NULL),
|
|
video_channel_(-1),
|
|
_average_jitter_buffer_delay_us(0),
|
|
least_required_delay_ms_(0),
|
|
_previousTimestamp(0),
|
|
_recPacketDelayMs(20),
|
|
_RxVadDetection(false),
|
|
_rxAgcIsEnabled(false),
|
|
_rxNsIsEnabled(false),
|
|
restored_packet_in_use_(false),
|
|
bitrate_controller_(
|
|
BitrateController::CreateBitrateController(Clock::GetRealTimeClock(),
|
|
true)),
|
|
rtcp_bandwidth_observer_(
|
|
bitrate_controller_->CreateRtcpBandwidthObserver()),
|
|
send_bitrate_observer_(new VoEBitrateObserver(this))
|
|
{
|
|
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::Channel() - ctor");
|
|
_inbandDtmfQueue.ResetDtmf();
|
|
_inbandDtmfGenerator.Init();
|
|
_outputAudioLevel.Clear();
|
|
|
|
RtpRtcp::Configuration configuration;
|
|
configuration.id = VoEModuleId(instanceId, channelId);
|
|
configuration.audio = true;
|
|
configuration.outgoing_transport = this;
|
|
configuration.rtcp_feedback = this;
|
|
configuration.audio_messages = this;
|
|
configuration.receive_statistics = rtp_receive_statistics_.get();
|
|
configuration.bandwidth_callback = rtcp_bandwidth_observer_.get();
|
|
|
|
_rtpRtcpModule.reset(RtpRtcp::CreateRtpRtcp(configuration));
|
|
|
|
statistics_proxy_.reset(new StatisticsProxy(_rtpRtcpModule->SSRC()));
|
|
rtp_receive_statistics_->RegisterRtcpStatisticsCallback(
|
|
statistics_proxy_.get());
|
|
|
|
Config audioproc_config;
|
|
audioproc_config.Set<ExperimentalAgc>(new ExperimentalAgc(false));
|
|
rx_audioproc_.reset(AudioProcessing::Create(audioproc_config));
|
|
}
|
|
|
|
Channel::~Channel()
|
|
{
|
|
rtp_receive_statistics_->RegisterRtcpStatisticsCallback(NULL);
|
|
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::~Channel() - dtor");
|
|
|
|
if (_outputExternalMedia)
|
|
{
|
|
DeRegisterExternalMediaProcessing(kPlaybackPerChannel);
|
|
}
|
|
if (channel_state_.Get().input_external_media)
|
|
{
|
|
DeRegisterExternalMediaProcessing(kRecordingPerChannel);
|
|
}
|
|
StopSend();
|
|
StopPlayout();
|
|
|
|
{
|
|
CriticalSectionScoped cs(&_fileCritSect);
|
|
if (_inputFilePlayerPtr)
|
|
{
|
|
_inputFilePlayerPtr->RegisterModuleFileCallback(NULL);
|
|
_inputFilePlayerPtr->StopPlayingFile();
|
|
FilePlayer::DestroyFilePlayer(_inputFilePlayerPtr);
|
|
_inputFilePlayerPtr = NULL;
|
|
}
|
|
if (_outputFilePlayerPtr)
|
|
{
|
|
_outputFilePlayerPtr->RegisterModuleFileCallback(NULL);
|
|
_outputFilePlayerPtr->StopPlayingFile();
|
|
FilePlayer::DestroyFilePlayer(_outputFilePlayerPtr);
|
|
_outputFilePlayerPtr = NULL;
|
|
}
|
|
if (_outputFileRecorderPtr)
|
|
{
|
|
_outputFileRecorderPtr->RegisterModuleFileCallback(NULL);
|
|
_outputFileRecorderPtr->StopRecording();
|
|
FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
|
|
_outputFileRecorderPtr = NULL;
|
|
}
|
|
}
|
|
|
|
// The order to safely shutdown modules in a channel is:
|
|
// 1. De-register callbacks in modules
|
|
// 2. De-register modules in process thread
|
|
// 3. Destroy modules
|
|
if (audio_coding_->RegisterTransportCallback(NULL) == -1)
|
|
{
|
|
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
|
|
VoEId(_instanceId,_channelId),
|
|
"~Channel() failed to de-register transport callback"
|
|
" (Audio coding module)");
|
|
}
|
|
if (audio_coding_->RegisterVADCallback(NULL) == -1)
|
|
{
|
|
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
|
|
VoEId(_instanceId,_channelId),
|
|
"~Channel() failed to de-register VAD callback"
|
|
" (Audio coding module)");
|
|
}
|
|
// De-register modules in process thread
|
|
if (_moduleProcessThreadPtr->DeRegisterModule(_rtpRtcpModule.get()) == -1)
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice,
|
|
VoEId(_instanceId,_channelId),
|
|
"~Channel() failed to deregister RTP/RTCP module");
|
|
}
|
|
// End of modules shutdown
|
|
|
|
// Delete other objects
|
|
if (vie_network_) {
|
|
vie_network_->Release();
|
|
vie_network_ = NULL;
|
|
}
|
|
RtpDump::DestroyRtpDump(&_rtpDumpIn);
|
|
RtpDump::DestroyRtpDump(&_rtpDumpOut);
|
|
delete &_callbackCritSect;
|
|
delete &_fileCritSect;
|
|
delete &volume_settings_critsect_;
|
|
}
|
|
|
|
int32_t
|
|
Channel::Init()
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::Init()");
|
|
|
|
channel_state_.Reset();
|
|
|
|
// --- Initial sanity
|
|
|
|
if ((_engineStatisticsPtr == NULL) ||
|
|
(_moduleProcessThreadPtr == NULL))
|
|
{
|
|
WEBRTC_TRACE(kTraceError, kTraceVoice,
|
|
VoEId(_instanceId,_channelId),
|
|
"Channel::Init() must call SetEngineInformation() first");
|
|
return -1;
|
|
}
|
|
|
|
// --- Add modules to process thread (for periodic schedulation)
|
|
|
|
const bool processThreadFail =
|
|
((_moduleProcessThreadPtr->RegisterModule(_rtpRtcpModule.get()) != 0) ||
|
|
false);
|
|
if (processThreadFail)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_CANNOT_INIT_CHANNEL, kTraceError,
|
|
"Channel::Init() modules not registered");
|
|
return -1;
|
|
}
|
|
// --- ACM initialization
|
|
|
|
if ((audio_coding_->InitializeReceiver() == -1) ||
|
|
#ifdef WEBRTC_CODEC_AVT
|
|
// out-of-band Dtmf tones are played out by default
|
|
(audio_coding_->SetDtmfPlayoutStatus(true) == -1) ||
|
|
#endif
|
|
(audio_coding_->InitializeSender() == -1))
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
|
|
"Channel::Init() unable to initialize the ACM - 1");
|
|
return -1;
|
|
}
|
|
|
|
// --- RTP/RTCP module initialization
|
|
|
|
// Ensure that RTCP is enabled by default for the created channel.
|
|
// Note that, the module will keep generating RTCP until it is explicitly
|
|
// disabled by the user.
|
|
// After StopListen (when no sockets exists), RTCP packets will no longer
|
|
// be transmitted since the Transport object will then be invalid.
|
|
telephone_event_handler_->SetTelephoneEventForwardToDecoder(true);
|
|
// RTCP is enabled by default.
|
|
if (_rtpRtcpModule->SetRTCPStatus(kRtcpCompound) == -1)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_RTP_RTCP_MODULE_ERROR, kTraceError,
|
|
"Channel::Init() RTP/RTCP module not initialized");
|
|
return -1;
|
|
}
|
|
|
|
// --- Register all permanent callbacks
|
|
const bool fail =
|
|
(audio_coding_->RegisterTransportCallback(this) == -1) ||
|
|
(audio_coding_->RegisterVADCallback(this) == -1);
|
|
|
|
if (fail)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_CANNOT_INIT_CHANNEL, kTraceError,
|
|
"Channel::Init() callbacks not registered");
|
|
return -1;
|
|
}
|
|
|
|
// --- Register all supported codecs to the receiving side of the
|
|
// RTP/RTCP module
|
|
|
|
CodecInst codec;
|
|
const uint8_t nSupportedCodecs = AudioCodingModule::NumberOfCodecs();
|
|
|
|
for (int idx = 0; idx < nSupportedCodecs; idx++)
|
|
{
|
|
// Open up the RTP/RTCP receiver for all supported codecs
|
|
if ((audio_coding_->Codec(idx, &codec) == -1) ||
|
|
(rtp_receiver_->RegisterReceivePayload(
|
|
codec.plname,
|
|
codec.pltype,
|
|
codec.plfreq,
|
|
codec.channels,
|
|
(codec.rate < 0) ? 0 : codec.rate) == -1))
|
|
{
|
|
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
|
|
VoEId(_instanceId,_channelId),
|
|
"Channel::Init() unable to register %s (%d/%d/%d/%d) "
|
|
"to RTP/RTCP receiver",
|
|
codec.plname, codec.pltype, codec.plfreq,
|
|
codec.channels, codec.rate);
|
|
}
|
|
else
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice,
|
|
VoEId(_instanceId,_channelId),
|
|
"Channel::Init() %s (%d/%d/%d/%d) has been added to "
|
|
"the RTP/RTCP receiver",
|
|
codec.plname, codec.pltype, codec.plfreq,
|
|
codec.channels, codec.rate);
|
|
}
|
|
|
|
// Ensure that PCMU is used as default codec on the sending side
|
|
if (!STR_CASE_CMP(codec.plname, "PCMU") && (codec.channels == 1))
|
|
{
|
|
SetSendCodec(codec);
|
|
}
|
|
|
|
// Register default PT for outband 'telephone-event'
|
|
if (!STR_CASE_CMP(codec.plname, "telephone-event"))
|
|
{
|
|
if ((_rtpRtcpModule->RegisterSendPayload(codec) == -1) ||
|
|
(audio_coding_->RegisterReceiveCodec(codec) == -1))
|
|
{
|
|
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
|
|
VoEId(_instanceId,_channelId),
|
|
"Channel::Init() failed to register outband "
|
|
"'telephone-event' (%d/%d) correctly",
|
|
codec.pltype, codec.plfreq);
|
|
}
|
|
}
|
|
|
|
if (!STR_CASE_CMP(codec.plname, "CN"))
|
|
{
|
|
if ((audio_coding_->RegisterSendCodec(codec) == -1) ||
|
|
(audio_coding_->RegisterReceiveCodec(codec) == -1) ||
|
|
(_rtpRtcpModule->RegisterSendPayload(codec) == -1))
|
|
{
|
|
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
|
|
VoEId(_instanceId,_channelId),
|
|
"Channel::Init() failed to register CN (%d/%d) "
|
|
"correctly - 1",
|
|
codec.pltype, codec.plfreq);
|
|
}
|
|
}
|
|
#ifdef WEBRTC_CODEC_RED
|
|
// Register RED to the receiving side of the ACM.
|
|
// We will not receive an OnInitializeDecoder() callback for RED.
|
|
if (!STR_CASE_CMP(codec.plname, "RED"))
|
|
{
|
|
if (audio_coding_->RegisterReceiveCodec(codec) == -1)
|
|
{
|
|
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
|
|
VoEId(_instanceId,_channelId),
|
|
"Channel::Init() failed to register RED (%d/%d) "
|
|
"correctly",
|
|
codec.pltype, codec.plfreq);
|
|
}
|
|
}
|
|
#endif
|
|
}
|
|
|
|
if (rx_audioproc_->noise_suppression()->set_level(kDefaultNsMode) != 0) {
|
|
LOG_FERR1(LS_ERROR, noise_suppression()->set_level, kDefaultNsMode);
|
|
return -1;
|
|
}
|
|
if (rx_audioproc_->gain_control()->set_mode(kDefaultRxAgcMode) != 0) {
|
|
LOG_FERR1(LS_ERROR, gain_control()->set_mode, kDefaultRxAgcMode);
|
|
return -1;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
int32_t
|
|
Channel::SetEngineInformation(Statistics& engineStatistics,
|
|
OutputMixer& outputMixer,
|
|
voe::TransmitMixer& transmitMixer,
|
|
ProcessThread& moduleProcessThread,
|
|
AudioDeviceModule& audioDeviceModule,
|
|
VoiceEngineObserver* voiceEngineObserver,
|
|
CriticalSectionWrapper* callbackCritSect)
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::SetEngineInformation()");
|
|
_engineStatisticsPtr = &engineStatistics;
|
|
_outputMixerPtr = &outputMixer;
|
|
_transmitMixerPtr = &transmitMixer,
|
|
_moduleProcessThreadPtr = &moduleProcessThread;
|
|
_audioDeviceModulePtr = &audioDeviceModule;
|
|
_voiceEngineObserverPtr = voiceEngineObserver;
|
|
_callbackCritSectPtr = callbackCritSect;
|
|
return 0;
|
|
}
|
|
|
|
int32_t
|
|
Channel::UpdateLocalTimeStamp()
|
|
{
|
|
|
|
_timeStamp += _audioFrame.samples_per_channel_;
|
|
return 0;
|
|
}
|
|
|
|
int32_t
|
|
Channel::StartPlayout()
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::StartPlayout()");
|
|
if (channel_state_.Get().playing)
|
|
{
|
|
return 0;
|
|
}
|
|
|
|
if (!_externalMixing) {
|
|
// Add participant as candidates for mixing.
|
|
if (_outputMixerPtr->SetMixabilityStatus(*this, true) != 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_AUDIO_CONF_MIX_MODULE_ERROR, kTraceError,
|
|
"StartPlayout() failed to add participant to mixer");
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
channel_state_.SetPlaying(true);
|
|
if (RegisterFilePlayingToMixer() != 0)
|
|
return -1;
|
|
|
|
return 0;
|
|
}
|
|
|
|
int32_t
|
|
Channel::StopPlayout()
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::StopPlayout()");
|
|
if (!channel_state_.Get().playing)
|
|
{
|
|
return 0;
|
|
}
|
|
|
|
if (!_externalMixing) {
|
|
// Remove participant as candidates for mixing
|
|
if (_outputMixerPtr->SetMixabilityStatus(*this, false) != 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_AUDIO_CONF_MIX_MODULE_ERROR, kTraceError,
|
|
"StopPlayout() failed to remove participant from mixer");
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
channel_state_.SetPlaying(false);
|
|
_outputAudioLevel.Clear();
|
|
|
|
return 0;
|
|
}
|
|
|
|
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);
|
|
|
|
if (_rtpRtcpModule->SetSendingStatus(true) != 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_RTP_RTCP_MODULE_ERROR, kTraceError,
|
|
"StartSend() RTP/RTCP failed to start sending");
|
|
CriticalSectionScoped cs(&_callbackCritSect);
|
|
channel_state_.SetSending(false);
|
|
return -1;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
int32_t
|
|
Channel::StopSend()
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::StopSend()");
|
|
if (!channel_state_.Get().sending)
|
|
{
|
|
return 0;
|
|
}
|
|
channel_state_.SetSending(false);
|
|
|
|
// Store the sequence number to be able to pick up the same sequence for
|
|
// the next StartSend(). This is needed for restarting device, otherwise
|
|
// it might cause libSRTP to complain about packets being replayed.
|
|
// TODO(xians): Remove this workaround after RtpRtcpModule's refactoring
|
|
// CL is landed. See issue
|
|
// https://code.google.com/p/webrtc/issues/detail?id=2111 .
|
|
send_sequence_number_ = _rtpRtcpModule->SequenceNumber();
|
|
|
|
// Reset sending SSRC and sequence number and triggers direct transmission
|
|
// of RTCP BYE
|
|
if (_rtpRtcpModule->SetSendingStatus(false) == -1 ||
|
|
_rtpRtcpModule->ResetSendDataCountersRTP() == -1)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_RTP_RTCP_MODULE_ERROR, kTraceWarning,
|
|
"StartSend() RTP/RTCP failed to stop sending");
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
int32_t
|
|
Channel::StartReceiving()
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::StartReceiving()");
|
|
if (channel_state_.Get().receiving)
|
|
{
|
|
return 0;
|
|
}
|
|
channel_state_.SetReceiving(true);
|
|
_numberOfDiscardedPackets = 0;
|
|
return 0;
|
|
}
|
|
|
|
int32_t
|
|
Channel::StopReceiving()
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::StopReceiving()");
|
|
if (!channel_state_.Get().receiving)
|
|
{
|
|
return 0;
|
|
}
|
|
|
|
channel_state_.SetReceiving(false);
|
|
return 0;
|
|
}
|
|
|
|
int32_t
|
|
Channel::SetNetEQPlayoutMode(NetEqModes mode)
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::SetNetEQPlayoutMode()");
|
|
AudioPlayoutMode playoutMode(voice);
|
|
switch (mode)
|
|
{
|
|
case kNetEqDefault:
|
|
playoutMode = voice;
|
|
break;
|
|
case kNetEqStreaming:
|
|
playoutMode = streaming;
|
|
break;
|
|
case kNetEqFax:
|
|
playoutMode = fax;
|
|
break;
|
|
case kNetEqOff:
|
|
playoutMode = off;
|
|
break;
|
|
}
|
|
if (audio_coding_->SetPlayoutMode(playoutMode) != 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
|
|
"SetNetEQPlayoutMode() failed to set playout mode");
|
|
return -1;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int32_t
|
|
Channel::GetNetEQPlayoutMode(NetEqModes& mode)
|
|
{
|
|
const AudioPlayoutMode playoutMode = audio_coding_->PlayoutMode();
|
|
switch (playoutMode)
|
|
{
|
|
case voice:
|
|
mode = kNetEqDefault;
|
|
break;
|
|
case streaming:
|
|
mode = kNetEqStreaming;
|
|
break;
|
|
case fax:
|
|
mode = kNetEqFax;
|
|
break;
|
|
case off:
|
|
mode = kNetEqOff;
|
|
}
|
|
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
|
|
VoEId(_instanceId,_channelId),
|
|
"Channel::GetNetEQPlayoutMode() => mode=%u", mode);
|
|
return 0;
|
|
}
|
|
|
|
int32_t
|
|
Channel::RegisterVoiceEngineObserver(VoiceEngineObserver& observer)
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::RegisterVoiceEngineObserver()");
|
|
CriticalSectionScoped cs(&_callbackCritSect);
|
|
|
|
if (_voiceEngineObserverPtr)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_INVALID_OPERATION, kTraceError,
|
|
"RegisterVoiceEngineObserver() observer already enabled");
|
|
return -1;
|
|
}
|
|
_voiceEngineObserverPtr = &observer;
|
|
return 0;
|
|
}
|
|
|
|
int32_t
|
|
Channel::DeRegisterVoiceEngineObserver()
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::DeRegisterVoiceEngineObserver()");
|
|
CriticalSectionScoped cs(&_callbackCritSect);
|
|
|
|
if (!_voiceEngineObserverPtr)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_INVALID_OPERATION, kTraceWarning,
|
|
"DeRegisterVoiceEngineObserver() observer already disabled");
|
|
return 0;
|
|
}
|
|
_voiceEngineObserverPtr = NULL;
|
|
return 0;
|
|
}
|
|
|
|
int32_t
|
|
Channel::GetSendCodec(CodecInst& codec)
|
|
{
|
|
return (audio_coding_->SendCodec(&codec));
|
|
}
|
|
|
|
int32_t
|
|
Channel::GetRecCodec(CodecInst& codec)
|
|
{
|
|
return (audio_coding_->ReceiveCodec(&codec));
|
|
}
|
|
|
|
int32_t
|
|
Channel::SetSendCodec(const CodecInst& codec)
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::SetSendCodec()");
|
|
|
|
if (audio_coding_->RegisterSendCodec(codec) != 0)
|
|
{
|
|
WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"SetSendCodec() failed to register codec to ACM");
|
|
return -1;
|
|
}
|
|
|
|
if (_rtpRtcpModule->RegisterSendPayload(codec) != 0)
|
|
{
|
|
_rtpRtcpModule->DeRegisterSendPayload(codec.pltype);
|
|
if (_rtpRtcpModule->RegisterSendPayload(codec) != 0)
|
|
{
|
|
WEBRTC_TRACE(
|
|
kTraceError, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"SetSendCodec() failed to register codec to"
|
|
" RTP/RTCP module");
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
if (_rtpRtcpModule->SetAudioPacketSize(codec.pacsize) != 0)
|
|
{
|
|
WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"SetSendCodec() failed to set audio packet size");
|
|
return -1;
|
|
}
|
|
|
|
bitrate_controller_->SetBitrateObserver(send_bitrate_observer_.get(),
|
|
codec.rate, 0, 0);
|
|
|
|
return 0;
|
|
}
|
|
|
|
void
|
|
Channel::OnNetworkChanged(const uint32_t bitrate_bps,
|
|
const uint8_t fraction_lost, // 0 - 255.
|
|
const uint32_t rtt) {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::OnNetworkChanged(bitrate_bps=%d, fration_lost=%d, rtt=%d)",
|
|
bitrate_bps, fraction_lost, rtt);
|
|
// Normalizes rate to 0 - 100.
|
|
if (audio_coding_->SetPacketLossRate(100 * fraction_lost / 255) != 0) {
|
|
_engineStatisticsPtr->SetLastError(VE_AUDIO_CODING_MODULE_ERROR,
|
|
kTraceError, "OnNetworkChanged() failed to set packet loss rate");
|
|
assert(false); // This should not happen.
|
|
}
|
|
}
|
|
|
|
int32_t
|
|
Channel::SetVADStatus(bool enableVAD, ACMVADMode mode, bool disableDTX)
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::SetVADStatus(mode=%d)", mode);
|
|
// To disable VAD, DTX must be disabled too
|
|
disableDTX = ((enableVAD == false) ? true : disableDTX);
|
|
if (audio_coding_->SetVAD(!disableDTX, enableVAD, mode) != 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
|
|
"SetVADStatus() failed to set VAD");
|
|
return -1;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int32_t
|
|
Channel::GetVADStatus(bool& enabledVAD, ACMVADMode& mode, bool& disabledDTX)
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::GetVADStatus");
|
|
if (audio_coding_->VAD(&disabledDTX, &enabledVAD, &mode) != 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
|
|
"GetVADStatus() failed to get VAD status");
|
|
return -1;
|
|
}
|
|
disabledDTX = !disabledDTX;
|
|
return 0;
|
|
}
|
|
|
|
int32_t
|
|
Channel::SetRecPayloadType(const CodecInst& codec)
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::SetRecPayloadType()");
|
|
|
|
if (channel_state_.Get().playing)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_ALREADY_PLAYING, kTraceError,
|
|
"SetRecPayloadType() unable to set PT while playing");
|
|
return -1;
|
|
}
|
|
if (channel_state_.Get().receiving)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_ALREADY_LISTENING, kTraceError,
|
|
"SetRecPayloadType() unable to set PT while listening");
|
|
return -1;
|
|
}
|
|
|
|
if (codec.pltype == -1)
|
|
{
|
|
// De-register the selected codec (RTP/RTCP module and ACM)
|
|
|
|
int8_t pltype(-1);
|
|
CodecInst rxCodec = codec;
|
|
|
|
// Get payload type for the given codec
|
|
rtp_payload_registry_->ReceivePayloadType(
|
|
rxCodec.plname,
|
|
rxCodec.plfreq,
|
|
rxCodec.channels,
|
|
(rxCodec.rate < 0) ? 0 : rxCodec.rate,
|
|
&pltype);
|
|
rxCodec.pltype = pltype;
|
|
|
|
if (rtp_receiver_->DeRegisterReceivePayload(pltype) != 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_RTP_RTCP_MODULE_ERROR,
|
|
kTraceError,
|
|
"SetRecPayloadType() RTP/RTCP-module deregistration "
|
|
"failed");
|
|
return -1;
|
|
}
|
|
if (audio_coding_->UnregisterReceiveCodec(rxCodec.pltype) != 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
|
|
"SetRecPayloadType() ACM deregistration failed - 1");
|
|
return -1;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
if (rtp_receiver_->RegisterReceivePayload(
|
|
codec.plname,
|
|
codec.pltype,
|
|
codec.plfreq,
|
|
codec.channels,
|
|
(codec.rate < 0) ? 0 : codec.rate) != 0)
|
|
{
|
|
// First attempt to register failed => de-register and try again
|
|
rtp_receiver_->DeRegisterReceivePayload(codec.pltype);
|
|
if (rtp_receiver_->RegisterReceivePayload(
|
|
codec.plname,
|
|
codec.pltype,
|
|
codec.plfreq,
|
|
codec.channels,
|
|
(codec.rate < 0) ? 0 : codec.rate) != 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_RTP_RTCP_MODULE_ERROR, kTraceError,
|
|
"SetRecPayloadType() RTP/RTCP-module registration failed");
|
|
return -1;
|
|
}
|
|
}
|
|
if (audio_coding_->RegisterReceiveCodec(codec) != 0)
|
|
{
|
|
audio_coding_->UnregisterReceiveCodec(codec.pltype);
|
|
if (audio_coding_->RegisterReceiveCodec(codec) != 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
|
|
"SetRecPayloadType() ACM registration failed - 1");
|
|
return -1;
|
|
}
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int32_t
|
|
Channel::GetRecPayloadType(CodecInst& codec)
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::GetRecPayloadType()");
|
|
int8_t payloadType(-1);
|
|
if (rtp_payload_registry_->ReceivePayloadType(
|
|
codec.plname,
|
|
codec.plfreq,
|
|
codec.channels,
|
|
(codec.rate < 0) ? 0 : codec.rate,
|
|
&payloadType) != 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_RTP_RTCP_MODULE_ERROR, kTraceWarning,
|
|
"GetRecPayloadType() failed to retrieve RX payload type");
|
|
return -1;
|
|
}
|
|
codec.pltype = payloadType;
|
|
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::GetRecPayloadType() => pltype=%u", codec.pltype);
|
|
return 0;
|
|
}
|
|
|
|
int32_t
|
|
Channel::SetSendCNPayloadType(int type, PayloadFrequencies frequency)
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::SetSendCNPayloadType()");
|
|
|
|
CodecInst codec;
|
|
int32_t samplingFreqHz(-1);
|
|
const int kMono = 1;
|
|
if (frequency == kFreq32000Hz)
|
|
samplingFreqHz = 32000;
|
|
else if (frequency == kFreq16000Hz)
|
|
samplingFreqHz = 16000;
|
|
|
|
if (audio_coding_->Codec("CN", &codec, samplingFreqHz, kMono) == -1)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
|
|
"SetSendCNPayloadType() failed to retrieve default CN codec "
|
|
"settings");
|
|
return -1;
|
|
}
|
|
|
|
// Modify the payload type (must be set to dynamic range)
|
|
codec.pltype = type;
|
|
|
|
if (audio_coding_->RegisterSendCodec(codec) != 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
|
|
"SetSendCNPayloadType() failed to register CN to ACM");
|
|
return -1;
|
|
}
|
|
|
|
if (_rtpRtcpModule->RegisterSendPayload(codec) != 0)
|
|
{
|
|
_rtpRtcpModule->DeRegisterSendPayload(codec.pltype);
|
|
if (_rtpRtcpModule->RegisterSendPayload(codec) != 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_RTP_RTCP_MODULE_ERROR, kTraceError,
|
|
"SetSendCNPayloadType() failed to register CN to RTP/RTCP "
|
|
"module");
|
|
return -1;
|
|
}
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int32_t Channel::RegisterExternalTransport(Transport& transport)
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
|
|
"Channel::RegisterExternalTransport()");
|
|
|
|
CriticalSectionScoped cs(&_callbackCritSect);
|
|
|
|
if (_externalTransport)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(VE_INVALID_OPERATION,
|
|
kTraceError,
|
|
"RegisterExternalTransport() external transport already enabled");
|
|
return -1;
|
|
}
|
|
_externalTransport = true;
|
|
_transportPtr = &transport;
|
|
return 0;
|
|
}
|
|
|
|
int32_t
|
|
Channel::DeRegisterExternalTransport()
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::DeRegisterExternalTransport()");
|
|
|
|
CriticalSectionScoped cs(&_callbackCritSect);
|
|
|
|
if (!_transportPtr)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_INVALID_OPERATION, kTraceWarning,
|
|
"DeRegisterExternalTransport() external transport already "
|
|
"disabled");
|
|
return 0;
|
|
}
|
|
_externalTransport = false;
|
|
_transportPtr = NULL;
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"DeRegisterExternalTransport() all transport is disabled");
|
|
return 0;
|
|
}
|
|
|
|
int32_t Channel::ReceivedRTPPacket(const int8_t* data, int32_t length,
|
|
const PacketTime& packet_time) {
|
|
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::ReceivedRTPPacket()");
|
|
|
|
// Store playout timestamp for the received RTP packet
|
|
UpdatePlayoutTimestamp(false);
|
|
|
|
// Dump the RTP packet to a file (if RTP dump is enabled).
|
|
if (_rtpDumpIn.DumpPacket((const uint8_t*)data,
|
|
(uint16_t)length) == -1) {
|
|
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
|
|
VoEId(_instanceId,_channelId),
|
|
"Channel::SendPacket() RTP dump to input file failed");
|
|
}
|
|
const uint8_t* received_packet = reinterpret_cast<const uint8_t*>(data);
|
|
RTPHeader header;
|
|
if (!rtp_header_parser_->Parse(received_packet, length, &header)) {
|
|
WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVoice, _channelId,
|
|
"Incoming packet: invalid RTP header");
|
|
return -1;
|
|
}
|
|
header.payload_type_frequency =
|
|
rtp_payload_registry_->GetPayloadTypeFrequency(header.payloadType);
|
|
if (header.payload_type_frequency < 0)
|
|
return -1;
|
|
bool in_order = IsPacketInOrder(header);
|
|
rtp_receive_statistics_->IncomingPacket(header, length,
|
|
IsPacketRetransmitted(header, in_order));
|
|
rtp_payload_registry_->SetIncomingPayloadType(header);
|
|
|
|
// Forward any packets to ViE bandwidth estimator, if enabled.
|
|
{
|
|
CriticalSectionScoped cs(&_callbackCritSect);
|
|
if (vie_network_) {
|
|
int64_t arrival_time_ms;
|
|
if (packet_time.timestamp != -1) {
|
|
arrival_time_ms = (packet_time.timestamp + 500) / 1000;
|
|
} else {
|
|
arrival_time_ms = TickTime::MillisecondTimestamp();
|
|
}
|
|
int payload_length = length - header.headerLength;
|
|
vie_network_->ReceivedBWEPacket(video_channel_, arrival_time_ms,
|
|
payload_length, header);
|
|
}
|
|
}
|
|
|
|
return ReceivePacket(received_packet, length, header, in_order) ? 0 : -1;
|
|
}
|
|
|
|
bool Channel::ReceivePacket(const uint8_t* packet,
|
|
int packet_length,
|
|
const RTPHeader& header,
|
|
bool in_order) {
|
|
if (rtp_payload_registry_->IsEncapsulated(header)) {
|
|
return HandleEncapsulation(packet, packet_length, header);
|
|
}
|
|
const uint8_t* payload = packet + header.headerLength;
|
|
int payload_length = packet_length - header.headerLength;
|
|
assert(payload_length >= 0);
|
|
PayloadUnion payload_specific;
|
|
if (!rtp_payload_registry_->GetPayloadSpecifics(header.payloadType,
|
|
&payload_specific)) {
|
|
return false;
|
|
}
|
|
return rtp_receiver_->IncomingRtpPacket(header, payload, payload_length,
|
|
payload_specific, in_order);
|
|
}
|
|
|
|
bool Channel::HandleEncapsulation(const uint8_t* packet,
|
|
int packet_length,
|
|
const RTPHeader& header) {
|
|
if (!rtp_payload_registry_->IsRtx(header))
|
|
return false;
|
|
|
|
// Remove the RTX header and parse the original RTP header.
|
|
if (packet_length < header.headerLength)
|
|
return false;
|
|
if (packet_length > kVoiceEngineMaxIpPacketSizeBytes)
|
|
return false;
|
|
if (restored_packet_in_use_) {
|
|
WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVoice, _channelId,
|
|
"Multiple RTX headers detected, dropping packet");
|
|
return false;
|
|
}
|
|
uint8_t* restored_packet_ptr = restored_packet_;
|
|
if (!rtp_payload_registry_->RestoreOriginalPacket(
|
|
&restored_packet_ptr, packet, &packet_length, rtp_receiver_->SSRC(),
|
|
header)) {
|
|
WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVoice, _channelId,
|
|
"Incoming RTX packet: invalid RTP header");
|
|
return false;
|
|
}
|
|
restored_packet_in_use_ = true;
|
|
bool ret = OnRecoveredPacket(restored_packet_ptr, packet_length);
|
|
restored_packet_in_use_ = false;
|
|
return ret;
|
|
}
|
|
|
|
bool Channel::IsPacketInOrder(const RTPHeader& header) const {
|
|
StreamStatistician* statistician =
|
|
rtp_receive_statistics_->GetStatistician(header.ssrc);
|
|
if (!statistician)
|
|
return false;
|
|
return statistician->IsPacketInOrder(header.sequenceNumber);
|
|
}
|
|
|
|
bool Channel::IsPacketRetransmitted(const RTPHeader& header,
|
|
bool in_order) const {
|
|
// Retransmissions are handled separately if RTX is enabled.
|
|
if (rtp_payload_registry_->RtxEnabled())
|
|
return false;
|
|
StreamStatistician* statistician =
|
|
rtp_receive_statistics_->GetStatistician(header.ssrc);
|
|
if (!statistician)
|
|
return false;
|
|
// Check if this is a retransmission.
|
|
uint16_t min_rtt = 0;
|
|
_rtpRtcpModule->RTT(rtp_receiver_->SSRC(), NULL, NULL, &min_rtt, NULL);
|
|
return !in_order &&
|
|
statistician->IsRetransmitOfOldPacket(header, min_rtt);
|
|
}
|
|
|
|
int32_t Channel::ReceivedRTCPPacket(const int8_t* data, int32_t length) {
|
|
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::ReceivedRTCPPacket()");
|
|
// Store playout timestamp for the received RTCP packet
|
|
UpdatePlayoutTimestamp(true);
|
|
|
|
// Dump the RTCP packet to a file (if RTP dump is enabled).
|
|
if (_rtpDumpIn.DumpPacket((const uint8_t*)data,
|
|
(uint16_t)length) == -1) {
|
|
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
|
|
VoEId(_instanceId,_channelId),
|
|
"Channel::SendPacket() RTCP dump to input file failed");
|
|
}
|
|
|
|
// Deliver RTCP packet to RTP/RTCP module for parsing
|
|
if (_rtpRtcpModule->IncomingRtcpPacket((const uint8_t*)data,
|
|
(uint16_t)length) == -1) {
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceWarning,
|
|
"Channel::IncomingRTPPacket() RTCP packet is invalid");
|
|
}
|
|
|
|
ntp_estimator_->UpdateRtcpTimestamp(rtp_receiver_->SSRC(),
|
|
_rtpRtcpModule.get());
|
|
return 0;
|
|
}
|
|
|
|
int Channel::StartPlayingFileLocally(const char* fileName,
|
|
bool loop,
|
|
FileFormats format,
|
|
int startPosition,
|
|
float volumeScaling,
|
|
int stopPosition,
|
|
const CodecInst* codecInst)
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::StartPlayingFileLocally(fileNameUTF8[]=%s, loop=%d,"
|
|
" format=%d, volumeScaling=%5.3f, startPosition=%d, "
|
|
"stopPosition=%d)", fileName, loop, format, volumeScaling,
|
|
startPosition, stopPosition);
|
|
|
|
if (channel_state_.Get().output_file_playing)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_ALREADY_PLAYING, kTraceError,
|
|
"StartPlayingFileLocally() is already playing");
|
|
return -1;
|
|
}
|
|
|
|
{
|
|
CriticalSectionScoped cs(&_fileCritSect);
|
|
|
|
if (_outputFilePlayerPtr)
|
|
{
|
|
_outputFilePlayerPtr->RegisterModuleFileCallback(NULL);
|
|
FilePlayer::DestroyFilePlayer(_outputFilePlayerPtr);
|
|
_outputFilePlayerPtr = NULL;
|
|
}
|
|
|
|
_outputFilePlayerPtr = FilePlayer::CreateFilePlayer(
|
|
_outputFilePlayerId, (const FileFormats)format);
|
|
|
|
if (_outputFilePlayerPtr == NULL)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_INVALID_ARGUMENT, kTraceError,
|
|
"StartPlayingFileLocally() filePlayer format is not correct");
|
|
return -1;
|
|
}
|
|
|
|
const uint32_t notificationTime(0);
|
|
|
|
if (_outputFilePlayerPtr->StartPlayingFile(
|
|
fileName,
|
|
loop,
|
|
startPosition,
|
|
volumeScaling,
|
|
notificationTime,
|
|
stopPosition,
|
|
(const CodecInst*)codecInst) != 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_BAD_FILE, kTraceError,
|
|
"StartPlayingFile() failed to start file playout");
|
|
_outputFilePlayerPtr->StopPlayingFile();
|
|
FilePlayer::DestroyFilePlayer(_outputFilePlayerPtr);
|
|
_outputFilePlayerPtr = NULL;
|
|
return -1;
|
|
}
|
|
_outputFilePlayerPtr->RegisterModuleFileCallback(this);
|
|
channel_state_.SetOutputFilePlaying(true);
|
|
}
|
|
|
|
if (RegisterFilePlayingToMixer() != 0)
|
|
return -1;
|
|
|
|
return 0;
|
|
}
|
|
|
|
int Channel::StartPlayingFileLocally(InStream* stream,
|
|
FileFormats format,
|
|
int startPosition,
|
|
float volumeScaling,
|
|
int stopPosition,
|
|
const CodecInst* codecInst)
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::StartPlayingFileLocally(format=%d,"
|
|
" volumeScaling=%5.3f, startPosition=%d, stopPosition=%d)",
|
|
format, volumeScaling, startPosition, stopPosition);
|
|
|
|
if(stream == NULL)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_BAD_FILE, kTraceError,
|
|
"StartPlayingFileLocally() NULL as input stream");
|
|
return -1;
|
|
}
|
|
|
|
|
|
if (channel_state_.Get().output_file_playing)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_ALREADY_PLAYING, kTraceError,
|
|
"StartPlayingFileLocally() is already playing");
|
|
return -1;
|
|
}
|
|
|
|
{
|
|
CriticalSectionScoped cs(&_fileCritSect);
|
|
|
|
// Destroy the old instance
|
|
if (_outputFilePlayerPtr)
|
|
{
|
|
_outputFilePlayerPtr->RegisterModuleFileCallback(NULL);
|
|
FilePlayer::DestroyFilePlayer(_outputFilePlayerPtr);
|
|
_outputFilePlayerPtr = NULL;
|
|
}
|
|
|
|
// Create the instance
|
|
_outputFilePlayerPtr = FilePlayer::CreateFilePlayer(
|
|
_outputFilePlayerId,
|
|
(const FileFormats)format);
|
|
|
|
if (_outputFilePlayerPtr == NULL)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_INVALID_ARGUMENT, kTraceError,
|
|
"StartPlayingFileLocally() filePlayer format isnot correct");
|
|
return -1;
|
|
}
|
|
|
|
const uint32_t notificationTime(0);
|
|
|
|
if (_outputFilePlayerPtr->StartPlayingFile(*stream, startPosition,
|
|
volumeScaling,
|
|
notificationTime,
|
|
stopPosition, codecInst) != 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(VE_BAD_FILE, kTraceError,
|
|
"StartPlayingFile() failed to "
|
|
"start file playout");
|
|
_outputFilePlayerPtr->StopPlayingFile();
|
|
FilePlayer::DestroyFilePlayer(_outputFilePlayerPtr);
|
|
_outputFilePlayerPtr = NULL;
|
|
return -1;
|
|
}
|
|
_outputFilePlayerPtr->RegisterModuleFileCallback(this);
|
|
channel_state_.SetOutputFilePlaying(true);
|
|
}
|
|
|
|
if (RegisterFilePlayingToMixer() != 0)
|
|
return -1;
|
|
|
|
return 0;
|
|
}
|
|
|
|
int Channel::StopPlayingFileLocally()
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::StopPlayingFileLocally()");
|
|
|
|
if (!channel_state_.Get().output_file_playing)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_INVALID_OPERATION, kTraceWarning,
|
|
"StopPlayingFileLocally() isnot playing");
|
|
return 0;
|
|
}
|
|
|
|
{
|
|
CriticalSectionScoped cs(&_fileCritSect);
|
|
|
|
if (_outputFilePlayerPtr->StopPlayingFile() != 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_STOP_RECORDING_FAILED, kTraceError,
|
|
"StopPlayingFile() could not stop playing");
|
|
return -1;
|
|
}
|
|
_outputFilePlayerPtr->RegisterModuleFileCallback(NULL);
|
|
FilePlayer::DestroyFilePlayer(_outputFilePlayerPtr);
|
|
_outputFilePlayerPtr = NULL;
|
|
channel_state_.SetOutputFilePlaying(false);
|
|
}
|
|
// _fileCritSect cannot be taken while calling
|
|
// SetAnonymousMixibilityStatus. Refer to comments in
|
|
// StartPlayingFileLocally(const char* ...) for more details.
|
|
if (_outputMixerPtr->SetAnonymousMixabilityStatus(*this, false) != 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_AUDIO_CONF_MIX_MODULE_ERROR, kTraceError,
|
|
"StopPlayingFile() failed to stop participant from playing as"
|
|
"file in the mixer");
|
|
return -1;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
int Channel::IsPlayingFileLocally() const
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::IsPlayingFileLocally()");
|
|
|
|
return channel_state_.Get().output_file_playing;
|
|
}
|
|
|
|
int Channel::RegisterFilePlayingToMixer()
|
|
{
|
|
// Return success for not registering for file playing to mixer if:
|
|
// 1. playing file before playout is started on that channel.
|
|
// 2. starting playout without file playing on that channel.
|
|
if (!channel_state_.Get().playing ||
|
|
!channel_state_.Get().output_file_playing)
|
|
{
|
|
return 0;
|
|
}
|
|
|
|
// |_fileCritSect| cannot be taken while calling
|
|
// SetAnonymousMixabilityStatus() since as soon as the participant is added
|
|
// frames can be pulled by the mixer. Since the frames are generated from
|
|
// the file, _fileCritSect will be taken. This would result in a deadlock.
|
|
if (_outputMixerPtr->SetAnonymousMixabilityStatus(*this, true) != 0)
|
|
{
|
|
channel_state_.SetOutputFilePlaying(false);
|
|
CriticalSectionScoped cs(&_fileCritSect);
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_AUDIO_CONF_MIX_MODULE_ERROR, kTraceError,
|
|
"StartPlayingFile() failed to add participant as file to mixer");
|
|
_outputFilePlayerPtr->StopPlayingFile();
|
|
FilePlayer::DestroyFilePlayer(_outputFilePlayerPtr);
|
|
_outputFilePlayerPtr = NULL;
|
|
return -1;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
int Channel::StartPlayingFileAsMicrophone(const char* fileName,
|
|
bool loop,
|
|
FileFormats format,
|
|
int startPosition,
|
|
float volumeScaling,
|
|
int stopPosition,
|
|
const CodecInst* codecInst)
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::StartPlayingFileAsMicrophone(fileNameUTF8[]=%s, "
|
|
"loop=%d, format=%d, volumeScaling=%5.3f, startPosition=%d, "
|
|
"stopPosition=%d)", fileName, loop, format, volumeScaling,
|
|
startPosition, stopPosition);
|
|
|
|
CriticalSectionScoped cs(&_fileCritSect);
|
|
|
|
if (channel_state_.Get().input_file_playing)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_ALREADY_PLAYING, kTraceWarning,
|
|
"StartPlayingFileAsMicrophone() filePlayer is playing");
|
|
return 0;
|
|
}
|
|
|
|
// Destroy the old instance
|
|
if (_inputFilePlayerPtr)
|
|
{
|
|
_inputFilePlayerPtr->RegisterModuleFileCallback(NULL);
|
|
FilePlayer::DestroyFilePlayer(_inputFilePlayerPtr);
|
|
_inputFilePlayerPtr = NULL;
|
|
}
|
|
|
|
// Create the instance
|
|
_inputFilePlayerPtr = FilePlayer::CreateFilePlayer(
|
|
_inputFilePlayerId, (const FileFormats)format);
|
|
|
|
if (_inputFilePlayerPtr == NULL)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_INVALID_ARGUMENT, kTraceError,
|
|
"StartPlayingFileAsMicrophone() filePlayer format isnot correct");
|
|
return -1;
|
|
}
|
|
|
|
const uint32_t notificationTime(0);
|
|
|
|
if (_inputFilePlayerPtr->StartPlayingFile(
|
|
fileName,
|
|
loop,
|
|
startPosition,
|
|
volumeScaling,
|
|
notificationTime,
|
|
stopPosition,
|
|
(const CodecInst*)codecInst) != 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_BAD_FILE, kTraceError,
|
|
"StartPlayingFile() failed to start file playout");
|
|
_inputFilePlayerPtr->StopPlayingFile();
|
|
FilePlayer::DestroyFilePlayer(_inputFilePlayerPtr);
|
|
_inputFilePlayerPtr = NULL;
|
|
return -1;
|
|
}
|
|
_inputFilePlayerPtr->RegisterModuleFileCallback(this);
|
|
channel_state_.SetInputFilePlaying(true);
|
|
|
|
return 0;
|
|
}
|
|
|
|
int Channel::StartPlayingFileAsMicrophone(InStream* stream,
|
|
FileFormats format,
|
|
int startPosition,
|
|
float volumeScaling,
|
|
int stopPosition,
|
|
const CodecInst* codecInst)
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::StartPlayingFileAsMicrophone(format=%d, "
|
|
"volumeScaling=%5.3f, startPosition=%d, stopPosition=%d)",
|
|
format, volumeScaling, startPosition, stopPosition);
|
|
|
|
if(stream == NULL)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_BAD_FILE, kTraceError,
|
|
"StartPlayingFileAsMicrophone NULL as input stream");
|
|
return -1;
|
|
}
|
|
|
|
CriticalSectionScoped cs(&_fileCritSect);
|
|
|
|
if (channel_state_.Get().input_file_playing)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_ALREADY_PLAYING, kTraceWarning,
|
|
"StartPlayingFileAsMicrophone() is playing");
|
|
return 0;
|
|
}
|
|
|
|
// Destroy the old instance
|
|
if (_inputFilePlayerPtr)
|
|
{
|
|
_inputFilePlayerPtr->RegisterModuleFileCallback(NULL);
|
|
FilePlayer::DestroyFilePlayer(_inputFilePlayerPtr);
|
|
_inputFilePlayerPtr = NULL;
|
|
}
|
|
|
|
// Create the instance
|
|
_inputFilePlayerPtr = FilePlayer::CreateFilePlayer(
|
|
_inputFilePlayerId, (const FileFormats)format);
|
|
|
|
if (_inputFilePlayerPtr == NULL)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_INVALID_ARGUMENT, kTraceError,
|
|
"StartPlayingInputFile() filePlayer format isnot correct");
|
|
return -1;
|
|
}
|
|
|
|
const uint32_t notificationTime(0);
|
|
|
|
if (_inputFilePlayerPtr->StartPlayingFile(*stream, startPosition,
|
|
volumeScaling, notificationTime,
|
|
stopPosition, codecInst) != 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(VE_BAD_FILE, kTraceError,
|
|
"StartPlayingFile() failed to start "
|
|
"file playout");
|
|
_inputFilePlayerPtr->StopPlayingFile();
|
|
FilePlayer::DestroyFilePlayer(_inputFilePlayerPtr);
|
|
_inputFilePlayerPtr = NULL;
|
|
return -1;
|
|
}
|
|
|
|
_inputFilePlayerPtr->RegisterModuleFileCallback(this);
|
|
channel_state_.SetInputFilePlaying(true);
|
|
|
|
return 0;
|
|
}
|
|
|
|
int Channel::StopPlayingFileAsMicrophone()
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::StopPlayingFileAsMicrophone()");
|
|
|
|
CriticalSectionScoped cs(&_fileCritSect);
|
|
|
|
if (!channel_state_.Get().input_file_playing)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_INVALID_OPERATION, kTraceWarning,
|
|
"StopPlayingFileAsMicrophone() isnot playing");
|
|
return 0;
|
|
}
|
|
|
|
if (_inputFilePlayerPtr->StopPlayingFile() != 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_STOP_RECORDING_FAILED, kTraceError,
|
|
"StopPlayingFile() could not stop playing");
|
|
return -1;
|
|
}
|
|
_inputFilePlayerPtr->RegisterModuleFileCallback(NULL);
|
|
FilePlayer::DestroyFilePlayer(_inputFilePlayerPtr);
|
|
_inputFilePlayerPtr = NULL;
|
|
channel_state_.SetInputFilePlaying(false);
|
|
|
|
return 0;
|
|
}
|
|
|
|
int Channel::IsPlayingFileAsMicrophone() const
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::IsPlayingFileAsMicrophone()");
|
|
return channel_state_.Get().input_file_playing;
|
|
}
|
|
|
|
int Channel::StartRecordingPlayout(const char* fileName,
|
|
const CodecInst* codecInst)
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::StartRecordingPlayout(fileName=%s)", fileName);
|
|
|
|
if (_outputFileRecording)
|
|
{
|
|
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,-1),
|
|
"StartRecordingPlayout() is already recording");
|
|
return 0;
|
|
}
|
|
|
|
FileFormats format;
|
|
const uint32_t notificationTime(0); // Not supported in VoE
|
|
CodecInst dummyCodec={100,"L16",16000,320,1,320000};
|
|
|
|
if ((codecInst != NULL) &&
|
|
((codecInst->channels < 1) || (codecInst->channels > 2)))
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_BAD_ARGUMENT, kTraceError,
|
|
"StartRecordingPlayout() invalid compression");
|
|
return(-1);
|
|
}
|
|
if(codecInst == NULL)
|
|
{
|
|
format = kFileFormatPcm16kHzFile;
|
|
codecInst=&dummyCodec;
|
|
}
|
|
else if((STR_CASE_CMP(codecInst->plname,"L16") == 0) ||
|
|
(STR_CASE_CMP(codecInst->plname,"PCMU") == 0) ||
|
|
(STR_CASE_CMP(codecInst->plname,"PCMA") == 0))
|
|
{
|
|
format = kFileFormatWavFile;
|
|
}
|
|
else
|
|
{
|
|
format = kFileFormatCompressedFile;
|
|
}
|
|
|
|
CriticalSectionScoped cs(&_fileCritSect);
|
|
|
|
// Destroy the old instance
|
|
if (_outputFileRecorderPtr)
|
|
{
|
|
_outputFileRecorderPtr->RegisterModuleFileCallback(NULL);
|
|
FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
|
|
_outputFileRecorderPtr = NULL;
|
|
}
|
|
|
|
_outputFileRecorderPtr = FileRecorder::CreateFileRecorder(
|
|
_outputFileRecorderId, (const FileFormats)format);
|
|
if (_outputFileRecorderPtr == NULL)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_INVALID_ARGUMENT, kTraceError,
|
|
"StartRecordingPlayout() fileRecorder format isnot correct");
|
|
return -1;
|
|
}
|
|
|
|
if (_outputFileRecorderPtr->StartRecordingAudioFile(
|
|
fileName, (const CodecInst&)*codecInst, notificationTime) != 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_BAD_FILE, kTraceError,
|
|
"StartRecordingAudioFile() failed to start file recording");
|
|
_outputFileRecorderPtr->StopRecording();
|
|
FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
|
|
_outputFileRecorderPtr = NULL;
|
|
return -1;
|
|
}
|
|
_outputFileRecorderPtr->RegisterModuleFileCallback(this);
|
|
_outputFileRecording = true;
|
|
|
|
return 0;
|
|
}
|
|
|
|
int Channel::StartRecordingPlayout(OutStream* stream,
|
|
const CodecInst* codecInst)
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::StartRecordingPlayout()");
|
|
|
|
if (_outputFileRecording)
|
|
{
|
|
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,-1),
|
|
"StartRecordingPlayout() is already recording");
|
|
return 0;
|
|
}
|
|
|
|
FileFormats format;
|
|
const uint32_t notificationTime(0); // Not supported in VoE
|
|
CodecInst dummyCodec={100,"L16",16000,320,1,320000};
|
|
|
|
if (codecInst != NULL && codecInst->channels != 1)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_BAD_ARGUMENT, kTraceError,
|
|
"StartRecordingPlayout() invalid compression");
|
|
return(-1);
|
|
}
|
|
if(codecInst == NULL)
|
|
{
|
|
format = kFileFormatPcm16kHzFile;
|
|
codecInst=&dummyCodec;
|
|
}
|
|
else if((STR_CASE_CMP(codecInst->plname,"L16") == 0) ||
|
|
(STR_CASE_CMP(codecInst->plname,"PCMU") == 0) ||
|
|
(STR_CASE_CMP(codecInst->plname,"PCMA") == 0))
|
|
{
|
|
format = kFileFormatWavFile;
|
|
}
|
|
else
|
|
{
|
|
format = kFileFormatCompressedFile;
|
|
}
|
|
|
|
CriticalSectionScoped cs(&_fileCritSect);
|
|
|
|
// Destroy the old instance
|
|
if (_outputFileRecorderPtr)
|
|
{
|
|
_outputFileRecorderPtr->RegisterModuleFileCallback(NULL);
|
|
FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
|
|
_outputFileRecorderPtr = NULL;
|
|
}
|
|
|
|
_outputFileRecorderPtr = FileRecorder::CreateFileRecorder(
|
|
_outputFileRecorderId, (const FileFormats)format);
|
|
if (_outputFileRecorderPtr == NULL)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_INVALID_ARGUMENT, kTraceError,
|
|
"StartRecordingPlayout() fileRecorder format isnot correct");
|
|
return -1;
|
|
}
|
|
|
|
if (_outputFileRecorderPtr->StartRecordingAudioFile(*stream, *codecInst,
|
|
notificationTime) != 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(VE_BAD_FILE, kTraceError,
|
|
"StartRecordingPlayout() failed to "
|
|
"start file recording");
|
|
_outputFileRecorderPtr->StopRecording();
|
|
FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
|
|
_outputFileRecorderPtr = NULL;
|
|
return -1;
|
|
}
|
|
|
|
_outputFileRecorderPtr->RegisterModuleFileCallback(this);
|
|
_outputFileRecording = true;
|
|
|
|
return 0;
|
|
}
|
|
|
|
int Channel::StopRecordingPlayout()
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
|
|
"Channel::StopRecordingPlayout()");
|
|
|
|
if (!_outputFileRecording)
|
|
{
|
|
WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
|
|
"StopRecordingPlayout() isnot recording");
|
|
return -1;
|
|
}
|
|
|
|
|
|
CriticalSectionScoped cs(&_fileCritSect);
|
|
|
|
if (_outputFileRecorderPtr->StopRecording() != 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_STOP_RECORDING_FAILED, kTraceError,
|
|
"StopRecording() could not stop recording");
|
|
return(-1);
|
|
}
|
|
_outputFileRecorderPtr->RegisterModuleFileCallback(NULL);
|
|
FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
|
|
_outputFileRecorderPtr = NULL;
|
|
_outputFileRecording = false;
|
|
|
|
return 0;
|
|
}
|
|
|
|
void
|
|
Channel::SetMixWithMicStatus(bool mix)
|
|
{
|
|
CriticalSectionScoped cs(&_fileCritSect);
|
|
_mixFileWithMicrophone=mix;
|
|
}
|
|
|
|
int
|
|
Channel::GetSpeechOutputLevel(uint32_t& level) const
|
|
{
|
|
int8_t currentLevel = _outputAudioLevel.Level();
|
|
level = static_cast<int32_t> (currentLevel);
|
|
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
|
|
VoEId(_instanceId,_channelId),
|
|
"GetSpeechOutputLevel() => level=%u", level);
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
Channel::GetSpeechOutputLevelFullRange(uint32_t& level) const
|
|
{
|
|
int16_t currentLevel = _outputAudioLevel.LevelFullRange();
|
|
level = static_cast<int32_t> (currentLevel);
|
|
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
|
|
VoEId(_instanceId,_channelId),
|
|
"GetSpeechOutputLevelFullRange() => level=%u", level);
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
Channel::SetMute(bool enable)
|
|
{
|
|
CriticalSectionScoped cs(&volume_settings_critsect_);
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::SetMute(enable=%d)", enable);
|
|
_mute = enable;
|
|
return 0;
|
|
}
|
|
|
|
bool
|
|
Channel::Mute() const
|
|
{
|
|
CriticalSectionScoped cs(&volume_settings_critsect_);
|
|
return _mute;
|
|
}
|
|
|
|
int
|
|
Channel::SetOutputVolumePan(float left, float right)
|
|
{
|
|
CriticalSectionScoped cs(&volume_settings_critsect_);
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::SetOutputVolumePan()");
|
|
_panLeft = left;
|
|
_panRight = right;
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
Channel::GetOutputVolumePan(float& left, float& right) const
|
|
{
|
|
CriticalSectionScoped cs(&volume_settings_critsect_);
|
|
left = _panLeft;
|
|
right = _panRight;
|
|
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
|
|
VoEId(_instanceId,_channelId),
|
|
"GetOutputVolumePan() => left=%3.2f, right=%3.2f", left, right);
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
Channel::SetChannelOutputVolumeScaling(float scaling)
|
|
{
|
|
CriticalSectionScoped cs(&volume_settings_critsect_);
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::SetChannelOutputVolumeScaling()");
|
|
_outputGain = scaling;
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
Channel::GetChannelOutputVolumeScaling(float& scaling) const
|
|
{
|
|
CriticalSectionScoped cs(&volume_settings_critsect_);
|
|
scaling = _outputGain;
|
|
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
|
|
VoEId(_instanceId,_channelId),
|
|
"GetChannelOutputVolumeScaling() => scaling=%3.2f", scaling);
|
|
return 0;
|
|
}
|
|
|
|
int Channel::SendTelephoneEventOutband(unsigned char eventCode,
|
|
int lengthMs, int attenuationDb,
|
|
bool playDtmfEvent)
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
|
|
"Channel::SendTelephoneEventOutband(..., playDtmfEvent=%d)",
|
|
playDtmfEvent);
|
|
|
|
_playOutbandDtmfEvent = playDtmfEvent;
|
|
|
|
if (_rtpRtcpModule->SendTelephoneEventOutband(eventCode, lengthMs,
|
|
attenuationDb) != 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_SEND_DTMF_FAILED,
|
|
kTraceWarning,
|
|
"SendTelephoneEventOutband() failed to send event");
|
|
return -1;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int Channel::SendTelephoneEventInband(unsigned char eventCode,
|
|
int lengthMs,
|
|
int attenuationDb,
|
|
bool playDtmfEvent)
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
|
|
"Channel::SendTelephoneEventInband(..., playDtmfEvent=%d)",
|
|
playDtmfEvent);
|
|
|
|
_playInbandDtmfEvent = playDtmfEvent;
|
|
_inbandDtmfQueue.AddDtmf(eventCode, lengthMs, attenuationDb);
|
|
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
Channel::SetDtmfPlayoutStatus(bool enable)
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::SetDtmfPlayoutStatus()");
|
|
if (audio_coding_->SetDtmfPlayoutStatus(enable) != 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_AUDIO_CODING_MODULE_ERROR, kTraceWarning,
|
|
"SetDtmfPlayoutStatus() failed to set Dtmf playout");
|
|
return -1;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
bool
|
|
Channel::DtmfPlayoutStatus() const
|
|
{
|
|
return audio_coding_->DtmfPlayoutStatus();
|
|
}
|
|
|
|
int
|
|
Channel::SetSendTelephoneEventPayloadType(unsigned char type)
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::SetSendTelephoneEventPayloadType()");
|
|
if (type > 127)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_INVALID_ARGUMENT, kTraceError,
|
|
"SetSendTelephoneEventPayloadType() invalid type");
|
|
return -1;
|
|
}
|
|
CodecInst codec = {};
|
|
codec.plfreq = 8000;
|
|
codec.pltype = type;
|
|
memcpy(codec.plname, "telephone-event", 16);
|
|
if (_rtpRtcpModule->RegisterSendPayload(codec) != 0)
|
|
{
|
|
_rtpRtcpModule->DeRegisterSendPayload(codec.pltype);
|
|
if (_rtpRtcpModule->RegisterSendPayload(codec) != 0) {
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_RTP_RTCP_MODULE_ERROR, kTraceError,
|
|
"SetSendTelephoneEventPayloadType() failed to register send"
|
|
"payload type");
|
|
return -1;
|
|
}
|
|
}
|
|
_sendTelephoneEventPayloadType = type;
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
Channel::GetSendTelephoneEventPayloadType(unsigned char& type)
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::GetSendTelephoneEventPayloadType()");
|
|
type = _sendTelephoneEventPayloadType;
|
|
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
|
|
VoEId(_instanceId,_channelId),
|
|
"GetSendTelephoneEventPayloadType() => type=%u", type);
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
Channel::UpdateRxVadDetection(AudioFrame& audioFrame)
|
|
{
|
|
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::UpdateRxVadDetection()");
|
|
|
|
int vadDecision = 1;
|
|
|
|
vadDecision = (audioFrame.vad_activity_ == AudioFrame::kVadActive)? 1 : 0;
|
|
|
|
if ((vadDecision != _oldVadDecision) && _rxVadObserverPtr)
|
|
{
|
|
OnRxVadDetected(vadDecision);
|
|
_oldVadDecision = vadDecision;
|
|
}
|
|
|
|
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::UpdateRxVadDetection() => vadDecision=%d",
|
|
vadDecision);
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
Channel::RegisterRxVadObserver(VoERxVadCallback &observer)
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::RegisterRxVadObserver()");
|
|
CriticalSectionScoped cs(&_callbackCritSect);
|
|
|
|
if (_rxVadObserverPtr)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_INVALID_OPERATION, kTraceError,
|
|
"RegisterRxVadObserver() observer already enabled");
|
|
return -1;
|
|
}
|
|
_rxVadObserverPtr = &observer;
|
|
_RxVadDetection = true;
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
Channel::DeRegisterRxVadObserver()
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::DeRegisterRxVadObserver()");
|
|
CriticalSectionScoped cs(&_callbackCritSect);
|
|
|
|
if (!_rxVadObserverPtr)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_INVALID_OPERATION, kTraceWarning,
|
|
"DeRegisterRxVadObserver() observer already disabled");
|
|
return 0;
|
|
}
|
|
_rxVadObserverPtr = NULL;
|
|
_RxVadDetection = false;
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
Channel::VoiceActivityIndicator(int &activity)
|
|
{
|
|
activity = _sendFrameType;
|
|
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::VoiceActivityIndicator(indicator=%d)", activity);
|
|
return 0;
|
|
}
|
|
|
|
#ifdef WEBRTC_VOICE_ENGINE_AGC
|
|
|
|
int
|
|
Channel::SetRxAgcStatus(bool enable, AgcModes mode)
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::SetRxAgcStatus(enable=%d, mode=%d)",
|
|
(int)enable, (int)mode);
|
|
|
|
GainControl::Mode agcMode = kDefaultRxAgcMode;
|
|
switch (mode)
|
|
{
|
|
case kAgcDefault:
|
|
break;
|
|
case kAgcUnchanged:
|
|
agcMode = rx_audioproc_->gain_control()->mode();
|
|
break;
|
|
case kAgcFixedDigital:
|
|
agcMode = GainControl::kFixedDigital;
|
|
break;
|
|
case kAgcAdaptiveDigital:
|
|
agcMode =GainControl::kAdaptiveDigital;
|
|
break;
|
|
default:
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_INVALID_ARGUMENT, kTraceError,
|
|
"SetRxAgcStatus() invalid Agc mode");
|
|
return -1;
|
|
}
|
|
|
|
if (rx_audioproc_->gain_control()->set_mode(agcMode) != 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_APM_ERROR, kTraceError,
|
|
"SetRxAgcStatus() failed to set Agc mode");
|
|
return -1;
|
|
}
|
|
if (rx_audioproc_->gain_control()->Enable(enable) != 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_APM_ERROR, kTraceError,
|
|
"SetRxAgcStatus() failed to set Agc state");
|
|
return -1;
|
|
}
|
|
|
|
_rxAgcIsEnabled = enable;
|
|
channel_state_.SetRxApmIsEnabled(_rxAgcIsEnabled || _rxNsIsEnabled);
|
|
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
Channel::GetRxAgcStatus(bool& enabled, AgcModes& mode)
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::GetRxAgcStatus(enable=?, mode=?)");
|
|
|
|
bool enable = rx_audioproc_->gain_control()->is_enabled();
|
|
GainControl::Mode agcMode =
|
|
rx_audioproc_->gain_control()->mode();
|
|
|
|
enabled = enable;
|
|
|
|
switch (agcMode)
|
|
{
|
|
case GainControl::kFixedDigital:
|
|
mode = kAgcFixedDigital;
|
|
break;
|
|
case GainControl::kAdaptiveDigital:
|
|
mode = kAgcAdaptiveDigital;
|
|
break;
|
|
default:
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_APM_ERROR, kTraceError,
|
|
"GetRxAgcStatus() invalid Agc mode");
|
|
return -1;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
Channel::SetRxAgcConfig(AgcConfig config)
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::SetRxAgcConfig()");
|
|
|
|
if (rx_audioproc_->gain_control()->set_target_level_dbfs(
|
|
config.targetLeveldBOv) != 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_APM_ERROR, kTraceError,
|
|
"SetRxAgcConfig() failed to set target peak |level|"
|
|
"(or envelope) of the Agc");
|
|
return -1;
|
|
}
|
|
if (rx_audioproc_->gain_control()->set_compression_gain_db(
|
|
config.digitalCompressionGaindB) != 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_APM_ERROR, kTraceError,
|
|
"SetRxAgcConfig() failed to set the range in |gain| the"
|
|
" digital compression stage may apply");
|
|
return -1;
|
|
}
|
|
if (rx_audioproc_->gain_control()->enable_limiter(
|
|
config.limiterEnable) != 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_APM_ERROR, kTraceError,
|
|
"SetRxAgcConfig() failed to set hard limiter to the signal");
|
|
return -1;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
Channel::GetRxAgcConfig(AgcConfig& config)
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::GetRxAgcConfig(config=%?)");
|
|
|
|
config.targetLeveldBOv =
|
|
rx_audioproc_->gain_control()->target_level_dbfs();
|
|
config.digitalCompressionGaindB =
|
|
rx_audioproc_->gain_control()->compression_gain_db();
|
|
config.limiterEnable =
|
|
rx_audioproc_->gain_control()->is_limiter_enabled();
|
|
|
|
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
|
|
VoEId(_instanceId,_channelId), "GetRxAgcConfig() => "
|
|
"targetLeveldBOv=%u, digitalCompressionGaindB=%u,"
|
|
" limiterEnable=%d",
|
|
config.targetLeveldBOv,
|
|
config.digitalCompressionGaindB,
|
|
config.limiterEnable);
|
|
|
|
return 0;
|
|
}
|
|
|
|
#endif // #ifdef WEBRTC_VOICE_ENGINE_AGC
|
|
|
|
#ifdef WEBRTC_VOICE_ENGINE_NR
|
|
|
|
int
|
|
Channel::SetRxNsStatus(bool enable, NsModes mode)
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::SetRxNsStatus(enable=%d, mode=%d)",
|
|
(int)enable, (int)mode);
|
|
|
|
NoiseSuppression::Level nsLevel = kDefaultNsMode;
|
|
switch (mode)
|
|
{
|
|
|
|
case kNsDefault:
|
|
break;
|
|
case kNsUnchanged:
|
|
nsLevel = rx_audioproc_->noise_suppression()->level();
|
|
break;
|
|
case kNsConference:
|
|
nsLevel = NoiseSuppression::kHigh;
|
|
break;
|
|
case kNsLowSuppression:
|
|
nsLevel = NoiseSuppression::kLow;
|
|
break;
|
|
case kNsModerateSuppression:
|
|
nsLevel = NoiseSuppression::kModerate;
|
|
break;
|
|
case kNsHighSuppression:
|
|
nsLevel = NoiseSuppression::kHigh;
|
|
break;
|
|
case kNsVeryHighSuppression:
|
|
nsLevel = NoiseSuppression::kVeryHigh;
|
|
break;
|
|
}
|
|
|
|
if (rx_audioproc_->noise_suppression()->set_level(nsLevel)
|
|
!= 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_APM_ERROR, kTraceError,
|
|
"SetRxNsStatus() failed to set NS level");
|
|
return -1;
|
|
}
|
|
if (rx_audioproc_->noise_suppression()->Enable(enable) != 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_APM_ERROR, kTraceError,
|
|
"SetRxNsStatus() failed to set NS state");
|
|
return -1;
|
|
}
|
|
|
|
_rxNsIsEnabled = enable;
|
|
channel_state_.SetRxApmIsEnabled(_rxAgcIsEnabled || _rxNsIsEnabled);
|
|
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
Channel::GetRxNsStatus(bool& enabled, NsModes& mode)
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::GetRxNsStatus(enable=?, mode=?)");
|
|
|
|
bool enable =
|
|
rx_audioproc_->noise_suppression()->is_enabled();
|
|
NoiseSuppression::Level ncLevel =
|
|
rx_audioproc_->noise_suppression()->level();
|
|
|
|
enabled = enable;
|
|
|
|
switch (ncLevel)
|
|
{
|
|
case NoiseSuppression::kLow:
|
|
mode = kNsLowSuppression;
|
|
break;
|
|
case NoiseSuppression::kModerate:
|
|
mode = kNsModerateSuppression;
|
|
break;
|
|
case NoiseSuppression::kHigh:
|
|
mode = kNsHighSuppression;
|
|
break;
|
|
case NoiseSuppression::kVeryHigh:
|
|
mode = kNsVeryHighSuppression;
|
|
break;
|
|
}
|
|
|
|
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
|
|
VoEId(_instanceId,_channelId),
|
|
"GetRxNsStatus() => enabled=%d, mode=%d", enabled, mode);
|
|
return 0;
|
|
}
|
|
|
|
#endif // #ifdef WEBRTC_VOICE_ENGINE_NR
|
|
|
|
int
|
|
Channel::RegisterRTCPObserver(VoERTCPObserver& observer)
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::RegisterRTCPObserver()");
|
|
CriticalSectionScoped cs(&_callbackCritSect);
|
|
|
|
if (_rtcpObserverPtr)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_INVALID_OPERATION, kTraceError,
|
|
"RegisterRTCPObserver() observer already enabled");
|
|
return -1;
|
|
}
|
|
|
|
_rtcpObserverPtr = &observer;
|
|
_rtcpObserver = true;
|
|
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
Channel::DeRegisterRTCPObserver()
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
|
|
"Channel::DeRegisterRTCPObserver()");
|
|
CriticalSectionScoped cs(&_callbackCritSect);
|
|
|
|
if (!_rtcpObserverPtr)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_INVALID_OPERATION, kTraceWarning,
|
|
"DeRegisterRTCPObserver() observer already disabled");
|
|
return 0;
|
|
}
|
|
|
|
_rtcpObserver = false;
|
|
_rtcpObserverPtr = NULL;
|
|
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
Channel::SetLocalSSRC(unsigned int ssrc)
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
|
|
"Channel::SetLocalSSRC()");
|
|
if (channel_state_.Get().sending)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_ALREADY_SENDING, kTraceError,
|
|
"SetLocalSSRC() already sending");
|
|
return -1;
|
|
}
|
|
_rtpRtcpModule->SetSSRC(ssrc);
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
Channel::GetLocalSSRC(unsigned int& ssrc)
|
|
{
|
|
ssrc = _rtpRtcpModule->SSRC();
|
|
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
|
|
VoEId(_instanceId,_channelId),
|
|
"GetLocalSSRC() => ssrc=%lu", ssrc);
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
Channel::GetRemoteSSRC(unsigned int& ssrc)
|
|
{
|
|
ssrc = rtp_receiver_->SSRC();
|
|
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
|
|
VoEId(_instanceId,_channelId),
|
|
"GetRemoteSSRC() => ssrc=%lu", ssrc);
|
|
return 0;
|
|
}
|
|
|
|
int Channel::SetSendAudioLevelIndicationStatus(bool enable, unsigned char id) {
|
|
_includeAudioLevelIndication = enable;
|
|
return SetSendRtpHeaderExtension(enable, kRtpExtensionAudioLevel, id);
|
|
}
|
|
|
|
int Channel::SetReceiveAudioLevelIndicationStatus(bool enable,
|
|
unsigned char id) {
|
|
rtp_header_parser_->DeregisterRtpHeaderExtension(
|
|
kRtpExtensionAudioLevel);
|
|
if (enable && !rtp_header_parser_->RegisterRtpHeaderExtension(
|
|
kRtpExtensionAudioLevel, id)) {
|
|
return -1;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int Channel::SetSendAbsoluteSenderTimeStatus(bool enable, unsigned char id) {
|
|
return SetSendRtpHeaderExtension(enable, kRtpExtensionAbsoluteSendTime, id);
|
|
}
|
|
|
|
int Channel::SetReceiveAbsoluteSenderTimeStatus(bool enable, unsigned char id) {
|
|
rtp_header_parser_->DeregisterRtpHeaderExtension(
|
|
kRtpExtensionAbsoluteSendTime);
|
|
if (enable && !rtp_header_parser_->RegisterRtpHeaderExtension(
|
|
kRtpExtensionAbsoluteSendTime, id)) {
|
|
return -1;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
Channel::SetRTCPStatus(bool enable)
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::SetRTCPStatus()");
|
|
if (_rtpRtcpModule->SetRTCPStatus(enable ?
|
|
kRtcpCompound : kRtcpOff) != 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_RTP_RTCP_MODULE_ERROR, kTraceError,
|
|
"SetRTCPStatus() failed to set RTCP status");
|
|
return -1;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
Channel::GetRTCPStatus(bool& enabled)
|
|
{
|
|
RTCPMethod method = _rtpRtcpModule->RTCP();
|
|
enabled = (method != kRtcpOff);
|
|
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
|
|
VoEId(_instanceId,_channelId),
|
|
"GetRTCPStatus() => enabled=%d", enabled);
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
Channel::SetRTCP_CNAME(const char cName[256])
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
|
|
"Channel::SetRTCP_CNAME()");
|
|
if (_rtpRtcpModule->SetCNAME(cName) != 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_RTP_RTCP_MODULE_ERROR, kTraceError,
|
|
"SetRTCP_CNAME() failed to set RTCP CNAME");
|
|
return -1;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
Channel::GetRTCP_CNAME(char cName[256])
|
|
{
|
|
if (_rtpRtcpModule->CNAME(cName) != 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_RTP_RTCP_MODULE_ERROR, kTraceError,
|
|
"GetRTCP_CNAME() failed to retrieve RTCP CNAME");
|
|
return -1;
|
|
}
|
|
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
|
|
VoEId(_instanceId, _channelId),
|
|
"GetRTCP_CNAME() => cName=%s", cName);
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
Channel::GetRemoteRTCP_CNAME(char cName[256])
|
|
{
|
|
if (cName == NULL)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_INVALID_ARGUMENT, kTraceError,
|
|
"GetRemoteRTCP_CNAME() invalid CNAME input buffer");
|
|
return -1;
|
|
}
|
|
char cname[RTCP_CNAME_SIZE];
|
|
const uint32_t remoteSSRC = rtp_receiver_->SSRC();
|
|
if (_rtpRtcpModule->RemoteCNAME(remoteSSRC, cname) != 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_CANNOT_RETRIEVE_CNAME, kTraceError,
|
|
"GetRemoteRTCP_CNAME() failed to retrieve remote RTCP CNAME");
|
|
return -1;
|
|
}
|
|
strcpy(cName, cname);
|
|
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
|
|
VoEId(_instanceId, _channelId),
|
|
"GetRemoteRTCP_CNAME() => cName=%s", cName);
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
Channel::GetRemoteRTCPData(
|
|
unsigned int& NTPHigh,
|
|
unsigned int& NTPLow,
|
|
unsigned int& timestamp,
|
|
unsigned int& playoutTimestamp,
|
|
unsigned int* jitter,
|
|
unsigned short* fractionLost)
|
|
{
|
|
// --- Information from sender info in received Sender Reports
|
|
|
|
RTCPSenderInfo senderInfo;
|
|
if (_rtpRtcpModule->RemoteRTCPStat(&senderInfo) != 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_RTP_RTCP_MODULE_ERROR, kTraceError,
|
|
"GetRemoteRTCPData() failed to retrieve sender info for remote "
|
|
"side");
|
|
return -1;
|
|
}
|
|
|
|
// We only utilize 12 out of 20 bytes in the sender info (ignores packet
|
|
// and octet count)
|
|
NTPHigh = senderInfo.NTPseconds;
|
|
NTPLow = senderInfo.NTPfraction;
|
|
timestamp = senderInfo.RTPtimeStamp;
|
|
|
|
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
|
|
VoEId(_instanceId, _channelId),
|
|
"GetRemoteRTCPData() => NTPHigh=%lu, NTPLow=%lu, "
|
|
"timestamp=%lu",
|
|
NTPHigh, NTPLow, timestamp);
|
|
|
|
// --- Locally derived information
|
|
|
|
// This value is updated on each incoming RTCP packet (0 when no packet
|
|
// has been received)
|
|
playoutTimestamp = playout_timestamp_rtcp_;
|
|
|
|
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
|
|
VoEId(_instanceId, _channelId),
|
|
"GetRemoteRTCPData() => playoutTimestamp=%lu",
|
|
playout_timestamp_rtcp_);
|
|
|
|
if (NULL != jitter || NULL != fractionLost)
|
|
{
|
|
// Get all RTCP receiver report blocks that have been received on this
|
|
// channel. If we receive RTP packets from a remote source we know the
|
|
// remote SSRC and use the report block from him.
|
|
// Otherwise use the first report block.
|
|
std::vector<RTCPReportBlock> remote_stats;
|
|
if (_rtpRtcpModule->RemoteRTCPStat(&remote_stats) != 0 ||
|
|
remote_stats.empty()) {
|
|
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
|
|
VoEId(_instanceId, _channelId),
|
|
"GetRemoteRTCPData() failed to measure statistics due"
|
|
" to lack of received RTP and/or RTCP packets");
|
|
return -1;
|
|
}
|
|
|
|
uint32_t remoteSSRC = rtp_receiver_->SSRC();
|
|
std::vector<RTCPReportBlock>::const_iterator it = remote_stats.begin();
|
|
for (; it != remote_stats.end(); ++it) {
|
|
if (it->remoteSSRC == remoteSSRC)
|
|
break;
|
|
}
|
|
|
|
if (it == remote_stats.end()) {
|
|
// If we have not received any RTCP packets from this SSRC it probably
|
|
// means that we have not received any RTP packets.
|
|
// Use the first received report block instead.
|
|
it = remote_stats.begin();
|
|
remoteSSRC = it->remoteSSRC;
|
|
}
|
|
|
|
if (jitter) {
|
|
*jitter = it->jitter;
|
|
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
|
|
VoEId(_instanceId, _channelId),
|
|
"GetRemoteRTCPData() => jitter = %lu", *jitter);
|
|
}
|
|
|
|
if (fractionLost) {
|
|
*fractionLost = it->fractionLost;
|
|
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
|
|
VoEId(_instanceId, _channelId),
|
|
"GetRemoteRTCPData() => fractionLost = %lu",
|
|
*fractionLost);
|
|
}
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
Channel::SendApplicationDefinedRTCPPacket(unsigned char subType,
|
|
unsigned int name,
|
|
const char* data,
|
|
unsigned short dataLengthInBytes)
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
|
|
"Channel::SendApplicationDefinedRTCPPacket()");
|
|
if (!channel_state_.Get().sending)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_NOT_SENDING, kTraceError,
|
|
"SendApplicationDefinedRTCPPacket() not sending");
|
|
return -1;
|
|
}
|
|
if (NULL == data)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_INVALID_ARGUMENT, kTraceError,
|
|
"SendApplicationDefinedRTCPPacket() invalid data value");
|
|
return -1;
|
|
}
|
|
if (dataLengthInBytes % 4 != 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_INVALID_ARGUMENT, kTraceError,
|
|
"SendApplicationDefinedRTCPPacket() invalid length value");
|
|
return -1;
|
|
}
|
|
RTCPMethod status = _rtpRtcpModule->RTCP();
|
|
if (status == kRtcpOff)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_RTCP_ERROR, kTraceError,
|
|
"SendApplicationDefinedRTCPPacket() RTCP is disabled");
|
|
return -1;
|
|
}
|
|
|
|
// Create and schedule the RTCP APP packet for transmission
|
|
if (_rtpRtcpModule->SetRTCPApplicationSpecificData(
|
|
subType,
|
|
name,
|
|
(const unsigned char*) data,
|
|
dataLengthInBytes) != 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_SEND_ERROR, kTraceError,
|
|
"SendApplicationDefinedRTCPPacket() failed to send RTCP packet");
|
|
return -1;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
Channel::GetRTPStatistics(
|
|
unsigned int& averageJitterMs,
|
|
unsigned int& maxJitterMs,
|
|
unsigned int& discardedPackets)
|
|
{
|
|
// The jitter statistics is updated for each received RTP packet and is
|
|
// based on received packets.
|
|
if (_rtpRtcpModule->RTCP() == kRtcpOff) {
|
|
// If RTCP is off, there is no timed thread in the RTCP module regularly
|
|
// generating new stats, trigger the update manually here instead.
|
|
StreamStatistician* statistician =
|
|
rtp_receive_statistics_->GetStatistician(rtp_receiver_->SSRC());
|
|
if (statistician) {
|
|
// Don't use returned statistics, use data from proxy instead so that
|
|
// max jitter can be fetched atomically.
|
|
RtcpStatistics s;
|
|
statistician->GetStatistics(&s, true);
|
|
}
|
|
}
|
|
|
|
ChannelStatistics stats = statistics_proxy_->GetStats();
|
|
const int32_t playoutFrequency = audio_coding_->PlayoutFrequency();
|
|
if (playoutFrequency > 0) {
|
|
// Scale RTP statistics given the current playout frequency
|
|
maxJitterMs = stats.max_jitter / (playoutFrequency / 1000);
|
|
averageJitterMs = stats.rtcp.jitter / (playoutFrequency / 1000);
|
|
}
|
|
|
|
discardedPackets = _numberOfDiscardedPackets;
|
|
|
|
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
|
|
VoEId(_instanceId, _channelId),
|
|
"GetRTPStatistics() => averageJitterMs = %lu, maxJitterMs = %lu,"
|
|
" discardedPackets = %lu)",
|
|
averageJitterMs, maxJitterMs, discardedPackets);
|
|
return 0;
|
|
}
|
|
|
|
int Channel::GetRemoteRTCPReportBlocks(
|
|
std::vector<ReportBlock>* report_blocks) {
|
|
if (report_blocks == NULL) {
|
|
_engineStatisticsPtr->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
|
|
"GetRemoteRTCPReportBlock()s invalid report_blocks.");
|
|
return -1;
|
|
}
|
|
|
|
// Get the report blocks from the latest received RTCP Sender or Receiver
|
|
// Report. Each element in the vector contains the sender's SSRC and a
|
|
// report block according to RFC 3550.
|
|
std::vector<RTCPReportBlock> rtcp_report_blocks;
|
|
if (_rtpRtcpModule->RemoteRTCPStat(&rtcp_report_blocks) != 0) {
|
|
_engineStatisticsPtr->SetLastError(VE_RTP_RTCP_MODULE_ERROR, kTraceError,
|
|
"GetRemoteRTCPReportBlocks() failed to read RTCP SR/RR report block.");
|
|
return -1;
|
|
}
|
|
|
|
if (rtcp_report_blocks.empty())
|
|
return 0;
|
|
|
|
std::vector<RTCPReportBlock>::const_iterator it = rtcp_report_blocks.begin();
|
|
for (; it != rtcp_report_blocks.end(); ++it) {
|
|
ReportBlock report_block;
|
|
report_block.sender_SSRC = it->remoteSSRC;
|
|
report_block.source_SSRC = it->sourceSSRC;
|
|
report_block.fraction_lost = it->fractionLost;
|
|
report_block.cumulative_num_packets_lost = it->cumulativeLost;
|
|
report_block.extended_highest_sequence_number = it->extendedHighSeqNum;
|
|
report_block.interarrival_jitter = it->jitter;
|
|
report_block.last_SR_timestamp = it->lastSR;
|
|
report_block.delay_since_last_SR = it->delaySinceLastSR;
|
|
report_blocks->push_back(report_block);
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
Channel::GetRTPStatistics(CallStatistics& stats)
|
|
{
|
|
// --- RtcpStatistics
|
|
|
|
// The jitter statistics is updated for each received RTP packet and is
|
|
// based on received packets.
|
|
RtcpStatistics statistics;
|
|
StreamStatistician* statistician =
|
|
rtp_receive_statistics_->GetStatistician(rtp_receiver_->SSRC());
|
|
if (!statistician || !statistician->GetStatistics(
|
|
&statistics, _rtpRtcpModule->RTCP() == kRtcpOff)) {
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_CANNOT_RETRIEVE_RTP_STAT, kTraceWarning,
|
|
"GetRTPStatistics() failed to read RTP statistics from the "
|
|
"RTP/RTCP module");
|
|
}
|
|
|
|
stats.fractionLost = statistics.fraction_lost;
|
|
stats.cumulativeLost = statistics.cumulative_lost;
|
|
stats.extendedMax = statistics.extended_max_sequence_number;
|
|
stats.jitterSamples = statistics.jitter;
|
|
|
|
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
|
|
VoEId(_instanceId, _channelId),
|
|
"GetRTPStatistics() => fractionLost=%lu, cumulativeLost=%lu,"
|
|
" extendedMax=%lu, jitterSamples=%li)",
|
|
stats.fractionLost, stats.cumulativeLost, stats.extendedMax,
|
|
stats.jitterSamples);
|
|
|
|
// --- RTT
|
|
|
|
uint16_t RTT(0);
|
|
RTCPMethod method = _rtpRtcpModule->RTCP();
|
|
if (method == kRtcpOff)
|
|
{
|
|
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
|
|
VoEId(_instanceId, _channelId),
|
|
"GetRTPStatistics() RTCP is disabled => valid RTT "
|
|
"measurements cannot be retrieved");
|
|
} else
|
|
{
|
|
// The remote SSRC will be zero if no RTP packet has been received.
|
|
uint32_t remoteSSRC = rtp_receiver_->SSRC();
|
|
if (remoteSSRC > 0)
|
|
{
|
|
uint16_t avgRTT(0);
|
|
uint16_t maxRTT(0);
|
|
uint16_t minRTT(0);
|
|
|
|
if (_rtpRtcpModule->RTT(remoteSSRC, &RTT, &avgRTT, &minRTT, &maxRTT)
|
|
!= 0)
|
|
{
|
|
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
|
|
VoEId(_instanceId, _channelId),
|
|
"GetRTPStatistics() failed to retrieve RTT from "
|
|
"the RTP/RTCP module");
|
|
}
|
|
} else
|
|
{
|
|
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
|
|
VoEId(_instanceId, _channelId),
|
|
"GetRTPStatistics() failed to measure RTT since no "
|
|
"RTP packets have been received yet");
|
|
}
|
|
}
|
|
|
|
stats.rttMs = static_cast<int> (RTT);
|
|
|
|
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
|
|
VoEId(_instanceId, _channelId),
|
|
"GetRTPStatistics() => rttMs=%d", stats.rttMs);
|
|
|
|
// --- Data counters
|
|
|
|
uint32_t bytesSent(0);
|
|
uint32_t packetsSent(0);
|
|
uint32_t bytesReceived(0);
|
|
uint32_t packetsReceived(0);
|
|
|
|
if (statistician) {
|
|
statistician->GetDataCounters(&bytesReceived, &packetsReceived);
|
|
}
|
|
|
|
if (_rtpRtcpModule->DataCountersRTP(&bytesSent,
|
|
&packetsSent) != 0)
|
|
{
|
|
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
|
|
VoEId(_instanceId, _channelId),
|
|
"GetRTPStatistics() failed to retrieve RTP datacounters =>"
|
|
" output will not be complete");
|
|
}
|
|
|
|
stats.bytesSent = bytesSent;
|
|
stats.packetsSent = packetsSent;
|
|
stats.bytesReceived = bytesReceived;
|
|
stats.packetsReceived = packetsReceived;
|
|
|
|
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
|
|
VoEId(_instanceId, _channelId),
|
|
"GetRTPStatistics() => bytesSent=%d, packetsSent=%d,"
|
|
" bytesReceived=%d, packetsReceived=%d)",
|
|
stats.bytesSent, stats.packetsSent, stats.bytesReceived,
|
|
stats.packetsReceived);
|
|
|
|
// --- Timestamps
|
|
{
|
|
CriticalSectionScoped lock(ts_stats_lock_.get());
|
|
stats.capture_start_ntp_time_ms_ = capture_start_ntp_time_ms_;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int Channel::SetREDStatus(bool enable, int redPayloadtype) {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
|
|
"Channel::SetREDStatus()");
|
|
|
|
if (enable) {
|
|
if (redPayloadtype < 0 || redPayloadtype > 127) {
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_PLTYPE_ERROR, kTraceError,
|
|
"SetREDStatus() invalid RED payload type");
|
|
return -1;
|
|
}
|
|
|
|
if (SetRedPayloadType(redPayloadtype) < 0) {
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_CODEC_ERROR, kTraceError,
|
|
"SetSecondarySendCodec() Failed to register RED ACM");
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
if (audio_coding_->SetREDStatus(enable) != 0) {
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
|
|
"SetREDStatus() failed to set RED state in the ACM");
|
|
return -1;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
Channel::GetREDStatus(bool& enabled, int& redPayloadtype)
|
|
{
|
|
enabled = audio_coding_->REDStatus();
|
|
if (enabled)
|
|
{
|
|
int8_t payloadType(0);
|
|
if (_rtpRtcpModule->SendREDPayloadType(payloadType) != 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_RTP_RTCP_MODULE_ERROR, kTraceError,
|
|
"GetREDStatus() failed to retrieve RED PT from RTP/RTCP "
|
|
"module");
|
|
return -1;
|
|
}
|
|
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
|
|
VoEId(_instanceId, _channelId),
|
|
"GetREDStatus() => enabled=%d, redPayloadtype=%d",
|
|
enabled, redPayloadtype);
|
|
return 0;
|
|
}
|
|
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
|
|
VoEId(_instanceId, _channelId),
|
|
"GetREDStatus() => enabled=%d", enabled);
|
|
return 0;
|
|
}
|
|
|
|
int Channel::SetCodecFECStatus(bool enable) {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
|
|
"Channel::SetCodecFECStatus()");
|
|
|
|
if (audio_coding_->SetCodecFEC(enable) != 0) {
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
|
|
"SetCodecFECStatus() failed to set FEC state");
|
|
return -1;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
bool Channel::GetCodecFECStatus() {
|
|
bool enabled = audio_coding_->CodecFEC();
|
|
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
|
|
VoEId(_instanceId, _channelId),
|
|
"GetCodecFECStatus() => enabled=%d", enabled);
|
|
return enabled;
|
|
}
|
|
|
|
void Channel::SetNACKStatus(bool enable, int maxNumberOfPackets) {
|
|
// None of these functions can fail.
|
|
_rtpRtcpModule->SetStorePacketsStatus(enable, maxNumberOfPackets);
|
|
rtp_receive_statistics_->SetMaxReorderingThreshold(maxNumberOfPackets);
|
|
rtp_receiver_->SetNACKStatus(enable ? kNackRtcp : kNackOff);
|
|
if (enable)
|
|
audio_coding_->EnableNack(maxNumberOfPackets);
|
|
else
|
|
audio_coding_->DisableNack();
|
|
}
|
|
|
|
// Called when we are missing one or more packets.
|
|
int Channel::ResendPackets(const uint16_t* sequence_numbers, int length) {
|
|
return _rtpRtcpModule->SendNACK(sequence_numbers, length);
|
|
}
|
|
|
|
int
|
|
Channel::StartRTPDump(const char fileNameUTF8[1024],
|
|
RTPDirections direction)
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
|
|
"Channel::StartRTPDump()");
|
|
if ((direction != kRtpIncoming) && (direction != kRtpOutgoing))
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_INVALID_ARGUMENT, kTraceError,
|
|
"StartRTPDump() invalid RTP direction");
|
|
return -1;
|
|
}
|
|
RtpDump* rtpDumpPtr = (direction == kRtpIncoming) ?
|
|
&_rtpDumpIn : &_rtpDumpOut;
|
|
if (rtpDumpPtr == NULL)
|
|
{
|
|
assert(false);
|
|
return -1;
|
|
}
|
|
if (rtpDumpPtr->IsActive())
|
|
{
|
|
rtpDumpPtr->Stop();
|
|
}
|
|
if (rtpDumpPtr->Start(fileNameUTF8) != 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_BAD_FILE, kTraceError,
|
|
"StartRTPDump() failed to create file");
|
|
return -1;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
Channel::StopRTPDump(RTPDirections direction)
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
|
|
"Channel::StopRTPDump()");
|
|
if ((direction != kRtpIncoming) && (direction != kRtpOutgoing))
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_INVALID_ARGUMENT, kTraceError,
|
|
"StopRTPDump() invalid RTP direction");
|
|
return -1;
|
|
}
|
|
RtpDump* rtpDumpPtr = (direction == kRtpIncoming) ?
|
|
&_rtpDumpIn : &_rtpDumpOut;
|
|
if (rtpDumpPtr == NULL)
|
|
{
|
|
assert(false);
|
|
return -1;
|
|
}
|
|
if (!rtpDumpPtr->IsActive())
|
|
{
|
|
return 0;
|
|
}
|
|
return rtpDumpPtr->Stop();
|
|
}
|
|
|
|
bool
|
|
Channel::RTPDumpIsActive(RTPDirections direction)
|
|
{
|
|
if ((direction != kRtpIncoming) &&
|
|
(direction != kRtpOutgoing))
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_INVALID_ARGUMENT, kTraceError,
|
|
"RTPDumpIsActive() invalid RTP direction");
|
|
return false;
|
|
}
|
|
RtpDump* rtpDumpPtr = (direction == kRtpIncoming) ?
|
|
&_rtpDumpIn : &_rtpDumpOut;
|
|
return rtpDumpPtr->IsActive();
|
|
}
|
|
|
|
void Channel::SetVideoEngineBWETarget(ViENetwork* vie_network,
|
|
int video_channel) {
|
|
CriticalSectionScoped cs(&_callbackCritSect);
|
|
if (vie_network_) {
|
|
vie_network_->Release();
|
|
vie_network_ = NULL;
|
|
}
|
|
video_channel_ = -1;
|
|
|
|
if (vie_network != NULL && video_channel != -1) {
|
|
vie_network_ = vie_network;
|
|
video_channel_ = video_channel;
|
|
}
|
|
}
|
|
|
|
uint32_t
|
|
Channel::Demultiplex(const AudioFrame& audioFrame)
|
|
{
|
|
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::Demultiplex()");
|
|
_audioFrame.CopyFrom(audioFrame);
|
|
_audioFrame.id_ = _channelId;
|
|
return 0;
|
|
}
|
|
|
|
void Channel::Demultiplex(const int16_t* audio_data,
|
|
int sample_rate,
|
|
int number_of_frames,
|
|
int number_of_channels) {
|
|
CodecInst codec;
|
|
GetSendCodec(codec);
|
|
|
|
if (!mono_recording_audio_.get()) {
|
|
// Temporary space for DownConvertToCodecFormat.
|
|
mono_recording_audio_.reset(new int16_t[kMaxMonoDataSizeSamples]);
|
|
}
|
|
DownConvertToCodecFormat(audio_data,
|
|
number_of_frames,
|
|
number_of_channels,
|
|
sample_rate,
|
|
codec.channels,
|
|
codec.plfreq,
|
|
mono_recording_audio_.get(),
|
|
&input_resampler_,
|
|
&_audioFrame);
|
|
}
|
|
|
|
uint32_t
|
|
Channel::PrepareEncodeAndSend(int mixingFrequency)
|
|
{
|
|
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::PrepareEncodeAndSend()");
|
|
|
|
if (_audioFrame.samples_per_channel_ == 0)
|
|
{
|
|
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::PrepareEncodeAndSend() invalid audio frame");
|
|
return -1;
|
|
}
|
|
|
|
if (channel_state_.Get().input_file_playing)
|
|
{
|
|
MixOrReplaceAudioWithFile(mixingFrequency);
|
|
}
|
|
|
|
bool is_muted = Mute(); // Cache locally as Mute() takes a lock.
|
|
if (is_muted) {
|
|
AudioFrameOperations::Mute(_audioFrame);
|
|
}
|
|
|
|
if (channel_state_.Get().input_external_media)
|
|
{
|
|
CriticalSectionScoped cs(&_callbackCritSect);
|
|
const bool isStereo = (_audioFrame.num_channels_ == 2);
|
|
if (_inputExternalMediaCallbackPtr)
|
|
{
|
|
_inputExternalMediaCallbackPtr->Process(
|
|
_channelId,
|
|
kRecordingPerChannel,
|
|
(int16_t*)_audioFrame.data_,
|
|
_audioFrame.samples_per_channel_,
|
|
_audioFrame.sample_rate_hz_,
|
|
isStereo);
|
|
}
|
|
}
|
|
|
|
InsertInbandDtmfTone();
|
|
|
|
if (_includeAudioLevelIndication) {
|
|
int length = _audioFrame.samples_per_channel_ * _audioFrame.num_channels_;
|
|
if (is_muted) {
|
|
rms_level_.ProcessMuted(length);
|
|
} else {
|
|
rms_level_.Process(_audioFrame.data_, length);
|
|
}
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
uint32_t
|
|
Channel::EncodeAndSend()
|
|
{
|
|
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::EncodeAndSend()");
|
|
|
|
assert(_audioFrame.num_channels_ <= 2);
|
|
if (_audioFrame.samples_per_channel_ == 0)
|
|
{
|
|
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::EncodeAndSend() invalid audio frame");
|
|
return -1;
|
|
}
|
|
|
|
_audioFrame.id_ = _channelId;
|
|
|
|
// --- Add 10ms of raw (PCM) audio data to the encoder @ 32kHz.
|
|
|
|
// The ACM resamples internally.
|
|
_audioFrame.timestamp_ = _timeStamp;
|
|
if (audio_coding_->Add10MsData((AudioFrame&)_audioFrame) != 0)
|
|
{
|
|
WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::EncodeAndSend() ACM encoding failed");
|
|
return -1;
|
|
}
|
|
|
|
_timeStamp += _audioFrame.samples_per_channel_;
|
|
|
|
// --- Encode if complete frame is ready
|
|
|
|
// This call will trigger AudioPacketizationCallback::SendData if encoding
|
|
// is done and payload is ready for packetization and transmission.
|
|
return audio_coding_->Process();
|
|
}
|
|
|
|
int Channel::RegisterExternalMediaProcessing(
|
|
ProcessingTypes type,
|
|
VoEMediaProcess& processObject)
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::RegisterExternalMediaProcessing()");
|
|
|
|
CriticalSectionScoped cs(&_callbackCritSect);
|
|
|
|
if (kPlaybackPerChannel == type)
|
|
{
|
|
if (_outputExternalMediaCallbackPtr)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_INVALID_OPERATION, kTraceError,
|
|
"Channel::RegisterExternalMediaProcessing() "
|
|
"output external media already enabled");
|
|
return -1;
|
|
}
|
|
_outputExternalMediaCallbackPtr = &processObject;
|
|
_outputExternalMedia = true;
|
|
}
|
|
else if (kRecordingPerChannel == type)
|
|
{
|
|
if (_inputExternalMediaCallbackPtr)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_INVALID_OPERATION, kTraceError,
|
|
"Channel::RegisterExternalMediaProcessing() "
|
|
"output external media already enabled");
|
|
return -1;
|
|
}
|
|
_inputExternalMediaCallbackPtr = &processObject;
|
|
channel_state_.SetInputExternalMedia(true);
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int Channel::DeRegisterExternalMediaProcessing(ProcessingTypes type)
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::DeRegisterExternalMediaProcessing()");
|
|
|
|
CriticalSectionScoped cs(&_callbackCritSect);
|
|
|
|
if (kPlaybackPerChannel == type)
|
|
{
|
|
if (!_outputExternalMediaCallbackPtr)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_INVALID_OPERATION, kTraceWarning,
|
|
"Channel::DeRegisterExternalMediaProcessing() "
|
|
"output external media already disabled");
|
|
return 0;
|
|
}
|
|
_outputExternalMedia = false;
|
|
_outputExternalMediaCallbackPtr = NULL;
|
|
}
|
|
else if (kRecordingPerChannel == type)
|
|
{
|
|
if (!_inputExternalMediaCallbackPtr)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_INVALID_OPERATION, kTraceWarning,
|
|
"Channel::DeRegisterExternalMediaProcessing() "
|
|
"input external media already disabled");
|
|
return 0;
|
|
}
|
|
channel_state_.SetInputExternalMedia(false);
|
|
_inputExternalMediaCallbackPtr = NULL;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
int Channel::SetExternalMixing(bool enabled) {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::SetExternalMixing(enabled=%d)", enabled);
|
|
|
|
if (channel_state_.Get().playing)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_INVALID_OPERATION, kTraceError,
|
|
"Channel::SetExternalMixing() "
|
|
"external mixing cannot be changed while playing.");
|
|
return -1;
|
|
}
|
|
|
|
_externalMixing = enabled;
|
|
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
Channel::GetNetworkStatistics(NetworkStatistics& stats)
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::GetNetworkStatistics()");
|
|
ACMNetworkStatistics acm_stats;
|
|
int return_value = audio_coding_->NetworkStatistics(&acm_stats);
|
|
if (return_value >= 0) {
|
|
memcpy(&stats, &acm_stats, sizeof(NetworkStatistics));
|
|
}
|
|
return return_value;
|
|
}
|
|
|
|
void Channel::GetDecodingCallStatistics(AudioDecodingCallStats* stats) const {
|
|
audio_coding_->GetDecodingCallStatistics(stats);
|
|
}
|
|
|
|
bool Channel::GetDelayEstimate(int* jitter_buffer_delay_ms,
|
|
int* playout_buffer_delay_ms) const {
|
|
if (_average_jitter_buffer_delay_us == 0) {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::GetDelayEstimate() no valid estimate.");
|
|
return false;
|
|
}
|
|
*jitter_buffer_delay_ms = (_average_jitter_buffer_delay_us + 500) / 1000 +
|
|
_recPacketDelayMs;
|
|
*playout_buffer_delay_ms = playout_delay_ms_;
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::GetDelayEstimate()");
|
|
return true;
|
|
}
|
|
|
|
int Channel::SetInitialPlayoutDelay(int delay_ms)
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::SetInitialPlayoutDelay()");
|
|
if ((delay_ms < kVoiceEngineMinMinPlayoutDelayMs) ||
|
|
(delay_ms > kVoiceEngineMaxMinPlayoutDelayMs))
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_INVALID_ARGUMENT, kTraceError,
|
|
"SetInitialPlayoutDelay() invalid min delay");
|
|
return -1;
|
|
}
|
|
if (audio_coding_->SetInitialPlayoutDelay(delay_ms) != 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
|
|
"SetInitialPlayoutDelay() failed to set min playout delay");
|
|
return -1;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
|
|
int
|
|
Channel::SetMinimumPlayoutDelay(int delayMs)
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::SetMinimumPlayoutDelay()");
|
|
if ((delayMs < kVoiceEngineMinMinPlayoutDelayMs) ||
|
|
(delayMs > kVoiceEngineMaxMinPlayoutDelayMs))
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_INVALID_ARGUMENT, kTraceError,
|
|
"SetMinimumPlayoutDelay() invalid min delay");
|
|
return -1;
|
|
}
|
|
if (audio_coding_->SetMinimumPlayoutDelay(delayMs) != 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
|
|
"SetMinimumPlayoutDelay() failed to set min playout delay");
|
|
return -1;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
void Channel::UpdatePlayoutTimestamp(bool rtcp) {
|
|
uint32_t playout_timestamp = 0;
|
|
|
|
if (audio_coding_->PlayoutTimestamp(&playout_timestamp) == -1) {
|
|
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::UpdatePlayoutTimestamp() failed to read playout"
|
|
" timestamp from the ACM");
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_CANNOT_RETRIEVE_VALUE, kTraceError,
|
|
"UpdatePlayoutTimestamp() failed to retrieve timestamp");
|
|
return;
|
|
}
|
|
|
|
uint16_t delay_ms = 0;
|
|
if (_audioDeviceModulePtr->PlayoutDelay(&delay_ms) == -1) {
|
|
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::UpdatePlayoutTimestamp() failed to read playout"
|
|
" delay from the ADM");
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_CANNOT_RETRIEVE_VALUE, kTraceError,
|
|
"UpdatePlayoutTimestamp() failed to retrieve playout delay");
|
|
return;
|
|
}
|
|
|
|
int32_t playout_frequency = audio_coding_->PlayoutFrequency();
|
|
CodecInst current_recive_codec;
|
|
if (audio_coding_->ReceiveCodec(¤t_recive_codec) == 0) {
|
|
if (STR_CASE_CMP("G722", current_recive_codec.plname) == 0) {
|
|
playout_frequency = 8000;
|
|
} else if (STR_CASE_CMP("opus", current_recive_codec.plname) == 0) {
|
|
playout_frequency = 48000;
|
|
}
|
|
}
|
|
|
|
jitter_buffer_playout_timestamp_ = playout_timestamp;
|
|
|
|
// Remove the playout delay.
|
|
playout_timestamp -= (delay_ms * (playout_frequency / 1000));
|
|
|
|
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::UpdatePlayoutTimestamp() => playoutTimestamp = %lu",
|
|
playout_timestamp);
|
|
|
|
if (rtcp) {
|
|
playout_timestamp_rtcp_ = playout_timestamp;
|
|
} else {
|
|
playout_timestamp_rtp_ = playout_timestamp;
|
|
}
|
|
playout_delay_ms_ = delay_ms;
|
|
}
|
|
|
|
int Channel::GetPlayoutTimestamp(unsigned int& timestamp) {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::GetPlayoutTimestamp()");
|
|
if (playout_timestamp_rtp_ == 0) {
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_CANNOT_RETRIEVE_VALUE, kTraceError,
|
|
"GetPlayoutTimestamp() failed to retrieve timestamp");
|
|
return -1;
|
|
}
|
|
timestamp = playout_timestamp_rtp_;
|
|
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
|
|
VoEId(_instanceId,_channelId),
|
|
"GetPlayoutTimestamp() => timestamp=%u", timestamp);
|
|
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;
|
|
}
|
|
if (_rtpRtcpModule->SetStartTimestamp(timestamp) != 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_RTP_RTCP_MODULE_ERROR, kTraceError,
|
|
"SetInitTimestamp() failed to set timestamp");
|
|
return -1;
|
|
}
|
|
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;
|
|
}
|
|
if (_rtpRtcpModule->SetSequenceNumber(sequenceNumber) != 0)
|
|
{
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_RTP_RTCP_MODULE_ERROR, kTraceError,
|
|
"SetInitSequenceNumber() failed to set sequence number");
|
|
return -1;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
Channel::GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::GetRtpRtcp()");
|
|
*rtpRtcpModule = _rtpRtcpModule.get();
|
|
*rtp_receiver = rtp_receiver_.get();
|
|
return 0;
|
|
}
|
|
|
|
// TODO(andrew): refactor Mix functions here and in transmit_mixer.cc to use
|
|
// a shared helper.
|
|
int32_t
|
|
Channel::MixOrReplaceAudioWithFile(int mixingFrequency)
|
|
{
|
|
scoped_ptr<int16_t[]> fileBuffer(new int16_t[640]);
|
|
int fileSamples(0);
|
|
|
|
{
|
|
CriticalSectionScoped cs(&_fileCritSect);
|
|
|
|
if (_inputFilePlayerPtr == NULL)
|
|
{
|
|
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
|
|
VoEId(_instanceId, _channelId),
|
|
"Channel::MixOrReplaceAudioWithFile() fileplayer"
|
|
" doesnt exist");
|
|
return -1;
|
|
}
|
|
|
|
if (_inputFilePlayerPtr->Get10msAudioFromFile(fileBuffer.get(),
|
|
fileSamples,
|
|
mixingFrequency) == -1)
|
|
{
|
|
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
|
|
VoEId(_instanceId, _channelId),
|
|
"Channel::MixOrReplaceAudioWithFile() file mixing "
|
|
"failed");
|
|
return -1;
|
|
}
|
|
if (fileSamples == 0)
|
|
{
|
|
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
|
|
VoEId(_instanceId, _channelId),
|
|
"Channel::MixOrReplaceAudioWithFile() file is ended");
|
|
return 0;
|
|
}
|
|
}
|
|
|
|
assert(_audioFrame.samples_per_channel_ == fileSamples);
|
|
|
|
if (_mixFileWithMicrophone)
|
|
{
|
|
// Currently file stream is always mono.
|
|
// TODO(xians): Change the code when FilePlayer supports real stereo.
|
|
MixWithSat(_audioFrame.data_,
|
|
_audioFrame.num_channels_,
|
|
fileBuffer.get(),
|
|
1,
|
|
fileSamples);
|
|
}
|
|
else
|
|
{
|
|
// Replace ACM audio with file.
|
|
// Currently file stream is always mono.
|
|
// TODO(xians): Change the code when FilePlayer supports real stereo.
|
|
_audioFrame.UpdateFrame(_channelId,
|
|
-1,
|
|
fileBuffer.get(),
|
|
fileSamples,
|
|
mixingFrequency,
|
|
AudioFrame::kNormalSpeech,
|
|
AudioFrame::kVadUnknown,
|
|
1);
|
|
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int32_t
|
|
Channel::MixAudioWithFile(AudioFrame& audioFrame,
|
|
int mixingFrequency)
|
|
{
|
|
assert(mixingFrequency <= 32000);
|
|
|
|
scoped_ptr<int16_t[]> fileBuffer(new int16_t[640]);
|
|
int fileSamples(0);
|
|
|
|
{
|
|
CriticalSectionScoped cs(&_fileCritSect);
|
|
|
|
if (_outputFilePlayerPtr == NULL)
|
|
{
|
|
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
|
|
VoEId(_instanceId, _channelId),
|
|
"Channel::MixAudioWithFile() file mixing failed");
|
|
return -1;
|
|
}
|
|
|
|
// We should get the frequency we ask for.
|
|
if (_outputFilePlayerPtr->Get10msAudioFromFile(fileBuffer.get(),
|
|
fileSamples,
|
|
mixingFrequency) == -1)
|
|
{
|
|
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
|
|
VoEId(_instanceId, _channelId),
|
|
"Channel::MixAudioWithFile() file mixing failed");
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
if (audioFrame.samples_per_channel_ == fileSamples)
|
|
{
|
|
// Currently file stream is always mono.
|
|
// TODO(xians): Change the code when FilePlayer supports real stereo.
|
|
MixWithSat(audioFrame.data_,
|
|
audioFrame.num_channels_,
|
|
fileBuffer.get(),
|
|
1,
|
|
fileSamples);
|
|
}
|
|
else
|
|
{
|
|
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::MixAudioWithFile() samples_per_channel_(%d) != "
|
|
"fileSamples(%d)",
|
|
audioFrame.samples_per_channel_, fileSamples);
|
|
return -1;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
Channel::InsertInbandDtmfTone()
|
|
{
|
|
// Check if we should start a new tone.
|
|
if (_inbandDtmfQueue.PendingDtmf() &&
|
|
!_inbandDtmfGenerator.IsAddingTone() &&
|
|
_inbandDtmfGenerator.DelaySinceLastTone() >
|
|
kMinTelephoneEventSeparationMs)
|
|
{
|
|
int8_t eventCode(0);
|
|
uint16_t lengthMs(0);
|
|
uint8_t attenuationDb(0);
|
|
|
|
eventCode = _inbandDtmfQueue.NextDtmf(&lengthMs, &attenuationDb);
|
|
_inbandDtmfGenerator.AddTone(eventCode, lengthMs, attenuationDb);
|
|
if (_playInbandDtmfEvent)
|
|
{
|
|
// Add tone to output mixer using a reduced length to minimize
|
|
// risk of echo.
|
|
_outputMixerPtr->PlayDtmfTone(eventCode, lengthMs - 80,
|
|
attenuationDb);
|
|
}
|
|
}
|
|
|
|
if (_inbandDtmfGenerator.IsAddingTone())
|
|
{
|
|
uint16_t frequency(0);
|
|
_inbandDtmfGenerator.GetSampleRate(frequency);
|
|
|
|
if (frequency != _audioFrame.sample_rate_hz_)
|
|
{
|
|
// Update sample rate of Dtmf tone since the mixing frequency
|
|
// has changed.
|
|
_inbandDtmfGenerator.SetSampleRate(
|
|
(uint16_t) (_audioFrame.sample_rate_hz_));
|
|
// Reset the tone to be added taking the new sample rate into
|
|
// account.
|
|
_inbandDtmfGenerator.ResetTone();
|
|
}
|
|
|
|
int16_t toneBuffer[320];
|
|
uint16_t toneSamples(0);
|
|
// Get 10ms tone segment and set time since last tone to zero
|
|
if (_inbandDtmfGenerator.Get10msTone(toneBuffer, toneSamples) == -1)
|
|
{
|
|
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
|
|
VoEId(_instanceId, _channelId),
|
|
"Channel::EncodeAndSend() inserting Dtmf failed");
|
|
return -1;
|
|
}
|
|
|
|
// Replace mixed audio with DTMF tone.
|
|
for (int sample = 0;
|
|
sample < _audioFrame.samples_per_channel_;
|
|
sample++)
|
|
{
|
|
for (int channel = 0;
|
|
channel < _audioFrame.num_channels_;
|
|
channel++)
|
|
{
|
|
const int index = sample * _audioFrame.num_channels_ + channel;
|
|
_audioFrame.data_[index] = toneBuffer[sample];
|
|
}
|
|
}
|
|
|
|
assert(_audioFrame.samples_per_channel_ == toneSamples);
|
|
} else
|
|
{
|
|
// Add 10ms to "delay-since-last-tone" counter
|
|
_inbandDtmfGenerator.UpdateDelaySinceLastTone();
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int32_t
|
|
Channel::SendPacketRaw(const void *data, int len, bool RTCP)
|
|
{
|
|
CriticalSectionScoped cs(&_callbackCritSect);
|
|
if (_transportPtr == NULL)
|
|
{
|
|
return -1;
|
|
}
|
|
if (!RTCP)
|
|
{
|
|
return _transportPtr->SendPacket(_channelId, data, len);
|
|
}
|
|
else
|
|
{
|
|
return _transportPtr->SendRTCPPacket(_channelId, data, len);
|
|
}
|
|
}
|
|
|
|
// Called for incoming RTP packets after successful RTP header parsing.
|
|
void Channel::UpdatePacketDelay(uint32_t rtp_timestamp,
|
|
uint16_t sequence_number) {
|
|
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::UpdatePacketDelay(timestamp=%lu, sequenceNumber=%u)",
|
|
rtp_timestamp, sequence_number);
|
|
|
|
// Get frequency of last received payload
|
|
int rtp_receive_frequency = audio_coding_->ReceiveFrequency();
|
|
|
|
CodecInst current_receive_codec;
|
|
if (audio_coding_->ReceiveCodec(¤t_receive_codec) != 0) {
|
|
return;
|
|
}
|
|
|
|
// Update the least required delay.
|
|
least_required_delay_ms_ = audio_coding_->LeastRequiredDelayMs();
|
|
|
|
if (STR_CASE_CMP("G722", current_receive_codec.plname) == 0) {
|
|
// Even though the actual sampling rate for G.722 audio is
|
|
// 16,000 Hz, the RTP clock rate for the G722 payload format is
|
|
// 8,000 Hz because that value was erroneously assigned in
|
|
// RFC 1890 and must remain unchanged for backward compatibility.
|
|
rtp_receive_frequency = 8000;
|
|
} else if (STR_CASE_CMP("opus", current_receive_codec.plname) == 0) {
|
|
// We are resampling Opus internally to 32,000 Hz until all our
|
|
// DSP routines can operate at 48,000 Hz, but the RTP clock
|
|
// rate for the Opus payload format is standardized to 48,000 Hz,
|
|
// because that is the maximum supported decoding sampling rate.
|
|
rtp_receive_frequency = 48000;
|
|
}
|
|
|
|
// |jitter_buffer_playout_timestamp_| updated in UpdatePlayoutTimestamp for
|
|
// every incoming packet.
|
|
uint32_t timestamp_diff_ms = (rtp_timestamp -
|
|
jitter_buffer_playout_timestamp_) / (rtp_receive_frequency / 1000);
|
|
if (!IsNewerTimestamp(rtp_timestamp, jitter_buffer_playout_timestamp_) ||
|
|
timestamp_diff_ms > (2 * kVoiceEngineMaxMinPlayoutDelayMs)) {
|
|
// If |jitter_buffer_playout_timestamp_| is newer than the incoming RTP
|
|
// timestamp, the resulting difference is negative, but is set to zero.
|
|
// This can happen when a network glitch causes a packet to arrive late,
|
|
// and during long comfort noise periods with clock drift.
|
|
timestamp_diff_ms = 0;
|
|
}
|
|
|
|
uint16_t packet_delay_ms = (rtp_timestamp - _previousTimestamp) /
|
|
(rtp_receive_frequency / 1000);
|
|
|
|
_previousTimestamp = rtp_timestamp;
|
|
|
|
if (timestamp_diff_ms == 0) return;
|
|
|
|
if (packet_delay_ms >= 10 && packet_delay_ms <= 60) {
|
|
_recPacketDelayMs = packet_delay_ms;
|
|
}
|
|
|
|
if (_average_jitter_buffer_delay_us == 0) {
|
|
_average_jitter_buffer_delay_us = timestamp_diff_ms * 1000;
|
|
return;
|
|
}
|
|
|
|
// Filter average delay value using exponential filter (alpha is
|
|
// 7/8). We derive 1000 *_average_jitter_buffer_delay_us here (reduces
|
|
// risk of rounding error) and compensate for it in GetDelayEstimate()
|
|
// later.
|
|
_average_jitter_buffer_delay_us = (_average_jitter_buffer_delay_us * 7 +
|
|
1000 * timestamp_diff_ms + 500) / 8;
|
|
}
|
|
|
|
void
|
|
Channel::RegisterReceiveCodecsToRTPModule()
|
|
{
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
|
"Channel::RegisterReceiveCodecsToRTPModule()");
|
|
|
|
|
|
CodecInst codec;
|
|
const uint8_t nSupportedCodecs = AudioCodingModule::NumberOfCodecs();
|
|
|
|
for (int idx = 0; idx < nSupportedCodecs; idx++)
|
|
{
|
|
// Open up the RTP/RTCP receiver for all supported codecs
|
|
if ((audio_coding_->Codec(idx, &codec) == -1) ||
|
|
(rtp_receiver_->RegisterReceivePayload(
|
|
codec.plname,
|
|
codec.pltype,
|
|
codec.plfreq,
|
|
codec.channels,
|
|
(codec.rate < 0) ? 0 : codec.rate) == -1))
|
|
{
|
|
WEBRTC_TRACE(
|
|
kTraceWarning,
|
|
kTraceVoice,
|
|
VoEId(_instanceId, _channelId),
|
|
"Channel::RegisterReceiveCodecsToRTPModule() unable"
|
|
" to register %s (%d/%d/%d/%d) to RTP/RTCP receiver",
|
|
codec.plname, codec.pltype, codec.plfreq,
|
|
codec.channels, codec.rate);
|
|
}
|
|
else
|
|
{
|
|
WEBRTC_TRACE(
|
|
kTraceInfo,
|
|
kTraceVoice,
|
|
VoEId(_instanceId, _channelId),
|
|
"Channel::RegisterReceiveCodecsToRTPModule() %s "
|
|
"(%d/%d/%d/%d) has been added to the RTP/RTCP "
|
|
"receiver",
|
|
codec.plname, codec.pltype, codec.plfreq,
|
|
codec.channels, codec.rate);
|
|
}
|
|
}
|
|
}
|
|
|
|
int Channel::SetSecondarySendCodec(const CodecInst& codec,
|
|
int red_payload_type) {
|
|
// Sanity check for payload type.
|
|
if (red_payload_type < 0 || red_payload_type > 127) {
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_PLTYPE_ERROR, kTraceError,
|
|
"SetRedPayloadType() invalid RED payload type");
|
|
return -1;
|
|
}
|
|
|
|
if (SetRedPayloadType(red_payload_type) < 0) {
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
|
|
"SetSecondarySendCodec() Failed to register RED ACM");
|
|
return -1;
|
|
}
|
|
if (audio_coding_->RegisterSecondarySendCodec(codec) < 0) {
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
|
|
"SetSecondarySendCodec() Failed to register secondary send codec in "
|
|
"ACM");
|
|
return -1;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
void Channel::RemoveSecondarySendCodec() {
|
|
audio_coding_->UnregisterSecondarySendCodec();
|
|
}
|
|
|
|
int Channel::GetSecondarySendCodec(CodecInst* codec) {
|
|
if (audio_coding_->SecondarySendCodec(codec) < 0) {
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
|
|
"GetSecondarySendCodec() Failed to get secondary sent codec from ACM");
|
|
return -1;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
// Assuming this method is called with valid payload type.
|
|
int Channel::SetRedPayloadType(int red_payload_type) {
|
|
CodecInst codec;
|
|
bool found_red = false;
|
|
|
|
// Get default RED settings from the ACM database
|
|
const int num_codecs = AudioCodingModule::NumberOfCodecs();
|
|
for (int idx = 0; idx < num_codecs; idx++) {
|
|
audio_coding_->Codec(idx, &codec);
|
|
if (!STR_CASE_CMP(codec.plname, "RED")) {
|
|
found_red = true;
|
|
break;
|
|
}
|
|
}
|
|
|
|
if (!found_red) {
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_CODEC_ERROR, kTraceError,
|
|
"SetRedPayloadType() RED is not supported");
|
|
return -1;
|
|
}
|
|
|
|
codec.pltype = red_payload_type;
|
|
if (audio_coding_->RegisterSendCodec(codec) < 0) {
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
|
|
"SetRedPayloadType() RED registration in ACM module failed");
|
|
return -1;
|
|
}
|
|
|
|
if (_rtpRtcpModule->SetSendREDPayloadType(red_payload_type) != 0) {
|
|
_engineStatisticsPtr->SetLastError(
|
|
VE_RTP_RTCP_MODULE_ERROR, kTraceError,
|
|
"SetRedPayloadType() RED registration in RTP/RTCP module failed");
|
|
return -1;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int Channel::SetSendRtpHeaderExtension(bool enable, RTPExtensionType type,
|
|
unsigned char id) {
|
|
int error = 0;
|
|
_rtpRtcpModule->DeregisterSendRtpHeaderExtension(type);
|
|
if (enable) {
|
|
error = _rtpRtcpModule->RegisterSendRtpHeaderExtension(type, id);
|
|
}
|
|
return error;
|
|
}
|
|
|
|
} // namespace voe
|
|
} // namespace webrtc
|