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- Clean up unused voice engine DTMF code following removal of VoEDtmf APIs.

Browse Source 
- Use better types in AudioSendStream::SendTelephoneEvent() and related methods.

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1722253002

Cr-Commit-Position: refs/heads/master@{#11927}
This commit is contained in:
solenberg 2016-03-09 03:32:44 -08:00 committed by Commit bot
parent 16daaa5a40
commit 8886c81658
21 changed files with 42 additions and 329 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

4
webrtc/audio/audio_send_stream.cc
View File

@ -125,8 +125,8 @@ bool AudioSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
return false;
}
bool AudioSendStream::SendTelephoneEvent(int payload_type, uint8_t event,
uint32_t duration_ms) {
bool AudioSendStream::SendTelephoneEvent(int payload_type, int event,
int duration_ms) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
return channel_proxy_->SetSendTelephoneEventPayloadType(payload_type) &&
channel_proxy_->SendTelephoneEventOutband(event, duration_ms);

4
webrtc/audio/audio_send_stream.h
View File

@ -40,8 +40,8 @@ class AudioSendStream final : public webrtc::AudioSendStream {
bool DeliverRtcp(const uint8_t* packet, size_t length) override;
// webrtc::AudioSendStream implementation.
bool SendTelephoneEvent(int payload_type, uint8_t event,
uint32_t duration_ms) override;
bool SendTelephoneEvent(int payload_type, int event,
int duration_ms) override;
webrtc::AudioSendStream::Stats GetStats() const override;
const webrtc::AudioSendStream::Config& config() const;

4
webrtc/audio/audio_send_stream_unittest.cc
View File

@ -46,8 +46,8 @@ const CallStatistics kCallStats = {
const CodecInst kCodecInst = {-121, "codec_name_send", 48000, -231, 0, -671};
const ReportBlock kReportBlock = {456, 780, 123, 567, 890, 132, 143, 13354};
const int kTelephoneEventPayloadType = 123;
const uint8_t kTelephoneEventCode = 45;
const uint32_t kTelephoneEventDuration = 6789;
const int kTelephoneEventCode = 45;
const int kTelephoneEventDuration = 6789;
struct ConfigHelper {
ConfigHelper()

4
webrtc/audio_send_stream.h
View File

@ -90,8 +90,8 @@ class AudioSendStream : public SendStream {
};
// TODO(solenberg): Make payload_type a config property instead.
virtual bool SendTelephoneEvent(int payload_type, uint8_t event,
uint32_t duration_ms) = 0;
virtual bool SendTelephoneEvent(int payload_type, int event,
int duration_ms) = 0;
virtual Stats GetStats() const = 0;
};
} // namespace webrtc

4
webrtc/media/engine/fakewebrtccall.cc
View File

@ -39,8 +39,8 @@ FakeAudioSendStream::TelephoneEvent
return latest_telephone_event_;
}
bool FakeAudioSendStream::SendTelephoneEvent(int payload_type, uint8_t event,
uint32_t duration_ms) {
bool FakeAudioSendStream::SendTelephoneEvent(int payload_type, int event,
int duration_ms) {
latest_telephone_event_.payload_type = payload_type;
latest_telephone_event_.event_code = event;
latest_telephone_event_.duration_ms = duration_ms;

8
webrtc/media/engine/fakewebrtccall.h
View File

@ -35,8 +35,8 @@ class FakeAudioSendStream final : public webrtc::AudioSendStream {
public:
struct TelephoneEvent {
int payload_type = -1;
uint8_t event_code = 0;
uint32_t duration_ms = 0;
int event_code = 0;
int duration_ms = 0;
};
explicit FakeAudioSendStream(const webrtc::AudioSendStream::Config& config);
@ -56,8 +56,8 @@ class FakeAudioSendStream final : public webrtc::AudioSendStream {
}
// webrtc::AudioSendStream implementation.
bool SendTelephoneEvent(int payload_type, uint8_t event,
uint32_t duration_ms) override;
bool SendTelephoneEvent(int payload_type, int event,
int duration_ms) override;
webrtc::AudioSendStream::Stats GetStats() const override;
TelephoneEvent latest_telephone_event_;

3
webrtc/media/engine/webrtcvoiceengine.cc
View File

@ -1178,8 +1178,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
RTC_CHECK(stream_);
}
bool SendTelephoneEvent(int payload_type, uint8_t event,
uint32_t duration_ms) {
bool SendTelephoneEvent(int payload_type, int event, int duration_ms) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK(stream_);
return stream_->SendTelephoneEvent(payload_type, event, duration_ms);

8
webrtc/modules/audio_coding/test/TwoWayCommunication.cc
View File

@ -50,14 +50,6 @@ TwoWayCommunication::~TwoWayCommunication() {
delete _channel_B2A;
delete _channelRef_A2B;
delete _channelRef_B2A;
#ifdef WEBRTC_DTMF_DETECTION
if (_dtmfDetectorA != NULL) {
delete _dtmfDetectorA;
}
if (_dtmfDetectorB != NULL) {
delete _dtmfDetectorB;
}
#endif
_inFileA.Close();
_inFileB.Close();
_outFileA.Close();

2
webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
View File

@ -322,7 +322,7 @@ TEST_F(RtpRtcpAudioTest, DTMF) {
(voice_codec.rate < 0) ? 0 : voice_codec.rate));
// Start DTMF test.
uint32_t timeStamp = 160;
int timeStamp = 160;
// Send a DTMF tone using RFC 2833 (4733).
for (int i = 0; i < 16; i++) {

3
webrtc/test/mock_voe_channel_proxy.h
View File

@ -43,8 +43,7 @@ class MockVoEChannelProxy : public voe::ChannelProxy {
MOCK_CONST_METHOD0(GetSpeechOutputLevelFullRange, int32_t());
MOCK_CONST_METHOD0(GetDelayEstimate, uint32_t());
MOCK_METHOD1(SetSendTelephoneEventPayloadType, bool(int payload_type));
MOCK_METHOD2(SendTelephoneEventOutband, bool(uint8_t event,
uint32_t duration_ms));
MOCK_METHOD2(SendTelephoneEventOutband, bool(int event, int duration_ms));
};
} // namespace test
} // namespace webrtc

2
webrtc/voice_engine/BUILD.gn
View File

@ -18,8 +18,6 @@ source_set("voice_engine") {
"channel_proxy.h",
"dtmf_inband.cc",
"dtmf_inband.h",
"dtmf_inband_queue.cc",
"dtmf_inband_queue.h",
"include/voe_audio_processing.h",
"include/voe_base.h",
"include/voe_codec.h",

128
webrtc/voice_engine/channel.cc
View File

@ -40,13 +40,11 @@
#include "webrtc/voice_engine/transmit_mixer.h"
#include "webrtc/voice_engine/utility.h"
#if defined(_WIN32)
#include <Qos.h>
#endif
namespace webrtc {
namespace voe {
const int kTelephoneEventAttenuationdB = 10;
class TransportFeedbackProxy : public TransportFeedbackObserver {
public:
TransportFeedbackProxy() : feedback_observer_(nullptr) {
@ -363,7 +361,7 @@ void Channel::OnPlayTelephoneEvent(uint8_t event,
" volume=%u)",
event, lengthMs, volume);
if (!_playOutbandDtmfEvent || (event > 15)) {
if (event > 15) {
// Ignore callback since feedback is disabled or event is not a
// Dtmf tone event.
return;
@ -761,14 +759,11 @@ Channel::Channel(int32_t channelId,
_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_(Clock::GetRealTimeClock()),
jitter_buffer_playout_timestamp_(0),
playout_timestamp_rtp_(0),
@ -796,8 +791,6 @@ Channel::Channel(int32_t channelId,
_panLeft(1.0f),
_panRight(1.0f),
_outputGain(1.0f),
_playOutbandDtmfEvent(false),
_playInbandDtmfEvent(false),
_lastLocalTimeStamp(0),
_lastPayloadType(0),
_includeAudioLevelIndication(false),
@ -830,8 +823,6 @@ Channel::Channel(int32_t channelId,
config.Get<NetEqFastAccelerate>().enabled;
audio_coding_.reset(AudioCodingModule::Create(acm_config));
_inbandDtmfQueue.ResetDtmf();
_inbandDtmfGenerator.Init();
_outputAudioLevel.Clear();
RtpRtcp::Configuration configuration;
@ -2212,21 +2203,18 @@ int Channel::GetChannelOutputVolumeScaling(float& scaling) const {
return 0;
}
int Channel::SendTelephoneEventOutband(unsigned char eventCode,
int lengthMs,
int attenuationDb,
bool playDtmfEvent) {
int Channel::SendTelephoneEventOutband(int event, int duration_ms) {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SendTelephoneEventOutband(..., playDtmfEvent=%d)",
playDtmfEvent);
"Channel::SendTelephoneEventOutband(...)");
RTC_DCHECK_LE(0, event);
RTC_DCHECK_GE(255, event);
RTC_DCHECK_LE(0, duration_ms);
RTC_DCHECK_GE(65535, duration_ms);
if (!Sending()) {
return -1;
}
_playOutbandDtmfEvent = playDtmfEvent;
if (_rtpRtcpModule->SendTelephoneEventOutband(eventCode, lengthMs,
attenuationDb) != 0) {
if (_rtpRtcpModule->SendTelephoneEventOutband(
event, duration_ms, kTelephoneEventAttenuationdB) != 0) {
_engineStatisticsPtr->SetLastError(
VE_SEND_DTMF_FAILED, kTraceWarning,
"SendTelephoneEventOutband() failed to send event");
@ -2235,32 +2223,14 @@ int Channel::SendTelephoneEventOutband(unsigned char eventCode,
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::SetSendTelephoneEventPayloadType(unsigned char type) {
int Channel::SetSendTelephoneEventPayloadType(int payload_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 = {};
RTC_DCHECK_LE(0, payload_type);
RTC_DCHECK_GE(127, payload_type);
CodecInst codec = {0};
codec.plfreq = 8000;
codec.pltype = type;
codec.pltype = payload_type;
memcpy(codec.plname, "telephone-event", 16);
if (_rtpRtcpModule->RegisterSendPayload(codec) != 0) {
_rtpRtcpModule->DeRegisterSendPayload(codec.pltype);
@ -2272,12 +2242,6 @@ int Channel::SetSendTelephoneEventPayloadType(unsigned char type) {
return -1;
}
}
_sendTelephoneEventPayloadType = type;
return 0;
}
int Channel::GetSendTelephoneEventPayloadType(unsigned char& type) {
type = _sendTelephoneEventPayloadType;
return 0;
}
@ -3027,8 +2991,6 @@ uint32_t Channel::PrepareEncodeAndSend(int mixingFrequency) {
}
}
InsertInbandDtmfTone();
if (_includeAudioLevelIndication) {
size_t length =
_audioFrame.samples_per_channel_ * _audioFrame.num_channels_;
@ -3348,64 +3310,6 @@ int32_t Channel::MixAudioWithFile(AudioFrame& audioFrame, int mixingFrequency) {
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 (size_t sample = 0; sample < _audioFrame.samples_per_channel_;
sample++) {
for (size_t channel = 0; channel < _audioFrame.num_channels_; channel++) {
const size_t 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;
}
void Channel::UpdatePlayoutTimestamp(bool rtcp) {
uint32_t playout_timestamp = 0;

23
webrtc/voice_engine/channel.h
View File

@ -25,8 +25,6 @@
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/utility/include/file_player.h"
#include "webrtc/modules/utility/include/file_recorder.h"
#include "webrtc/voice_engine/dtmf_inband.h"
#include "webrtc/voice_engine/dtmf_inband_queue.h"
#include "webrtc/voice_engine/include/voe_audio_processing.h"
#include "webrtc/voice_engine/include/voe_network.h"
#include "webrtc/voice_engine/level_indicator.h"
@ -296,17 +294,9 @@ class Channel
// VoEVideoSyncExtended
int GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const;
// VoEDtmf
int SendTelephoneEventOutband(unsigned char eventCode,
int lengthMs,
int attenuationDb,
bool playDtmfEvent);
int SendTelephoneEventInband(unsigned char eventCode,
int lengthMs,
int attenuationDb,
bool playDtmfEvent);
int SetSendTelephoneEventPayloadType(unsigned char type);
int GetSendTelephoneEventPayloadType(unsigned char& type);
// DTMF
int SendTelephoneEventOutband(int event, int duration_ms);
int SetSendTelephoneEventPayloadType(int payload_type);
// VoEAudioProcessingImpl
int UpdateRxVadDetection(AudioFrame& audioFrame);
@ -464,7 +454,6 @@ class Channel
bool IsPacketInOrder(const RTPHeader& header) const;
bool IsPacketRetransmitted(const RTPHeader& header, bool in_order) const;
int ResendPackets(const uint16_t* sequence_numbers, int length);
int InsertInbandDtmfTone();
int32_t MixOrReplaceAudioWithFile(int mixingFrequency);
int32_t MixAudioWithFile(AudioFrame& audioFrame, int mixingFrequency);
void UpdatePlayoutTimestamp(bool rtcp);
@ -510,13 +499,10 @@ class Channel
int _outputFilePlayerId;
int _outputFileRecorderId;
bool _outputFileRecording;
DtmfInbandQueue _inbandDtmfQueue;
DtmfInband _inbandDtmfGenerator;
bool _outputExternalMedia;
VoEMediaProcess* _inputExternalMediaCallbackPtr;
VoEMediaProcess* _outputExternalMediaCallbackPtr;
uint32_t _timeStamp;
uint8_t _sendTelephoneEventPayloadType;
RemoteNtpTimeEstimator ntp_estimator_ GUARDED_BY(ts_stats_lock_);
@ -560,9 +546,6 @@ class Channel
float _panLeft;
float _panRight;
float _outputGain;
// VoEDtmf
bool _playOutbandDtmfEvent;
bool _playInbandDtmfEvent;
// VoeRTP_RTCP
uint32_t _lastLocalTimeStamp;
int8_t _lastPayloadType;

6
webrtc/voice_engine/channel_proxy.cc
View File

@ -148,11 +148,9 @@ bool ChannelProxy::SetSendTelephoneEventPayloadType(int payload_type) {
return channel()->SetSendTelephoneEventPayloadType(payload_type) == 0;
}
bool ChannelProxy::SendTelephoneEventOutband(uint8_t event,
uint32_t duration_ms) {
bool ChannelProxy::SendTelephoneEventOutband(int event, int duration_ms) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
return
channel()->SendTelephoneEventOutband(event, duration_ms, 10, false) == 0;
return channel()->SendTelephoneEventOutband(event, duration_ms) == 0;
}
void ChannelProxy::SetSink(std::unique_ptr<AudioSinkInterface> sink) {

2
webrtc/voice_engine/channel_proxy.h
View File

@ -68,7 +68,7 @@ class ChannelProxy {
virtual uint32_t GetDelayEstimate() const;
virtual bool SetSendTelephoneEventPayloadType(int payload_type);
virtual bool SendTelephoneEventOutband(uint8_t event, uint32_t duration_ms);
virtual bool SendTelephoneEventOutband(int event, int duration_ms);
virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink);

3
webrtc/voice_engine/dtmf_inband.h
View File

@ -17,6 +17,9 @@
namespace webrtc {
// TODO(solenberg): Used as a DTMF tone generator in voe::OutputMixer. Pull out
// the one in NetEq and use that instead? We don't need several
// implemenations of this.
class DtmfInband
{
public:

86
webrtc/voice_engine/dtmf_inband_queue.cc
View File

@ -1,86 +0,0 @@
/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/voice_engine/dtmf_inband_queue.h"
namespace webrtc {
DtmfInbandQueue::DtmfInbandQueue(int32_t id):
_id(id),
_nextEmptyIndex(0)
{
memset(_DtmfKey,0, sizeof(_DtmfKey));
memset(_DtmfLen,0, sizeof(_DtmfLen));
memset(_DtmfLevel,0, sizeof(_DtmfLevel));
}
DtmfInbandQueue::~DtmfInbandQueue()
{
}
int
DtmfInbandQueue::AddDtmf(uint8_t key, uint16_t len, uint8_t level)
{
rtc::CritScope lock(&_DtmfCritsect);
if (_nextEmptyIndex >= kDtmfInbandMax)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_id,-1),
"DtmfInbandQueue::AddDtmf() unable to add Dtmf tone");
return -1;
}
int32_t index = _nextEmptyIndex;
_DtmfKey[index] = key;
_DtmfLen[index] = len;
_DtmfLevel[index] = level;
_nextEmptyIndex++;
return 0;
}
int8_t
DtmfInbandQueue::NextDtmf(uint16_t* len, uint8_t* level)
{
rtc::CritScope lock(&_DtmfCritsect);
if(!PendingDtmf())
{
return -1;
}
int8_t nextDtmf = _DtmfKey[0];
*len=_DtmfLen[0];
*level=_DtmfLevel[0];
memmove(&(_DtmfKey[0]), &(_DtmfKey[1]),
_nextEmptyIndex*sizeof(uint8_t));
memmove(&(_DtmfLen[0]), &(_DtmfLen[1]),
_nextEmptyIndex*sizeof(uint16_t));
memmove(&(_DtmfLevel[0]), &(_DtmfLevel[1]),
_nextEmptyIndex*sizeof(uint8_t));
_nextEmptyIndex--;
return nextDtmf;
}
bool
DtmfInbandQueue::PendingDtmf()
{
rtc::CritScope lock(&_DtmfCritsect);
return _nextEmptyIndex > 0;
}
void
DtmfInbandQueue::ResetDtmf()
{
rtc::CritScope lock(&_DtmfCritsect);
_nextEmptyIndex = 0;
}
} // namespace webrtc

50
webrtc/voice_engine/dtmf_inband_queue.h
View File

@ -1,50 +0,0 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_VOICE_ENGINE_DTMF_INBAND_QUEUE_H
#define WEBRTC_VOICE_ENGINE_DTMF_INBAND_QUEUE_H
#include "webrtc/base/criticalsection.h"
#include "webrtc/typedefs.h"
#include "webrtc/voice_engine/voice_engine_defines.h"
namespace webrtc {
class DtmfInbandQueue
{
public:
DtmfInbandQueue(int32_t id);
virtual ~DtmfInbandQueue();
int AddDtmf(uint8_t DtmfKey, uint16_t len, uint8_t level);
int8_t NextDtmf(uint16_t* len, uint8_t* level);
bool PendingDtmf();
void ResetDtmf();
private:
enum {kDtmfInbandMax = 20};
int32_t _id;
rtc::CriticalSection _DtmfCritsect;
uint8_t _nextEmptyIndex;
uint8_t _DtmfKey[kDtmfInbandMax];
uint16_t _DtmfLen[kDtmfInbandMax];
uint8_t _DtmfLevel[kDtmfInbandMax];
};
} // namespace webrtc
#endif // WEBRTC_VOICE_ENGINE_DTMF_INBAND_QUEUE_H

21
webrtc/voice_engine/transmit_mixer.cc
View File

@ -205,7 +205,6 @@ TransmitMixer::TransmitMixer(uint32_t instanceId) :
external_postproc_ptr_(NULL),
external_preproc_ptr_(NULL),
_mute(false),
_remainingMuteMicTimeMs(0),
stereo_codec_(false),
swap_stereo_channels_(false)
{
@ -359,17 +358,6 @@ TransmitMixer::PrepareDemux(const void* audioSamples,
TypingDetection(keyPressed);
#endif
// --- Mute during DTMF tone if direct feedback is enabled
if (_remainingMuteMicTimeMs > 0)
{
AudioFrameOperations::Mute(_audioFrame);
_remainingMuteMicTimeMs -= 10;
if (_remainingMuteMicTimeMs < 0)
{
_remainingMuteMicTimeMs = 0;
}
}
// --- Mute signal
if (_mute)
{
@ -477,15 +465,6 @@ uint32_t TransmitMixer::CaptureLevel() const
return _captureLevel;
}
void
TransmitMixer::UpdateMuteMicrophoneTime(uint32_t lengthMs)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
"TransmitMixer::UpdateMuteMicrophoneTime(lengthMs=%d)",
lengthMs);
_remainingMuteMicTimeMs = lengthMs;
}
int32_t
TransmitMixer::StopSend()
{

4
webrtc/voice_engine/transmit_mixer.h
View File

@ -75,9 +75,6 @@ public:
int32_t StopSend();
// VoEDtmf
void UpdateMuteMicrophoneTime(uint32_t lengthMs);
// VoEExternalMedia
int RegisterExternalMediaProcessing(VoEMediaProcess* object,
ProcessingTypes type);
@ -226,7 +223,6 @@ private:
VoEMediaProcess* external_postproc_ptr_;
VoEMediaProcess* external_preproc_ptr_;
bool _mute;
int32_t _remainingMuteMicTimeMs;
bool stereo_codec_;
bool swap_stereo_channels_;
};

2
webrtc/voice_engine/voice_engine.gyp
View File

@ -54,8 +54,6 @@
'channel_proxy.h',
'dtmf_inband.cc',
'dtmf_inband.h',
'dtmf_inband_queue.cc',
'dtmf_inband_queue.h',
'level_indicator.cc',
'level_indicator.h',
'monitor_module.cc',