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Remove deprecated AudioCodingModule::Destroy.

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Have Channel hold a pointer rather than reference, and shorten the name.

TESTED=trybots
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2256004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4820 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
andrew@webrtc.org 2013-09-23 23:02:24 +00:00
parent 1112c30e1e
commit eb524d997b
7 changed files with 78 additions and 110 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

5
webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc
View File

@ -49,7 +49,7 @@ class AcmReceiverTest : public AudioPacketizationCallback,
void SetUp() {
ASSERT_TRUE(receiver_.get() != NULL);
ASSERT_TRUE(acm_ != NULL);
ASSERT_TRUE(acm_.get() != NULL);
for (int n = 0; n < ACMCodecDB::kNumCodecs; n++) {
ASSERT_EQ(0, ACMCodecDB::Codec(n, &codecs_[n]));
}
@ -69,7 +69,6 @@ class AcmReceiverTest : public AudioPacketizationCallback,
}
void TearDown() {
AudioCodingModule::Destroy(acm_);
}
void InsertOnePacketOfSilence(int codec_id) {
@ -145,7 +144,7 @@ class AcmReceiverTest : public AudioPacketizationCallback,
scoped_ptr<AcmReceiver> receiver_;
CodecInst codecs_[ACMCodecDB::kMaxNumCodecs];
AudioCodingModule* acm_;
scoped_ptr<AudioCodingModule> acm_;
WebRtcRTPHeader rtp_header_;
uint32_t timestamp_;
bool packet_sent_; // Set when SendData is called reset when inserting audio.

5
webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc
View File

@ -28,11 +28,6 @@ AudioCodingModule* AudioCodingModule::Create(int id, Clock* clock) {
return new acm1::AudioCodingModuleImpl(id, clock);
}
// Destroy module
void AudioCodingModule::Destroy(AudioCodingModule* module) {
delete module;
}
// Get number of supported codecs
int AudioCodingModule::NumberOfCodecs() {
return ACMCodecDB::kNumCodecs;

5
webrtc/modules/audio_coding/main/interface/audio_coding_module.h
View File

@ -82,16 +82,13 @@ class AudioCodingModule: public Module {
// Creation and destruction of a ACM.
//
// The second method is used for testing where a simulated clock can be
// injected into ACM. ACM will take the owner ship of the object clock and
// injected into ACM. ACM will take the ownership of the object clock and
// delete it when destroyed.
//
static AudioCodingModule* Create(int id);
static AudioCodingModule* Create(int id, Clock* clock);
virtual ~AudioCodingModule() {};
// TODO(ajm): Deprecated. Remove all calls to this unneeded method.
static void Destroy(AudioCodingModule* module);
///////////////////////////////////////////////////////////////////////////
// Utility functions
//

1
webrtc/modules/utility/source/coder.cc
View File

@ -28,7 +28,6 @@ AudioCoder::AudioCoder(uint32_t instanceID)
AudioCoder::~AudioCoder()
{
AudioCodingModule::Destroy(_acm);
}
int32_t AudioCoder::SetEncodeCodec(const CodecInst& codecInst,

3
webrtc/modules/utility/source/coder.h
View File

@ -13,6 +13,7 @@
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@ -49,7 +50,7 @@ protected:
const RTPFragmentationHeader* fragmentation);
private:
AudioCodingModule* _acm;
scoped_ptr<AudioCodingModule> _acm;
CodecInst _receiveCodec;

139
webrtc/voice_engine/channel.cc
View File

@ -469,11 +469,11 @@ Channel::OnInitializeDecoder(
receiveCodec.rate = rate;
strncpy(receiveCodec.plname, payloadName, RTP_PAYLOAD_NAME_SIZE - 1);
_audioCodingModule.Codec(payloadName, &dummyCodec, frequency, channels);
audio_coding_->Codec(payloadName, &dummyCodec, frequency, channels);
receiveCodec.pacsize = dummyCodec.pacsize;
// Register the new codec to the ACM
if (_audioCodingModule.RegisterReceiveCodec(receiveCodec) == -1)
if (audio_coding_->RegisterReceiveCodec(receiveCodec) == -1)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
VoEId(_instanceId, _channelId),
@ -625,9 +625,9 @@ Channel::OnReceivedPayloadData(const uint8_t* payloadData,
}
// Push the incoming payload (parsed and ready for decoding) into the ACM
if (_audioCodingModule.IncomingPacket(payloadData,
payloadSize,
*rtpHeader) != 0)
if (audio_coding_->IncomingPacket(payloadData,
payloadSize,
*rtpHeader) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CODING_MODULE_ERROR, kTraceWarning,
@ -643,7 +643,7 @@ Channel::OnReceivedPayloadData(const uint8_t* payloadData,
_rtpRtcpModule->RTT(rtp_receiver_->SSRC(), &round_trip_time,
NULL, NULL, NULL);
std::vector<uint16_t> nack_list = _audioCodingModule.GetNackList(
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
@ -674,8 +674,8 @@ int32_t Channel::GetAudioFrame(int32_t id, AudioFrame& audioFrame)
"Channel::GetAudioFrame(id=%d)", id);
// Get 10ms raw PCM data from the ACM (mixer limits output frequency)
if (_audioCodingModule.PlayoutData10Ms(audioFrame.sample_rate_hz_,
&audioFrame) == -1)
if (audio_coding_->PlayoutData10Ms(audioFrame.sample_rate_hz_,
&audioFrame) == -1)
{
WEBRTC_TRACE(kTraceError, kTraceVoice,
VoEId(_instanceId,_channelId),
@ -782,12 +782,12 @@ Channel::NeededFrequency(int32_t id)
int highestNeeded = 0;
// Determine highest needed receive frequency
int32_t receiveFrequency = _audioCodingModule.ReceiveFrequency();
int32_t receiveFrequency = audio_coding_->ReceiveFrequency();
// Return the bigger of playout and receive frequency in the ACM.
if (_audioCodingModule.PlayoutFrequency() > receiveFrequency)
if (audio_coding_->PlayoutFrequency() > receiveFrequency)
{
highestNeeded = _audioCodingModule.PlayoutFrequency();
highestNeeded = audio_coding_->PlayoutFrequency();
}
else
{
@ -917,7 +917,7 @@ Channel::Channel(int32_t channelId,
VoEModuleId(instanceId, channelId), Clock::GetRealTimeClock(), this,
this, this, rtp_payload_registry_.get())),
telephone_event_handler_(rtp_receiver_->GetTelephoneEventHandler()),
_audioCodingModule(*config.Get<AudioCodingModuleFactory>().Create(
audio_coding_(config.Get<AudioCodingModuleFactory>().Create(
VoEModuleId(instanceId, channelId))),
_rtpDumpIn(*RtpDump::CreateRtpDump()),
_rtpDumpOut(*RtpDump::CreateRtpDump()),
@ -1071,14 +1071,14 @@ Channel::~Channel()
// 1. De-register callbacks in modules
// 2. De-register modules in process thread
// 3. Destroy modules
if (_audioCodingModule.RegisterTransportCallback(NULL) == -1)
if (audio_coding_->RegisterTransportCallback(NULL) == -1)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
VoEId(_instanceId,_channelId),
"~Channel() failed to de-register transport callback"
" (Audio coding module)");
}
if (_audioCodingModule.RegisterVADCallback(NULL) == -1)
if (audio_coding_->RegisterVADCallback(NULL) == -1)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
VoEId(_instanceId,_channelId),
@ -1092,10 +1092,6 @@ Channel::~Channel()
VoEId(_instanceId,_channelId),
"~Channel() failed to deregister RTP/RTCP module");
}
// Destroy modules
AudioCodingModule::Destroy(&_audioCodingModule);
// End of modules shutdown
// Delete other objects
@ -1140,12 +1136,12 @@ Channel::Init()
}
// --- ACM initialization
if ((_audioCodingModule.InitializeReceiver() == -1) ||
if ((audio_coding_->InitializeReceiver() == -1) ||
#ifdef WEBRTC_CODEC_AVT
// out-of-band Dtmf tones are played out by default
(_audioCodingModule.SetDtmfPlayoutStatus(true) == -1) ||
(audio_coding_->SetDtmfPlayoutStatus(true) == -1) ||
#endif
(_audioCodingModule.InitializeSender() == -1))
(audio_coding_->InitializeSender() == -1))
{
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
@ -1172,8 +1168,8 @@ Channel::Init()
// --- Register all permanent callbacks
const bool fail =
(_audioCodingModule.RegisterTransportCallback(this) == -1) ||
(_audioCodingModule.RegisterVADCallback(this) == -1);
(audio_coding_->RegisterTransportCallback(this) == -1) ||
(audio_coding_->RegisterVADCallback(this) == -1);
if (fail)
{
@ -1192,7 +1188,7 @@ Channel::Init()
for (int idx = 0; idx < nSupportedCodecs; idx++)
{
// Open up the RTP/RTCP receiver for all supported codecs
if ((_audioCodingModule.Codec(idx, &codec) == -1) ||
if ((audio_coding_->Codec(idx, &codec) == -1) ||
(rtp_receiver_->RegisterReceivePayload(
codec.plname,
codec.pltype,
@ -1227,7 +1223,7 @@ Channel::Init()
if (!STR_CASE_CMP(codec.plname, "telephone-event"))
{
if ((_rtpRtcpModule->RegisterSendPayload(codec) == -1) ||
(_audioCodingModule.RegisterReceiveCodec(codec) == -1))
(audio_coding_->RegisterReceiveCodec(codec) == -1))
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
VoEId(_instanceId,_channelId),
@ -1239,8 +1235,8 @@ Channel::Init()
if (!STR_CASE_CMP(codec.plname, "CN"))
{
if ((_audioCodingModule.RegisterSendCodec(codec) == -1) ||
(_audioCodingModule.RegisterReceiveCodec(codec) == -1) ||
if ((audio_coding_->RegisterSendCodec(codec) == -1) ||
(audio_coding_->RegisterReceiveCodec(codec) == -1) ||
(_rtpRtcpModule->RegisterSendPayload(codec) == -1))
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
@ -1255,7 +1251,7 @@ Channel::Init()
// We will not receive an OnInitializeDecoder() callback for RED.
if (!STR_CASE_CMP(codec.plname, "RED"))
{
if (_audioCodingModule.RegisterReceiveCodec(codec) == -1)
if (audio_coding_->RegisterReceiveCodec(codec) == -1)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
VoEId(_instanceId,_channelId),
@ -1537,7 +1533,7 @@ Channel::SetNetEQPlayoutMode(NetEqModes mode)
playoutMode = off;
break;
}
if (_audioCodingModule.SetPlayoutMode(playoutMode) != 0)
if (audio_coding_->SetPlayoutMode(playoutMode) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
@ -1550,7 +1546,7 @@ Channel::SetNetEQPlayoutMode(NetEqModes mode)
int32_t
Channel::GetNetEQPlayoutMode(NetEqModes& mode)
{
const AudioPlayoutMode playoutMode = _audioCodingModule.PlayoutMode();
const AudioPlayoutMode playoutMode = audio_coding_->PlayoutMode();
switch (playoutMode)
{
case voice:
@ -1655,13 +1651,13 @@ Channel::DeRegisterVoiceEngineObserver()
int32_t
Channel::GetSendCodec(CodecInst& codec)
{
return (_audioCodingModule.SendCodec(&codec));
return (audio_coding_->SendCodec(&codec));
}
int32_t
Channel::GetRecCodec(CodecInst& codec)
{
return (_audioCodingModule.ReceiveCodec(&codec));
return (audio_coding_->ReceiveCodec(&codec));
}
int32_t
@ -1670,7 +1666,7 @@ Channel::SetSendCodec(const CodecInst& codec)
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SetSendCodec()");
if (_audioCodingModule.RegisterSendCodec(codec) != 0)
if (audio_coding_->RegisterSendCodec(codec) != 0)
{
WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,_channelId),
"SetSendCodec() failed to register codec to ACM");
@ -1707,7 +1703,7 @@ Channel::SetVADStatus(bool enableVAD, ACMVADMode mode, bool disableDTX)
"Channel::SetVADStatus(mode=%d)", mode);
// To disable VAD, DTX must be disabled too
disableDTX = ((enableVAD == false) ? true : disableDTX);
if (_audioCodingModule.SetVAD(!disableDTX, enableVAD, mode) != 0)
if (audio_coding_->SetVAD(!disableDTX, enableVAD, mode) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
@ -1722,7 +1718,7 @@ Channel::GetVADStatus(bool& enabledVAD, ACMVADMode& mode, bool& disabledDTX)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::GetVADStatus");
if (_audioCodingModule.VAD(&disabledDTX, &enabledVAD, &mode) != 0)
if (audio_coding_->VAD(&disabledDTX, &enabledVAD, &mode) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
@ -1779,7 +1775,7 @@ Channel::SetRecPayloadType(const CodecInst& codec)
"failed");
return -1;
}
if (_audioCodingModule.UnregisterReceiveCodec(rxCodec.pltype) != 0)
if (audio_coding_->UnregisterReceiveCodec(rxCodec.pltype) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
@ -1811,10 +1807,10 @@ Channel::SetRecPayloadType(const CodecInst& codec)
return -1;
}
}
if (_audioCodingModule.RegisterReceiveCodec(codec) != 0)
if (audio_coding_->RegisterReceiveCodec(codec) != 0)
{
_audioCodingModule.UnregisterReceiveCodec(codec.pltype);
if (_audioCodingModule.RegisterReceiveCodec(codec) != 0)
audio_coding_->UnregisterReceiveCodec(codec.pltype);
if (audio_coding_->RegisterReceiveCodec(codec) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
@ -1904,7 +1900,7 @@ Channel::SetSendCNPayloadType(int type, PayloadFrequencies frequency)
else if (frequency == kFreq16000Hz)
samplingFreqHz = 16000;
if (_audioCodingModule.Codec("CN", &codec, samplingFreqHz, kMono) == -1)
if (audio_coding_->Codec("CN", &codec, samplingFreqHz, kMono) == -1)
{
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
@ -1916,7 +1912,7 @@ Channel::SetSendCNPayloadType(int type, PayloadFrequencies frequency)
// Modify the payload type (must be set to dynamic range)
codec.pltype = type;
if (_audioCodingModule.RegisterSendCodec(codec) != 0)
if (audio_coding_->RegisterSendCodec(codec) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
@ -1946,7 +1942,7 @@ Channel::SetISACInitTargetRate(int rateBps, bool useFixedFrameSize)
"Channel::SetISACInitTargetRate()");
CodecInst sendCodec;
if (_audioCodingModule.SendCodec(&sendCodec) == -1)
if (audio_coding_->SendCodec(&sendCodec) == -1)
{
_engineStatisticsPtr->SetLastError(
VE_CODEC_ERROR, kTraceError,
@ -1995,7 +1991,7 @@ Channel::SetISACInitTargetRate(int rateBps, bool useFixedFrameSize)
initFrameSizeMsec = (uint8_t)(sendCodec.pacsize / 32); // 30ms
}
if (_audioCodingModule.ConfigISACBandwidthEstimator(
if (audio_coding_->ConfigISACBandwidthEstimator(
initFrameSizeMsec, rateBps, useFixedFrameSize) == -1)
{
_engineStatisticsPtr->SetLastError(
@ -2014,7 +2010,7 @@ Channel::SetISACMaxRate(int rateBps)
"Channel::SetISACMaxRate()");
CodecInst sendCodec;
if (_audioCodingModule.SendCodec(&sendCodec) == -1)
if (audio_coding_->SendCodec(&sendCodec) == -1)
{
_engineStatisticsPtr->SetLastError(
VE_CODEC_ERROR, kTraceError,
@ -2061,7 +2057,7 @@ Channel::SetISACMaxRate(int rateBps)
// Set the maximum instantaneous rate of iSAC (works for both adaptive
// and non-adaptive mode)
if (_audioCodingModule.SetISACMaxRate(rateBps) == -1)
if (audio_coding_->SetISACMaxRate(rateBps) == -1)
{
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
@ -2078,7 +2074,7 @@ Channel::SetISACMaxPayloadSize(int sizeBytes)
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SetISACMaxPayloadSize()");
CodecInst sendCodec;
if (_audioCodingModule.SendCodec(&sendCodec) == -1)
if (audio_coding_->SendCodec(&sendCodec) == -1)
{
_engineStatisticsPtr->SetLastError(
VE_CODEC_ERROR, kTraceError,
@ -2122,7 +2118,7 @@ Channel::SetISACMaxPayloadSize(int sizeBytes)
return -1;
}
if (_audioCodingModule.SetISACMaxPayloadSize(sizeBytes) == -1)
if (audio_coding_->SetISACMaxPayloadSize(sizeBytes) == -1)
{
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
@ -3151,7 +3147,7 @@ Channel::SetDtmfPlayoutStatus(bool enable)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::SetDtmfPlayoutStatus()");
if (_audioCodingModule.SetDtmfPlayoutStatus(enable) != 0)
if (audio_coding_->SetDtmfPlayoutStatus(enable) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CODING_MODULE_ERROR, kTraceWarning,
@ -3164,7 +3160,7 @@ Channel::SetDtmfPlayoutStatus(bool enable)
bool
Channel::DtmfPlayoutStatus() const
{
return _audioCodingModule.DtmfPlayoutStatus();
return audio_coding_->DtmfPlayoutStatus();
}
int
@ -3962,8 +3958,7 @@ Channel::GetRTPStatistics(
"RTP/RTCP module");
}
const int32_t playoutFrequency =
_audioCodingModule.PlayoutFrequency();
const int32_t playoutFrequency = audio_coding_->PlayoutFrequency();
if (playoutFrequency > 0)
{
// Scale RTP statistics given the current playout frequency
@ -4169,7 +4164,7 @@ int Channel::SetFECStatus(bool enable, int redPayloadtype) {
}
}
if (_audioCodingModule.SetFECStatus(enable) != 0) {
if (audio_coding_->SetFECStatus(enable) != 0) {
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
"SetFECStatus() failed to set FEC state in the ACM");
@ -4181,7 +4176,7 @@ int Channel::SetFECStatus(bool enable, int redPayloadtype) {
int
Channel::GetFECStatus(bool& enabled, int& redPayloadtype)
{
enabled = _audioCodingModule.FECStatus();
enabled = audio_coding_->FECStatus();
if (enabled)
{
int8_t payloadType(0);
@ -4211,9 +4206,9 @@ void Channel::SetNACKStatus(bool enable, int maxNumberOfPackets) {
rtp_receive_statistics_->SetMaxReorderingThreshold(maxNumberOfPackets);
rtp_receiver_->SetNACKStatus(enable ? kNackRtcp : kNackOff);
if (enable)
_audioCodingModule.EnableNack(maxNumberOfPackets);
audio_coding_->EnableNack(maxNumberOfPackets);
else
_audioCodingModule.DisableNack();
audio_coding_->DisableNack();
}
// Called when we are missing one or more packets.
@ -4525,7 +4520,7 @@ Channel::EncodeAndSend()
// The ACM resamples internally.
_audioFrame.timestamp_ = _timeStamp;
if (_audioCodingModule.Add10MsData((AudioFrame&)_audioFrame) != 0)
if (audio_coding_->Add10MsData((AudioFrame&)_audioFrame) != 0)
{
WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::EncodeAndSend() ACM encoding failed");
@ -4538,7 +4533,7 @@ Channel::EncodeAndSend()
// This call will trigger AudioPacketizationCallback::SendData if encoding
// is done and payload is ready for packetization and transmission.
return _audioCodingModule.Process();
return audio_coding_->Process();
}
int Channel::RegisterExternalMediaProcessing(
@ -4702,7 +4697,7 @@ Channel::GetNetworkStatistics(NetworkStatistics& stats)
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::GetNetworkStatistics()");
ACMNetworkStatistics acm_stats;
int return_value = _audioCodingModule.NetworkStatistics(&acm_stats);
int return_value = audio_coding_->NetworkStatistics(&acm_stats);
if (return_value >= 0) {
memcpy(&stats, &acm_stats, sizeof(NetworkStatistics));
}
@ -4736,7 +4731,7 @@ int Channel::SetInitialPlayoutDelay(int delay_ms)
"SetInitialPlayoutDelay() invalid min delay");
return -1;
}
if (_audioCodingModule.SetInitialPlayoutDelay(delay_ms) != 0)
if (audio_coding_->SetInitialPlayoutDelay(delay_ms) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
@ -4760,7 +4755,7 @@ Channel::SetMinimumPlayoutDelay(int delayMs)
"SetMinimumPlayoutDelay() invalid min delay");
return -1;
}
if (_audioCodingModule.SetMinimumPlayoutDelay(delayMs) != 0)
if (audio_coding_->SetMinimumPlayoutDelay(delayMs) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
@ -4773,7 +4768,7 @@ Channel::SetMinimumPlayoutDelay(int delayMs)
void Channel::UpdatePlayoutTimestamp(bool rtcp) {
uint32_t playout_timestamp = 0;
if (_audioCodingModule.PlayoutTimestamp(&playout_timestamp) == -1) {
if (audio_coding_->PlayoutTimestamp(&playout_timestamp) == -1) {
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::UpdatePlayoutTimestamp() failed to read playout"
" timestamp from the ACM");
@ -4794,9 +4789,9 @@ void Channel::UpdatePlayoutTimestamp(bool rtcp) {
return;
}
int32_t playout_frequency = _audioCodingModule.PlayoutFrequency();
int32_t playout_frequency = audio_coding_->PlayoutFrequency();
CodecInst current_recive_codec;
if (_audioCodingModule.ReceiveCodec(&current_recive_codec) == 0) {
if (audio_coding_->ReceiveCodec(&current_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) {
@ -5132,15 +5127,15 @@ void Channel::UpdatePacketDelay(uint32_t rtp_timestamp,
rtp_timestamp, sequence_number);
// Get frequency of last received payload
int rtp_receive_frequency = _audioCodingModule.ReceiveFrequency();
int rtp_receive_frequency = audio_coding_->ReceiveFrequency();
CodecInst current_receive_codec;
if (_audioCodingModule.ReceiveCodec(&current_receive_codec) != 0) {
if (audio_coding_->ReceiveCodec(&current_receive_codec) != 0) {
return;
}
// Update the least required delay.
least_required_delay_ms_ = _audioCodingModule.LeastRequiredDelayMs();
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
@ -5202,7 +5197,7 @@ Channel::RegisterReceiveCodecsToRTPModule()
for (int idx = 0; idx < nSupportedCodecs; idx++)
{
// Open up the RTP/RTCP receiver for all supported codecs
if ((_audioCodingModule.Codec(idx, &codec) == -1) ||
if ((audio_coding_->Codec(idx, &codec) == -1) ||
(rtp_receiver_->RegisterReceivePayload(
codec.plname,
codec.pltype,
@ -5265,7 +5260,7 @@ int Channel::SetSecondarySendCodec(const CodecInst& codec,
"SetSecondarySendCodec() Failed to register RED ACM");
return -1;
}
if (_audioCodingModule.RegisterSecondarySendCodec(codec) < 0) {
if (audio_coding_->RegisterSecondarySendCodec(codec) < 0) {
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
"SetSecondarySendCodec() Failed to register secondary send codec in "
@ -5277,11 +5272,11 @@ int Channel::SetSecondarySendCodec(const CodecInst& codec,
}
void Channel::RemoveSecondarySendCodec() {
_audioCodingModule.UnregisterSecondarySendCodec();
audio_coding_->UnregisterSecondarySendCodec();
}
int Channel::GetSecondarySendCodec(CodecInst* codec) {
if (_audioCodingModule.SecondarySendCodec(codec) < 0) {
if (audio_coding_->SecondarySendCodec(codec) < 0) {
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
"GetSecondarySendCodec() Failed to get secondary sent codec from ACM");
@ -5298,7 +5293,7 @@ int Channel::SetRedPayloadType(int red_payload_type) {
// Get default RED settings from the ACM database
const int num_codecs = AudioCodingModule::NumberOfCodecs();
for (int idx = 0; idx < num_codecs; idx++) {
_audioCodingModule.Codec(idx, &codec);
audio_coding_->Codec(idx, &codec);
if (!STR_CASE_CMP(codec.plname, "RED")) {
found_red = true;
break;
@ -5313,7 +5308,7 @@ int Channel::SetRedPayloadType(int red_payload_type) {
}
codec.pltype = red_payload_type;
if (_audioCodingModule.RegisterSendCodec(codec) < 0) {
if (audio_coding_->RegisterSendCodec(codec) < 0) {
_engineStatisticsPtr->SetLastError(
VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
"SetRedPayloadType() RED registration in ACM module failed");

30
webrtc/voice_engine/channel.h
View File

@ -33,8 +33,8 @@
#include "webrtc/voice_engine/include/voe_dtmf.h"
#endif
namespace webrtc
{
namespace webrtc {
class AudioDeviceModule;
class Config;
class CriticalSectionWrapper;
@ -56,8 +56,8 @@ struct CallStatistics;
struct ReportBlock;
struct SenderInfo;
namespace voe
{
namespace voe {
class Statistics;
class TransmitMixer;
class OutputMixer;
@ -77,7 +77,6 @@ class Channel:
public:
enum {KNumSocketThreads = 1};
enum {KNumberOfSocketBuffers = 8};
public:
virtual ~Channel();
static int32_t CreateChannel(Channel*& channel,
int32_t channelId,
@ -95,7 +94,6 @@ public:
CriticalSectionWrapper* callbackCritSect);
int32_t UpdateLocalTimeStamp();
public:
// API methods
// VoEBase
@ -288,7 +286,6 @@ public:
unsigned short payloadSize);
uint32_t LastRemoteTimeStamp() { return _lastRemoteTimeStamp; }
public:
// From AudioPacketizationCallback in the ACM
int32_t SendData(FrameType frameType,
uint8_t payloadType,
@ -299,10 +296,8 @@ public:
// From ACMVADCallback in the ACM
int32_t InFrameType(int16_t frameType);
public:
int32_t OnRxVadDetected(int vadDecision);
public:
// From RtpData in the RTP/RTCP module
int32_t OnReceivedPayloadData(const uint8_t* payloadData,
uint16_t payloadSize,
@ -310,7 +305,6 @@ public:
bool OnRecoveredPacket(const uint8_t* packet, int packet_length);
public:
// From RtpFeedback in the RTP/RTCP module
int32_t OnInitializeDecoder(
int32_t id,
@ -335,7 +329,6 @@ public:
void ResetStatistics(uint32_t ssrc);
public:
// From RtcpFeedback in the RTP/RTCP module
void OnApplicationDataReceived(int32_t id,
uint8_t subType,
@ -343,7 +336,6 @@ public:
uint16_t length,
const uint8_t* data);
public:
// From RtpAudioFeedback in the RTP/RTCP module
void OnReceivedTelephoneEvent(int32_t id,
uint8_t event,
@ -354,21 +346,17 @@ public:
uint16_t lengthMs,
uint8_t volume);
public:
// From Transport (called by the RTP/RTCP module)
int SendPacket(int /*channel*/, const void *data, int len);
int SendRTCPPacket(int /*channel*/, const void *data, int len);
public:
// From MixerParticipant
int32_t GetAudioFrame(int32_t id, AudioFrame& audioFrame);
int32_t NeededFrequency(int32_t id);
public:
// From MonitorObserver
void OnPeriodicProcess();
public:
// From FileCallback
void PlayNotification(int32_t id,
uint32_t durationMs);
@ -377,7 +365,6 @@ public:
void PlayFileEnded(int32_t id);
void RecordFileEnded(int32_t id);
public:
uint32_t InstanceId() const
{
return _instanceId;
@ -457,23 +444,21 @@ private:
int ApmProcessRx(AudioFrame& audioFrame);
int SetRedPayloadType(int red_payload_type);
private:
CriticalSectionWrapper& _fileCritSect;
CriticalSectionWrapper& _callbackCritSect;
uint32_t _instanceId;
int32_t _channelId;
private:
scoped_ptr<RtpHeaderParser> rtp_header_parser_;
scoped_ptr<RTPPayloadRegistry> rtp_payload_registry_;
scoped_ptr<ReceiveStatistics> rtp_receive_statistics_;
scoped_ptr<RtpReceiver> rtp_receiver_;
TelephoneEventHandler* telephone_event_handler_;
scoped_ptr<RtpRtcp> _rtpRtcpModule;
AudioCodingModule& _audioCodingModule;
scoped_ptr<AudioCodingModule> audio_coding_;
RtpDump& _rtpDumpIn;
RtpDump& _rtpDumpOut;
private:
AudioLevel _outputAudioLevel;
bool _externalTransport;
AudioFrame _audioFrame;
@ -509,7 +494,6 @@ private:
uint16_t send_sequence_number_;
uint8_t restored_packet_[kVoiceEngineMaxIpPacketSizeBytes];
private:
// uses
Statistics* _engineStatisticsPtr;
OutputMixer* _outputMixerPtr;
@ -527,7 +511,6 @@ private:
int32_t _sendFrameType; // Send data is voice, 1-voice, 0-otherwise
VoERTPObserver* _rtpObserverPtr;
VoERTCPObserver* _rtcpObserverPtr;
private:
// VoEBase
bool _outputIsOnHold;
bool _externalPlayout;
@ -581,7 +564,6 @@ private:
};
} // namespace voe
} // namespace webrtc
#endif // WEBRTC_VOICE_ENGINE_CHANNEL_H