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/*
* libjingle
* Copyright 2010 Google Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
*
* 1. Redistributions of source code must retain the above copyright notice,
* this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright notice,
* this list of conditions and the following disclaimer in the documentation
* and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_
#define TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_
#include <list>
#include <map>
#include <vector>
#include "talk/media/base/codec.h"
#include "talk/media/base/rtputils.h"
#include "talk/media/webrtc/fakewebrtccommon.h"
#include "talk/media/webrtc/webrtcvoe.h"
#include "webrtc/base/basictypes.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/stringutils.h"
#include "webrtc/config.h"
#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
namespace cricket {
static const int kOpusBandwidthNb = 4000;
static const int kOpusBandwidthMb = 6000;
static const int kOpusBandwidthWb = 8000;
static const int kOpusBandwidthSwb = 12000;
static const int kOpusBandwidthFb = 20000;
#define WEBRTC_CHECK_CHANNEL(channel) \
if (channels_.find(channel) == channels_.end()) return -1;
class FakeAudioProcessing : public webrtc::AudioProcessing {
public:
FakeAudioProcessing() : experimental_ns_enabled_(false) {}
WEBRTC_STUB(Initialize, ())
WEBRTC_STUB(Initialize, (
int input_sample_rate_hz,
int output_sample_rate_hz,
int reverse_sample_rate_hz,
webrtc::AudioProcessing::ChannelLayout input_layout,
webrtc::AudioProcessing::ChannelLayout output_layout,
webrtc::AudioProcessing::ChannelLayout reverse_layout));
WEBRTC_STUB(Initialize, (
const webrtc::ProcessingConfig& processing_config));
WEBRTC_VOID_FUNC(SetExtraOptions, (const webrtc::Config& config)) {
experimental_ns_enabled_ = config.Get<webrtc::ExperimentalNs>().enabled;
}
WEBRTC_STUB_CONST(input_sample_rate_hz, ());
WEBRTC_STUB_CONST(proc_sample_rate_hz, ());
WEBRTC_STUB_CONST(proc_split_sample_rate_hz, ());
WEBRTC_STUB_CONST(num_input_channels, ());
WEBRTC_STUB_CONST(num_output_channels, ());
WEBRTC_STUB_CONST(num_reverse_channels, ());
WEBRTC_VOID_STUB(set_output_will_be_muted, (bool muted));
WEBRTC_STUB(ProcessStream, (webrtc::AudioFrame* frame));
WEBRTC_STUB(ProcessStream, (
const float* const* src,
size_t samples_per_channel,
int input_sample_rate_hz,
webrtc::AudioProcessing::ChannelLayout input_layout,
int output_sample_rate_hz,
webrtc::AudioProcessing::ChannelLayout output_layout,
float* const* dest));
WEBRTC_STUB(ProcessStream,
(const float* const* src,
const webrtc::StreamConfig& input_config,
const webrtc::StreamConfig& output_config,
float* const* dest));
WEBRTC_STUB(AnalyzeReverseStream, (webrtc::AudioFrame* frame));
WEBRTC_STUB(ProcessReverseStream, (webrtc::AudioFrame * frame));
WEBRTC_STUB(AnalyzeReverseStream, (
const float* const* data,
size_t samples_per_channel,
int sample_rate_hz,
webrtc::AudioProcessing::ChannelLayout layout));
WEBRTC_STUB(ProcessReverseStream,
(const float* const* src,
const webrtc::StreamConfig& reverse_input_config,
const webrtc::StreamConfig& reverse_output_config,
float* const* dest));
WEBRTC_STUB(set_stream_delay_ms, (int delay));
WEBRTC_STUB_CONST(stream_delay_ms, ());
WEBRTC_BOOL_STUB_CONST(was_stream_delay_set, ());
WEBRTC_VOID_STUB(set_stream_key_pressed, (bool key_pressed));
WEBRTC_VOID_STUB(set_delay_offset_ms, (int offset));
WEBRTC_STUB_CONST(delay_offset_ms, ());
WEBRTC_STUB(StartDebugRecording, (const char filename[kMaxFilenameSize]));
WEBRTC_STUB(StartDebugRecording, (FILE* handle));
WEBRTC_STUB(StopDebugRecording, ());
WEBRTC_VOID_STUB(UpdateHistogramsOnCallEnd, ());
webrtc::EchoCancellation* echo_cancellation() const override { return NULL; }
webrtc::EchoControlMobile* echo_control_mobile() const override {
return NULL;
}
webrtc::GainControl* gain_control() const override { return NULL; }
webrtc::HighPassFilter* high_pass_filter() const override { return NULL; }
webrtc::LevelEstimator* level_estimator() const override { return NULL; }
webrtc::NoiseSuppression* noise_suppression() const override { return NULL; }
webrtc::VoiceDetection* voice_detection() const override { return NULL; }
bool experimental_ns_enabled() {
return experimental_ns_enabled_;
}
private:
bool experimental_ns_enabled_;
};
class FakeWebRtcVoiceEngine
: public webrtc::VoEAudioProcessing,
public webrtc::VoEBase, public webrtc::VoECodec,
public webrtc::VoEHardware,
public webrtc::VoENetwork, public webrtc::VoERTP_RTCP,
public webrtc::VoEVolumeControl {
public:
struct Channel {
explicit Channel()
: external_transport(false),
send(false),
playout(false),
volume_scale(1.0),
vad(false),
codec_fec(false),
max_encoding_bandwidth(0),
opus_dtx(false),
red(false),
nack(false),
cn8_type(13),
cn16_type(105),
red_type(117),
nack_max_packets(0),
send_ssrc(0),
associate_send_channel(-1),
recv_codecs(),
neteq_capacity(-1),
neteq_fast_accelerate(false) {
memset(&send_codec, 0, sizeof(send_codec));
}
bool external_transport;
bool send;
bool playout;
float volume_scale;
bool vad;
bool codec_fec;
int max_encoding_bandwidth;
bool opus_dtx;
bool red;
bool nack;
int cn8_type;
int cn16_type;
int red_type;
int nack_max_packets;
uint32_t send_ssrc;
int associate_send_channel;
std::vector<webrtc::CodecInst> recv_codecs;
webrtc::CodecInst send_codec;
webrtc::PacketTime last_rtp_packet_time;
std::list<std::string> packets;
int neteq_capacity;
bool neteq_fast_accelerate;
};
FakeWebRtcVoiceEngine()
: inited_(false),
last_channel_(-1),
fail_create_channel_(false),
num_set_send_codecs_(0),
ec_enabled_(false),
ec_metrics_enabled_(false),
cng_enabled_(false),
ns_enabled_(false),
agc_enabled_(false),
highpass_filter_enabled_(false),
stereo_swapping_enabled_(false),
typing_detection_enabled_(false),
ec_mode_(webrtc::kEcDefault),
aecm_mode_(webrtc::kAecmSpeakerphone),
ns_mode_(webrtc::kNsDefault),
agc_mode_(webrtc::kAgcDefault),
observer_(NULL),
playout_fail_channel_(-1),
send_fail_channel_(-1),
recording_sample_rate_(-1),
playout_sample_rate_(-1) {
memset(&agc_config_, 0, sizeof(agc_config_));
}
~FakeWebRtcVoiceEngine() {
RTC_CHECK(channels_.empty());
}
bool ec_metrics_enabled() const { return ec_metrics_enabled_; }
bool IsInited() const { return inited_; }
int GetLastChannel() const { return last_channel_; }
int GetNumChannels() const { return static_cast<int>(channels_.size()); }
uint32_t GetLocalSSRC(int channel) {
return channels_[channel]->send_ssrc;
}
bool GetPlayout(int channel) {
return channels_[channel]->playout;
}
bool GetSend(int channel) {
return channels_[channel]->send;
}
bool GetVAD(int channel) {
return channels_[channel]->vad;
}
bool GetOpusDtx(int channel) {
return channels_[channel]->opus_dtx;
}
bool GetRED(int channel) {
return channels_[channel]->red;
}
bool GetCodecFEC(int channel) {
return channels_[channel]->codec_fec;
}
int GetMaxEncodingBandwidth(int channel) {
return channels_[channel]->max_encoding_bandwidth;
}
bool GetNACK(int channel) {
return channels_[channel]->nack;
}
int GetNACKMaxPackets(int channel) {
return channels_[channel]->nack_max_packets;
}
const webrtc::PacketTime& GetLastRtpPacketTime(int channel) {
RTC_DCHECK(channels_.find(channel) != channels_.end());
return channels_[channel]->last_rtp_packet_time;
}
int GetSendCNPayloadType(int channel, bool wideband) {
return (wideband) ?
channels_[channel]->cn16_type :
channels_[channel]->cn8_type;
}
int GetSendREDPayloadType(int channel) {
return channels_[channel]->red_type;
}
bool CheckPacket(int channel, const void* data, size_t len) {
bool result = !CheckNoPacket(channel);
if (result) {
std::string packet = channels_[channel]->packets.front();
result = (packet == std::string(static_cast<const char*>(data), len));
channels_[channel]->packets.pop_front();
}
return result;
}
bool CheckNoPacket(int channel) {
return channels_[channel]->packets.empty();
}
void TriggerCallbackOnError(int channel_num, int err_code) {
RTC_DCHECK(observer_ != NULL);
observer_->CallbackOnError(channel_num, err_code);
}
void set_playout_fail_channel(int channel) {
playout_fail_channel_ = channel;
}
void set_send_fail_channel(int channel) {
send_fail_channel_ = channel;
}
void set_fail_create_channel(bool fail_create_channel) {
fail_create_channel_ = fail_create_channel;
}
int AddChannel(const webrtc::Config& config) {
if (fail_create_channel_) {
return -1;
}
Channel* ch = new Channel();
auto db = webrtc::acm2::RentACodec::Database();
ch->recv_codecs.assign(db.begin(), db.end());
if (config.Get<webrtc::NetEqCapacityConfig>().enabled) {
ch->neteq_capacity = config.Get<webrtc::NetEqCapacityConfig>().capacity;
}
ch->neteq_fast_accelerate =
config.Get<webrtc::NetEqFastAccelerate>().enabled;
channels_[++last_channel_] = ch;
return last_channel_;
}
int GetNumSetSendCodecs() const { return num_set_send_codecs_; }
int GetAssociateSendChannel(int channel) {
return channels_[channel]->associate_send_channel;
}
WEBRTC_STUB(Release, ());
// webrtc::VoEBase
WEBRTC_FUNC(RegisterVoiceEngineObserver, (
webrtc::VoiceEngineObserver& observer)) {
observer_ = &observer;
return 0;
}
WEBRTC_STUB(DeRegisterVoiceEngineObserver, ());
WEBRTC_FUNC(Init, (webrtc::AudioDeviceModule* adm,
webrtc::AudioProcessing* audioproc)) {
inited_ = true;
return 0;
}
WEBRTC_FUNC(Terminate, ()) {
inited_ = false;
return 0;
}
webrtc::AudioProcessing* audio_processing() override {
return &audio_processing_;
}
WEBRTC_FUNC(CreateChannel, ()) {
webrtc::Config empty_config;
return AddChannel(empty_config);
}
WEBRTC_FUNC(CreateChannel, (const webrtc::Config& config)) {
return AddChannel(config);
}
WEBRTC_FUNC(DeleteChannel, (int channel)) {
WEBRTC_CHECK_CHANNEL(channel);
for (const auto& ch : channels_) {
if (ch.second->associate_send_channel == channel) {
ch.second->associate_send_channel = -1;
}
}
delete channels_[channel];
channels_.erase(channel);
return 0;
}
WEBRTC_STUB(StartReceive, (int channel));
WEBRTC_FUNC(StartPlayout, (int channel)) {
if (playout_fail_channel_ != channel) {
WEBRTC_CHECK_CHANNEL(channel);
channels_[channel]->playout = true;
return 0;
} else {
// When playout_fail_channel_ == channel, fail the StartPlayout on this
// channel.
return -1;
}
}
WEBRTC_FUNC(StartSend, (int channel)) {
if (send_fail_channel_ != channel) {
WEBRTC_CHECK_CHANNEL(channel);
channels_[channel]->send = true;
return 0;
} else {
// When send_fail_channel_ == channel, fail the StartSend on this
// channel.
return -1;
}
}
WEBRTC_STUB(StopReceive, (int channel));
WEBRTC_FUNC(StopPlayout, (int channel)) {
WEBRTC_CHECK_CHANNEL(channel);
channels_[channel]->playout = false;
return 0;
}
WEBRTC_FUNC(StopSend, (int channel)) {
WEBRTC_CHECK_CHANNEL(channel);
channels_[channel]->send = false;
return 0;
}
WEBRTC_STUB(GetVersion, (char version[1024]));
WEBRTC_STUB(LastError, ());
WEBRTC_FUNC(AssociateSendChannel, (int channel,
int accociate_send_channel)) {
WEBRTC_CHECK_CHANNEL(channel);
channels_[channel]->associate_send_channel = accociate_send_channel;
return 0;
}
webrtc::RtcEventLog* GetEventLog() { return nullptr; }
// webrtc::VoECodec
WEBRTC_STUB(NumOfCodecs, ());
WEBRTC_STUB(GetCodec, (int index, webrtc::CodecInst& codec));
WEBRTC_FUNC(SetSendCodec, (int channel, const webrtc::CodecInst& codec)) {
WEBRTC_CHECK_CHANNEL(channel);
// To match the behavior of the real implementation.
if (_stricmp(codec.plname, "telephone-event") == 0 ||
_stricmp(codec.plname, "audio/telephone-event") == 0 ||
_stricmp(codec.plname, "CN") == 0 ||
_stricmp(codec.plname, "red") == 0 ) {
return -1;
}
channels_[channel]->send_codec = codec;
++num_set_send_codecs_;
return 0;
}
WEBRTC_FUNC(GetSendCodec, (int channel, webrtc::CodecInst& codec)) {
WEBRTC_CHECK_CHANNEL(channel);
codec = channels_[channel]->send_codec;
return 0;
}
WEBRTC_STUB(SetBitRate, (int channel, int bitrate_bps));
WEBRTC_STUB(GetRecCodec, (int channel, webrtc::CodecInst& codec));
WEBRTC_FUNC(SetRecPayloadType, (int channel,
const webrtc::CodecInst& codec)) {
WEBRTC_CHECK_CHANNEL(channel);
Channel* ch = channels_[channel];
if (ch->playout)
return -1; // Channel is in use.
// Check if something else already has this slot.
if (codec.pltype != -1) {
for (std::vector<webrtc::CodecInst>::iterator it =
ch->recv_codecs.begin(); it != ch->recv_codecs.end(); ++it) {
if (it->pltype == codec.pltype &&
_stricmp(it->plname, codec.plname) != 0) {
return -1;
}
}
}
// Otherwise try to find this codec and update its payload type.
int result = -1; // not found
for (std::vector<webrtc::CodecInst>::iterator it = ch->recv_codecs.begin();
it != ch->recv_codecs.end(); ++it) {
if (strcmp(it->plname, codec.plname) == 0 &&
it->plfreq == codec.plfreq &&
it->channels == codec.channels) {
it->pltype = codec.pltype;
result = 0;
}
}
return result;
}
WEBRTC_FUNC(SetSendCNPayloadType, (int channel, int type,
webrtc::PayloadFrequencies frequency)) {
WEBRTC_CHECK_CHANNEL(channel);
if (frequency == webrtc::kFreq8000Hz) {
channels_[channel]->cn8_type = type;
} else if (frequency == webrtc::kFreq16000Hz) {
channels_[channel]->cn16_type = type;
}
return 0;
}
WEBRTC_FUNC(GetRecPayloadType, (int channel, webrtc::CodecInst& codec)) {
WEBRTC_CHECK_CHANNEL(channel);
Channel* ch = channels_[channel];
for (std::vector<webrtc::CodecInst>::iterator it = ch->recv_codecs.begin();
it != ch->recv_codecs.end(); ++it) {
if (strcmp(it->plname, codec.plname) == 0 &&
it->plfreq == codec.plfreq &&
it->channels == codec.channels &&
it->pltype != -1) {
codec.pltype = it->pltype;
return 0;
}
}
return -1; // not found
}
WEBRTC_FUNC(SetVADStatus, (int channel, bool enable, webrtc::VadModes mode,
bool disableDTX)) {
WEBRTC_CHECK_CHANNEL(channel);
if (channels_[channel]->send_codec.channels == 2) {
// Replicating VoE behavior; VAD cannot be enabled for stereo.
return -1;
}
channels_[channel]->vad = enable;
return 0;
}
WEBRTC_STUB(GetVADStatus, (int channel, bool& enabled,
webrtc::VadModes& mode, bool& disabledDTX));
WEBRTC_FUNC(SetFECStatus, (int channel, bool enable)) {
WEBRTC_CHECK_CHANNEL(channel);
if (_stricmp(channels_[channel]->send_codec.plname, "opus") != 0) {
// Return -1 if current send codec is not Opus.
// TODO(minyue): Excludes other codecs if they support inband FEC.
return -1;
}
channels_[channel]->codec_fec = enable;
return 0;
}
WEBRTC_FUNC(GetFECStatus, (int channel, bool& enable)) {
WEBRTC_CHECK_CHANNEL(channel);
enable = channels_[channel]->codec_fec;
return 0;
}
WEBRTC_FUNC(SetOpusMaxPlaybackRate, (int channel, int frequency_hz)) {
WEBRTC_CHECK_CHANNEL(channel);
if (_stricmp(channels_[channel]->send_codec.plname, "opus") != 0) {
// Return -1 if current send codec is not Opus.
return -1;
}
if (frequency_hz <= 8000)
channels_[channel]->max_encoding_bandwidth = kOpusBandwidthNb;
else if (frequency_hz <= 12000)
channels_[channel]->max_encoding_bandwidth = kOpusBandwidthMb;
else if (frequency_hz <= 16000)
channels_[channel]->max_encoding_bandwidth = kOpusBandwidthWb;
else if (frequency_hz <= 24000)
channels_[channel]->max_encoding_bandwidth = kOpusBandwidthSwb;
else
channels_[channel]->max_encoding_bandwidth = kOpusBandwidthFb;
return 0;
}
WEBRTC_FUNC(SetOpusDtx, (int channel, bool enable_dtx)) {
WEBRTC_CHECK_CHANNEL(channel);
if (_stricmp(channels_[channel]->send_codec.plname, "opus") != 0) {
// Return -1 if current send codec is not Opus.
return -1;
}
channels_[channel]->opus_dtx = enable_dtx;
return 0;
}
// webrtc::VoEHardware
WEBRTC_STUB(GetNumOfRecordingDevices, (int& num));
WEBRTC_STUB(GetNumOfPlayoutDevices, (int& num));
WEBRTC_STUB(GetRecordingDeviceName, (int i, char* name, char* guid));
WEBRTC_STUB(GetPlayoutDeviceName, (int i, char* name, char* guid));
WEBRTC_STUB(SetRecordingDevice, (int, webrtc::StereoChannel));
WEBRTC_STUB(SetPlayoutDevice, (int));
WEBRTC_STUB(SetAudioDeviceLayer, (webrtc::AudioLayers));
WEBRTC_STUB(GetAudioDeviceLayer, (webrtc::AudioLayers&));
WEBRTC_FUNC(SetRecordingSampleRate, (unsigned int samples_per_sec)) {
recording_sample_rate_ = samples_per_sec;
return 0;
}
WEBRTC_FUNC_CONST(RecordingSampleRate, (unsigned int* samples_per_sec)) {
*samples_per_sec = recording_sample_rate_;
return 0;
}
WEBRTC_FUNC(SetPlayoutSampleRate, (unsigned int samples_per_sec)) {
playout_sample_rate_ = samples_per_sec;
return 0;
}
WEBRTC_FUNC_CONST(PlayoutSampleRate, (unsigned int* samples_per_sec)) {
*samples_per_sec = playout_sample_rate_;
return 0;
}
WEBRTC_STUB(EnableBuiltInAEC, (bool enable));
virtual bool BuiltInAECIsAvailable() const { return false; }
WEBRTC_STUB(EnableBuiltInAGC, (bool enable));
virtual bool BuiltInAGCIsAvailable() const { return false; }
WEBRTC_STUB(EnableBuiltInNS, (bool enable));
virtual bool BuiltInNSIsAvailable() const { return false; }
// webrtc::VoENetwork
WEBRTC_FUNC(RegisterExternalTransport, (int channel,
webrtc::Transport& transport)) {
WEBRTC_CHECK_CHANNEL(channel);
channels_[channel]->external_transport = true;
return 0;
}
WEBRTC_FUNC(DeRegisterExternalTransport, (int channel)) {
WEBRTC_CHECK_CHANNEL(channel);
channels_[channel]->external_transport = false;
return 0;
}
WEBRTC_FUNC(ReceivedRTPPacket, (int channel, const void* data,
size_t length)) {
WEBRTC_CHECK_CHANNEL(channel);
if (!channels_[channel]->external_transport) return -1;
channels_[channel]->packets.push_back(
std::string(static_cast<const char*>(data), length));
return 0;
}
WEBRTC_FUNC(ReceivedRTPPacket, (int channel, const void* data,
size_t length,
const webrtc::PacketTime& packet_time)) {
WEBRTC_CHECK_CHANNEL(channel);
if (ReceivedRTPPacket(channel, data, length) == -1) {
return -1;
}
channels_[channel]->last_rtp_packet_time = packet_time;
return 0;
}
WEBRTC_STUB(ReceivedRTCPPacket, (int channel, const void* data,
size_t length));
// webrtc::VoERTP_RTCP
WEBRTC_FUNC(SetLocalSSRC, (int channel, unsigned int ssrc)) {
WEBRTC_CHECK_CHANNEL(channel);
channels_[channel]->send_ssrc = ssrc;
return 0;
}
WEBRTC_STUB(GetLocalSSRC, (int channel, unsigned int& ssrc));
WEBRTC_STUB(GetRemoteSSRC, (int channel, unsigned int& ssrc));
WEBRTC_STUB(SetSendAudioLevelIndicationStatus, (int channel, bool enable,
unsigned char id));
WEBRTC_STUB(SetReceiveAudioLevelIndicationStatus, (int channel, bool enable,
unsigned char id));
WEBRTC_STUB(SetSendAbsoluteSenderTimeStatus, (int channel, bool enable,
unsigned char id));
WEBRTC_STUB(SetReceiveAbsoluteSenderTimeStatus, (int channel, bool enable,
unsigned char id));
WEBRTC_STUB(SetRTCPStatus, (int channel, bool enable));
WEBRTC_STUB(GetRTCPStatus, (int channel, bool& enabled));
WEBRTC_STUB(SetRTCP_CNAME, (int channel, const char cname[256]));
WEBRTC_STUB(GetRTCP_CNAME, (int channel, char cname[256]));
WEBRTC_STUB(GetRemoteRTCP_CNAME, (int channel, char* cname));
WEBRTC_STUB(GetRemoteRTCPData, (int channel, unsigned int& NTPHigh,
unsigned int& NTPLow,
unsigned int& timestamp,
unsigned int& playoutTimestamp,
unsigned int* jitter,
unsigned short* fractionLost));
WEBRTC_STUB(GetRemoteRTCPReportBlocks,
(int channel, std::vector<webrtc::ReportBlock>* receive_blocks));
WEBRTC_STUB(GetRTPStatistics, (int channel, unsigned int& averageJitterMs,
unsigned int& maxJitterMs,
unsigned int& discardedPackets));
WEBRTC_STUB(GetRTCPStatistics, (int channel, webrtc::CallStatistics& stats));
WEBRTC_FUNC(SetREDStatus, (int channel, bool enable, int redPayloadtype)) {
WEBRTC_CHECK_CHANNEL(channel);
channels_[channel]->red = enable;
channels_[channel]->red_type = redPayloadtype;
return 0;
}
WEBRTC_FUNC(GetREDStatus, (int channel, bool& enable, int& redPayloadtype)) {
WEBRTC_CHECK_CHANNEL(channel);
enable = channels_[channel]->red;
redPayloadtype = channels_[channel]->red_type;
return 0;
}
WEBRTC_FUNC(SetNACKStatus, (int channel, bool enable, int maxNoPackets)) {
WEBRTC_CHECK_CHANNEL(channel);
channels_[channel]->nack = enable;
channels_[channel]->nack_max_packets = maxNoPackets;
return 0;
}
// webrtc::VoEVolumeControl
WEBRTC_STUB(SetSpeakerVolume, (unsigned int));
WEBRTC_STUB(GetSpeakerVolume, (unsigned int&));
WEBRTC_STUB(SetMicVolume, (unsigned int));
WEBRTC_STUB(GetMicVolume, (unsigned int&));
WEBRTC_STUB(SetInputMute, (int, bool));
WEBRTC_STUB(GetInputMute, (int, bool&));
WEBRTC_STUB(GetSpeechInputLevel, (unsigned int&));
WEBRTC_STUB(GetSpeechOutputLevel, (int, unsigned int&));
WEBRTC_STUB(GetSpeechInputLevelFullRange, (unsigned int&));
WEBRTC_STUB(GetSpeechOutputLevelFullRange, (int, unsigned int&));
WEBRTC_FUNC(SetChannelOutputVolumeScaling, (int channel, float scale)) {
WEBRTC_CHECK_CHANNEL(channel);
channels_[channel]->volume_scale= scale;
return 0;
}
WEBRTC_FUNC(GetChannelOutputVolumeScaling, (int channel, float& scale)) {
WEBRTC_CHECK_CHANNEL(channel);
scale = channels_[channel]->volume_scale;
return 0;
}
WEBRTC_STUB(SetOutputVolumePan, (int channel, float left, float right));
WEBRTC_STUB(GetOutputVolumePan, (int channel, float& left, float& right));
// webrtc::VoEAudioProcessing
WEBRTC_FUNC(SetNsStatus, (bool enable, webrtc::NsModes mode)) {
ns_enabled_ = enable;
ns_mode_ = mode;
return 0;
}
WEBRTC_FUNC(GetNsStatus, (bool& enabled, webrtc::NsModes& mode)) {
enabled = ns_enabled_;
mode = ns_mode_;
return 0;
}
WEBRTC_FUNC(SetAgcStatus, (bool enable, webrtc::AgcModes mode)) {
agc_enabled_ = enable;
agc_mode_ = mode;
return 0;
}
WEBRTC_FUNC(GetAgcStatus, (bool& enabled, webrtc::AgcModes& mode)) {
enabled = agc_enabled_;
mode = agc_mode_;
return 0;
}
WEBRTC_FUNC(SetAgcConfig, (webrtc::AgcConfig config)) {
agc_config_ = config;
return 0;
}
WEBRTC_FUNC(GetAgcConfig, (webrtc::AgcConfig& config)) {
config = agc_config_;
return 0;
}
WEBRTC_FUNC(SetEcStatus, (bool enable, webrtc::EcModes mode)) {
ec_enabled_ = enable;
ec_mode_ = mode;
return 0;
}
WEBRTC_FUNC(GetEcStatus, (bool& enabled, webrtc::EcModes& mode)) {
enabled = ec_enabled_;
mode = ec_mode_;
return 0;
}
WEBRTC_STUB(EnableDriftCompensation, (bool enable))
WEBRTC_BOOL_STUB(DriftCompensationEnabled, ())
WEBRTC_VOID_STUB(SetDelayOffsetMs, (int offset))
WEBRTC_STUB(DelayOffsetMs, ());
WEBRTC_FUNC(SetAecmMode, (webrtc::AecmModes mode, bool enableCNG)) {
aecm_mode_ = mode;
cng_enabled_ = enableCNG;
return 0;
}
WEBRTC_FUNC(GetAecmMode, (webrtc::AecmModes& mode, bool& enabledCNG)) {
mode = aecm_mode_;
enabledCNG = cng_enabled_;
return 0;
}
WEBRTC_STUB(SetRxNsStatus, (int channel, bool enable, webrtc::NsModes mode));
WEBRTC_STUB(GetRxNsStatus, (int channel, bool& enabled,
webrtc::NsModes& mode));
WEBRTC_STUB(SetRxAgcStatus, (int channel, bool enable,
webrtc::AgcModes mode));
WEBRTC_STUB(GetRxAgcStatus, (int channel, bool& enabled,
webrtc::AgcModes& mode));
WEBRTC_STUB(SetRxAgcConfig, (int channel, webrtc::AgcConfig config));
WEBRTC_STUB(GetRxAgcConfig, (int channel, webrtc::AgcConfig& config));
WEBRTC_STUB(RegisterRxVadObserver, (int, webrtc::VoERxVadCallback&));
WEBRTC_STUB(DeRegisterRxVadObserver, (int channel));
WEBRTC_STUB(VoiceActivityIndicator, (int channel));
WEBRTC_FUNC(SetEcMetricsStatus, (bool enable)) {
ec_metrics_enabled_ = enable;
return 0;
}
WEBRTC_STUB(GetEcMetricsStatus, (bool& enabled));
WEBRTC_STUB(GetEchoMetrics, (int& ERL, int& ERLE, int& RERL, int& A_NLP));
WEBRTC_STUB(GetEcDelayMetrics, (int& delay_median, int& delay_std,
float& fraction_poor_delays));
WEBRTC_STUB(StartDebugRecording, (const char* fileNameUTF8));
WEBRTC_STUB(StartDebugRecording, (FILE* handle));
WEBRTC_STUB(StopDebugRecording, ());
WEBRTC_FUNC(SetTypingDetectionStatus, (bool enable)) {
typing_detection_enabled_ = enable;
return 0;
}
WEBRTC_FUNC(GetTypingDetectionStatus, (bool& enabled)) {
enabled = typing_detection_enabled_;
return 0;
}
WEBRTC_STUB(TimeSinceLastTyping, (int& seconds));
WEBRTC_STUB(SetTypingDetectionParameters, (int timeWindow,
int costPerTyping,
int reportingThreshold,
int penaltyDecay,
int typeEventDelay));
int EnableHighPassFilter(bool enable) {
highpass_filter_enabled_ = enable;
return 0;
}
bool IsHighPassFilterEnabled() {
return highpass_filter_enabled_;
}
bool IsStereoChannelSwappingEnabled() {
return stereo_swapping_enabled_;
}
void EnableStereoChannelSwapping(bool enable) {
stereo_swapping_enabled_ = enable;
}
int GetNetEqCapacity() const {
auto ch = channels_.find(last_channel_);
ASSERT(ch != channels_.end());
return ch->second->neteq_capacity;
}
bool GetNetEqFastAccelerate() const {
auto ch = channels_.find(last_channel_);
ASSERT(ch != channels_.end());
return ch->second->neteq_fast_accelerate;
}
private:
bool inited_;
int last_channel_;
std::map<int, Channel*> channels_;
bool fail_create_channel_;
int num_set_send_codecs_; // how many times we call SetSendCodec().
bool ec_enabled_;
bool ec_metrics_enabled_;
bool cng_enabled_;
bool ns_enabled_;
bool agc_enabled_;
bool highpass_filter_enabled_;
bool stereo_swapping_enabled_;
bool typing_detection_enabled_;
webrtc::EcModes ec_mode_;
webrtc::AecmModes aecm_mode_;
webrtc::NsModes ns_mode_;
webrtc::AgcModes agc_mode_;
webrtc::AgcConfig agc_config_;
webrtc::VoiceEngineObserver* observer_;
int playout_fail_channel_;
int send_fail_channel_;
int recording_sample_rate_;
int playout_sample_rate_;
FakeAudioProcessing audio_processing_;
};
} // namespace cricket
#endif // TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_