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/*
* Copyright (c) 2010 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 MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_
#define MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_
#include <map>
#include <vector>
#include "media/engine/webrtcvoe.h"
#include "rtc_base/checks.h"
namespace webrtc {
namespace voe {
class TransmitMixer;
} // namespace voe
} // namespace webrtc
namespace cricket {
#define WEBRTC_CHECK_CHANNEL(channel) \
if (channels_.find(channel) == channels_.end()) return -1;
#define WEBRTC_STUB(method, args) \
int method args override { return 0; }
#define WEBRTC_FUNC(method, args) int method args override
class FakeWebRtcVoiceEngine : public webrtc::VoEBase {
public:
struct Channel {
std::vector<webrtc::CodecInst> recv_codecs;
size_t neteq_capacity = 0;
bool neteq_fast_accelerate = false;
};
explicit FakeWebRtcVoiceEngine(webrtc::voe::TransmitMixer* transmit_mixer)
: transmit_mixer_(transmit_mixer) {}
~FakeWebRtcVoiceEngine() override {
RTC_CHECK(channels_.empty());
}
bool IsInited() const { return inited_; }
int GetLastChannel() const { return last_channel_; }
int GetNumChannels() const { return static_cast<int>(channels_.size()); }
void set_fail_create_channel(bool fail_create_channel) {
fail_create_channel_ = fail_create_channel;
}
WEBRTC_STUB(Release, ());
// webrtc::VoEBase
WEBRTC_FUNC(Init,
(webrtc::AudioDeviceModule* adm,
webrtc::AudioProcessing* audioproc,
const rtc::scoped_refptr<webrtc::AudioDecoderFactory>&
decoder_factory)) {
inited_ = true;
return 0;
}
WEBRTC_FUNC(Terminate, ()) {
inited_ = false;
return 0;
}
webrtc::AudioDeviceModule* audio_device_module() override {
return nullptr;
}
webrtc::voe::TransmitMixer* transmit_mixer() override {
return transmit_mixer_;
}
WEBRTC_FUNC(CreateChannel, ()) {
return CreateChannel(webrtc::VoEBase::ChannelConfig());
}
WEBRTC_FUNC(CreateChannel, (const webrtc::VoEBase::ChannelConfig& config)) {
if (fail_create_channel_) {
return -1;
}
Channel* ch = new Channel();
ch->neteq_capacity = config.acm_config.neteq_config.max_packets_in_buffer;
ch->neteq_fast_accelerate =
config.acm_config.neteq_config.enable_fast_accelerate;
channels_[++last_channel_] = ch;
return last_channel_;
}
WEBRTC_FUNC(DeleteChannel, (int channel)) {
WEBRTC_CHECK_CHANNEL(channel);
delete channels_[channel];
channels_.erase(channel);
return 0;
}
WEBRTC_STUB(StartPlayout, (int channel));
WEBRTC_STUB(StartSend, (int channel));
WEBRTC_STUB(StopPlayout, (int channel));
WEBRTC_STUB(StopSend, (int channel));
size_t GetNetEqCapacity() const {
auto ch = channels_.find(last_channel_);
RTC_DCHECK(ch != channels_.end());
return ch->second->neteq_capacity;
}
bool GetNetEqFastAccelerate() const {
auto ch = channels_.find(last_channel_);
RTC_CHECK(ch != channels_.end());
return ch->second->neteq_fast_accelerate;
}
private:
bool inited_ = false;
int last_channel_ = -1;
std::map<int, Channel*> channels_;
bool fail_create_channel_ = false;
webrtc::voe::TransmitMixer* transmit_mixer_ = nullptr;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(FakeWebRtcVoiceEngine);
};
} // namespace cricket
#endif // MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_