diff options
Diffstat (limited to 'talk/media/webrtc/fakewebrtcvoiceengine.h')
| -rw-r--r-- | talk/media/webrtc/fakewebrtcvoiceengine.h | 269 |
1 files changed, 32 insertions, 237 deletions
diff --git a/talk/media/webrtc/fakewebrtcvoiceengine.h b/talk/media/webrtc/fakewebrtcvoiceengine.h index 2405e07b5f..65ba927cc5 100644 --- a/talk/media/webrtc/fakewebrtcvoiceengine.h +++ b/talk/media/webrtc/fakewebrtcvoiceengine.h @@ -41,19 +41,11 @@ #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 char kFakeDefaultDeviceName[] = "Fake Default"; -static const int kFakeDefaultDeviceId = -1; -static const char kFakeDeviceName[] = "Fake Device"; -#ifdef WIN32 -static const int kFakeDeviceId = 0; -#else -static const int kFakeDeviceId = 1; -#endif - static const int kOpusBandwidthNb = 4000; static const int kOpusBandwidthMb = 6000; static const int kOpusBandwidthWb = 8000; @@ -63,18 +55,6 @@ static const int kOpusBandwidthFb = 20000; #define WEBRTC_CHECK_CHANNEL(channel) \ if (channels_.find(channel) == channels_.end()) return -1; -#define WEBRTC_ASSERT_CHANNEL(channel) \ - RTC_DCHECK(channels_.find(channel) != channels_.end()); - -// Verify the header extension ID, if enabled, is within the bounds specified in -// [RFC5285]: 1-14 inclusive. -#define WEBRTC_CHECK_HEADER_EXTENSION_ID(enable, id) \ - do { \ - if (enable && (id < 1 || id > 14)) { \ - return -1; \ - } \ - } while (0); - class FakeAudioProcessing : public webrtc::AudioProcessing { public: FakeAudioProcessing() : experimental_ns_enabled_(false) {} @@ -94,11 +74,13 @@ class FakeAudioProcessing : public webrtc::AudioProcessing { 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, ()); + size_t num_input_channels() const override { return 0; } + size_t num_proc_channels() const override { return 0; } + size_t num_output_channels() const override { return 0; } + size_t num_reverse_channels() const override { return 0; } WEBRTC_VOID_STUB(set_output_will_be_muted, (bool muted)); WEBRTC_STUB(ProcessStream, (webrtc::AudioFrame* frame)); WEBRTC_STUB(ProcessStream, ( @@ -156,20 +138,11 @@ class FakeAudioProcessing : public webrtc::AudioProcessing { class FakeWebRtcVoiceEngine : public webrtc::VoEAudioProcessing, - public webrtc::VoEBase, public webrtc::VoECodec, public webrtc::VoEDtmf, + public webrtc::VoEBase, public webrtc::VoECodec, public webrtc::VoEHardware, public webrtc::VoENetwork, public webrtc::VoERTP_RTCP, public webrtc::VoEVolumeControl { public: - struct DtmfInfo { - DtmfInfo() - : dtmf_event_code(-1), - dtmf_out_of_band(false), - dtmf_length_ms(-1) {} - int dtmf_event_code; - bool dtmf_out_of_band; - int dtmf_length_ms; - }; struct Channel { explicit Channel() : external_transport(false), @@ -184,15 +157,11 @@ class FakeWebRtcVoiceEngine nack(false), cn8_type(13), cn16_type(105), - dtmf_type(106), red_type(117), nack_max_packets(0), send_ssrc(0), - send_audio_level_ext_(-1), - receive_audio_level_ext_(-1), - send_absolute_sender_time_ext_(-1), - receive_absolute_sender_time_ext_(-1), associate_send_channel(-1), + recv_codecs(), neteq_capacity(-1), neteq_fast_accelerate(false) { memset(&send_codec, 0, sizeof(send_codec)); @@ -209,16 +178,10 @@ class FakeWebRtcVoiceEngine bool nack; int cn8_type; int cn16_type; - int dtmf_type; int red_type; int nack_max_packets; uint32_t send_ssrc; - int send_audio_level_ext_; - int receive_audio_level_ext_; - int send_absolute_sender_time_ext_; - int receive_absolute_sender_time_ext_; int associate_send_channel; - DtmfInfo dtmf_info; std::vector<webrtc::CodecInst> recv_codecs; webrtc::CodecInst send_codec; webrtc::PacketTime last_rtp_packet_time; @@ -227,13 +190,10 @@ class FakeWebRtcVoiceEngine bool neteq_fast_accelerate; }; - FakeWebRtcVoiceEngine(const cricket::AudioCodec* const* codecs, - int num_codecs) + FakeWebRtcVoiceEngine() : inited_(false), last_channel_(-1), fail_create_channel_(false), - codecs_(codecs), - num_codecs_(num_codecs), num_set_send_codecs_(0), ec_enabled_(false), ec_metrics_enabled_(false), @@ -255,26 +215,13 @@ class FakeWebRtcVoiceEngine memset(&agc_config_, 0, sizeof(agc_config_)); } ~FakeWebRtcVoiceEngine() { - // Ought to have all been deleted by the WebRtcVoiceMediaChannel - // destructors, but just in case ... - for (std::map<int, Channel*>::const_iterator i = channels_.begin(); - i != channels_.end(); ++i) { - delete i->second; - } + 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 GetChannelFromLocalSsrc(uint32_t local_ssrc) const { - for (std::map<int, Channel*>::const_iterator iter = channels_.begin(); - iter != channels_.end(); ++iter) { - if (local_ssrc == iter->second->send_ssrc) - return iter->first; - } - return -1; - } int GetNumChannels() const { return static_cast<int>(channels_.size()); } uint32_t GetLocalSSRC(int channel) { return channels_[channel]->send_ssrc; @@ -307,7 +254,7 @@ class FakeWebRtcVoiceEngine return channels_[channel]->nack_max_packets; } const webrtc::PacketTime& GetLastRtpPacketTime(int channel) { - WEBRTC_ASSERT_CHANNEL(channel); + RTC_DCHECK(channels_.find(channel) != channels_.end()); return channels_[channel]->last_rtp_packet_time; } int GetSendCNPayloadType(int channel, bool wideband) { @@ -315,9 +262,6 @@ class FakeWebRtcVoiceEngine channels_[channel]->cn16_type : channels_[channel]->cn8_type; } - int GetSendTelephoneEventPayloadType(int channel) { - return channels_[channel]->dtmf_type; - } int GetSendREDPayloadType(int channel) { return channels_[channel]->red_type; } @@ -351,11 +295,8 @@ class FakeWebRtcVoiceEngine return -1; } Channel* ch = new Channel(); - for (int i = 0; i < NumOfCodecs(); ++i) { - webrtc::CodecInst codec; - GetCodec(i, codec); - ch->recv_codecs.push_back(codec); - } + 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; } @@ -364,24 +305,6 @@ class FakeWebRtcVoiceEngine channels_[++last_channel_] = ch; return last_channel_; } - int GetSendRtpExtensionId(int channel, const std::string& extension) { - WEBRTC_ASSERT_CHANNEL(channel); - if (extension == kRtpAudioLevelHeaderExtension) { - return channels_[channel]->send_audio_level_ext_; - } else if (extension == kRtpAbsoluteSenderTimeHeaderExtension) { - return channels_[channel]->send_absolute_sender_time_ext_; - } - return -1; - } - int GetReceiveRtpExtensionId(int channel, const std::string& extension) { - WEBRTC_ASSERT_CHANNEL(channel); - if (extension == kRtpAudioLevelHeaderExtension) { - return channels_[channel]->receive_audio_level_ext_; - } else if (extension == kRtpAbsoluteSenderTimeHeaderExtension) { - return channels_[channel]->receive_absolute_sender_time_ext_; - } - return -1; - } int GetNumSetSendCodecs() const { return num_set_send_codecs_; } @@ -473,22 +396,8 @@ class FakeWebRtcVoiceEngine webrtc::RtcEventLog* GetEventLog() { return nullptr; } // webrtc::VoECodec - WEBRTC_FUNC(NumOfCodecs, ()) { - return num_codecs_; - } - WEBRTC_FUNC(GetCodec, (int index, webrtc::CodecInst& codec)) { - if (index < 0 || index >= NumOfCodecs()) { - return -1; - } - const cricket::AudioCodec& c(*codecs_[index]); - codec.pltype = c.id; - rtc::strcpyn(codec.plname, sizeof(codec.plname), c.name.c_str()); - codec.plfreq = c.clockrate; - codec.pacsize = 0; - codec.channels = c.channels; - codec.rate = c.bitrate; - return 0; - } + 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. @@ -526,16 +435,17 @@ class FakeWebRtcVoiceEngine } } // 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->plfreq == codec.plfreq && + it->channels == codec.channels) { it->pltype = codec.pltype; - it->channels = codec.channels; - return 0; + result = 0; } } - return -1; // not found + return result; } WEBRTC_FUNC(SetSendCNPayloadType, (int channel, int type, webrtc::PayloadFrequencies frequency)) { @@ -620,46 +530,11 @@ class FakeWebRtcVoiceEngine return 0; } - // webrtc::VoEDtmf - WEBRTC_FUNC(SendTelephoneEvent, (int channel, int event_code, - bool out_of_band = true, int length_ms = 160, int attenuation_db = 10)) { - channels_[channel]->dtmf_info.dtmf_event_code = event_code; - channels_[channel]->dtmf_info.dtmf_out_of_band = out_of_band; - channels_[channel]->dtmf_info.dtmf_length_ms = length_ms; - return 0; - } - - WEBRTC_FUNC(SetSendTelephoneEventPayloadType, - (int channel, unsigned char type)) { - channels_[channel]->dtmf_type = type; - return 0; - }; - WEBRTC_STUB(GetSendTelephoneEventPayloadType, - (int channel, unsigned char& type)); - - WEBRTC_STUB(SetDtmfFeedbackStatus, (bool enable, bool directFeedback)); - WEBRTC_STUB(GetDtmfFeedbackStatus, (bool& enabled, bool& directFeedback)); - - WEBRTC_FUNC(PlayDtmfTone, - (int event_code, int length_ms = 200, int attenuation_db = 10)) { - dtmf_info_.dtmf_event_code = event_code; - dtmf_info_.dtmf_length_ms = length_ms; - return 0; - } - // webrtc::VoEHardware - WEBRTC_FUNC(GetNumOfRecordingDevices, (int& num)) { - return GetNumDevices(num); - } - WEBRTC_FUNC(GetNumOfPlayoutDevices, (int& num)) { - return GetNumDevices(num); - } - WEBRTC_FUNC(GetRecordingDeviceName, (int i, char* name, char* guid)) { - return GetDeviceName(i, name, guid); - } - WEBRTC_FUNC(GetPlayoutDeviceName, (int i, char* name, char* guid)) { - return GetDeviceName(i, name, guid); - } + 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)); @@ -729,35 +604,14 @@ class FakeWebRtcVoiceEngine } WEBRTC_STUB(GetLocalSSRC, (int channel, unsigned int& ssrc)); WEBRTC_STUB(GetRemoteSSRC, (int channel, unsigned int& ssrc)); - WEBRTC_FUNC(SetSendAudioLevelIndicationStatus, (int channel, bool enable, - unsigned char id)) { - WEBRTC_CHECK_CHANNEL(channel); - WEBRTC_CHECK_HEADER_EXTENSION_ID(enable, id); - channels_[channel]->send_audio_level_ext_ = (enable) ? id : -1; - return 0; - } - WEBRTC_FUNC(SetReceiveAudioLevelIndicationStatus, (int channel, bool enable, - unsigned char id)) { - WEBRTC_CHECK_CHANNEL(channel); - WEBRTC_CHECK_HEADER_EXTENSION_ID(enable, id); - channels_[channel]->receive_audio_level_ext_ = (enable) ? id : -1; - return 0; - } - WEBRTC_FUNC(SetSendAbsoluteSenderTimeStatus, (int channel, bool enable, - unsigned char id)) { - WEBRTC_CHECK_CHANNEL(channel); - WEBRTC_CHECK_HEADER_EXTENSION_ID(enable, id); - channels_[channel]->send_absolute_sender_time_ext_ = (enable) ? id : -1; - return 0; - } - WEBRTC_FUNC(SetReceiveAbsoluteSenderTimeStatus, (int channel, bool enable, - unsigned char id)) { - WEBRTC_CHECK_CHANNEL(channel); - WEBRTC_CHECK_HEADER_EXTENSION_ID(enable, id); - channels_[channel]->receive_absolute_sender_time_ext_ = (enable) ? id : -1; - return 0; - } - + 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])); @@ -776,22 +630,12 @@ class FakeWebRtcVoiceEngine unsigned int& discardedPackets)); WEBRTC_STUB(GetRTCPStatistics, (int channel, webrtc::CallStatistics& stats)); WEBRTC_FUNC(SetREDStatus, (int channel, bool enable, int redPayloadtype)) { - return SetFECStatus(channel, enable, redPayloadtype); - } - // TODO(minyue): remove the below function when transition to SetREDStatus - // is finished. - WEBRTC_FUNC(SetFECStatus, (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)) { - return GetFECStatus(channel, enable, redPayloadtype); - } - // TODO(minyue): remove the below function when transition to GetREDStatus - // is finished. - WEBRTC_FUNC(GetFECStatus, (int channel, bool& enable, int& redPayloadtype)) { WEBRTC_CHECK_CHANNEL(channel); enable = channels_[channel]->red; redPayloadtype = channels_[channel]->red_type; @@ -937,15 +781,6 @@ class FakeWebRtcVoiceEngine void EnableStereoChannelSwapping(bool enable) { stereo_swapping_enabled_ = enable; } - bool WasSendTelephoneEventCalled(int channel, int event_code, int length_ms) { - return (channels_[channel]->dtmf_info.dtmf_event_code == event_code && - channels_[channel]->dtmf_info.dtmf_out_of_band == true && - channels_[channel]->dtmf_info.dtmf_length_ms == length_ms); - } - bool WasPlayDtmfToneCalled(int event_code, int length_ms) { - return (dtmf_info_.dtmf_event_code == event_code && - dtmf_info_.dtmf_length_ms == length_ms); - } int GetNetEqCapacity() const { auto ch = channels_.find(last_channel_); ASSERT(ch != channels_.end()); @@ -958,47 +793,10 @@ class FakeWebRtcVoiceEngine } private: - int GetNumDevices(int& num) { -#ifdef WIN32 - num = 1; -#else - // On non-Windows platforms VE adds a special entry for the default device, - // so if there is one physical device then there are two entries in the - // list. - num = 2; -#endif - return 0; - } - - int GetDeviceName(int i, char* name, char* guid) { - const char *s; -#ifdef WIN32 - if (0 == i) { - s = kFakeDeviceName; - } else { - return -1; - } -#else - // See comment above. - if (0 == i) { - s = kFakeDefaultDeviceName; - } else if (1 == i) { - s = kFakeDeviceName; - } else { - return -1; - } -#endif - strcpy(name, s); - guid[0] = '\0'; - return 0; - } - bool inited_; int last_channel_; std::map<int, Channel*> channels_; bool fail_create_channel_; - const cricket::AudioCodec* const* codecs_; - int num_codecs_; int num_set_send_codecs_; // how many times we call SetSendCodec(). bool ec_enabled_; bool ec_metrics_enabled_; @@ -1018,12 +816,9 @@ class FakeWebRtcVoiceEngine int send_fail_channel_; int recording_sample_rate_; int playout_sample_rate_; - DtmfInfo dtmf_info_; FakeAudioProcessing audio_processing_; }; -#undef WEBRTC_CHECK_HEADER_EXTENSION_ID - } // namespace cricket #endif // TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_ |