Index: webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc |
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc |
index 879af49bd235c7b2d5eaa19c49419ed6df69bdc5..467e749ccecd73609a83086bdee3d8e22774337b 100644 |
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc |
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc |
@@ -803,287 +803,4 @@ void AudioCodingModuleImpl::GetDecodingCallStatistics( |
} |
} // namespace acm2 |
- |
-AudioCodingImpl::AudioCodingImpl(const Config& config) { |
- AudioCodingModule::Config config_old = config.ToOldConfig(); |
- acm_old_.reset(new acm2::AudioCodingModuleImpl(config_old)); |
- acm_old_->RegisterTransportCallback(config.transport); |
- acm_old_->RegisterVADCallback(config.vad_callback); |
- if (config.initial_playout_delay_ms > 0) { |
- acm_old_->SetInitialPlayoutDelay(config.initial_playout_delay_ms); |
- } |
- playout_frequency_hz_ = config.playout_frequency_hz; |
-} |
- |
-AudioCodingImpl::~AudioCodingImpl() = default; |
- |
-bool AudioCodingImpl::RegisterSendCodec(AudioEncoder* send_codec) { |
- FATAL() << "Not implemented yet."; |
- return false; |
-} |
- |
-bool AudioCodingImpl::RegisterSendCodec(int encoder_type, |
- uint8_t payload_type, |
- int frame_size_samples) { |
- std::string codec_name; |
- int sample_rate_hz; |
- int channels; |
- if (!MapCodecTypeToParameters( |
- encoder_type, &codec_name, &sample_rate_hz, &channels)) { |
- return false; |
- } |
- webrtc::CodecInst codec; |
- AudioCodingModule::Codec( |
- codec_name.c_str(), &codec, sample_rate_hz, channels); |
- codec.pltype = payload_type; |
- if (frame_size_samples > 0) { |
- codec.pacsize = frame_size_samples; |
- } |
- return acm_old_->RegisterSendCodec(codec) == 0; |
-} |
- |
-const AudioEncoder* AudioCodingImpl::GetSenderInfo() const { |
- FATAL() << "Not implemented yet."; |
- return reinterpret_cast<const AudioEncoder*>(NULL); |
-} |
- |
-const CodecInst* AudioCodingImpl::GetSenderCodecInst() { |
- if (acm_old_->SendCodec(¤t_send_codec_) != 0) { |
- return NULL; |
- } |
- return ¤t_send_codec_; |
-} |
- |
-int AudioCodingImpl::Add10MsAudio(const AudioFrame& audio_frame) { |
- acm2::AudioCodingModuleImpl::InputData input_data; |
- CriticalSectionScoped lock(acm_old_->acm_crit_sect_.get()); |
- if (acm_old_->Add10MsDataInternal(audio_frame, &input_data) != 0) |
- return -1; |
- return acm_old_->Encode(input_data); |
-} |
- |
-const ReceiverInfo* AudioCodingImpl::GetReceiverInfo() const { |
- FATAL() << "Not implemented yet."; |
- return reinterpret_cast<const ReceiverInfo*>(NULL); |
-} |
- |
-bool AudioCodingImpl::RegisterReceiveCodec(AudioDecoder* receive_codec) { |
- FATAL() << "Not implemented yet."; |
- return false; |
-} |
- |
-bool AudioCodingImpl::RegisterReceiveCodec(int decoder_type, |
- uint8_t payload_type) { |
- std::string codec_name; |
- int sample_rate_hz; |
- int channels; |
- if (!MapCodecTypeToParameters( |
- decoder_type, &codec_name, &sample_rate_hz, &channels)) { |
- return false; |
- } |
- webrtc::CodecInst codec; |
- AudioCodingModule::Codec( |
- codec_name.c_str(), &codec, sample_rate_hz, channels); |
- codec.pltype = payload_type; |
- return acm_old_->RegisterReceiveCodec(codec) == 0; |
-} |
- |
-bool AudioCodingImpl::InsertPacket(const uint8_t* incoming_payload, |
- size_t payload_len_bytes, |
- const WebRtcRTPHeader& rtp_info) { |
- return acm_old_->IncomingPacket( |
- incoming_payload, payload_len_bytes, rtp_info) == 0; |
-} |
- |
-bool AudioCodingImpl::InsertPayload(const uint8_t* incoming_payload, |
- size_t payload_len_byte, |
- uint8_t payload_type, |
- uint32_t timestamp) { |
- FATAL() << "Not implemented yet."; |
- return false; |
-} |
- |
-bool AudioCodingImpl::SetMinimumPlayoutDelay(int time_ms) { |
- FATAL() << "Not implemented yet."; |
- return false; |
-} |
- |
-bool AudioCodingImpl::SetMaximumPlayoutDelay(int time_ms) { |
- FATAL() << "Not implemented yet."; |
- return false; |
-} |
- |
-int AudioCodingImpl::LeastRequiredDelayMs() const { |
- FATAL() << "Not implemented yet."; |
- return -1; |
-} |
- |
-bool AudioCodingImpl::PlayoutTimestamp(uint32_t* timestamp) { |
- FATAL() << "Not implemented yet."; |
- return false; |
-} |
- |
-bool AudioCodingImpl::Get10MsAudio(AudioFrame* audio_frame) { |
- return acm_old_->PlayoutData10Ms(playout_frequency_hz_, audio_frame) == 0; |
-} |
- |
-bool AudioCodingImpl::GetNetworkStatistics( |
- NetworkStatistics* network_statistics) { |
- FATAL() << "Not implemented yet."; |
- return false; |
-} |
- |
-bool AudioCodingImpl::EnableNack(size_t max_nack_list_size) { |
- FATAL() << "Not implemented yet."; |
- return false; |
-} |
- |
-void AudioCodingImpl::DisableNack() { |
- // A bug in the linker of Visual Studio 2013 Update 3 prevent us from using |
- // FATAL() here, if we do so then the linker hang when the WPO is turned on. |
- // TODO(sebmarchand): Re-evaluate this when we upgrade the toolchain. |
-} |
- |
-bool AudioCodingImpl::SetVad(bool enable_dtx, |
- bool enable_vad, |
- ACMVADMode vad_mode) { |
- return acm_old_->SetVAD(enable_dtx, enable_vad, vad_mode) == 0; |
-} |
- |
-std::vector<uint16_t> AudioCodingImpl::GetNackList( |
- int round_trip_time_ms) const { |
- return acm_old_->GetNackList(round_trip_time_ms); |
-} |
- |
-void AudioCodingImpl::GetDecodingCallStatistics( |
- AudioDecodingCallStats* call_stats) const { |
- acm_old_->GetDecodingCallStatistics(call_stats); |
-} |
- |
-bool AudioCodingImpl::MapCodecTypeToParameters(int codec_type, |
- std::string* codec_name, |
- int* sample_rate_hz, |
- int* channels) { |
- switch (codec_type) { |
- case acm2::ACMCodecDB::kPCM16B: |
- *codec_name = "L16"; |
- *sample_rate_hz = 8000; |
- *channels = 1; |
- break; |
- case acm2::ACMCodecDB::kPCM16Bwb: |
- *codec_name = "L16"; |
- *sample_rate_hz = 16000; |
- *channels = 1; |
- break; |
- case acm2::ACMCodecDB::kPCM16Bswb32kHz: |
- *codec_name = "L16"; |
- *sample_rate_hz = 32000; |
- *channels = 1; |
- break; |
- case acm2::ACMCodecDB::kPCM16B_2ch: |
- *codec_name = "L16"; |
- *sample_rate_hz = 8000; |
- *channels = 2; |
- break; |
- case acm2::ACMCodecDB::kPCM16Bwb_2ch: |
- *codec_name = "L16"; |
- *sample_rate_hz = 16000; |
- *channels = 2; |
- break; |
- case acm2::ACMCodecDB::kPCM16Bswb32kHz_2ch: |
- *codec_name = "L16"; |
- *sample_rate_hz = 32000; |
- *channels = 2; |
- break; |
-#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) |
- case acm2::ACMCodecDB::kISAC: |
- *codec_name = "ISAC"; |
- *sample_rate_hz = 16000; |
- *channels = 1; |
- break; |
-#endif |
-#ifdef WEBRTC_CODEC_ISAC |
- case acm2::ACMCodecDB::kISACSWB: |
- *codec_name = "ISAC"; |
- *sample_rate_hz = 32000; |
- *channels = 1; |
- break; |
-#endif |
-#ifdef WEBRTC_CODEC_ILBC |
- case acm2::ACMCodecDB::kILBC: |
- *codec_name = "ILBC"; |
- *sample_rate_hz = 8000; |
- *channels = 1; |
- break; |
-#endif |
- case acm2::ACMCodecDB::kPCMA: |
- *codec_name = "PCMA"; |
- *sample_rate_hz = 8000; |
- *channels = 1; |
- break; |
- case acm2::ACMCodecDB::kPCMA_2ch: |
- *codec_name = "PCMA"; |
- *sample_rate_hz = 8000; |
- *channels = 2; |
- break; |
- case acm2::ACMCodecDB::kPCMU: |
- *codec_name = "PCMU"; |
- *sample_rate_hz = 8000; |
- *channels = 1; |
- break; |
- case acm2::ACMCodecDB::kPCMU_2ch: |
- *codec_name = "PCMU"; |
- *sample_rate_hz = 8000; |
- *channels = 2; |
- break; |
-#ifdef WEBRTC_CODEC_G722 |
- case acm2::ACMCodecDB::kG722: |
- *codec_name = "G722"; |
- *sample_rate_hz = 16000; |
- *channels = 1; |
- break; |
- case acm2::ACMCodecDB::kG722_2ch: |
- *codec_name = "G722"; |
- *sample_rate_hz = 16000; |
- *channels = 2; |
- break; |
-#endif |
-#ifdef WEBRTC_CODEC_OPUS |
- case acm2::ACMCodecDB::kOpus: |
- *codec_name = "opus"; |
- *sample_rate_hz = 48000; |
- *channels = 2; |
- break; |
-#endif |
- case acm2::ACMCodecDB::kCNNB: |
- *codec_name = "CN"; |
- *sample_rate_hz = 8000; |
- *channels = 1; |
- break; |
- case acm2::ACMCodecDB::kCNWB: |
- *codec_name = "CN"; |
- *sample_rate_hz = 16000; |
- *channels = 1; |
- break; |
- case acm2::ACMCodecDB::kCNSWB: |
- *codec_name = "CN"; |
- *sample_rate_hz = 32000; |
- *channels = 1; |
- break; |
- case acm2::ACMCodecDB::kRED: |
- *codec_name = "red"; |
- *sample_rate_hz = 8000; |
- *channels = 1; |
- break; |
- case acm2::ACMCodecDB::kAVT: |
- *codec_name = "telephone-event"; |
- *sample_rate_hz = 8000; |
- *channels = 1; |
- break; |
- default: |
- FATAL() << "Codec type " << codec_type << " not supported."; |
- } |
- return true; |
-} |
- |
} // namespace webrtc |