Starting to implement the new ACM API
The new implementation class is called AudioCodingImpl, and will in the
end replace AudioCodingModuleImpl.
This is work in progress.
BUG=3520
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17359004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7103 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc
index eca909c..9b61d33 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc
@@ -94,4 +94,8 @@
}
}
+AudioCoding* AudioCoding::Create(const Config& config) {
+ return new AudioCodingImpl(config);
+}
+
} // namespace webrtc
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 381f020..98e2558 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
@@ -14,6 +14,7 @@
#include <stdlib.h>
#include <vector>
+#include "webrtc/base/checks.h"
#include "webrtc/engine_configurations.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
@@ -2053,4 +2054,255 @@
} // namespace acm2
+bool AudioCodingImpl::RegisterSendCodec(AudioEncoder* send_codec) {
+ FATAL() << "Not implemented yet.";
+}
+
+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.";
+}
+
+int AudioCodingImpl::Add10MsAudio(const AudioFrame& audio_frame) {
+ if (acm_old_->Add10MsData(audio_frame) != 0) {
+ return -1;
+ }
+ return acm_old_->Process();
+}
+
+const ReceiverInfo* AudioCodingImpl::GetReceiverInfo() const {
+ FATAL() << "Not implemented yet.";
+}
+
+bool AudioCodingImpl::RegisterReceiveCodec(AudioDecoder* receive_codec) {
+ FATAL() << "Not implemented yet.";
+}
+
+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,
+ int32_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,
+ int32_t payload_len_byte,
+ uint8_t payload_type,
+ uint32_t timestamp) {
+ FATAL() << "Not implemented yet.";
+}
+
+bool AudioCodingImpl::SetMinimumPlayoutDelay(int time_ms) {
+ FATAL() << "Not implemented yet.";
+}
+
+bool AudioCodingImpl::SetMaximumPlayoutDelay(int time_ms) {
+ FATAL() << "Not implemented yet.";
+}
+
+int AudioCodingImpl::LeastRequiredDelayMs() const {
+ FATAL() << "Not implemented yet.";
+}
+
+bool AudioCodingImpl::PlayoutTimestamp(uint32_t* timestamp) {
+ FATAL() << "Not implemented yet.";
+}
+
+bool AudioCodingImpl::Get10MsAudio(AudioFrame* audio_frame) {
+ return acm_old_->PlayoutData10Ms(playout_frequency_hz_, audio_frame) == 0;
+}
+
+bool AudioCodingImpl::NetworkStatistics(
+ ACMNetworkStatistics* network_statistics) {
+ FATAL() << "Not implemented yet.";
+}
+
+bool AudioCodingImpl::EnableNack(size_t max_nack_list_size) {
+ FATAL() << "Not implemented yet.";
+}
+
+void AudioCodingImpl::DisableNack() {
+ FATAL() << "Not implemented yet.";
+}
+
+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) {
+#ifdef WEBRTC_CODEC_PCM16
+ 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;
+#endif
+#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;
+ case acm2::ACMCodecDB::kISACFB:
+ *codec_name = "ISAC";
+ *sample_rate_hz = 48000;
+ *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;
+#ifdef WEBRTC_CODEC_AVT
+ case acm2::ACMCodecDB::kAVT:
+ *codec_name = "telephone-event";
+ *sample_rate_hz = 8000;
+ *channels = 1;
+ break;
+#endif
+ default:
+ FATAL() << "Codec type " << codec_type << " not supported.";
+ }
+ return true;
+}
+
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
index 635b4dc..41c2894 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
@@ -382,6 +382,88 @@
} // namespace acm2
+class AudioCodingImpl : public AudioCoding {
+ public:
+ AudioCodingImpl(const Config& config) {
+ AudioCodingModule::Config config_old;
+ config_old.id = 0;
+ config_old.neteq_config = config.neteq_config;
+ config_old.clock = config.clock;
+ acm_old_.reset(new acm2::AudioCodingModuleImpl(config_old));
+ acm_old_->RegisterTransportCallback(config.transport);
+ acm_old_->RegisterVADCallback(config.vad_callback);
+ acm_old_->SetDtmfPlayoutStatus(config.play_dtmf);
+ if (config.initial_playout_delay_ms > 0) {
+ acm_old_->SetInitialPlayoutDelay(config.initial_playout_delay_ms);
+ }
+ playout_frequency_hz_ = config.playout_frequency_hz;
+ }
+
+ virtual ~AudioCodingImpl() OVERRIDE {};
+
+ virtual bool RegisterSendCodec(AudioEncoder* send_codec) OVERRIDE;
+
+ virtual bool RegisterSendCodec(int encoder_type,
+ uint8_t payload_type,
+ int frame_size_samples = 0) OVERRIDE;
+
+ virtual const AudioEncoder* GetSenderInfo() const OVERRIDE;
+
+ virtual int Add10MsAudio(const AudioFrame& audio_frame) OVERRIDE;
+
+ virtual const ReceiverInfo* GetReceiverInfo() const OVERRIDE;
+
+ virtual bool RegisterReceiveCodec(AudioDecoder* receive_codec) OVERRIDE;
+
+ virtual bool RegisterReceiveCodec(int decoder_type,
+ uint8_t payload_type) OVERRIDE;
+
+ virtual bool InsertPacket(const uint8_t* incoming_payload,
+ int32_t payload_len_bytes,
+ const WebRtcRTPHeader& rtp_info) OVERRIDE;
+
+ virtual bool InsertPayload(const uint8_t* incoming_payload,
+ int32_t payload_len_byte,
+ uint8_t payload_type,
+ uint32_t timestamp) OVERRIDE;
+
+ virtual bool SetMinimumPlayoutDelay(int time_ms) OVERRIDE;
+
+ virtual bool SetMaximumPlayoutDelay(int time_ms) OVERRIDE;
+
+ virtual int LeastRequiredDelayMs() const OVERRIDE;
+
+ virtual bool PlayoutTimestamp(uint32_t* timestamp) OVERRIDE;
+
+ virtual bool Get10MsAudio(AudioFrame* audio_frame) OVERRIDE;
+
+ virtual bool NetworkStatistics(
+ ACMNetworkStatistics* network_statistics) OVERRIDE;
+
+ virtual bool EnableNack(size_t max_nack_list_size) OVERRIDE;
+
+ virtual void DisableNack() OVERRIDE;
+
+ virtual std::vector<uint16_t> GetNackList(
+ int round_trip_time_ms) const OVERRIDE;
+
+ virtual void GetDecodingCallStatistics(
+ AudioDecodingCallStats* call_stats) const OVERRIDE;
+
+ private:
+ // Temporary method to be used during redesign phase.
+ // Maps |codec_type| (a value from the anonymous enum in acm2::ACMCodecDB) to
+ // |codec_name|, |sample_rate_hz|, and |channels|.
+ // TODO(henrik.lundin) Remove this when no longer needed.
+ static bool MapCodecTypeToParameters(int codec_type,
+ std::string* codec_name,
+ int* sample_rate_hz,
+ int* channels);
+
+ scoped_ptr<acm2::AudioCodingModuleImpl> acm_old_;
+ int playout_frequency_hz_;
+};
+
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_AUDIO_CODING_MODULE_IMPL_H_
