Index: webrtc/modules/utility/source/coder.cc |
diff --git a/webrtc/modules/utility/source/coder.cc b/webrtc/modules/utility/source/coder.cc |
new file mode 100644 |
index 0000000000000000000000000000000000000000..f2ae43eb108a939f261ae17646a05c26fd307643 |
--- /dev/null |
+++ b/webrtc/modules/utility/source/coder.cc |
@@ -0,0 +1,116 @@ |
+/* |
+ * Copyright (c) 2012 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. |
+ */ |
+ |
+#include "webrtc/common_types.h" |
+#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h" |
+#include "webrtc/modules/include/module_common_types.h" |
+#include "webrtc/modules/utility/source/coder.h" |
+ |
+namespace webrtc { |
+namespace { |
+AudioCodingModule::Config GetAcmConfig(uint32_t id) { |
+ AudioCodingModule::Config config; |
+ // This class does not handle muted output. |
+ config.neteq_config.enable_muted_state = false; |
+ config.id = id; |
+ config.decoder_factory = CreateBuiltinAudioDecoderFactory(); |
+ return config; |
+} |
+} // namespace |
+ |
+AudioCoder::AudioCoder(uint32_t instance_id) |
+ : acm_(AudioCodingModule::Create(GetAcmConfig(instance_id))), |
+ receive_codec_(), |
+ encode_timestamp_(0), |
+ encoded_data_(nullptr), |
+ encoded_length_in_bytes_(0), |
+ decode_timestamp_(0) { |
+ acm_->InitializeReceiver(); |
+ acm_->RegisterTransportCallback(this); |
+} |
+ |
+AudioCoder::~AudioCoder() {} |
+ |
+int32_t AudioCoder::SetEncodeCodec(const CodecInst& codec_inst) { |
+ const bool success = codec_manager_.RegisterEncoder(codec_inst) && |
+ codec_manager_.MakeEncoder(&rent_a_codec_, acm_.get()); |
+ return success ? 0 : -1; |
+} |
+ |
+int32_t AudioCoder::SetDecodeCodec(const CodecInst& codec_inst) { |
+ if (acm_->RegisterReceiveCodec(codec_inst, [&] { |
+ return rent_a_codec_.RentIsacDecoder(codec_inst.plfreq); |
+ }) == -1) { |
+ return -1; |
+ } |
+ memcpy(&receive_codec_, &codec_inst, sizeof(CodecInst)); |
+ return 0; |
+} |
+ |
+int32_t AudioCoder::Decode(AudioFrame& decoded_audio, |
+ uint32_t samp_freq_hz, |
+ const int8_t* incoming_payload, |
+ size_t payload_length) { |
+ if (payload_length > 0) { |
+ const uint8_t payload_type = receive_codec_.pltype; |
+ decode_timestamp_ += receive_codec_.pacsize; |
+ if (acm_->IncomingPayload((const uint8_t*)incoming_payload, payload_length, |
+ payload_type, decode_timestamp_) == -1) { |
+ return -1; |
+ } |
+ } |
+ bool muted; |
+ int32_t ret = |
+ acm_->PlayoutData10Ms((uint16_t)samp_freq_hz, &decoded_audio, &muted); |
+ RTC_DCHECK(!muted); |
+ return ret; |
+} |
+ |
+int32_t AudioCoder::PlayoutData(AudioFrame& decoded_audio, |
+ uint16_t& samp_freq_hz) { |
+ bool muted; |
+ int32_t ret = acm_->PlayoutData10Ms(samp_freq_hz, &decoded_audio, &muted); |
+ RTC_DCHECK(!muted); |
+ return ret; |
+} |
+ |
+int32_t AudioCoder::Encode(const AudioFrame& audio, |
+ int8_t* encoded_data, |
+ size_t& encoded_length_in_bytes) { |
+ // Fake a timestamp in case audio doesn't contain a correct timestamp. |
+ // Make a local copy of the audio frame since audio is const |
+ AudioFrame audio_frame; |
+ audio_frame.CopyFrom(audio); |
+ audio_frame.timestamp_ = encode_timestamp_; |
+ encode_timestamp_ += static_cast<uint32_t>(audio_frame.samples_per_channel_); |
+ |
+ // For any codec with a frame size that is longer than 10 ms the encoded |
+ // length in bytes should be zero until a a full frame has been encoded. |
+ encoded_length_in_bytes_ = 0; |
+ if (acm_->Add10MsData((AudioFrame&)audio_frame) == -1) { |
+ return -1; |
+ } |
+ encoded_data_ = encoded_data; |
+ encoded_length_in_bytes = encoded_length_in_bytes_; |
+ return 0; |
+} |
+ |
+int32_t AudioCoder::SendData(FrameType /* frame_type */, |
+ uint8_t /* payload_type */, |
+ uint32_t /* time_stamp */, |
+ const uint8_t* payload_data, |
+ size_t payload_size, |
+ const RTPFragmentationHeader* /* fragmentation*/) { |
+ memcpy(encoded_data_, payload_data, sizeof(uint8_t) * payload_size); |
+ encoded_length_in_bytes_ = payload_size; |
+ return 0; |
+} |
+ |
+} // namespace webrtc |