Index: webrtc/modules/utility/source/coder.cc |
diff --git a/webrtc/modules/utility/source/coder.cc b/webrtc/modules/utility/source/coder.cc |
deleted file mode 100644 |
index 71f969097fd218365b015d00c6a3b3b8fbeaf8b2..0000000000000000000000000000000000000000 |
--- a/webrtc/modules/utility/source/coder.cc |
+++ /dev/null |
@@ -1,116 +0,0 @@ |
-/* |
- * 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/audio_format_conversion.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.pltype, |
- CodecInstToSdp(codec_inst))) { |
- 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 |