Revert of Move FilePlayer and FileRecorder to Voice Engine (patchset #6 id:100001 of https://codereview.webrtc.org/2240163002/ )
Reason for revert:
Breaks downstream code, so revert again. Yay.
Original issue's description:
> Move FilePlayer and FileRecorder to Voice Engine
>
> Because Voice Engine was the only user.
>
> (This is a re-land of https://codereview.webrtc.org/2037623002, which
> had to be reverted.)
>
> NOPRESUBMIT=True
>
> Committed: https://crrev.com/dc65ea29b3270ad418050658ad962ddd33ee70c1
> Cr-Commit-Position: refs/heads/master@{#13757}
TBR=perkj@webrtc.org,kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review-Url: https://codereview.webrtc.org/2245153002
Cr-Commit-Position: refs/heads/master@{#13758}
diff --git a/webrtc/modules/utility/source/coder.cc b/webrtc/modules/utility/source/coder.cc
new file mode 100644
index 0000000..f2ae43e
--- /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