| Index: webrtc/voice_engine/coder.cc
|
| diff --git a/webrtc/voice_engine/coder.cc b/webrtc/voice_engine/coder.cc
|
| deleted file mode 100644
|
| index 6337e103a14707bd2264cf5dff51e2c3fd70140e..0000000000000000000000000000000000000000
|
| --- a/webrtc/voice_engine/coder.cc
|
| +++ /dev/null
|
| @@ -1,118 +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/voice_engine/coder.h"
|
| -
|
| -#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
|
| -#include "webrtc/common_types.h"
|
| -#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
|
| -#include "webrtc/modules/include/module_common_types.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)
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| - : acm_(AudioCodingModule::Create(GetAcmConfig(instance_id))),
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| - receive_codec_(),
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| - encode_timestamp_(0),
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| - encoded_data_(nullptr),
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| - encoded_length_in_bytes_(0),
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| - decode_timestamp_(0) {
|
| - acm_->InitializeReceiver();
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| - acm_->RegisterTransportCallback(this);
|
| -}
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| -
|
| -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,
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| - uint32_t samp_freq_hz,
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| - const int8_t* incoming_payload,
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| - size_t payload_length) {
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| - 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,
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| - payload_type, decode_timestamp_) == -1) {
|
| - return -1;
|
| - }
|
| - }
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| - bool muted;
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| - 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,
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| - uint16_t samp_freq_hz) {
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| - bool muted;
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| - int32_t ret = acm_->PlayoutData10Ms(samp_freq_hz, decoded_audio, &muted);
|
| - RTC_DCHECK(!muted);
|
| - return ret;
|
| -}
|
| -
|
| -int32_t AudioCoder::Encode(const AudioFrame& audio,
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| - int8_t* encoded_data,
|
| - size_t* encoded_length_in_bytes) {
|
| - // Fake a timestamp in case audio doesn't contain a correct timestamp.
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| - // Make a local copy of the audio frame since audio is const
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| - AudioFrame audio_frame;
|
| - audio_frame.CopyFrom(audio);
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| - audio_frame.timestamp_ = encode_timestamp_;
|
| - encode_timestamp_ += static_cast<uint32_t>(audio_frame.samples_per_channel_);
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| -
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| - // For any codec with a frame size that is longer than 10 ms the encoded
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| - // length in bytes should be zero until a a full frame has been encoded.
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| - encoded_length_in_bytes_ = 0;
|
| - encoded_data_ = encoded_data;
|
| - if (acm_->Add10MsData((AudioFrame&)audio_frame) == -1) {
|
| - return -1;
|
| - }
|
| -
|
| - *encoded_length_in_bytes = encoded_length_in_bytes_;
|
| - return 0;
|
| -}
|
| -
|
| -int32_t AudioCoder::SendData(FrameType /* frame_type */,
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| - uint8_t /* payload_type */,
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| - uint32_t /* time_stamp */,
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| - const uint8_t* payload_data,
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| - 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
|
|
|