blob: f72d03b887de9356df67d44bb20f7d8963f868c5 [file] [log] [blame]
/*
* 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