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/*
* Copyright (c) 2015 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 "modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
#include <memory>
#include <utility>
#include "modules/audio_coding/codecs/ilbc/ilbc.h"
#include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
namespace webrtc {
AudioDecoderIlbcImpl::AudioDecoderIlbcImpl() {
WebRtcIlbcfix_DecoderCreate(&dec_state_);
WebRtcIlbcfix_Decoderinit30Ms(dec_state_);
}
AudioDecoderIlbcImpl::~AudioDecoderIlbcImpl() {
WebRtcIlbcfix_DecoderFree(dec_state_);
}
bool AudioDecoderIlbcImpl::HasDecodePlc() const {
return true;
}
int AudioDecoderIlbcImpl::DecodeInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
RTC_DCHECK_EQ(sample_rate_hz, 8000);
int16_t temp_type = 1; // Default is speech.
int ret = WebRtcIlbcfix_Decode(dec_state_, encoded, encoded_len, decoded,
&temp_type);
*speech_type = ConvertSpeechType(temp_type);
return ret;
}
size_t AudioDecoderIlbcImpl::DecodePlc(size_t num_frames, int16_t* decoded) {
return WebRtcIlbcfix_NetEqPlc(dec_state_, decoded, num_frames);
}
void AudioDecoderIlbcImpl::Reset() {
WebRtcIlbcfix_Decoderinit30Ms(dec_state_);
}
std::vector<AudioDecoder::ParseResult> AudioDecoderIlbcImpl::ParsePayload(
rtc::Buffer&& payload,
uint32_t timestamp) {
std::vector<ParseResult> results;
size_t bytes_per_frame;
int timestamps_per_frame;
if (payload.size() >= 950) {
RTC_LOG(LS_WARNING)
<< "AudioDecoderIlbcImpl::ParsePayload: Payload too large";
return results;
}
if (payload.size() % 38 == 0) {
// 20 ms frames.
bytes_per_frame = 38;
timestamps_per_frame = 160;
} else if (payload.size() % 50 == 0) {
// 30 ms frames.
bytes_per_frame = 50;
timestamps_per_frame = 240;
} else {
RTC_LOG(LS_WARNING)
<< "AudioDecoderIlbcImpl::ParsePayload: Invalid payload";
return results;
}
RTC_DCHECK_EQ(0, payload.size() % bytes_per_frame);
if (payload.size() == bytes_per_frame) {
std::unique_ptr<EncodedAudioFrame> frame(
new LegacyEncodedAudioFrame(this, std::move(payload)));
results.emplace_back(timestamp, 0, std::move(frame));
} else {
size_t byte_offset;
uint32_t timestamp_offset;
for (byte_offset = 0, timestamp_offset = 0; byte_offset < payload.size();
byte_offset += bytes_per_frame,
timestamp_offset += timestamps_per_frame) {
std::unique_ptr<EncodedAudioFrame> frame(new LegacyEncodedAudioFrame(
this, rtc::Buffer(payload.data() + byte_offset, bytes_per_frame)));
results.emplace_back(timestamp + timestamp_offset, 0, std::move(frame));
}
}
return results;
}
int AudioDecoderIlbcImpl::SampleRateHz() const {
return 8000;
}
size_t AudioDecoderIlbcImpl::Channels() const {
return 1;
}
} // namespace webrtc
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