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/*
* Copyright (c) 2019 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.
*/
#ifndef MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_CODER_OPUS_COMMON_H_
#define MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_CODER_OPUS_COMMON_H_
#include <string>
#include <utility>
#include <vector>
#include "absl/strings/string_view.h"
#include "absl/types/optional.h"
#include "api/audio_codecs/audio_decoder.h"
#include "api/audio_codecs/audio_format.h"
#include "rtc_base/string_to_number.h"
namespace webrtc {
absl::optional<std::string> GetFormatParameter(const SdpAudioFormat& format,
absl::string_view param);
template <typename T>
absl::optional<T> GetFormatParameter(const SdpAudioFormat& format,
absl::string_view param) {
return rtc::StringToNumber<T>(GetFormatParameter(format, param).value_or(""));
}
template <>
absl::optional<std::vector<unsigned char>> GetFormatParameter(
const SdpAudioFormat& format,
absl::string_view param);
class OpusFrame : public AudioDecoder::EncodedAudioFrame {
public:
OpusFrame(AudioDecoder* decoder,
rtc::Buffer&& payload,
bool is_primary_payload)
: decoder_(decoder),
payload_(std::move(payload)),
is_primary_payload_(is_primary_payload) {}
size_t Duration() const override {
int ret;
if (is_primary_payload_) {
ret = decoder_->PacketDuration(payload_.data(), payload_.size());
} else {
ret = decoder_->PacketDurationRedundant(payload_.data(), payload_.size());
}
return (ret < 0) ? 0 : static_cast<size_t>(ret);
}
bool IsDtxPacket() const override { return payload_.size() <= 2; }
absl::optional<DecodeResult> Decode(
rtc::ArrayView<int16_t> decoded) const override {
AudioDecoder::SpeechType speech_type = AudioDecoder::kSpeech;
int ret;
if (is_primary_payload_) {
ret = decoder_->Decode(
payload_.data(), payload_.size(), decoder_->SampleRateHz(),
decoded.size() * sizeof(int16_t), decoded.data(), &speech_type);
} else {
ret = decoder_->DecodeRedundant(
payload_.data(), payload_.size(), decoder_->SampleRateHz(),
decoded.size() * sizeof(int16_t), decoded.data(), &speech_type);
}
if (ret < 0)
return absl::nullopt;
return DecodeResult{static_cast<size_t>(ret), speech_type};
}
private:
AudioDecoder* const decoder_;
const rtc::Buffer payload_;
const bool is_primary_payload_;
};
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_CODER_OPUS_COMMON_H_