| /* |
| * 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 "api/audio_codecs/audio_decoder.h" |
| |
| #include <assert.h> |
| |
| #include <memory> |
| #include <utility> |
| |
| #include "api/array_view.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/sanitizer.h" |
| #include "rtc_base/trace_event.h" |
| |
| namespace webrtc { |
| |
| namespace { |
| |
| class OldStyleEncodedFrame final : public AudioDecoder::EncodedAudioFrame { |
| public: |
| OldStyleEncodedFrame(AudioDecoder* decoder, rtc::Buffer&& payload) |
| : decoder_(decoder), payload_(std::move(payload)) {} |
| |
| size_t Duration() const override { |
| const int ret = decoder_->PacketDuration(payload_.data(), payload_.size()); |
| return ret < 0 ? 0 : static_cast<size_t>(ret); |
| } |
| |
| absl::optional<DecodeResult> Decode( |
| rtc::ArrayView<int16_t> decoded) const override { |
| auto speech_type = AudioDecoder::kSpeech; |
| const int ret = decoder_->Decode( |
| payload_.data(), payload_.size(), decoder_->SampleRateHz(), |
| decoded.size() * sizeof(int16_t), decoded.data(), &speech_type); |
| return ret < 0 ? absl::nullopt |
| : absl::optional<DecodeResult>( |
| {static_cast<size_t>(ret), speech_type}); |
| } |
| |
| private: |
| AudioDecoder* const decoder_; |
| const rtc::Buffer payload_; |
| }; |
| |
| } // namespace |
| |
| bool AudioDecoder::EncodedAudioFrame::IsDtxPacket() const { |
| return false; |
| } |
| |
| AudioDecoder::ParseResult::ParseResult() = default; |
| AudioDecoder::ParseResult::ParseResult(ParseResult&& b) = default; |
| AudioDecoder::ParseResult::ParseResult(uint32_t timestamp, |
| int priority, |
| std::unique_ptr<EncodedAudioFrame> frame) |
| : timestamp(timestamp), priority(priority), frame(std::move(frame)) { |
| RTC_DCHECK_GE(priority, 0); |
| } |
| |
| AudioDecoder::ParseResult::~ParseResult() = default; |
| |
| AudioDecoder::ParseResult& AudioDecoder::ParseResult::operator=( |
| ParseResult&& b) = default; |
| |
| std::vector<AudioDecoder::ParseResult> AudioDecoder::ParsePayload( |
| rtc::Buffer&& payload, |
| uint32_t timestamp) { |
| std::vector<ParseResult> results; |
| std::unique_ptr<EncodedAudioFrame> frame( |
| new OldStyleEncodedFrame(this, std::move(payload))); |
| results.emplace_back(timestamp, 0, std::move(frame)); |
| return results; |
| } |
| |
| int AudioDecoder::Decode(const uint8_t* encoded, |
| size_t encoded_len, |
| int sample_rate_hz, |
| size_t max_decoded_bytes, |
| int16_t* decoded, |
| SpeechType* speech_type) { |
| TRACE_EVENT0("webrtc", "AudioDecoder::Decode"); |
| rtc::MsanCheckInitialized(rtc::MakeArrayView(encoded, encoded_len)); |
| int duration = PacketDuration(encoded, encoded_len); |
| if (duration >= 0 && |
| duration * Channels() * sizeof(int16_t) > max_decoded_bytes) { |
| return -1; |
| } |
| return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded, |
| speech_type); |
| } |
| |
| int AudioDecoder::DecodeRedundant(const uint8_t* encoded, |
| size_t encoded_len, |
| int sample_rate_hz, |
| size_t max_decoded_bytes, |
| int16_t* decoded, |
| SpeechType* speech_type) { |
| TRACE_EVENT0("webrtc", "AudioDecoder::DecodeRedundant"); |
| rtc::MsanCheckInitialized(rtc::MakeArrayView(encoded, encoded_len)); |
| int duration = PacketDurationRedundant(encoded, encoded_len); |
| if (duration >= 0 && |
| duration * Channels() * sizeof(int16_t) > max_decoded_bytes) { |
| return -1; |
| } |
| return DecodeRedundantInternal(encoded, encoded_len, sample_rate_hz, decoded, |
| speech_type); |
| } |
| |
| int AudioDecoder::DecodeRedundantInternal(const uint8_t* encoded, |
| size_t encoded_len, |
| int sample_rate_hz, |
| int16_t* decoded, |
| SpeechType* speech_type) { |
| return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded, |
| speech_type); |
| } |
| |
| bool AudioDecoder::HasDecodePlc() const { |
| return false; |
| } |
| |
| size_t AudioDecoder::DecodePlc(size_t num_frames, int16_t* decoded) { |
| return 0; |
| } |
| |
| // TODO(bugs.webrtc.org/9676): Remove default implementation. |
| void AudioDecoder::GeneratePlc(size_t /*requested_samples_per_channel*/, |
| rtc::BufferT<int16_t>* /*concealment_audio*/) {} |
| |
| int AudioDecoder::ErrorCode() { |
| return 0; |
| } |
| |
| int AudioDecoder::PacketDuration(const uint8_t* encoded, |
| size_t encoded_len) const { |
| return kNotImplemented; |
| } |
| |
| int AudioDecoder::PacketDurationRedundant(const uint8_t* encoded, |
| size_t encoded_len) const { |
| return kNotImplemented; |
| } |
| |
| bool AudioDecoder::PacketHasFec(const uint8_t* encoded, |
| size_t encoded_len) const { |
| return false; |
| } |
| |
| AudioDecoder::SpeechType AudioDecoder::ConvertSpeechType(int16_t type) { |
| switch (type) { |
| case 0: // TODO(hlundin): Both iSAC and Opus return 0 for speech. |
| case 1: |
| return kSpeech; |
| case 2: |
| return kComfortNoise; |
| default: |
| RTC_NOTREACHED(); |
| return kSpeech; |
| } |
| } |
| |
| } // namespace webrtc |