Index: webrtc/modules/audio_coding/codecs/audio_decoder.cc |
diff --git a/webrtc/modules/audio_coding/codecs/audio_decoder.cc b/webrtc/modules/audio_coding/codecs/audio_decoder.cc |
deleted file mode 100644 |
index afa5115d5a1740036c566ded3c9c9019b78bb233..0000000000000000000000000000000000000000 |
--- a/webrtc/modules/audio_coding/codecs/audio_decoder.cc |
+++ /dev/null |
@@ -1,129 +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/modules/audio_coding/codecs/audio_decoder.h" |
- |
-#include <assert.h> |
-#include <memory> |
-#include <utility> |
- |
-#include "webrtc/base/array_view.h" |
-#include "webrtc/base/checks.h" |
-#include "webrtc/base/sanitizer.h" |
-#include "webrtc/base/trace_event.h" |
-#include "webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h" |
- |
-namespace webrtc { |
- |
-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 LegacyEncodedAudioFrame(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; |
-} |
- |
-int AudioDecoder::IncomingPacket(const uint8_t* payload, |
- size_t payload_len, |
- uint16_t rtp_sequence_number, |
- uint32_t rtp_timestamp, |
- uint32_t arrival_timestamp) { |
- return 0; |
-} |
- |
-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: |
- assert(false); |
- return kSpeech; |
- } |
-} |
- |
-} // namespace webrtc |