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Unified Diff: webrtc/modules/audio_coding/codecs/audio_decoder.cc

Issue 2668523004: Move AudioDecoder and related stuff to the api/ directory (Closed)
Patch Set: more review fixes Created 3 years, 10 months ago
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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
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