d7b7ae8bda
Also removes virtual from VideoDecoder::Decode and updated mocks and tests accordingly to use VideoDecoder::DecodeInternal instead. BUG=webrtc:5167 R=henrik.lundin@webrtc.org Review URL: https://codereview.webrtc.org/1512483003 . Cr-Commit-Position: refs/heads/master@{#10935}
104 lines
3.4 KiB
C++
104 lines
3.4 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
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#include <assert.h>
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#include "webrtc/base/checks.h"
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#include "webrtc/base/trace_event.h"
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namespace webrtc {
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int AudioDecoder::Decode(const uint8_t* encoded, size_t encoded_len,
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int sample_rate_hz, size_t max_decoded_bytes,
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int16_t* decoded, SpeechType* speech_type) {
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TRACE_EVENT0("webrtc", "AudioDecoder::Decode");
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int duration = PacketDuration(encoded, encoded_len);
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if (duration >= 0 &&
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duration * Channels() * sizeof(int16_t) > max_decoded_bytes) {
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return -1;
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}
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return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded,
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speech_type);
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}
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int AudioDecoder::DecodeRedundant(const uint8_t* encoded, size_t encoded_len,
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int sample_rate_hz, size_t max_decoded_bytes,
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int16_t* decoded, SpeechType* speech_type) {
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TRACE_EVENT0("webrtc", "AudioDecoder::DecodeRedundant");
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int duration = PacketDurationRedundant(encoded, encoded_len);
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if (duration >= 0 &&
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duration * Channels() * sizeof(int16_t) > max_decoded_bytes) {
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return -1;
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}
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return DecodeRedundantInternal(encoded, encoded_len, sample_rate_hz, decoded,
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speech_type);
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}
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int AudioDecoder::DecodeRedundantInternal(const uint8_t* encoded,
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size_t encoded_len,
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int sample_rate_hz, int16_t* decoded,
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SpeechType* speech_type) {
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return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded,
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speech_type);
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}
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bool AudioDecoder::HasDecodePlc() const { return false; }
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size_t AudioDecoder::DecodePlc(size_t num_frames, int16_t* decoded) {
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return 0;
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}
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int AudioDecoder::IncomingPacket(const uint8_t* payload,
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size_t payload_len,
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uint16_t rtp_sequence_number,
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uint32_t rtp_timestamp,
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uint32_t arrival_timestamp) {
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return 0;
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}
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int AudioDecoder::ErrorCode() { return 0; }
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int AudioDecoder::PacketDuration(const uint8_t* encoded,
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size_t encoded_len) const {
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return kNotImplemented;
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}
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int AudioDecoder::PacketDurationRedundant(const uint8_t* encoded,
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size_t encoded_len) const {
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return kNotImplemented;
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}
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bool AudioDecoder::PacketHasFec(const uint8_t* encoded,
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size_t encoded_len) const {
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return false;
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}
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CNG_dec_inst* AudioDecoder::CngDecoderInstance() {
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FATAL() << "Not a CNG decoder";
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return NULL;
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}
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AudioDecoder::SpeechType AudioDecoder::ConvertSpeechType(int16_t type) {
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switch (type) {
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case 0: // TODO(hlundin): Both iSAC and Opus return 0 for speech.
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case 1:
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return kSpeech;
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case 2:
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return kComfortNoise;
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default:
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assert(false);
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return kSpeech;
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}
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}
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} // namespace webrtc
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