1eda4e3db6
This should be safe to land now that issue 4143 was resolved (in r8492). This change effectively reverts 8488. TBR=kwiberg@webrtc.org Original commit message: This CL changes the way the decoder sample rate is set and updated. In practice, it only concerns the iSAC (float) codec. One single iSAC decoder instance is used for both wideband and super-wideband decoding, and the instance must be told to switch output frequency if the payload type changes. This used to be done through a call to UpdateDecoderSampleRate, but is now instead done in the Decode call as an extra parameter. Review URL: https://webrtc-codereview.appspot.com/39289004 Cr-Commit-Position: refs/heads/master@{#8496} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8496 4adac7df-926f-26a2-2b94-8c16560cd09d
75 lines
2.2 KiB
C++
75 lines
2.2 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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namespace webrtc {
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int AudioDecoder::DecodeRedundant(const uint8_t* encoded,
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size_t encoded_len,
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int sample_rate_hz,
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int16_t* decoded,
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SpeechType* speech_type) {
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return Decode(encoded, encoded_len, sample_rate_hz, decoded, speech_type);
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}
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bool AudioDecoder::HasDecodePlc() const { return false; }
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int AudioDecoder::DecodePlc(int num_frames, int16_t* decoded) { return -1; }
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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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