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
287 lines
9.2 KiB
C++
287 lines
9.2 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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#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_AUDIO_DECODER_IMPL_H_
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#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_AUDIO_DECODER_IMPL_H_
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#include <assert.h>
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#ifndef AUDIO_DECODER_UNITTEST
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// If this is compiled as a part of the audio_deoder_unittest, the codec
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// selection is made in the gypi file instead of in engine_configurations.h.
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#include "webrtc/engine_configurations.h"
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#endif
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
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#include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h"
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#ifdef WEBRTC_CODEC_G722
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#include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h"
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#endif
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#ifdef WEBRTC_CODEC_ILBC
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#include "webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h"
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#endif
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#ifdef WEBRTC_CODEC_OPUS
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#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
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#endif
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class AudioDecoderPcmU : public AudioDecoder {
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public:
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AudioDecoderPcmU() {}
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virtual int Decode(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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virtual int Init() { return 0; }
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virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len) const;
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private:
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DISALLOW_COPY_AND_ASSIGN(AudioDecoderPcmU);
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};
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class AudioDecoderPcmA : public AudioDecoder {
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public:
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AudioDecoderPcmA() {}
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virtual int Decode(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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virtual int Init() { return 0; }
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virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len) const;
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private:
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DISALLOW_COPY_AND_ASSIGN(AudioDecoderPcmA);
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};
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class AudioDecoderPcmUMultiCh : public AudioDecoderPcmU {
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public:
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explicit AudioDecoderPcmUMultiCh(size_t channels) : AudioDecoderPcmU() {
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assert(channels > 0);
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channels_ = channels;
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}
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private:
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DISALLOW_COPY_AND_ASSIGN(AudioDecoderPcmUMultiCh);
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};
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class AudioDecoderPcmAMultiCh : public AudioDecoderPcmA {
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public:
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explicit AudioDecoderPcmAMultiCh(size_t channels) : AudioDecoderPcmA() {
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assert(channels > 0);
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channels_ = channels;
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}
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private:
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DISALLOW_COPY_AND_ASSIGN(AudioDecoderPcmAMultiCh);
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};
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#ifdef WEBRTC_CODEC_PCM16
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// This class handles all four types (i.e., sample rates) of PCM16B codecs.
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// The type is specified in the constructor parameter |type|.
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class AudioDecoderPcm16B : public AudioDecoder {
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public:
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AudioDecoderPcm16B();
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virtual int Decode(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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virtual int Init() { return 0; }
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virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len) const;
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private:
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DISALLOW_COPY_AND_ASSIGN(AudioDecoderPcm16B);
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};
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// This class handles all four types (i.e., sample rates) of PCM16B codecs.
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// The type is specified in the constructor parameter |type|, and the number
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// of channels is derived from the type.
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class AudioDecoderPcm16BMultiCh : public AudioDecoderPcm16B {
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public:
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explicit AudioDecoderPcm16BMultiCh(int num_channels);
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private:
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DISALLOW_COPY_AND_ASSIGN(AudioDecoderPcm16BMultiCh);
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};
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#endif
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#ifdef WEBRTC_CODEC_ILBC
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class AudioDecoderIlbc : public AudioDecoder {
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public:
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AudioDecoderIlbc();
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virtual ~AudioDecoderIlbc();
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virtual int Decode(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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virtual bool HasDecodePlc() const { return true; }
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virtual int DecodePlc(int num_frames, int16_t* decoded);
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virtual int Init();
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private:
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IlbcDecoderInstance* dec_state_;
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DISALLOW_COPY_AND_ASSIGN(AudioDecoderIlbc);
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};
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#endif
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#ifdef WEBRTC_CODEC_G722
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class AudioDecoderG722 : public AudioDecoder {
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public:
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AudioDecoderG722();
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virtual ~AudioDecoderG722();
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virtual int Decode(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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virtual bool HasDecodePlc() const { return false; }
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virtual int Init();
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virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len) const;
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private:
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G722DecInst* dec_state_;
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DISALLOW_COPY_AND_ASSIGN(AudioDecoderG722);
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};
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class AudioDecoderG722Stereo : public AudioDecoder {
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public:
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AudioDecoderG722Stereo();
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virtual ~AudioDecoderG722Stereo();
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virtual int Decode(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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virtual int Init();
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private:
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// Splits the stereo-interleaved payload in |encoded| into separate payloads
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// for left and right channels. The separated payloads are written to
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// |encoded_deinterleaved|, which must hold at least |encoded_len| samples.
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// The left channel starts at offset 0, while the right channel starts at
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// offset encoded_len / 2 into |encoded_deinterleaved|.
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void SplitStereoPacket(const uint8_t* encoded, size_t encoded_len,
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uint8_t* encoded_deinterleaved);
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G722DecInst* dec_state_left_;
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G722DecInst* dec_state_right_;
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DISALLOW_COPY_AND_ASSIGN(AudioDecoderG722Stereo);
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};
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#endif
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#ifdef WEBRTC_CODEC_OPUS
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class AudioDecoderOpus : public AudioDecoder {
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public:
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explicit AudioDecoderOpus(int num_channels);
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virtual ~AudioDecoderOpus();
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virtual int Decode(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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virtual int 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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virtual int Init();
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virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len) const;
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virtual int PacketDurationRedundant(const uint8_t* encoded,
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size_t encoded_len) const;
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virtual bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const;
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private:
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OpusDecInst* dec_state_;
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DISALLOW_COPY_AND_ASSIGN(AudioDecoderOpus);
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};
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#endif
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// AudioDecoderCng is a special type of AudioDecoder. It inherits from
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// AudioDecoder just to fit in the DecoderDatabase. None of the class methods
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// should be used, except constructor, destructor, and accessors.
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// TODO(hlundin): Consider the possibility to create a super-class to
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// AudioDecoder that is stored in DecoderDatabase. Then AudioDecoder and a
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// specific CngDecoder class could both inherit from that class.
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class AudioDecoderCng : public AudioDecoder {
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public:
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explicit AudioDecoderCng();
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virtual ~AudioDecoderCng();
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virtual int Decode(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 -1;
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}
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virtual int Init();
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virtual int 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) { return -1; }
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virtual CNG_dec_inst* CngDecoderInstance() OVERRIDE { return dec_state_; }
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private:
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CNG_dec_inst* dec_state_;
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DISALLOW_COPY_AND_ASSIGN(AudioDecoderCng);
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};
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enum NetEqDecoder {
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kDecoderPCMu,
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kDecoderPCMa,
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kDecoderPCMu_2ch,
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kDecoderPCMa_2ch,
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kDecoderILBC,
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kDecoderISAC,
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kDecoderISACswb,
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kDecoderISACfb,
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kDecoderPCM16B,
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kDecoderPCM16Bwb,
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kDecoderPCM16Bswb32kHz,
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kDecoderPCM16Bswb48kHz,
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kDecoderPCM16B_2ch,
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kDecoderPCM16Bwb_2ch,
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kDecoderPCM16Bswb32kHz_2ch,
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kDecoderPCM16Bswb48kHz_2ch,
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kDecoderPCM16B_5ch,
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kDecoderG722,
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kDecoderG722_2ch,
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kDecoderRED,
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kDecoderAVT,
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kDecoderCNGnb,
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kDecoderCNGwb,
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kDecoderCNGswb32kHz,
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kDecoderCNGswb48kHz,
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kDecoderArbitrary,
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kDecoderOpus,
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kDecoderOpus_2ch,
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};
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// Returns true if |codec_type| is supported.
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bool CodecSupported(NetEqDecoder codec_type);
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// Returns the sample rate for |codec_type|.
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int CodecSampleRateHz(NetEqDecoder codec_type);
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// Creates an AudioDecoder object of type |codec_type|. Returns NULL for for
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// unsupported codecs, and when creating an AudioDecoder is not applicable
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// (e.g., for RED and DTMF/AVT types).
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AudioDecoder* CreateAudioDecoder(NetEqDecoder codec_type);
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_AUDIO_DECODER_IMPL_H_
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