f45c8ca88b
This effectively reverts r8249. This new class inherits from ACMGenericCodec. The purpose is to wrap AudioEncoder objects into an ACMGenericCodec interface. This is a temporary construction that will be used during the ACM redesign work. BUG=4228 COAUTHOR=kwiberg@webrtc.org TBR=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/38919004 Cr-Commit-Position: refs/heads/master@{#8255} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8255 4adac7df-926f-26a2-2b94-8c16560cd09d
106 lines
4.2 KiB
C++
106 lines
4.2 KiB
C++
/*
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* Copyright (c) 2014 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_CODECS_AUDIO_ENCODER_H_
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#define WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_
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#include <algorithm>
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#include <vector>
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#include "webrtc/typedefs.h"
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namespace webrtc {
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// This is the interface class for encoders in AudioCoding module. Each codec
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// type must have an implementation of this class.
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class AudioEncoder {
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public:
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struct EncodedInfoLeaf {
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EncodedInfoLeaf()
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: encoded_bytes(0), encoded_timestamp(0), payload_type(0) {}
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size_t encoded_bytes;
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uint32_t encoded_timestamp;
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int payload_type;
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};
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// This is the main struct for auxiliary encoding information. Each encoded
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// packet should be accompanied by one EncodedInfo struct, containing the
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// total number of |encoded_bytes|, the |encoded_timestamp| and the
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// |payload_type|. If the packet contains redundant encodings, the |redundant|
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// vector will be populated with EncodedInfoLeaf structs. Each struct in the
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// vector represents one encoding; the order of structs in the vector is the
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// same as the order in which the actual payloads are written to the byte
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// stream. When EncoderInfoLeaf structs are present in the vector, the main
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// struct's |encoded_bytes| will be the sum of all the |encoded_bytes| in the
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// vector.
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struct EncodedInfo : public EncodedInfoLeaf {
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EncodedInfo();
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~EncodedInfo();
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std::vector<EncodedInfoLeaf> redundant;
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};
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virtual ~AudioEncoder() {}
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// Accepts one 10 ms block of input audio (i.e., sample_rate_hz() / 100 *
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// num_channels() samples). Multi-channel audio must be sample-interleaved.
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// If successful, the encoder produces zero or more bytes of output in
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// |encoded|, and provides the number of encoded bytes in |encoded_bytes|.
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// In case of error, false is returned, otherwise true. It is an error for the
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// encoder to attempt to produce more than |max_encoded_bytes| bytes of
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// output.
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bool Encode(uint32_t rtp_timestamp,
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const int16_t* audio,
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size_t num_samples_per_channel,
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size_t max_encoded_bytes,
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uint8_t* encoded,
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EncodedInfo* info);
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// Return the input sample rate in Hz and the number of input channels.
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// These are constants set at instantiation time.
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virtual int sample_rate_hz() const = 0;
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virtual int num_channels() const = 0;
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// Returns the rate with which the RTP timestamps are updated. By default,
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// this is the same as sample_rate_hz().
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virtual int rtp_timestamp_rate_hz() const;
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// Returns the number of 10 ms frames the encoder will put in the next
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// packet. This value may only change when Encode() outputs a packet; i.e.,
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// the encoder may vary the number of 10 ms frames from packet to packet, but
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// it must decide the length of the next packet no later than when outputting
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// the preceding packet.
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virtual int Num10MsFramesInNextPacket() const = 0;
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// Returns the maximum value that can be returned by
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// Num10MsFramesInNextPacket().
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virtual int Max10MsFramesInAPacket() const = 0;
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// Changes the target bitrate. The implementation is free to alter this value,
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// e.g., if the desired value is outside the valid range.
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virtual void SetTargetBitrate(int bits_per_second) {}
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// Tells the implementation what the projected packet loss rate is. The rate
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// is in the range [0.0, 1.0]. This rate is typically used to adjust channel
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// coding efforts, such as FEC.
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virtual void SetProjectedPacketLossRate(double fraction) {}
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protected:
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virtual bool EncodeInternal(uint32_t rtp_timestamp,
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const int16_t* audio,
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size_t max_encoded_bytes,
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uint8_t* encoded,
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EncodedInfo* info) = 0;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_
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