3b79daff14
BUG=3926 R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/35469004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7883 4adac7df-926f-26a2-2b94-8c16560cd09d
85 lines
3.1 KiB
C++
85 lines
3.1 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 "webrtc/base/checks.h"
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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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// codec type must have an implementation of this class.
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class AudioEncoder {
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public:
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struct EncodedInfo {
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EncodedInfo() : 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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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 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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CHECK_EQ(num_samples_per_channel,
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static_cast<size_t>(sample_rate_hz() / 100));
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bool ret = EncodeInternal(timestamp,
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audio,
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max_encoded_bytes,
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encoded,
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info);
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CHECK_LE(info->encoded_bytes, max_encoded_bytes);
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return ret;
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}
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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 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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protected:
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virtual bool EncodeInternal(uint32_t 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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