decd9306ae
Rename this accessor function to reflect its new, slightly changed meaning. The reason for the change is that some codecs (iSAC) vary the number of 10 ms frames from packet to packet, and so can't return a truly constant value. BUG=3926 R=henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31849004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7556 4adac7df-926f-26a2-2b94-8c16560cd09d
76 lines
2.9 KiB
C++
76 lines
2.9 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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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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size_t* encoded_bytes,
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uint32_t* encoded_timestamp) {
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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 = Encode(timestamp,
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audio,
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max_encoded_bytes,
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encoded,
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encoded_bytes,
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encoded_timestamp);
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CHECK_LE(*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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protected:
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virtual bool Encode(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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size_t* encoded_bytes,
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uint32_t* encoded_timestamp) = 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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