264e66f7a5
BUG=3926 TBR=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/24029004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7464 4adac7df-926f-26a2-2b94-8c16560cd09d
68 lines
2.5 KiB
C++
68 lines
2.5 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 <limits>
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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 returns the number of bytes. In case of error, -1 is
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// returned. It is an error for the encoder to attempt to produce more than
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// |max_encoded_bytes| bytes of output.
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ssize_t Encode(uint32_t timestamp,
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const int16_t* audio,
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size_t num_samples,
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size_t max_encoded_bytes,
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uint8_t* encoded,
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uint32_t* encoded_timestamp) {
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CHECK_EQ(num_samples,
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static_cast<size_t>(sample_rate_hz() / 100 * num_channels()));
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ssize_t num_bytes =
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Encode(timestamp, audio, max_encoded_bytes, encoded, encoded_timestamp);
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CHECK_LE(num_bytes,
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static_cast<ssize_t>(std::min(
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max_encoded_bytes,
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static_cast<size_t>(std::numeric_limits<ssize_t>::max()))));
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return num_bytes;
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}
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// Returns the input sample rate in Hz, the number of input channels, and the
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// number of 10 ms frames the encoder puts in one output packet. These are
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// 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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virtual int num_10ms_frames_per_packet() const = 0;
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protected:
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virtual ssize_t 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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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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