| /* |
| * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_ |
| #define WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_ |
| |
| #include <algorithm> |
| |
| #include "webrtc/base/checks.h" |
| #include "webrtc/typedefs.h" |
| |
| namespace webrtc { |
| |
| // This is the interface class for encoders in AudioCoding module. Each codec |
| // codec type must have an implementation of this class. |
| class AudioEncoder { |
| public: |
| struct EncodedInfo { |
| uint32_t encoded_timestamp; |
| }; |
| |
| virtual ~AudioEncoder() {} |
| |
| // Accepts one 10 ms block of input audio (i.e., sample_rate_hz() / 100 * |
| // num_channels() samples). Multi-channel audio must be sample-interleaved. |
| // If successful, the encoder produces zero or more bytes of output in |
| // |encoded|, and provides the number of encoded bytes in |encoded_bytes|. |
| // In case of error, false is returned, otherwise true. It is an error for the |
| // encoder to attempt to produce more than |max_encoded_bytes| bytes of |
| // output. |
| bool Encode(uint32_t timestamp, |
| const int16_t* audio, |
| size_t num_samples_per_channel, |
| size_t max_encoded_bytes, |
| uint8_t* encoded, |
| size_t* encoded_bytes, |
| EncodedInfo* info) { |
| CHECK_EQ(num_samples_per_channel, |
| static_cast<size_t>(sample_rate_hz() / 100)); |
| bool ret = Encode(timestamp, |
| audio, |
| max_encoded_bytes, |
| encoded, |
| encoded_bytes, |
| info); |
| CHECK_LE(*encoded_bytes, max_encoded_bytes); |
| return ret; |
| } |
| |
| // Return the input sample rate in Hz and the number of input channels. |
| // These are constants set at instantiation time. |
| virtual int sample_rate_hz() const = 0; |
| virtual int num_channels() const = 0; |
| |
| // Returns the number of 10 ms frames the encoder will put in the next |
| // packet. This value may only change when Encode() outputs a packet; i.e., |
| // the encoder may vary the number of 10 ms frames from packet to packet, but |
| // it must decide the length of the next packet no later than when outputting |
| // the preceding packet. |
| virtual int Num10MsFramesInNextPacket() const = 0; |
| |
| protected: |
| virtual bool Encode(uint32_t timestamp, |
| const int16_t* audio, |
| size_t max_encoded_bytes, |
| uint8_t* encoded, |
| size_t* encoded_bytes, |
| EncodedInfo* info) = 0; |
| }; |
| |
| } // namespace webrtc |
| #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_ |