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Diffstat (limited to 'webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h')
-rw-r--r-- | webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h | 63 |
1 files changed, 63 insertions, 0 deletions
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h b/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h new file mode 100644 index 0000000000..102a274642 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h @@ -0,0 +1,63 @@ +/* + * 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_ILBC_AUDIO_ENCODER_ILBC_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_ + +#include "webrtc/base/scoped_ptr.h" +#include "webrtc/modules/audio_coding/codecs/audio_encoder.h" +#include "webrtc/modules/audio_coding/codecs/ilbc/ilbc.h" + +namespace webrtc { + +struct CodecInst; + +class AudioEncoderIlbc final : public AudioEncoder { + public: + struct Config { + bool IsOk() const; + + int payload_type = 102; + int frame_size_ms = 30; // Valid values are 20, 30, 40, and 60 ms. + // Note that frame size 40 ms produces encodings with two 20 ms frames in + // them, and frame size 60 ms consists of two 30 ms frames. + }; + + explicit AudioEncoderIlbc(const Config& config); + explicit AudioEncoderIlbc(const CodecInst& codec_inst); + ~AudioEncoderIlbc() override; + + size_t MaxEncodedBytes() const override; + int SampleRateHz() const override; + size_t NumChannels() const override; + size_t Num10MsFramesInNextPacket() const override; + size_t Max10MsFramesInAPacket() const override; + int GetTargetBitrate() const override; + EncodedInfo EncodeInternal(uint32_t rtp_timestamp, + rtc::ArrayView<const int16_t> audio, + size_t max_encoded_bytes, + uint8_t* encoded) override; + void Reset() override; + + private: + size_t RequiredOutputSizeBytes() const; + + static const size_t kMaxSamplesPerPacket = 480; + const Config config_; + const size_t num_10ms_frames_per_packet_; + size_t num_10ms_frames_buffered_; + uint32_t first_timestamp_in_buffer_; + int16_t input_buffer_[kMaxSamplesPerPacket]; + IlbcEncoderInstance* encoder_; + RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderIlbc); +}; + +} // namespace webrtc +#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_ |