Index: webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h |
diff --git a/webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h b/webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h |
deleted file mode 100644 |
index aad75a18e3c0188ff3fb8db3939f2df5e9c06fd3..0000000000000000000000000000000000000000 |
--- a/webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h |
+++ /dev/null |
@@ -1,73 +0,0 @@ |
-/* |
- * 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_G722_INCLUDE_AUDIO_ENCODER_G722_H_ |
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_INCLUDE_AUDIO_ENCODER_G722_H_ |
- |
-#include "webrtc/base/buffer.h" |
-#include "webrtc/base/scoped_ptr.h" |
-#include "webrtc/modules/audio_coding/codecs/audio_encoder.h" |
-#include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h" |
- |
-namespace webrtc { |
- |
-struct CodecInst; |
- |
-class AudioEncoderG722 final : public AudioEncoder { |
- public: |
- struct Config { |
- bool IsOk() const; |
- |
- int payload_type = 9; |
- int frame_size_ms = 20; |
- int num_channels = 1; |
- }; |
- |
- explicit AudioEncoderG722(const Config& config); |
- explicit AudioEncoderG722(const CodecInst& codec_inst); |
- ~AudioEncoderG722() override; |
- |
- size_t MaxEncodedBytes() const override; |
- int SampleRateHz() const override; |
- int NumChannels() const override; |
- int RtpTimestampRateHz() 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: |
- // The encoder state for one channel. |
- struct EncoderState { |
- G722EncInst* encoder; |
- rtc::scoped_ptr<int16_t[]> speech_buffer; // Queued up for encoding. |
- rtc::Buffer encoded_buffer; // Already encoded. |
- EncoderState(); |
- ~EncoderState(); |
- }; |
- |
- size_t SamplesPerChannel() const; |
- |
- const int num_channels_; |
- const int payload_type_; |
- const size_t num_10ms_frames_per_packet_; |
- size_t num_10ms_frames_buffered_; |
- uint32_t first_timestamp_in_buffer_; |
- const rtc::scoped_ptr<EncoderState[]> encoders_; |
- rtc::Buffer interleave_buffer_; |
- RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderG722); |
-}; |
- |
-} // namespace webrtc |
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_INCLUDE_AUDIO_ENCODER_G722_H_ |