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10542f21c8
Mechanically generated by running this command: tools_webrtc/do-renames.sh update all-renames.txt && git cl format Then manually updating: tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc Bug: webrtc:10159 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833 Reviewed-on: https://webrtc-review.googlesource.com/c/115653 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26226}
65 lines
2.2 KiB
C++
65 lines
2.2 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 MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_
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#define MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_
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#include <memory>
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#include "api/audio_codecs/audio_encoder.h"
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#include "api/audio_codecs/g722/audio_encoder_g722_config.h"
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#include "modules/audio_coding/codecs/g722/g722_interface.h"
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#include "rtc_base/buffer.h"
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#include "rtc_base/constructor_magic.h"
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namespace webrtc {
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class AudioEncoderG722Impl final : public AudioEncoder {
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public:
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AudioEncoderG722Impl(const AudioEncoderG722Config& config, int payload_type);
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~AudioEncoderG722Impl() override;
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int SampleRateHz() const override;
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size_t NumChannels() const override;
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int RtpTimestampRateHz() const override;
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size_t Num10MsFramesInNextPacket() const override;
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size_t Max10MsFramesInAPacket() const override;
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int GetTargetBitrate() const override;
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void Reset() override;
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protected:
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EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
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rtc::ArrayView<const int16_t> audio,
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rtc::Buffer* encoded) override;
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private:
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// The encoder state for one channel.
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struct EncoderState {
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G722EncInst* encoder;
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std::unique_ptr<int16_t[]> speech_buffer; // Queued up for encoding.
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rtc::Buffer encoded_buffer; // Already encoded.
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EncoderState();
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~EncoderState();
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};
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size_t SamplesPerChannel() const;
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const size_t num_channels_;
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const int payload_type_;
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const size_t num_10ms_frames_per_packet_;
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size_t num_10ms_frames_buffered_;
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uint32_t first_timestamp_in_buffer_;
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const std::unique_ptr<EncoderState[]> encoders_;
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rtc::Buffer interleave_buffer_;
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RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderG722Impl);
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_
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