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d970807e0c
The type rtc::scoped_refptr<T> is now part of api/. Please include it from api/scoped_refptr.h. More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o. Bug: webrtc:9887, webrtc:8205 No-Try: True Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf Reviewed-on: https://webrtc-review.googlesource.com/c/119041 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26414}
93 lines
3.2 KiB
C++
93 lines
3.2 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_ISAC_AUDIO_ENCODER_ISAC_T_H_
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#define MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_
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#include <vector>
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#include "api/audio_codecs/audio_encoder.h"
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#include "api/scoped_refptr.h"
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#include "modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
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#include "rtc_base/constructor_magic.h"
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namespace webrtc {
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template <typename T>
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class AudioEncoderIsacT final : public AudioEncoder {
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public:
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// Allowed combinations of sample rate, frame size, and bit rate are
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// - 16000 Hz, 30 ms, 10000-32000 bps
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// - 16000 Hz, 60 ms, 10000-32000 bps
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// - 32000 Hz, 30 ms, 10000-56000 bps (if T has super-wideband support)
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struct Config {
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bool IsOk() const;
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rtc::scoped_refptr<LockedIsacBandwidthInfo> bwinfo;
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int payload_type = 103;
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int sample_rate_hz = 16000;
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int frame_size_ms = 30;
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int bit_rate = kDefaultBitRate; // Limit on the short-term average bit
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// rate, in bits/s.
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int max_payload_size_bytes = -1;
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int max_bit_rate = -1;
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// If true, the encoder will dynamically adjust frame size and bit rate;
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// the configured values are then merely the starting point.
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bool adaptive_mode = false;
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// In adaptive mode, prevent adaptive changes to the frame size. (Not used
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// in nonadaptive mode.)
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bool enforce_frame_size = false;
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};
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explicit AudioEncoderIsacT(const Config& config);
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~AudioEncoderIsacT() override;
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int SampleRateHz() const override;
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size_t NumChannels() 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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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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void Reset() override;
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private:
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// This value is taken from STREAM_SIZE_MAX_60 for iSAC float (60 ms) and
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// STREAM_MAXW16_60MS for iSAC fix (60 ms).
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static const size_t kSufficientEncodeBufferSizeBytes = 400;
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static const int kDefaultBitRate = 32000;
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// Recreate the iSAC encoder instance with the given settings, and save them.
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void RecreateEncoderInstance(const Config& config);
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Config config_;
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typename T::instance_type* isac_state_ = nullptr;
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rtc::scoped_refptr<LockedIsacBandwidthInfo> bwinfo_;
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// Have we accepted input but not yet emitted it in a packet?
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bool packet_in_progress_ = false;
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// Timestamp of the first input of the currently in-progress packet.
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uint32_t packet_timestamp_;
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// Timestamp of the previously encoded packet.
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uint32_t last_encoded_timestamp_;
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RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderIsacT);
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_
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