mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 22:00:47 +01:00
48b32b748e
This appears unused. If deleted, other code related to isac bandwidth estimation becomes unused and may be deleted in followup cls. Bug: webrtc:10098 Change-Id: Ifeac2e90de895b12c337ea28cc33704350b9abf4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153667 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29252}
81 lines
2.7 KiB
C++
81 lines
2.7 KiB
C++
/*
|
|
* 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 MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_
|
|
#define MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_
|
|
|
|
#include <vector>
|
|
|
|
#include "api/audio_codecs/audio_encoder.h"
|
|
#include "api/scoped_refptr.h"
|
|
#include "rtc_base/constructor_magic.h"
|
|
|
|
namespace webrtc {
|
|
|
|
template <typename T>
|
|
class AudioEncoderIsacT final : public AudioEncoder {
|
|
public:
|
|
// Allowed combinations of sample rate, frame size, and bit rate are
|
|
// - 16000 Hz, 30 ms, 10000-32000 bps
|
|
// - 16000 Hz, 60 ms, 10000-32000 bps
|
|
// - 32000 Hz, 30 ms, 10000-56000 bps (if T has super-wideband support)
|
|
struct Config {
|
|
bool IsOk() const;
|
|
|
|
int payload_type = 103;
|
|
int sample_rate_hz = 16000;
|
|
int frame_size_ms = 30;
|
|
int bit_rate = kDefaultBitRate; // Limit on the short-term average bit
|
|
// rate, in bits/s.
|
|
int max_payload_size_bytes = -1;
|
|
int max_bit_rate = -1;
|
|
};
|
|
|
|
explicit AudioEncoderIsacT(const Config& config);
|
|
~AudioEncoderIsacT() 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 EncodeImpl(uint32_t rtp_timestamp,
|
|
rtc::ArrayView<const int16_t> audio,
|
|
rtc::Buffer* encoded) override;
|
|
void Reset() override;
|
|
|
|
private:
|
|
// This value is taken from STREAM_SIZE_MAX_60 for iSAC float (60 ms) and
|
|
// STREAM_MAXW16_60MS for iSAC fix (60 ms).
|
|
static const size_t kSufficientEncodeBufferSizeBytes = 400;
|
|
|
|
static const int kDefaultBitRate = 32000;
|
|
|
|
// Recreate the iSAC encoder instance with the given settings, and save them.
|
|
void RecreateEncoderInstance(const Config& config);
|
|
|
|
Config config_;
|
|
typename T::instance_type* isac_state_ = nullptr;
|
|
|
|
// Have we accepted input but not yet emitted it in a packet?
|
|
bool packet_in_progress_ = false;
|
|
|
|
// Timestamp of the first input of the currently in-progress packet.
|
|
uint32_t packet_timestamp_;
|
|
|
|
// Timestamp of the previously encoded packet.
|
|
uint32_t last_encoded_timestamp_;
|
|
|
|
RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderIsacT);
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_
|