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36274f9158
This is a reland of1dbe30c7e8Original change's description: > Reland "Default enable WebRTC-SendSideBwe-WithOverhead." > > This is a reland of87c1950841> > Original change's description: > > Default enable WebRTC-SendSideBwe-WithOverhead. > > > > Bug: webrtc:6762 > > Change-Id: I18ace06a33b3b60d5a19796d4769f70cd977d604 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188801 > > Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > Reviewed-by: Ali Tofigh <alito@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#32472} > > Bug: webrtc:6762 > Change-Id: Icf096a8755d29600a13bd08b1f22f5a79de21e90 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190143 > Reviewed-by: Ali Tofigh <alito@webrtc.org> > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32492} Bug: webrtc:6762 Change-Id: I6d79894a213fc42d2338409e7513247725881b1a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191221 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Ali Tofigh <alito@webrtc.org> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32534}
108 lines
3.9 KiB
C++
108 lines
3.9 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 <utility>
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#include <vector>
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#include "absl/types/optional.h"
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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 "api/units/time_delta.h"
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#include "rtc_base/constructor_magic.h"
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#include "system_wrappers/include/field_trial.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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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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};
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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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void SetTargetBitrate(int target_bps) override;
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void OnReceivedTargetAudioBitrate(int target_bps) override;
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void OnReceivedUplinkBandwidth(
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int target_audio_bitrate_bps,
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absl::optional<int64_t> bwe_period_ms) override;
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void OnReceivedUplinkAllocation(BitrateAllocationUpdate update) override;
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void OnReceivedOverhead(size_t overhead_bytes_per_packet) 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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absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
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const 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 constexpr int kDefaultBitRate = 32000;
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static constexpr int kMinBitrateBps = 10000;
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static constexpr int MaxBitrateBps(int sample_rate_hz) {
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return sample_rate_hz == 32000 ? 56000 : 32000;
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}
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void SetTargetBitrate(int target_bps, bool subtract_per_packet_overhead);
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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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// 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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// Cache the value of the "WebRTC-SendSideBwe-WithOverhead" field trial.
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const bool send_side_bwe_with_overhead_ =
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!field_trial::IsDisabled("WebRTC-SendSideBwe-WithOverhead");
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// When we send a packet, expect this many bytes of headers to be added to it.
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// Start out with a reasonable default that we can use until we receive a real
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// value.
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DataSize overhead_per_packet_ = DataSize::Bytes(28);
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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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