zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Delete support for enabling adaptive isac mode

Browse Source 
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}
This commit is contained in:
Niels Möller
2019-09-19 15:21:54 +02:00
committed by Commit Bot
parent b5e4785464
commit 48b32b748e
8 changed files with 6 additions and 126 deletions
Download Patch File Download Diff File Expand all files Collapse all files

4
modules/audio_coding/BUILD.gn
View File

@ -370,15 +370,13 @@ rtc_source_set("ilbc_c") {
]
}
rtc_static_library("isac_common") {
rtc_source_set("isac_common") {
poisonous = [ "audio_codecs" ]
sources = [
"codecs/isac/audio_decoder_isac_t.h",
"codecs/isac/audio_decoder_isac_t_impl.h",
"codecs/isac/audio_encoder_isac_t.h",
"codecs/isac/audio_encoder_isac_t_impl.h",
"codecs/isac/locked_bandwidth_info.cc",
"codecs/isac/locked_bandwidth_info.h",
]
deps = [
":isac_bwinfo",

3
modules/audio_coding/codecs/isac/audio_decoder_isac_t.h
View File

@ -16,7 +16,6 @@
#include "absl/types/optional.h"
#include "api/audio_codecs/audio_decoder.h"
#include "api/scoped_refptr.h"
#include "modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
@ -26,7 +25,6 @@ class AudioDecoderIsacT final : public AudioDecoder {
public:
struct Config {
bool IsOk() const;
rtc::scoped_refptr<LockedIsacBandwidthInfo> bwinfo;
int sample_rate_hz = 16000;
};
explicit AudioDecoderIsacT(const Config& config);
@ -52,7 +50,6 @@ class AudioDecoderIsacT final : public AudioDecoder {
private:
typename T::instance_type* isac_state_;
int sample_rate_hz_;
rtc::scoped_refptr<LockedIsacBandwidthInfo> bwinfo_;
RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderIsacT);
};

12
modules/audio_coding/codecs/isac/audio_decoder_isac_t_impl.h
View File

@ -22,16 +22,11 @@ bool AudioDecoderIsacT<T>::Config::IsOk() const {
template <typename T>
AudioDecoderIsacT<T>::AudioDecoderIsacT(const Config& config)
: sample_rate_hz_(config.sample_rate_hz), bwinfo_(config.bwinfo) {
: sample_rate_hz_(config.sample_rate_hz) {
RTC_CHECK(config.IsOk()) << "Unsupported sample rate "
<< config.sample_rate_hz;
RTC_CHECK_EQ(0, T::Create(&isac_state_));
T::DecoderInit(isac_state_);
if (bwinfo_) {
IsacBandwidthInfo bi;
T::GetBandwidthInfo(isac_state_, &bi);
bwinfo_->Set(bi);
}
RTC_CHECK_EQ(0, T::SetDecSampRate(isac_state_, sample_rate_hz_));
}
@ -78,11 +73,6 @@ int AudioDecoderIsacT<T>::IncomingPacket(const uint8_t* payload,
int ret = T::UpdateBwEstimate(isac_state_, payload, payload_len,
rtp_sequence_number, rtp_timestamp,
arrival_timestamp);
if (bwinfo_) {
IsacBandwidthInfo bwinfo;
T::GetBandwidthInfo(isac_state_, &bwinfo);
bwinfo_->Set(bwinfo);
}
return ret;
}

12
modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
View File

@ -15,7 +15,6 @@
#include "api/audio_codecs/audio_encoder.h"
#include "api/scoped_refptr.h"
#include "modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
@ -30,8 +29,6 @@ class AudioEncoderIsacT final : public AudioEncoder {
struct Config {
bool IsOk() const;
rtc::scoped_refptr<LockedIsacBandwidthInfo> bwinfo;
int payload_type = 103;
int sample_rate_hz = 16000;
int frame_size_ms = 30;
@ -39,14 +36,6 @@ class AudioEncoderIsacT final : public AudioEncoder {
// rate, in bits/s.
int max_payload_size_bytes = -1;
int max_bit_rate = -1;
// If true, the encoder will dynamically adjust frame size and bit rate;
// the configured values are then merely the starting point.
bool adaptive_mode = false;
// In adaptive mode, prevent adaptive changes to the frame size. (Not used
// in nonadaptive mode.)
bool enforce_frame_size = false;
};
explicit AudioEncoderIsacT(const Config& config);
@ -74,7 +63,6 @@ class AudioEncoderIsacT final : public AudioEncoder {
Config config_;
typename T::instance_type* isac_state_ = nullptr;
rtc::scoped_refptr<LockedIsacBandwidthInfo> bwinfo_;
// Have we accepted input but not yet emitted it in a packet?
bool packet_in_progress_ = false;

19
modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
View File

@ -21,8 +21,7 @@ bool AudioEncoderIsacT<T>::Config::IsOk() const {
return false;
if (max_payload_size_bytes < 120 && max_payload_size_bytes != -1)
return false;
if (adaptive_mode && !bwinfo)
return false;
switch (sample_rate_hz) {
case 16000:
if (max_bit_rate > 53400)
@ -78,8 +77,6 @@ size_t AudioEncoderIsacT<T>::Max10MsFramesInAPacket() const {
template <typename T>
int AudioEncoderIsacT<T>::GetTargetBitrate() const {
if (config_.adaptive_mode)
return -1;
return config_.bit_rate == 0 ? kDefaultBitRate : config_.bit_rate;
}
@ -93,11 +90,6 @@ AudioEncoder::EncodedInfo AudioEncoderIsacT<T>::EncodeImpl(
packet_in_progress_ = true;
packet_timestamp_ = rtp_timestamp;
}
if (bwinfo_) {
IsacBandwidthInfo bwinfo = bwinfo_->Get();
T::SetBandwidthInfo(isac_state_, &bwinfo);
}
size_t encoded_bytes = encoded->AppendData(
kSufficientEncodeBufferSizeBytes, [&](rtc::ArrayView<uint8_t> encoded) {
int r = T::Encode(isac_state_, audio.data(), encoded.data());
@ -131,19 +123,14 @@ template <typename T>
void AudioEncoderIsacT<T>::RecreateEncoderInstance(const Config& config) {
RTC_CHECK(config.IsOk());
packet_in_progress_ = false;
bwinfo_ = config.bwinfo;
if (isac_state_)
RTC_CHECK_EQ(0, T::Free(isac_state_));
RTC_CHECK_EQ(0, T::Create(&isac_state_));
RTC_CHECK_EQ(0, T::EncoderInit(isac_state_, config.adaptive_mode ? 0 : 1));
RTC_CHECK_EQ(0, T::EncoderInit(isac_state_, 1));
RTC_CHECK_EQ(0, T::SetEncSampRate(isac_state_, config.sample_rate_hz));
const int bit_rate = config.bit_rate == 0 ? kDefaultBitRate : config.bit_rate;
if (config.adaptive_mode) {
RTC_CHECK_EQ(0, T::ControlBwe(isac_state_, bit_rate, config.frame_size_ms,
config.enforce_frame_size));
} else {
RTC_CHECK_EQ(0, T::Control(isac_state_, bit_rate, config.frame_size_ms));
}
if (config.max_payload_size_bytes != -1)
RTC_CHECK_EQ(
0, T::SetMaxPayloadSize(isac_state_, config.max_payload_size_bytes));

21
modules/audio_coding/codecs/isac/locked_bandwidth_info.cc
View File

@ -1,21 +0,0 @@
/*
* Copyright (c) 2015 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.
*/
#include "modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
namespace webrtc {
LockedIsacBandwidthInfo::LockedIsacBandwidthInfo() : ref_count_(0) {
bwinfo_.in_use = 0;
}
LockedIsacBandwidthInfo::~LockedIsacBandwidthInfo() = default;
} // namespace webrtc

56
modules/audio_coding/codecs/isac/locked_bandwidth_info.h
View File

@ -1,56 +0,0 @@
/*
* Copyright (c) 2015 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_LOCKED_BANDWIDTH_INFO_H_
#define MODULES_AUDIO_CODING_CODECS_ISAC_LOCKED_BANDWIDTH_INFO_H_
#include "modules/audio_coding/codecs/isac/bandwidth_info.h"
#include "rtc_base/atomic_ops.h"
#include "rtc_base/critical_section.h"
#include "rtc_base/thread_annotations.h"
namespace webrtc {
// An IsacBandwidthInfo that's safe to access from multiple threads because
// it's protected by a mutex.
class LockedIsacBandwidthInfo final {
public:
LockedIsacBandwidthInfo();
~LockedIsacBandwidthInfo();
IsacBandwidthInfo Get() const {
rtc::CritScope lock(&lock_);
return bwinfo_;
}
void Set(const IsacBandwidthInfo& bwinfo) {
rtc::CritScope lock(&lock_);
bwinfo_ = bwinfo;
}
int AddRef() const { return rtc::AtomicOps::Increment(&ref_count_); }
int Release() const {
const int count = rtc::AtomicOps::Decrement(&ref_count_);
if (count == 0) {
delete this;
}
return count;
}
private:
mutable volatile int ref_count_;
rtc::CriticalSection lock_;
IsacBandwidthInfo bwinfo_ RTC_GUARDED_BY(lock_);
};
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_CODECS_ISAC_LOCKED_BANDWIDTH_INFO_H_

3
modules/audio_coding/neteq/audio_decoder_unittest.cc
View File

@ -353,7 +353,6 @@ class AudioDecoderIsacFloatTest : public AudioDecoderTest {
AudioEncoderIsacFloatImpl::Config config;
config.payload_type = payload_type_;
config.sample_rate_hz = codec_input_rate_hz_;
config.adaptive_mode = false;
config.frame_size_ms =
1000 * static_cast<int>(frame_size_) / codec_input_rate_hz_;
audio_encoder_.reset(new AudioEncoderIsacFloatImpl(config));
@ -373,7 +372,6 @@ class AudioDecoderIsacSwbTest : public AudioDecoderTest {
AudioEncoderIsacFloatImpl::Config config;
config.payload_type = payload_type_;
config.sample_rate_hz = codec_input_rate_hz_;
config.adaptive_mode = false;
config.frame_size_ms =
1000 * static_cast<int>(frame_size_) / codec_input_rate_hz_;
audio_encoder_.reset(new AudioEncoderIsacFloatImpl(config));
@ -393,7 +391,6 @@ class AudioDecoderIsacFixTest : public AudioDecoderTest {
AudioEncoderIsacFixImpl::Config config;
config.payload_type = payload_type_;
config.sample_rate_hz = codec_input_rate_hz_;
config.adaptive_mode = false;
config.frame_size_ms =
1000 * static_cast<int>(frame_size_) / codec_input_rate_hz_;
audio_encoder_.reset(new AudioEncoderIsacFixImpl(config));