mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-14 06:10:40 +01:00
cd2a92f8e0
This is not used and adds a lot of maintenance overhead to the code since it requires that the transport feedback adapter communicates directly with audio send stream. This also means that the packet loss tracker used as input for this can be removed and a lot of wiring up code overall. Bug: webrtc:9883 Change-Id: I25689fb622ed89cbb378c27212a159485f5f53be Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156502 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29667}
135 lines
4.2 KiB
C++
135 lines
4.2 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.
|
|
*/
|
|
|
|
#include "modules/audio_coding/codecs/red/audio_encoder_copy_red.h"
|
|
|
|
#include <string.h>
|
|
|
|
#include <utility>
|
|
#include <vector>
|
|
|
|
#include "rtc_base/checks.h"
|
|
|
|
namespace webrtc {
|
|
|
|
AudioEncoderCopyRed::Config::Config() = default;
|
|
AudioEncoderCopyRed::Config::Config(Config&&) = default;
|
|
AudioEncoderCopyRed::Config::~Config() = default;
|
|
|
|
AudioEncoderCopyRed::AudioEncoderCopyRed(Config&& config)
|
|
: speech_encoder_(std::move(config.speech_encoder)),
|
|
red_payload_type_(config.payload_type) {
|
|
RTC_CHECK(speech_encoder_) << "Speech encoder not provided.";
|
|
}
|
|
|
|
AudioEncoderCopyRed::~AudioEncoderCopyRed() = default;
|
|
|
|
int AudioEncoderCopyRed::SampleRateHz() const {
|
|
return speech_encoder_->SampleRateHz();
|
|
}
|
|
|
|
size_t AudioEncoderCopyRed::NumChannels() const {
|
|
return speech_encoder_->NumChannels();
|
|
}
|
|
|
|
int AudioEncoderCopyRed::RtpTimestampRateHz() const {
|
|
return speech_encoder_->RtpTimestampRateHz();
|
|
}
|
|
|
|
size_t AudioEncoderCopyRed::Num10MsFramesInNextPacket() const {
|
|
return speech_encoder_->Num10MsFramesInNextPacket();
|
|
}
|
|
|
|
size_t AudioEncoderCopyRed::Max10MsFramesInAPacket() const {
|
|
return speech_encoder_->Max10MsFramesInAPacket();
|
|
}
|
|
|
|
int AudioEncoderCopyRed::GetTargetBitrate() const {
|
|
return speech_encoder_->GetTargetBitrate();
|
|
}
|
|
|
|
AudioEncoder::EncodedInfo AudioEncoderCopyRed::EncodeImpl(
|
|
uint32_t rtp_timestamp,
|
|
rtc::ArrayView<const int16_t> audio,
|
|
rtc::Buffer* encoded) {
|
|
const size_t primary_offset = encoded->size();
|
|
EncodedInfo info = speech_encoder_->Encode(rtp_timestamp, audio, encoded);
|
|
|
|
RTC_CHECK(info.redundant.empty()) << "Cannot use nested redundant encoders.";
|
|
RTC_DCHECK_EQ(encoded->size() - primary_offset, info.encoded_bytes);
|
|
|
|
if (info.encoded_bytes > 0) {
|
|
// |info| will be implicitly cast to an EncodedInfoLeaf struct, effectively
|
|
// discarding the (empty) vector of redundant information. This is
|
|
// intentional.
|
|
info.redundant.push_back(info);
|
|
RTC_DCHECK_EQ(info.redundant.size(), 1);
|
|
if (secondary_info_.encoded_bytes > 0) {
|
|
encoded->AppendData(secondary_encoded_);
|
|
info.redundant.push_back(secondary_info_);
|
|
RTC_DCHECK_EQ(info.redundant.size(), 2);
|
|
}
|
|
// Save primary to secondary.
|
|
secondary_encoded_.SetData(encoded->data() + primary_offset,
|
|
info.encoded_bytes);
|
|
secondary_info_ = info;
|
|
RTC_DCHECK_EQ(info.speech, info.redundant[0].speech);
|
|
}
|
|
// Update main EncodedInfo.
|
|
info.payload_type = red_payload_type_;
|
|
info.encoded_bytes = 0;
|
|
for (std::vector<EncodedInfoLeaf>::const_iterator it = info.redundant.begin();
|
|
it != info.redundant.end(); ++it) {
|
|
info.encoded_bytes += it->encoded_bytes;
|
|
}
|
|
return info;
|
|
}
|
|
|
|
void AudioEncoderCopyRed::Reset() {
|
|
speech_encoder_->Reset();
|
|
secondary_encoded_.Clear();
|
|
secondary_info_.encoded_bytes = 0;
|
|
}
|
|
|
|
bool AudioEncoderCopyRed::SetFec(bool enable) {
|
|
return speech_encoder_->SetFec(enable);
|
|
}
|
|
|
|
bool AudioEncoderCopyRed::SetDtx(bool enable) {
|
|
return speech_encoder_->SetDtx(enable);
|
|
}
|
|
|
|
bool AudioEncoderCopyRed::SetApplication(Application application) {
|
|
return speech_encoder_->SetApplication(application);
|
|
}
|
|
|
|
void AudioEncoderCopyRed::SetMaxPlaybackRate(int frequency_hz) {
|
|
speech_encoder_->SetMaxPlaybackRate(frequency_hz);
|
|
}
|
|
|
|
rtc::ArrayView<std::unique_ptr<AudioEncoder>>
|
|
AudioEncoderCopyRed::ReclaimContainedEncoders() {
|
|
return rtc::ArrayView<std::unique_ptr<AudioEncoder>>(&speech_encoder_, 1);
|
|
}
|
|
|
|
void AudioEncoderCopyRed::OnReceivedUplinkPacketLossFraction(
|
|
float uplink_packet_loss_fraction) {
|
|
speech_encoder_->OnReceivedUplinkPacketLossFraction(
|
|
uplink_packet_loss_fraction);
|
|
}
|
|
|
|
void AudioEncoderCopyRed::OnReceivedUplinkBandwidth(
|
|
int target_audio_bitrate_bps,
|
|
absl::optional<int64_t> bwe_period_ms) {
|
|
speech_encoder_->OnReceivedUplinkBandwidth(target_audio_bitrate_bps,
|
|
bwe_period_ms);
|
|
}
|
|
|
|
} // namespace webrtc
|