mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 05:40:42 +01:00
d93b7b91e0
Bug: webrtc:14763 Change-Id: I9615a9ce41c9b577c4ebd4cdcc9885bfbc5dac48 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293040 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39706}
91 lines
3 KiB
C++
91 lines
3 KiB
C++
/*
|
|
* Copyright (c) 2016 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/neteq/tools/encode_neteq_input.h"
|
|
|
|
#include <utility>
|
|
|
|
#include "rtc_base/checks.h"
|
|
#include "rtc_base/numerics/safe_conversions.h"
|
|
|
|
namespace webrtc {
|
|
namespace test {
|
|
|
|
EncodeNetEqInput::EncodeNetEqInput(std::unique_ptr<Generator> generator,
|
|
std::unique_ptr<AudioEncoder> encoder,
|
|
int64_t input_duration_ms)
|
|
: generator_(std::move(generator)),
|
|
encoder_(std::move(encoder)),
|
|
input_duration_ms_(input_duration_ms) {
|
|
event_ = GetNextEvent();
|
|
}
|
|
|
|
EncodeNetEqInput::~EncodeNetEqInput() = default;
|
|
|
|
std::unique_ptr<NetEqInput::Event> EncodeNetEqInput::PopEvent() {
|
|
std::unique_ptr<NetEqInput::Event> event_to_return = std::move(event_);
|
|
event_ = GetNextEvent();
|
|
return event_to_return;
|
|
}
|
|
|
|
bool EncodeNetEqInput::ended() const {
|
|
return next_output_event_ms_ > input_duration_ms_ + kOutputPeriodMs;
|
|
}
|
|
|
|
absl::optional<RTPHeader> EncodeNetEqInput::NextHeader() const {
|
|
RTC_DCHECK(packet_data_);
|
|
return packet_data_->header;
|
|
}
|
|
|
|
std::unique_ptr<NetEqInput::Event> EncodeNetEqInput::GetNextEvent() {
|
|
std::unique_ptr<NetEqInput::Event> event;
|
|
if (packet_data_ == nullptr) {
|
|
CreatePacket();
|
|
}
|
|
if (next_output_event_ms_ < packet_data_->timestamp_ms_) {
|
|
event = std::make_unique<GetAudio>(next_output_event_ms_);
|
|
next_output_event_ms_ += kOutputPeriodMs;
|
|
return event;
|
|
}
|
|
event = std::move(packet_data_);
|
|
CreatePacket();
|
|
return event;
|
|
}
|
|
|
|
void EncodeNetEqInput::CreatePacket() {
|
|
// Create a new PacketData object.
|
|
packet_data_.reset(new NetEqInput::PacketData);
|
|
RTC_DCHECK_EQ(packet_data_->payload.size(), 0);
|
|
|
|
// Loop until we get a packet.
|
|
AudioEncoder::EncodedInfo info;
|
|
RTC_DCHECK(!info.send_even_if_empty);
|
|
int num_blocks = 0;
|
|
while (packet_data_->payload.size() == 0 && !info.send_even_if_empty) {
|
|
const size_t num_samples = rtc::CheckedDivExact(
|
|
static_cast<int>(encoder_->SampleRateHz() * kOutputPeriodMs), 1000);
|
|
|
|
info = encoder_->Encode(rtp_timestamp_, generator_->Generate(num_samples),
|
|
&packet_data_->payload);
|
|
|
|
rtp_timestamp_ += rtc::dchecked_cast<uint32_t>(
|
|
num_samples * encoder_->RtpTimestampRateHz() /
|
|
encoder_->SampleRateHz());
|
|
++num_blocks;
|
|
}
|
|
packet_data_->header.timestamp = info.encoded_timestamp;
|
|
packet_data_->header.payloadType = info.payload_type;
|
|
packet_data_->header.sequenceNumber = sequence_number_++;
|
|
packet_data_->timestamp_ms_ = next_packet_time_ms_;
|
|
next_packet_time_ms_ += num_blocks * kOutputPeriodMs;
|
|
}
|
|
|
|
} // namespace test
|
|
} // namespace webrtc
|