diff options
Diffstat (limited to 'third_party/libwebrtc/modules/audio_coding/neteq/tools/encode_neteq_input.cc')
-rw-r--r-- | third_party/libwebrtc/modules/audio_coding/neteq/tools/encode_neteq_input.cc | 94 |
1 files changed, 94 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/encode_neteq_input.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/encode_neteq_input.cc new file mode 100644 index 0000000000..87b987ddb6 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/encode_neteq_input.cc @@ -0,0 +1,94 @@ +/* + * 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) { + CreatePacket(); +} + +EncodeNetEqInput::~EncodeNetEqInput() = default; + +absl::optional<int64_t> EncodeNetEqInput::NextPacketTime() const { + RTC_DCHECK(packet_data_); + return static_cast<int64_t>(packet_data_->time_ms); +} + +absl::optional<int64_t> EncodeNetEqInput::NextOutputEventTime() const { + return next_output_event_ms_; +} + +std::unique_ptr<NetEqInput::PacketData> EncodeNetEqInput::PopPacket() { + RTC_DCHECK(packet_data_); + // Grab the packet to return... + std::unique_ptr<PacketData> packet_to_return = std::move(packet_data_); + // ... and line up the next packet for future use. + CreatePacket(); + + return packet_to_return; +} + +void EncodeNetEqInput::AdvanceOutputEvent() { + next_output_event_ms_ += kOutputPeriodMs; +} + +bool EncodeNetEqInput::ended() const { + return next_output_event_ms_ > input_duration_ms_; +} + +absl::optional<RTPHeader> EncodeNetEqInput::NextHeader() const { + RTC_DCHECK(packet_data_); + return packet_data_->header; +} + +void EncodeNetEqInput::CreatePacket() { + // Create a new PacketData object. + RTC_DCHECK(!packet_data_); + 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_->time_ms = next_packet_time_ms_; + next_packet_time_ms_ += num_blocks * kOutputPeriodMs; +} + +} // namespace test +} // namespace webrtc |