| /* |
| * 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 "webrtc/api/audio_codecs/audio_encoder.h" |
| |
| #include "webrtc/rtc_base/checks.h" |
| #include "webrtc/rtc_base/trace_event.h" |
| |
| namespace webrtc { |
| |
| ANAStats::ANAStats() = default; |
| ANAStats::~ANAStats() = default; |
| ANAStats::ANAStats(const ANAStats&) = default; |
| |
| AudioEncoder::EncodedInfo::EncodedInfo() = default; |
| AudioEncoder::EncodedInfo::EncodedInfo(const EncodedInfo&) = default; |
| AudioEncoder::EncodedInfo::EncodedInfo(EncodedInfo&&) = default; |
| AudioEncoder::EncodedInfo::~EncodedInfo() = default; |
| AudioEncoder::EncodedInfo& AudioEncoder::EncodedInfo::operator=( |
| const EncodedInfo&) = default; |
| AudioEncoder::EncodedInfo& AudioEncoder::EncodedInfo::operator=(EncodedInfo&&) = |
| default; |
| |
| int AudioEncoder::RtpTimestampRateHz() const { |
| return SampleRateHz(); |
| } |
| |
| AudioEncoder::EncodedInfo AudioEncoder::Encode( |
| uint32_t rtp_timestamp, |
| rtc::ArrayView<const int16_t> audio, |
| rtc::Buffer* encoded) { |
| TRACE_EVENT0("webrtc", "AudioEncoder::Encode"); |
| RTC_CHECK_EQ(audio.size(), |
| static_cast<size_t>(NumChannels() * SampleRateHz() / 100)); |
| |
| const size_t old_size = encoded->size(); |
| EncodedInfo info = EncodeImpl(rtp_timestamp, audio, encoded); |
| RTC_CHECK_EQ(encoded->size() - old_size, info.encoded_bytes); |
| return info; |
| } |
| |
| bool AudioEncoder::SetFec(bool enable) { |
| return !enable; |
| } |
| |
| bool AudioEncoder::SetDtx(bool enable) { |
| return !enable; |
| } |
| |
| bool AudioEncoder::GetDtx() const { |
| return false; |
| } |
| |
| bool AudioEncoder::SetApplication(Application application) { |
| return false; |
| } |
| |
| void AudioEncoder::SetMaxPlaybackRate(int frequency_hz) {} |
| |
| void AudioEncoder::SetTargetBitrate(int target_bps) {} |
| |
| rtc::ArrayView<std::unique_ptr<AudioEncoder>> |
| AudioEncoder::ReclaimContainedEncoders() { |
| return nullptr; |
| } |
| |
| bool AudioEncoder::EnableAudioNetworkAdaptor(const std::string& config_string, |
| RtcEventLog* event_log) { |
| return false; |
| } |
| |
| void AudioEncoder::DisableAudioNetworkAdaptor() {} |
| |
| void AudioEncoder::OnReceivedUplinkPacketLossFraction( |
| float uplink_packet_loss_fraction) {} |
| |
| void AudioEncoder::OnReceivedUplinkRecoverablePacketLossFraction( |
| float uplink_recoverable_packet_loss_fraction) {} |
| |
| void AudioEncoder::OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) { |
| OnReceivedUplinkBandwidth(target_audio_bitrate_bps, rtc::Optional<int64_t>()); |
| } |
| |
| void AudioEncoder::OnReceivedUplinkBandwidth( |
| int target_audio_bitrate_bps, |
| rtc::Optional<int64_t> bwe_period_ms) {} |
| |
| void AudioEncoder::OnReceivedRtt(int rtt_ms) {} |
| |
| void AudioEncoder::OnReceivedOverhead(size_t overhead_bytes_per_packet) {} |
| |
| void AudioEncoder::SetReceiverFrameLengthRange(int min_frame_length_ms, |
| int max_frame_length_ms) {} |
| |
| ANAStats AudioEncoder::GetANAStats() const { |
| return ANAStats(); |
| } |
| |
| } // namespace webrtc |