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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
106 lines
3.2 KiB
C++
106 lines
3.2 KiB
C++
/*
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* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "api/audio_codecs/audio_encoder.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/trace_event.h"
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namespace webrtc {
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ANAStats::ANAStats() = default;
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ANAStats::~ANAStats() = default;
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ANAStats::ANAStats(const ANAStats&) = default;
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AudioEncoder::EncodedInfo::EncodedInfo() = default;
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AudioEncoder::EncodedInfo::EncodedInfo(const EncodedInfo&) = default;
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AudioEncoder::EncodedInfo::EncodedInfo(EncodedInfo&&) = default;
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AudioEncoder::EncodedInfo::~EncodedInfo() = default;
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AudioEncoder::EncodedInfo& AudioEncoder::EncodedInfo::operator=(
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const EncodedInfo&) = default;
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AudioEncoder::EncodedInfo& AudioEncoder::EncodedInfo::operator=(EncodedInfo&&) =
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default;
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int AudioEncoder::RtpTimestampRateHz() const {
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return SampleRateHz();
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}
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AudioEncoder::EncodedInfo AudioEncoder::Encode(
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uint32_t rtp_timestamp,
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rtc::ArrayView<const int16_t> audio,
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rtc::Buffer* encoded) {
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TRACE_EVENT0("webrtc", "AudioEncoder::Encode");
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RTC_CHECK_EQ(audio.size(),
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static_cast<size_t>(NumChannels() * SampleRateHz() / 100));
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const size_t old_size = encoded->size();
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EncodedInfo info = EncodeImpl(rtp_timestamp, audio, encoded);
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RTC_CHECK_EQ(encoded->size() - old_size, info.encoded_bytes);
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return info;
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}
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bool AudioEncoder::SetFec(bool enable) {
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return !enable;
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}
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bool AudioEncoder::SetDtx(bool enable) {
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return !enable;
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}
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bool AudioEncoder::GetDtx() const {
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return false;
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}
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bool AudioEncoder::SetApplication(Application application) {
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return false;
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}
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void AudioEncoder::SetMaxPlaybackRate(int frequency_hz) {}
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void AudioEncoder::SetTargetBitrate(int target_bps) {}
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rtc::ArrayView<std::unique_ptr<AudioEncoder>>
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AudioEncoder::ReclaimContainedEncoders() {
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return nullptr;
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}
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bool AudioEncoder::EnableAudioNetworkAdaptor(const std::string& config_string,
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RtcEventLog* event_log) {
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return false;
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}
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void AudioEncoder::DisableAudioNetworkAdaptor() {}
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void AudioEncoder::OnReceivedUplinkPacketLossFraction(
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float uplink_packet_loss_fraction) {}
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void AudioEncoder::OnReceivedUplinkRecoverablePacketLossFraction(
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float uplink_recoverable_packet_loss_fraction) {}
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void AudioEncoder::OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) {
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OnReceivedUplinkBandwidth(target_audio_bitrate_bps, rtc::Optional<int64_t>());
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}
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void AudioEncoder::OnReceivedUplinkBandwidth(
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int target_audio_bitrate_bps,
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rtc::Optional<int64_t> bwe_period_ms) {}
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void AudioEncoder::OnReceivedRtt(int rtt_ms) {}
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void AudioEncoder::OnReceivedOverhead(size_t overhead_bytes_per_packet) {}
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void AudioEncoder::SetReceiverFrameLengthRange(int min_frame_length_ms,
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int max_frame_length_ms) {}
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ANAStats AudioEncoder::GetANAStats() const {
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return ANAStats();
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}
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} // namespace webrtc
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