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https://github.com/mollyim/webrtc.git
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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}
118 lines
3.5 KiB
C++
118 lines
3.5 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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RTC_NOTREACHED();
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}
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void AudioEncoder::OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) {
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OnReceivedUplinkBandwidth(target_audio_bitrate_bps, absl::nullopt);
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}
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void AudioEncoder::OnReceivedUplinkBandwidth(
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int target_audio_bitrate_bps,
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absl::optional<int64_t> bwe_period_ms) {}
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void AudioEncoder::OnReceivedUplinkAllocation(BitrateAllocationUpdate update) {
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OnReceivedUplinkBandwidth(update.target_bitrate.bps(),
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update.bwe_period.ms());
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}
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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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absl::optional<std::pair<TimeDelta, TimeDelta>>
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AudioEncoder::GetFrameLengthRange() const {
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return absl::nullopt;
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}
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} // namespace webrtc
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