3f81fcd2e8
If the specification for the speech encoder hasn't changed, we should reuse it instead of recreating it. Otherwise, we lose its state. (This problem was originally discovered because AudioEncoderOpus instances would forget that they were supposed to be using DTX.) BUG=webrtc:6020, chromium:622647 Review-Url: https://codereview.webrtc.org/2089183002 Cr-Commit-Position: refs/heads/master@{#13273}
67 lines
2.0 KiB
C++
67 lines
2.0 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 "webrtc/modules/audio_coding/codecs/audio_encoder.h"
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#include "webrtc/base/checks.h"
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#include "webrtc/base/trace_event.h"
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namespace webrtc {
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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::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::SetProjectedPacketLossRate(double fraction) {}
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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() { return nullptr; }
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} // namespace webrtc
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