4f43fcfd49
Renamed the new variant of EncodeInternal to EncodeImpl, so that subclasses implementing one of the EncodeInternal don't need to explicitly contain 'using AudioEncoder::EncodeInternal' to avoid their implementation hiding the other variant of EncodeInternal. This causes a warning (treated as an error) when building using GCC. Review URL: https://codereview.webrtc.org/1764583003 Cr-Commit-Position: refs/heads/master@{#11868}
108 lines
3.2 KiB
C++
108 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 "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() = 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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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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size_t max_encoded_bytes,
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uint8_t* encoded) {
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return DEPRECATED_Encode(rtp_timestamp, audio, max_encoded_bytes, encoded);
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}
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AudioEncoder::EncodedInfo AudioEncoder::DEPRECATED_Encode(
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uint32_t rtp_timestamp,
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rtc::ArrayView<const int16_t> audio,
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size_t max_encoded_bytes,
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uint8_t* 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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EncodedInfo info =
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EncodeInternal(rtp_timestamp, audio, max_encoded_bytes, encoded);
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RTC_CHECK_LE(info.encoded_bytes, max_encoded_bytes);
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return info;
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}
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AudioEncoder::EncodedInfo AudioEncoder::EncodeImpl(
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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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{
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EncodedInfo info;
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encoded->AppendData(MaxEncodedBytes(), [&] (rtc::ArrayView<uint8_t> encoded) {
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info = EncodeInternal(rtp_timestamp, audio,
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encoded.size(), encoded.data());
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return info.encoded_bytes;
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});
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return info;
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}
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AudioEncoder::EncodedInfo AudioEncoder::EncodeInternal(
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uint32_t rtp_timestamp,
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rtc::ArrayView<const int16_t> audio,
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size_t max_encoded_bytes,
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uint8_t* encoded)
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{
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rtc::Buffer temp_buffer;
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EncodedInfo info = EncodeImpl(rtp_timestamp, audio, &temp_buffer);
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RTC_DCHECK_LE(temp_buffer.size(), max_encoded_bytes);
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std::memcpy(encoded, temp_buffer.data(), 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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} // namespace webrtc
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