10a029e952
For backwards compatibility, I've added kept the old interface to Encode() and EncodeInternal and created default implementations of both variants of EncodeInternal(), each calling the other. At least one of the variants must be implemented in a subclass or we'll run out of stack and explode. Would be nice if we could catch that before runtime. :/ The new interface to EncodeInternal() is protected, since it should never be called from the outside. Was unable to mark the old EncodeInternal() as RTC_DEPRECATED, since the default implementaion of the new variant needs to call it to work around old implementations. The old Encode() variant is deprecated, at least. Added a test for backwards compatibility in audio_encoder_unittest.cc. For the added test I broke out MockEncodeHelper from audio_encoder_copy_red_unittest.cc and renamed it MockAudioEncoderHelper. Review URL: https://codereview.webrtc.org/1725143003 Cr-Commit-Position: refs/heads/master@{#11823}
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 = EncodeInternal(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::EncodeInternal(
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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 = EncodeInternal(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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