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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}
108 lines
3.4 KiB
C++
108 lines
3.4 KiB
C++
/*
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* Copyright (c) 2015 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 "modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
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#include <utility>
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#include "modules/audio_coding/codecs/ilbc/ilbc.h"
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#include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/logging.h"
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namespace webrtc {
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AudioDecoderIlbcImpl::AudioDecoderIlbcImpl() {
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WebRtcIlbcfix_DecoderCreate(&dec_state_);
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WebRtcIlbcfix_Decoderinit30Ms(dec_state_);
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}
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AudioDecoderIlbcImpl::~AudioDecoderIlbcImpl() {
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WebRtcIlbcfix_DecoderFree(dec_state_);
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}
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bool AudioDecoderIlbcImpl::HasDecodePlc() const {
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return true;
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}
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int AudioDecoderIlbcImpl::DecodeInternal(const uint8_t* encoded,
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size_t encoded_len,
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int sample_rate_hz,
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int16_t* decoded,
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SpeechType* speech_type) {
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RTC_DCHECK_EQ(sample_rate_hz, 8000);
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int16_t temp_type = 1; // Default is speech.
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int ret = WebRtcIlbcfix_Decode(dec_state_, encoded, encoded_len, decoded,
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&temp_type);
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*speech_type = ConvertSpeechType(temp_type);
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return ret;
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}
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size_t AudioDecoderIlbcImpl::DecodePlc(size_t num_frames, int16_t* decoded) {
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return WebRtcIlbcfix_NetEqPlc(dec_state_, decoded, num_frames);
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}
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void AudioDecoderIlbcImpl::Reset() {
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WebRtcIlbcfix_Decoderinit30Ms(dec_state_);
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}
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std::vector<AudioDecoder::ParseResult> AudioDecoderIlbcImpl::ParsePayload(
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rtc::Buffer&& payload,
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uint32_t timestamp) {
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std::vector<ParseResult> results;
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size_t bytes_per_frame;
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int timestamps_per_frame;
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if (payload.size() >= 950) {
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LOG(LS_WARNING) << "AudioDecoderIlbcImpl::ParsePayload: Payload too large";
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return results;
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}
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if (payload.size() % 38 == 0) {
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// 20 ms frames.
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bytes_per_frame = 38;
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timestamps_per_frame = 160;
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} else if (payload.size() % 50 == 0) {
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// 30 ms frames.
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bytes_per_frame = 50;
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timestamps_per_frame = 240;
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} else {
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LOG(LS_WARNING) << "AudioDecoderIlbcImpl::ParsePayload: Invalid payload";
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return results;
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}
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RTC_DCHECK_EQ(0, payload.size() % bytes_per_frame);
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if (payload.size() == bytes_per_frame) {
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std::unique_ptr<EncodedAudioFrame> frame(
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new LegacyEncodedAudioFrame(this, std::move(payload)));
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results.emplace_back(timestamp, 0, std::move(frame));
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} else {
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size_t byte_offset;
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uint32_t timestamp_offset;
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for (byte_offset = 0, timestamp_offset = 0;
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byte_offset < payload.size();
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byte_offset += bytes_per_frame,
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timestamp_offset += timestamps_per_frame) {
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std::unique_ptr<EncodedAudioFrame> frame(new LegacyEncodedAudioFrame(
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this, rtc::Buffer(payload.data() + byte_offset, bytes_per_frame)));
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results.emplace_back(timestamp + timestamp_offset, 0, std::move(frame));
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}
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}
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return results;
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}
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int AudioDecoderIlbcImpl::SampleRateHz() const {
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return 8000;
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}
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size_t AudioDecoderIlbcImpl::Channels() const {
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return 1;
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}
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} // namespace webrtc
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