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Diffstat (limited to '')
-rw-r--r-- | third_party/libwebrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc | 110 |
1 files changed, 110 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc new file mode 100644 index 0000000000..57b5abbe23 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc @@ -0,0 +1,110 @@ +/* + * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h" + +#include <memory> +#include <utility> + +#include "modules/audio_coding/codecs/ilbc/ilbc.h" +#include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h" +#include "rtc_base/checks.h" +#include "rtc_base/logging.h" + +namespace webrtc { + +AudioDecoderIlbcImpl::AudioDecoderIlbcImpl() { + WebRtcIlbcfix_DecoderCreate(&dec_state_); + WebRtcIlbcfix_Decoderinit30Ms(dec_state_); +} + +AudioDecoderIlbcImpl::~AudioDecoderIlbcImpl() { + WebRtcIlbcfix_DecoderFree(dec_state_); +} + +bool AudioDecoderIlbcImpl::HasDecodePlc() const { + return true; +} + +int AudioDecoderIlbcImpl::DecodeInternal(const uint8_t* encoded, + size_t encoded_len, + int sample_rate_hz, + int16_t* decoded, + SpeechType* speech_type) { + RTC_DCHECK_EQ(sample_rate_hz, 8000); + int16_t temp_type = 1; // Default is speech. + int ret = WebRtcIlbcfix_Decode(dec_state_, encoded, encoded_len, decoded, + &temp_type); + *speech_type = ConvertSpeechType(temp_type); + return ret; +} + +size_t AudioDecoderIlbcImpl::DecodePlc(size_t num_frames, int16_t* decoded) { + return WebRtcIlbcfix_NetEqPlc(dec_state_, decoded, num_frames); +} + +void AudioDecoderIlbcImpl::Reset() { + WebRtcIlbcfix_Decoderinit30Ms(dec_state_); +} + +std::vector<AudioDecoder::ParseResult> AudioDecoderIlbcImpl::ParsePayload( + rtc::Buffer&& payload, + uint32_t timestamp) { + std::vector<ParseResult> results; + size_t bytes_per_frame; + int timestamps_per_frame; + if (payload.size() >= 950) { + RTC_LOG(LS_WARNING) + << "AudioDecoderIlbcImpl::ParsePayload: Payload too large"; + return results; + } + if (payload.size() % 38 == 0) { + // 20 ms frames. + bytes_per_frame = 38; + timestamps_per_frame = 160; + } else if (payload.size() % 50 == 0) { + // 30 ms frames. + bytes_per_frame = 50; + timestamps_per_frame = 240; + } else { + RTC_LOG(LS_WARNING) + << "AudioDecoderIlbcImpl::ParsePayload: Invalid payload"; + return results; + } + + RTC_DCHECK_EQ(0, payload.size() % bytes_per_frame); + if (payload.size() == bytes_per_frame) { + std::unique_ptr<EncodedAudioFrame> frame( + new LegacyEncodedAudioFrame(this, std::move(payload))); + results.emplace_back(timestamp, 0, std::move(frame)); + } else { + size_t byte_offset; + uint32_t timestamp_offset; + for (byte_offset = 0, timestamp_offset = 0; byte_offset < payload.size(); + byte_offset += bytes_per_frame, + timestamp_offset += timestamps_per_frame) { + std::unique_ptr<EncodedAudioFrame> frame(new LegacyEncodedAudioFrame( + this, rtc::Buffer(payload.data() + byte_offset, bytes_per_frame))); + results.emplace_back(timestamp + timestamp_offset, 0, std::move(frame)); + } + } + + return results; +} + +int AudioDecoderIlbcImpl::SampleRateHz() const { + return 8000; +} + +size_t AudioDecoderIlbcImpl::Channels() const { + return 1; +} + +} // namespace webrtc |