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Diffstat (limited to 'third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc')
-rw-r--r-- | third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc | 102 |
1 files changed, 102 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc b/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc new file mode 100644 index 0000000000..46ac671b30 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc @@ -0,0 +1,102 @@ +/* + * 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/g711/audio_decoder_pcm.h" + +#include <utility> + +#include "modules/audio_coding/codecs/g711/g711_interface.h" +#include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h" + +namespace webrtc { + +void AudioDecoderPcmU::Reset() {} + +std::vector<AudioDecoder::ParseResult> AudioDecoderPcmU::ParsePayload( + rtc::Buffer&& payload, + uint32_t timestamp) { + return LegacyEncodedAudioFrame::SplitBySamples( + this, std::move(payload), timestamp, 8 * num_channels_, 8); +} + +int AudioDecoderPcmU::SampleRateHz() const { + return 8000; +} + +size_t AudioDecoderPcmU::Channels() const { + return num_channels_; +} + +int AudioDecoderPcmU::DecodeInternal(const uint8_t* encoded, + size_t encoded_len, + int sample_rate_hz, + int16_t* decoded, + SpeechType* speech_type) { + RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz); + // Adjust the encoded length down to ensure the same number of samples in each + // channel. + const size_t encoded_len_adjusted = + PacketDuration(encoded, encoded_len) * + Channels(); // 1 byte per sample per channel + int16_t temp_type = 1; // Default is speech. + size_t ret = + WebRtcG711_DecodeU(encoded, encoded_len_adjusted, decoded, &temp_type); + *speech_type = ConvertSpeechType(temp_type); + return static_cast<int>(ret); +} + +int AudioDecoderPcmU::PacketDuration(const uint8_t* encoded, + size_t encoded_len) const { + // One encoded byte per sample per channel. + return static_cast<int>(encoded_len / Channels()); +} + +void AudioDecoderPcmA::Reset() {} + +std::vector<AudioDecoder::ParseResult> AudioDecoderPcmA::ParsePayload( + rtc::Buffer&& payload, + uint32_t timestamp) { + return LegacyEncodedAudioFrame::SplitBySamples( + this, std::move(payload), timestamp, 8 * num_channels_, 8); +} + +int AudioDecoderPcmA::SampleRateHz() const { + return 8000; +} + +size_t AudioDecoderPcmA::Channels() const { + return num_channels_; +} + +int AudioDecoderPcmA::DecodeInternal(const uint8_t* encoded, + size_t encoded_len, + int sample_rate_hz, + int16_t* decoded, + SpeechType* speech_type) { + RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz); + // Adjust the encoded length down to ensure the same number of samples in each + // channel. + const size_t encoded_len_adjusted = + PacketDuration(encoded, encoded_len) * + Channels(); // 1 byte per sample per channel + int16_t temp_type = 1; // Default is speech. + size_t ret = + WebRtcG711_DecodeA(encoded, encoded_len_adjusted, decoded, &temp_type); + *speech_type = ConvertSpeechType(temp_type); + return static_cast<int>(ret); +} + +int AudioDecoderPcmA::PacketDuration(const uint8_t* encoded, + size_t encoded_len) const { + // One encoded byte per sample per channel. + return static_cast<int>(encoded_len / Channels()); +} + +} // namespace webrtc |