blob: e600d2d56c1ab5947e8c5ce02cb782086ab83c76 [file] [log] [blame]
/*
* 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 "webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h"
#include "webrtc/base/checks.h"
#include "webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
namespace webrtc {
AudioDecoderPcm16B::AudioDecoderPcm16B(int sample_rate_hz, size_t num_channels)
: sample_rate_hz_(sample_rate_hz), num_channels_(num_channels) {
RTC_DCHECK(sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
sample_rate_hz == 32000 || sample_rate_hz == 48000)
<< "Unsupported sample rate " << sample_rate_hz;
RTC_DCHECK_GE(num_channels, 1u);
}
void AudioDecoderPcm16B::Reset() {}
int AudioDecoderPcm16B::SampleRateHz() const {
return sample_rate_hz_;
}
size_t AudioDecoderPcm16B::Channels() const {
return num_channels_;
}
int AudioDecoderPcm16B::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_, sample_rate_hz);
size_t ret = WebRtcPcm16b_Decode(encoded, encoded_len, decoded);
*speech_type = ConvertSpeechType(1);
return static_cast<int>(ret);
}
std::vector<AudioDecoder::ParseResult> AudioDecoderPcm16B::ParsePayload(
rtc::Buffer&& payload,
uint32_t timestamp,
bool is_primary) {
const int samples_per_ms = rtc::CheckedDivExact(sample_rate_hz_, 1000);
return LegacyEncodedAudioFrame::SplitBySamples(
this, std::move(payload), timestamp, is_primary,
samples_per_ms * 2 * num_channels_, samples_per_ms);
}
int AudioDecoderPcm16B::PacketDuration(const uint8_t* encoded,
size_t encoded_len) const {
// Two encoded byte per sample per channel.
return static_cast<int>(encoded_len / (2 * Channels()));
}
} // namespace webrtc