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/*
* Copyright (c) 2014 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/g711/include/audio_encoder_pcm.h"
#include <limits>
#include "webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h"
namespace webrtc {
namespace {
int16_t NumSamplesPerFrame(int num_channels,
int frame_size_ms,
int sample_rate_hz) {
int samples_per_frame = num_channels * frame_size_ms * sample_rate_hz / 1000;
CHECK_LE(samples_per_frame, std::numeric_limits<int16_t>::max())
<< "Frame size too large.";
return static_cast<int16_t>(samples_per_frame);
}
} // namespace
AudioEncoderPcm::AudioEncoderPcm(const Config& config, int sample_rate_hz)
: sample_rate_hz_(sample_rate_hz),
num_channels_(config.num_channels),
payload_type_(config.payload_type),
num_10ms_frames_per_packet_(config.frame_size_ms / 10),
full_frame_samples_(NumSamplesPerFrame(config.num_channels,
config.frame_size_ms,
sample_rate_hz_)),
first_timestamp_in_buffer_(0) {
CHECK_GT(sample_rate_hz, 0) << "Sample rate must be larger than 0 Hz";
CHECK_EQ(config.frame_size_ms % 10, 0)
<< "Frame size must be an integer multiple of 10 ms.";
speech_buffer_.reserve(full_frame_samples_);
}
AudioEncoderPcm::~AudioEncoderPcm() {
}
int AudioEncoderPcm::sample_rate_hz() const {
return sample_rate_hz_;
}
int AudioEncoderPcm::num_channels() const {
return num_channels_;
}
int AudioEncoderPcm::Num10MsFramesInNextPacket() const {
return num_10ms_frames_per_packet_;
}
int AudioEncoderPcm::Max10MsFramesInAPacket() const {
return num_10ms_frames_per_packet_;
}
bool AudioEncoderPcm::EncodeInternal(uint32_t timestamp,
const int16_t* audio,
size_t max_encoded_bytes,
uint8_t* encoded,
EncodedInfo* info) {
const int num_samples = sample_rate_hz() / 100 * num_channels();
if (speech_buffer_.empty()) {
first_timestamp_in_buffer_ = timestamp;
}
for (int i = 0; i < num_samples; ++i) {
speech_buffer_.push_back(audio[i]);
}
if (speech_buffer_.size() < static_cast<size_t>(full_frame_samples_)) {
info->encoded_bytes = 0;
return true;
}
CHECK_EQ(speech_buffer_.size(), static_cast<size_t>(full_frame_samples_));
int16_t ret = EncodeCall(&speech_buffer_[0], full_frame_samples_, encoded);
speech_buffer_.clear();
info->encoded_timestamp = first_timestamp_in_buffer_;
info->payload_type = payload_type_;
if (ret < 0)
return false;
info->encoded_bytes = static_cast<size_t>(ret);
return true;
}
int16_t AudioEncoderPcmA::EncodeCall(const int16_t* audio,
size_t input_len,
uint8_t* encoded) {
return WebRtcG711_EncodeA(const_cast<int16_t*>(audio),
static_cast<int16_t>(input_len),
reinterpret_cast<int16_t*>(encoded));
}
int16_t AudioEncoderPcmU::EncodeCall(const int16_t* audio,
size_t input_len,
uint8_t* encoded) {
return WebRtcG711_EncodeU(const_cast<int16_t*>(audio),
static_cast<int16_t>(input_len),
reinterpret_cast<int16_t*>(encoded));
}
} // namespace webrtc
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