8dc21dc238
BUG=3926 R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/28129004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7801 4adac7df-926f-26a2-2b94-8c16560cd09d
101 lines
3.6 KiB
C++
101 lines
3.6 KiB
C++
/*
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* Copyright (c) 2014 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 "webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h"
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#include <limits>
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#include "webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h"
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namespace webrtc {
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namespace {
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int16_t NumSamplesPerFrame(int num_channels,
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int frame_size_ms,
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int sample_rate_hz) {
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int samples_per_frame = num_channels * frame_size_ms * sample_rate_hz / 1000;
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CHECK_LE(samples_per_frame, std::numeric_limits<int16_t>::max())
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<< "Frame size too large.";
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return static_cast<int16_t>(samples_per_frame);
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}
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} // namespace
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AudioEncoderPcm::AudioEncoderPcm(const Config& config)
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: num_channels_(config.num_channels),
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payload_type_(config.payload_type),
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num_10ms_frames_per_packet_(config.frame_size_ms / 10),
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full_frame_samples_(NumSamplesPerFrame(num_channels_,
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config.frame_size_ms,
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kSampleRateHz)),
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first_timestamp_in_buffer_(0) {
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CHECK_EQ(config.frame_size_ms % 10, 0)
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<< "Frame size must be an integer multiple of 10 ms.";
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speech_buffer_.reserve(full_frame_samples_);
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}
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AudioEncoderPcm::~AudioEncoderPcm() {
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}
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int AudioEncoderPcm::sample_rate_hz() const {
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return kSampleRateHz;
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}
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int AudioEncoderPcm::num_channels() const {
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return num_channels_;
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}
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int AudioEncoderPcm::Num10MsFramesInNextPacket() const {
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return num_10ms_frames_per_packet_;
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}
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bool AudioEncoderPcm::EncodeInternal(uint32_t timestamp,
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const int16_t* audio,
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size_t max_encoded_bytes,
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uint8_t* encoded,
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size_t* encoded_bytes,
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EncodedInfo* info) {
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const int num_samples = sample_rate_hz() / 100 * num_channels();
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if (speech_buffer_.empty()) {
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first_timestamp_in_buffer_ = timestamp;
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}
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for (int i = 0; i < num_samples; ++i) {
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speech_buffer_.push_back(audio[i]);
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}
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if (speech_buffer_.size() < static_cast<size_t>(full_frame_samples_)) {
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*encoded_bytes = 0;
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return true;
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}
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CHECK_EQ(speech_buffer_.size(), static_cast<size_t>(full_frame_samples_));
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int16_t ret = EncodeCall(&speech_buffer_[0], full_frame_samples_, encoded);
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speech_buffer_.clear();
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info->encoded_timestamp = first_timestamp_in_buffer_;
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info->payload_type = payload_type_;
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if (ret < 0)
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return false;
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*encoded_bytes = static_cast<size_t>(ret);
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return true;
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}
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int16_t AudioEncoderPcmA::EncodeCall(const int16_t* audio,
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size_t input_len,
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uint8_t* encoded) {
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return WebRtcG711_EncodeA(const_cast<int16_t*>(audio),
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static_cast<int16_t>(input_len),
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reinterpret_cast<int16_t*>(encoded));
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}
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int16_t AudioEncoderPcmU::EncodeCall(const int16_t* audio,
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size_t input_len,
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uint8_t* encoded) {
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return WebRtcG711_EncodeU(const_cast<int16_t*>(audio),
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static_cast<int16_t>(input_len),
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reinterpret_cast<int16_t*>(encoded));
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}
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} // namespace webrtc
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