decd9306ae
Rename this accessor function to reflect its new, slightly changed meaning. The reason for the change is that some codecs (iSAC) vary the number of 10 ms frames from packet to packet, and so can't return a truly constant value. BUG=3926 R=henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31849004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7556 4adac7df-926f-26a2-2b94-8c16560cd09d
105 lines
3.1 KiB
C++
105 lines
3.1 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/opus/interface/audio_encoder_opus.h"
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#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
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namespace webrtc {
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namespace {
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// We always encode at 48 kHz.
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const int kSampleRateHz = 48000;
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int DivExact(int a, int b) {
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CHECK_EQ(a % b, 0);
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return a / b;
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}
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int16_t ClampInt16(size_t x) {
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return static_cast<int16_t>(
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std::min(x, static_cast<size_t>(std::numeric_limits<int16_t>::max())));
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}
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int16_t CastInt16(size_t x) {
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DCHECK_LE(x, static_cast<size_t>(std::numeric_limits<int16_t>::max()));
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return static_cast<int16_t>(x);
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}
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} // namespace
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AudioEncoderOpus::Config::Config() : frame_size_ms(20), num_channels(1) {}
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bool AudioEncoderOpus::Config::IsOk() const {
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if (frame_size_ms <= 0 || frame_size_ms % 10 != 0)
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return false;
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if (num_channels <= 0)
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return false;
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return true;
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}
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AudioEncoderOpus::AudioEncoderOpus(const Config& config)
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: num_10ms_frames_per_packet_(DivExact(config.frame_size_ms, 10)),
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num_channels_(config.num_channels),
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samples_per_10ms_frame_(DivExact(kSampleRateHz, 100) * num_channels_) {
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CHECK(config.IsOk());
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input_buffer_.reserve(num_10ms_frames_per_packet_ * samples_per_10ms_frame_);
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CHECK_EQ(0, WebRtcOpus_EncoderCreate(&inst_, num_channels_));
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}
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AudioEncoderOpus::~AudioEncoderOpus() {
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CHECK_EQ(0, WebRtcOpus_EncoderFree(inst_));
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}
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int AudioEncoderOpus::sample_rate_hz() const {
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return kSampleRateHz;
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}
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int AudioEncoderOpus::num_channels() const {
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return num_channels_;
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}
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int AudioEncoderOpus::Num10MsFramesInNextPacket() const {
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return num_10ms_frames_per_packet_;
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}
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bool AudioEncoderOpus::Encode(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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uint32_t* encoded_timestamp) {
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if (input_buffer_.empty())
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first_timestamp_in_buffer_ = timestamp;
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input_buffer_.insert(input_buffer_.end(), audio,
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audio + samples_per_10ms_frame_);
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if (input_buffer_.size() < (static_cast<size_t>(num_10ms_frames_per_packet_) *
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samples_per_10ms_frame_)) {
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*encoded_bytes = 0;
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return true;
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}
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CHECK_EQ(input_buffer_.size(),
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static_cast<size_t>(num_10ms_frames_per_packet_) *
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samples_per_10ms_frame_);
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int16_t r = WebRtcOpus_Encode(
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inst_, &input_buffer_[0],
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DivExact(CastInt16(input_buffer_.size()), num_channels_),
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ClampInt16(max_encoded_bytes), encoded);
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input_buffer_.clear();
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if (r < 0)
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return false;
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*encoded_bytes = r;
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*encoded_timestamp = first_timestamp_in_buffer_;
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return true;
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}
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} // namespace webrtc
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