9afaee74ab
Old review at: https://webrtc-codereview.appspot.com/43839004/ R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/45769004 Cr-Commit-Position: refs/heads/master@{#8788} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8788 4adac7df-926f-26a2-2b94-8c16560cd09d
116 lines
3.5 KiB
C++
116 lines
3.5 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/ilbc/interface/audio_encoder_ilbc.h"
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#include <cstring>
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#include <limits>
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#include "webrtc/base/checks.h"
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#include "webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h"
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namespace webrtc {
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namespace {
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const int kSampleRateHz = 8000;
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} // namespace
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AudioEncoderIlbc::AudioEncoderIlbc(const Config& config)
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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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num_10ms_frames_buffered_(0) {
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CHECK(config.frame_size_ms == 20 || config.frame_size_ms == 30 ||
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config.frame_size_ms == 40 || config.frame_size_ms == 60)
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<< "Frame size must be 20, 30, 40, or 60 ms.";
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DCHECK_LE(kSampleRateHz / 100 * num_10ms_frames_per_packet_,
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kMaxSamplesPerPacket);
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CHECK_EQ(0, WebRtcIlbcfix_EncoderCreate(&encoder_));
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const int encoder_frame_size_ms = config.frame_size_ms > 30
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? config.frame_size_ms / 2
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: config.frame_size_ms;
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CHECK_EQ(0, WebRtcIlbcfix_EncoderInit(encoder_, encoder_frame_size_ms));
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}
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AudioEncoderIlbc::~AudioEncoderIlbc() {
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CHECK_EQ(0, WebRtcIlbcfix_EncoderFree(encoder_));
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}
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int AudioEncoderIlbc::SampleRateHz() const {
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return kSampleRateHz;
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}
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int AudioEncoderIlbc::NumChannels() const {
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return 1;
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}
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size_t AudioEncoderIlbc::MaxEncodedBytes() const {
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return RequiredOutputSizeBytes();
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}
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int AudioEncoderIlbc::Num10MsFramesInNextPacket() const {
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return num_10ms_frames_per_packet_;
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}
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int AudioEncoderIlbc::Max10MsFramesInAPacket() const {
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return num_10ms_frames_per_packet_;
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}
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AudioEncoder::EncodedInfo AudioEncoderIlbc::EncodeInternal(
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uint32_t rtp_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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DCHECK_GE(max_encoded_bytes, RequiredOutputSizeBytes());
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// Save timestamp if starting a new packet.
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if (num_10ms_frames_buffered_ == 0)
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first_timestamp_in_buffer_ = rtp_timestamp;
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// Buffer input.
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std::memcpy(input_buffer_ + kSampleRateHz / 100 * num_10ms_frames_buffered_,
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audio,
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kSampleRateHz / 100 * sizeof(audio[0]));
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// If we don't yet have enough buffered input for a whole packet, we're done
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// for now.
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if (++num_10ms_frames_buffered_ < num_10ms_frames_per_packet_) {
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return EncodedInfo();
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}
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// Encode buffered input.
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DCHECK_EQ(num_10ms_frames_buffered_, num_10ms_frames_per_packet_);
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num_10ms_frames_buffered_ = 0;
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const int output_len = WebRtcIlbcfix_Encode(
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encoder_,
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input_buffer_,
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kSampleRateHz / 100 * num_10ms_frames_per_packet_,
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encoded);
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CHECK_GE(output_len, 0);
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EncodedInfo info;
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info.encoded_bytes = output_len;
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DCHECK_EQ(info.encoded_bytes, RequiredOutputSizeBytes());
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info.encoded_timestamp = first_timestamp_in_buffer_;
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info.payload_type = payload_type_;
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return info;
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}
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size_t AudioEncoderIlbc::RequiredOutputSizeBytes() const {
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switch (num_10ms_frames_per_packet_) {
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case 2: return 38;
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case 3: return 50;
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case 4: return 2 * 38;
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case 6: return 2 * 50;
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default: FATAL();
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}
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}
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} // namespace webrtc
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