a1fa491334
This patch fixes the internal AudioCoder output buffer setting to be set prior it will be used within callback from ACM BUG=webrtc:7462 Review-Url: https://codereview.webrtc.org/2806933002 Cr-Commit-Position: refs/heads/master@{#17800}
119 lines
4.1 KiB
C++
119 lines
4.1 KiB
C++
/*
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* Copyright (c) 2012 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/voice_engine/coder.h"
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#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
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#include "webrtc/common_types.h"
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#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
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#include "webrtc/modules/include/module_common_types.h"
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namespace webrtc {
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namespace {
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AudioCodingModule::Config GetAcmConfig(uint32_t id) {
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AudioCodingModule::Config config;
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// This class does not handle muted output.
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config.neteq_config.enable_muted_state = false;
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config.id = id;
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config.decoder_factory = CreateBuiltinAudioDecoderFactory();
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return config;
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}
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} // namespace
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AudioCoder::AudioCoder(uint32_t instance_id)
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: acm_(AudioCodingModule::Create(GetAcmConfig(instance_id))),
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receive_codec_(),
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encode_timestamp_(0),
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encoded_data_(nullptr),
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encoded_length_in_bytes_(0),
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decode_timestamp_(0) {
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acm_->InitializeReceiver();
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acm_->RegisterTransportCallback(this);
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}
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AudioCoder::~AudioCoder() {}
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int32_t AudioCoder::SetEncodeCodec(const CodecInst& codec_inst) {
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const bool success = codec_manager_.RegisterEncoder(codec_inst) &&
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codec_manager_.MakeEncoder(&rent_a_codec_, acm_.get());
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return success ? 0 : -1;
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}
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int32_t AudioCoder::SetDecodeCodec(const CodecInst& codec_inst) {
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if (!acm_->RegisterReceiveCodec(codec_inst.pltype,
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CodecInstToSdp(codec_inst))) {
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return -1;
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}
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memcpy(&receive_codec_, &codec_inst, sizeof(CodecInst));
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return 0;
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}
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int32_t AudioCoder::Decode(AudioFrame* decoded_audio,
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uint32_t samp_freq_hz,
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const int8_t* incoming_payload,
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size_t payload_length) {
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if (payload_length > 0) {
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const uint8_t payload_type = receive_codec_.pltype;
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decode_timestamp_ += receive_codec_.pacsize;
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if (acm_->IncomingPayload((const uint8_t*)incoming_payload, payload_length,
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payload_type, decode_timestamp_) == -1) {
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return -1;
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}
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}
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bool muted;
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int32_t ret =
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acm_->PlayoutData10Ms((uint16_t)samp_freq_hz, decoded_audio, &muted);
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RTC_DCHECK(!muted);
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return ret;
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}
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int32_t AudioCoder::PlayoutData(AudioFrame* decoded_audio,
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uint16_t samp_freq_hz) {
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bool muted;
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int32_t ret = acm_->PlayoutData10Ms(samp_freq_hz, decoded_audio, &muted);
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RTC_DCHECK(!muted);
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return ret;
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}
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int32_t AudioCoder::Encode(const AudioFrame& audio,
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int8_t* encoded_data,
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size_t* encoded_length_in_bytes) {
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// Fake a timestamp in case audio doesn't contain a correct timestamp.
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// Make a local copy of the audio frame since audio is const
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AudioFrame audio_frame;
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audio_frame.CopyFrom(audio);
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audio_frame.timestamp_ = encode_timestamp_;
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encode_timestamp_ += static_cast<uint32_t>(audio_frame.samples_per_channel_);
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// For any codec with a frame size that is longer than 10 ms the encoded
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// length in bytes should be zero until a a full frame has been encoded.
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encoded_length_in_bytes_ = 0;
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encoded_data_ = encoded_data;
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if (acm_->Add10MsData((AudioFrame&)audio_frame) == -1) {
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return -1;
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}
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*encoded_length_in_bytes = encoded_length_in_bytes_;
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return 0;
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}
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int32_t AudioCoder::SendData(FrameType /* frame_type */,
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uint8_t /* payload_type */,
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uint32_t /* time_stamp */,
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const uint8_t* payload_data,
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size_t payload_size,
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const RTPFragmentationHeader* /* fragmentation*/) {
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memcpy(encoded_data_, payload_data, sizeof(uint8_t) * payload_size);
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encoded_length_in_bytes_ = payload_size;
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return 0;
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}
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} // namespace webrtc
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