Starting to implement the new ACM API
The new implementation class is called AudioCodingImpl, and will in the end replace AudioCodingModuleImpl. This is work in progress. BUG=3520 R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/17359004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7103 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrik.lundin@webrtc.org
@ -94,4 +94,8 @@ bool AudioCodingModule::IsCodecValid(const CodecInst& codec) {
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}
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}
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}
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}
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AudioCoding* AudioCoding::Create(const Config& config) {
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return new AudioCodingImpl(config);
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}
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} // namespace webrtc
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} // namespace webrtc
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@ -14,6 +14,7 @@
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#include <stdlib.h>
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#include <stdlib.h>
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#include <vector>
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#include <vector>
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#include "webrtc/base/checks.h"
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#include "webrtc/engine_configurations.h"
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#include "webrtc/engine_configurations.h"
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#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
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#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
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#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
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#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
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@ -2053,4 +2054,255 @@ void AudioCodingModuleImpl::GetDecodingCallStatistics(
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} // namespace acm2
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} // namespace acm2
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bool AudioCodingImpl::RegisterSendCodec(AudioEncoder* send_codec) {
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FATAL() << "Not implemented yet.";
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}
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bool AudioCodingImpl::RegisterSendCodec(int encoder_type,
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uint8_t payload_type,
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int frame_size_samples) {
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std::string codec_name;
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int sample_rate_hz;
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int channels;
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if (!MapCodecTypeToParameters(
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encoder_type, &codec_name, &sample_rate_hz, &channels)) {
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return false;
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}
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webrtc::CodecInst codec;
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AudioCodingModule::Codec(
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codec_name.c_str(), &codec, sample_rate_hz, channels);
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codec.pltype = payload_type;
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if (frame_size_samples > 0) {
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codec.pacsize = frame_size_samples;
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}
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return acm_old_->RegisterSendCodec(codec) == 0;
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}
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const AudioEncoder* AudioCodingImpl::GetSenderInfo() const {
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FATAL() << "Not implemented yet.";
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}
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int AudioCodingImpl::Add10MsAudio(const AudioFrame& audio_frame) {
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if (acm_old_->Add10MsData(audio_frame) != 0) {
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return -1;
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}
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return acm_old_->Process();
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}
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const ReceiverInfo* AudioCodingImpl::GetReceiverInfo() const {
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FATAL() << "Not implemented yet.";
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}
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bool AudioCodingImpl::RegisterReceiveCodec(AudioDecoder* receive_codec) {
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FATAL() << "Not implemented yet.";
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}
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bool AudioCodingImpl::RegisterReceiveCodec(int decoder_type,
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uint8_t payload_type) {
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std::string codec_name;
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int sample_rate_hz;
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int channels;
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if (!MapCodecTypeToParameters(
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decoder_type, &codec_name, &sample_rate_hz, &channels)) {
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return false;
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}
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webrtc::CodecInst codec;
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AudioCodingModule::Codec(
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codec_name.c_str(), &codec, sample_rate_hz, channels);
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codec.pltype = payload_type;
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return acm_old_->RegisterReceiveCodec(codec) == 0;
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}
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bool AudioCodingImpl::InsertPacket(const uint8_t* incoming_payload,
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int32_t payload_len_bytes,
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const WebRtcRTPHeader& rtp_info) {
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return acm_old_->IncomingPacket(
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incoming_payload, payload_len_bytes, rtp_info) == 0;
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}
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bool AudioCodingImpl::InsertPayload(const uint8_t* incoming_payload,
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int32_t payload_len_byte,
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uint8_t payload_type,
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uint32_t timestamp) {
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FATAL() << "Not implemented yet.";
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}
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bool AudioCodingImpl::SetMinimumPlayoutDelay(int time_ms) {
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FATAL() << "Not implemented yet.";
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}
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bool AudioCodingImpl::SetMaximumPlayoutDelay(int time_ms) {
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FATAL() << "Not implemented yet.";
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}
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int AudioCodingImpl::LeastRequiredDelayMs() const {
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FATAL() << "Not implemented yet.";
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}
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bool AudioCodingImpl::PlayoutTimestamp(uint32_t* timestamp) {
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FATAL() << "Not implemented yet.";
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}
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bool AudioCodingImpl::Get10MsAudio(AudioFrame* audio_frame) {
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return acm_old_->PlayoutData10Ms(playout_frequency_hz_, audio_frame) == 0;
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}
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bool AudioCodingImpl::NetworkStatistics(
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ACMNetworkStatistics* network_statistics) {
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FATAL() << "Not implemented yet.";
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}
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bool AudioCodingImpl::EnableNack(size_t max_nack_list_size) {
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FATAL() << "Not implemented yet.";
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}
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void AudioCodingImpl::DisableNack() {
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FATAL() << "Not implemented yet.";
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}
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std::vector<uint16_t> AudioCodingImpl::GetNackList(
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int round_trip_time_ms) const {
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return acm_old_->GetNackList(round_trip_time_ms);
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}
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void AudioCodingImpl::GetDecodingCallStatistics(
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AudioDecodingCallStats* call_stats) const {
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acm_old_->GetDecodingCallStatistics(call_stats);
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}
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bool AudioCodingImpl::MapCodecTypeToParameters(int codec_type,
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std::string* codec_name,
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int* sample_rate_hz,
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int* channels) {
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switch (codec_type) {
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#ifdef WEBRTC_CODEC_PCM16
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case acm2::ACMCodecDB::kPCM16B:
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*codec_name = "L16";
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*sample_rate_hz = 8000;
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*channels = 1;
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break;
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case acm2::ACMCodecDB::kPCM16Bwb:
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*codec_name = "L16";
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*sample_rate_hz = 16000;
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*channels = 1;
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break;
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case acm2::ACMCodecDB::kPCM16Bswb32kHz:
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*codec_name = "L16";
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*sample_rate_hz = 32000;
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*channels = 1;
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break;
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case acm2::ACMCodecDB::kPCM16B_2ch:
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*codec_name = "L16";
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*sample_rate_hz = 8000;
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*channels = 2;
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break;
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case acm2::ACMCodecDB::kPCM16Bwb_2ch:
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*codec_name = "L16";
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*sample_rate_hz = 16000;
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*channels = 2;
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break;
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case acm2::ACMCodecDB::kPCM16Bswb32kHz_2ch:
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*codec_name = "L16";
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*sample_rate_hz = 32000;
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*channels = 2;
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break;
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#endif
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#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
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case acm2::ACMCodecDB::kISAC:
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*codec_name = "ISAC";
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*sample_rate_hz = 16000;
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*channels = 1;
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break;
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#endif
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#ifdef WEBRTC_CODEC_ISAC
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case acm2::ACMCodecDB::kISACSWB:
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*codec_name = "ISAC";
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*sample_rate_hz = 32000;
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*channels = 1;
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break;
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case acm2::ACMCodecDB::kISACFB:
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*codec_name = "ISAC";
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*sample_rate_hz = 48000;
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*channels = 1;
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break;
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#endif
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#ifdef WEBRTC_CODEC_ILBC
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case acm2::ACMCodecDB::kILBC:
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*codec_name = "ILBC";
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*sample_rate_hz = 8000;
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*channels = 1;
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break;
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#endif
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case acm2::ACMCodecDB::kPCMA:
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*codec_name = "PCMA";
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*sample_rate_hz = 8000;
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*channels = 1;
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break;
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case acm2::ACMCodecDB::kPCMA_2ch:
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*codec_name = "PCMA";
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*sample_rate_hz = 8000;
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*channels = 2;
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break;
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case acm2::ACMCodecDB::kPCMU:
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*codec_name = "PCMU";
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*sample_rate_hz = 8000;
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*channels = 1;
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break;
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case acm2::ACMCodecDB::kPCMU_2ch:
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*codec_name = "PCMU";
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*sample_rate_hz = 8000;
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*channels = 2;
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break;
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#ifdef WEBRTC_CODEC_G722
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case acm2::ACMCodecDB::kG722:
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*codec_name = "G722";
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*sample_rate_hz = 16000;
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*channels = 1;
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break;
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case acm2::ACMCodecDB::kG722_2ch:
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*codec_name = "G722";
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*sample_rate_hz = 16000;
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*channels = 2;
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break;
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#endif
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#ifdef WEBRTC_CODEC_OPUS
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case acm2::ACMCodecDB::kOpus:
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*codec_name = "opus";
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*sample_rate_hz = 48000;
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*channels = 2;
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break;
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#endif
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case acm2::ACMCodecDB::kCNNB:
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*codec_name = "CN";
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*sample_rate_hz = 8000;
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*channels = 1;
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break;
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case acm2::ACMCodecDB::kCNWB:
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*codec_name = "CN";
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*sample_rate_hz = 16000;
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*channels = 1;
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break;
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case acm2::ACMCodecDB::kCNSWB:
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*codec_name = "CN";
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*sample_rate_hz = 32000;
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*channels = 1;
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break;
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case acm2::ACMCodecDB::kRED:
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*codec_name = "red";
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*sample_rate_hz = 8000;
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*channels = 1;
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break;
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#ifdef WEBRTC_CODEC_AVT
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case acm2::ACMCodecDB::kAVT:
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*codec_name = "telephone-event";
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*sample_rate_hz = 8000;
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*channels = 1;
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break;
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#endif
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default:
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FATAL() << "Codec type " << codec_type << " not supported.";
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}
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return true;
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}
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} // namespace webrtc
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} // namespace webrtc
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@ -382,6 +382,88 @@ class AudioCodingModuleImpl : public AudioCodingModule {
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} // namespace acm2
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} // namespace acm2
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class AudioCodingImpl : public AudioCoding {
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public:
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AudioCodingImpl(const Config& config) {
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AudioCodingModule::Config config_old;
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config_old.id = 0;
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config_old.neteq_config = config.neteq_config;
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config_old.clock = config.clock;
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acm_old_.reset(new acm2::AudioCodingModuleImpl(config_old));
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acm_old_->RegisterTransportCallback(config.transport);
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acm_old_->RegisterVADCallback(config.vad_callback);
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acm_old_->SetDtmfPlayoutStatus(config.play_dtmf);
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if (config.initial_playout_delay_ms > 0) {
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acm_old_->SetInitialPlayoutDelay(config.initial_playout_delay_ms);
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}
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playout_frequency_hz_ = config.playout_frequency_hz;
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}
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virtual ~AudioCodingImpl() OVERRIDE {};
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virtual bool RegisterSendCodec(AudioEncoder* send_codec) OVERRIDE;
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virtual bool RegisterSendCodec(int encoder_type,
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uint8_t payload_type,
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int frame_size_samples = 0) OVERRIDE;
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virtual const AudioEncoder* GetSenderInfo() const OVERRIDE;
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virtual int Add10MsAudio(const AudioFrame& audio_frame) OVERRIDE;
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virtual const ReceiverInfo* GetReceiverInfo() const OVERRIDE;
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virtual bool RegisterReceiveCodec(AudioDecoder* receive_codec) OVERRIDE;
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virtual bool RegisterReceiveCodec(int decoder_type,
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uint8_t payload_type) OVERRIDE;
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virtual bool InsertPacket(const uint8_t* incoming_payload,
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int32_t payload_len_bytes,
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const WebRtcRTPHeader& rtp_info) OVERRIDE;
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virtual bool InsertPayload(const uint8_t* incoming_payload,
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int32_t payload_len_byte,
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uint8_t payload_type,
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uint32_t timestamp) OVERRIDE;
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virtual bool SetMinimumPlayoutDelay(int time_ms) OVERRIDE;
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virtual bool SetMaximumPlayoutDelay(int time_ms) OVERRIDE;
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virtual int LeastRequiredDelayMs() const OVERRIDE;
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virtual bool PlayoutTimestamp(uint32_t* timestamp) OVERRIDE;
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virtual bool Get10MsAudio(AudioFrame* audio_frame) OVERRIDE;
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virtual bool NetworkStatistics(
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ACMNetworkStatistics* network_statistics) OVERRIDE;
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virtual bool EnableNack(size_t max_nack_list_size) OVERRIDE;
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virtual void DisableNack() OVERRIDE;
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virtual std::vector<uint16_t> GetNackList(
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int round_trip_time_ms) const OVERRIDE;
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virtual void GetDecodingCallStatistics(
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AudioDecodingCallStats* call_stats) const OVERRIDE;
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private:
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// Temporary method to be used during redesign phase.
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// Maps |codec_type| (a value from the anonymous enum in acm2::ACMCodecDB) to
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// |codec_name|, |sample_rate_hz|, and |channels|.
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// TODO(henrik.lundin) Remove this when no longer needed.
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static bool MapCodecTypeToParameters(int codec_type,
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std::string* codec_name,
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int* sample_rate_hz,
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int* channels);
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scoped_ptr<acm2::AudioCodingModuleImpl> acm_old_;
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int playout_frequency_hz_;
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};
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} // namespace webrtc
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_AUDIO_CODING_MODULE_IMPL_H_
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#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_AUDIO_CODING_MODULE_IMPL_H_
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Block a user