diff options
Diffstat (limited to 'chromium/media/cast/audio_receiver/audio_decoder.cc')
| -rw-r--r-- | chromium/media/cast/audio_receiver/audio_decoder.cc | 161 |
1 files changed, 0 insertions, 161 deletions
diff --git a/chromium/media/cast/audio_receiver/audio_decoder.cc b/chromium/media/cast/audio_receiver/audio_decoder.cc deleted file mode 100644 index a761a5a84de..00000000000 --- a/chromium/media/cast/audio_receiver/audio_decoder.cc +++ /dev/null @@ -1,161 +0,0 @@ -// Copyright 2013 The Chromium Authors. All rights reserved. -// Use of this source code is governed by a BSD-style license that can be -// found in the LICENSE file. - -#include "base/logging.h" -#include "media/cast/audio_receiver/audio_decoder.h" - -#include "third_party/webrtc/modules/audio_coding/main/interface/audio_coding_module.h" -#include "third_party/webrtc/modules/interface/module_common_types.h" - -namespace media { -namespace cast { - -AudioDecoder::AudioDecoder(scoped_refptr<CastEnvironment> cast_environment, - const AudioReceiverConfig& audio_config, - RtpPayloadFeedback* incoming_payload_feedback) - : cast_environment_(cast_environment), - audio_decoder_(webrtc::AudioCodingModule::Create(0)), - cast_message_builder_(cast_environment->Clock(), - incoming_payload_feedback, &frame_id_map_, audio_config.incoming_ssrc, - true, 0), - have_received_packets_(false), - last_played_out_timestamp_(0) { - audio_decoder_->InitializeReceiver(); - - webrtc::CodecInst receive_codec; - switch (audio_config.codec) { - case kPcm16: - receive_codec.pltype = audio_config.rtp_payload_type; - strncpy(receive_codec.plname, "L16", 4); - receive_codec.plfreq = audio_config.frequency; - receive_codec.pacsize = -1; - receive_codec.channels = audio_config.channels; - receive_codec.rate = -1; - break; - case kOpus: - receive_codec.pltype = audio_config.rtp_payload_type; - strncpy(receive_codec.plname, "opus", 5); - receive_codec.plfreq = audio_config.frequency; - receive_codec.pacsize = -1; - receive_codec.channels = audio_config.channels; - receive_codec.rate = -1; - break; - case kExternalAudio: - NOTREACHED() << "Codec must be specified for audio decoder"; - break; - } - if (audio_decoder_->RegisterReceiveCodec(receive_codec) != 0) { - NOTREACHED() << "Failed to register receive codec"; - } - - audio_decoder_->SetMaximumPlayoutDelay(audio_config.rtp_max_delay_ms); - audio_decoder_->SetPlayoutMode(webrtc::streaming); -} - -AudioDecoder::~AudioDecoder() {} - -bool AudioDecoder::GetRawAudioFrame(int number_of_10ms_blocks, - int desired_frequency, - PcmAudioFrame* audio_frame, - uint32* rtp_timestamp) { - DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::AUDIO_DECODER)); - // We don't care about the race case where a packet arrives at the same time - // as this function in called. The data will be there the next time this - // function is called. - lock_.Acquire(); - // Get a local copy under lock. - bool have_received_packets = have_received_packets_; - lock_.Release(); - - if (!have_received_packets) return false; - - audio_frame->samples.clear(); - - for (int i = 0; i < number_of_10ms_blocks; ++i) { - webrtc::AudioFrame webrtc_audio_frame; - if (0 != audio_decoder_->PlayoutData10Ms(desired_frequency, - &webrtc_audio_frame)) { - return false; - } - if (webrtc_audio_frame.speech_type_ == webrtc::AudioFrame::kPLCCNG || - webrtc_audio_frame.speech_type_ == webrtc::AudioFrame::kUndefined) { - // We are only interested in real decoded audio. - return false; - } - audio_frame->frequency = webrtc_audio_frame.sample_rate_hz_; - audio_frame->channels = webrtc_audio_frame.num_channels_; - - if (i == 0) { - // Use the timestamp from the first 10ms block. - if (0 != audio_decoder_->PlayoutTimestamp(rtp_timestamp)) { - return false; - } - lock_.Acquire(); - last_played_out_timestamp_ = *rtp_timestamp; - lock_.Release(); - } - int samples_per_10ms = webrtc_audio_frame.samples_per_channel_; - - audio_frame->samples.insert( - audio_frame->samples.end(), - &webrtc_audio_frame.data_[0], - &webrtc_audio_frame.data_[samples_per_10ms * audio_frame->channels]); - } - return true; -} - -void AudioDecoder::IncomingParsedRtpPacket(const uint8* payload_data, - size_t payload_size, - const RtpCastHeader& rtp_header) { - DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN)); - DCHECK_LE(payload_size, kIpPacketSize); - audio_decoder_->IncomingPacket(payload_data, static_cast<int32>(payload_size), - rtp_header.webrtc); - lock_.Acquire(); - have_received_packets_ = true; - uint32 last_played_out_timestamp = last_played_out_timestamp_; - lock_.Release(); - - bool complete = false; - if (!frame_id_map_.InsertPacket(rtp_header, &complete)) return; - if (!complete) return; - - cast_message_builder_.CompleteFrameReceived(rtp_header.frame_id, - rtp_header.is_key_frame); - - frame_id_rtp_timestamp_map_[rtp_header.frame_id] = - rtp_header.webrtc.header.timestamp; - - if (last_played_out_timestamp == 0) return; // Nothing is played out yet. - - uint32 latest_frame_id_to_remove = 0; - bool frame_to_remove = false; - - FrameIdRtpTimestampMap::iterator it = frame_id_rtp_timestamp_map_.begin(); - while (it != frame_id_rtp_timestamp_map_.end()) { - if (IsNewerRtpTimestamp(it->second, last_played_out_timestamp)) { - break; - } - frame_to_remove = true; - latest_frame_id_to_remove = it->first; - frame_id_rtp_timestamp_map_.erase(it); - it = frame_id_rtp_timestamp_map_.begin(); - } - if (!frame_to_remove) return; - - frame_id_map_.RemoveOldFrames(latest_frame_id_to_remove); -} - -bool AudioDecoder::TimeToSendNextCastMessage(base::TimeTicks* time_to_send) { - DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN)); - return cast_message_builder_.TimeToSendNextCastMessage(time_to_send); -} - -void AudioDecoder::SendCastMessage() { - DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN)); - cast_message_builder_.UpdateCastMessage(); -} - -} // namespace cast -} // namespace media |