From 54b856b21c6e7a3d5dbfd2b0e56efb739431971a Mon Sep 17 00:00:00 2001
From: Nicolas Pope <nwpope@utu.fi>
Date: Sun, 5 Jul 2020 20:55:13 +0300
Subject: [PATCH] WIP fix aggregate bug
---
applications/tools/simple_viewer/main.cpp | 3 ++-
components/audio/src/software_decoder.cpp | 3 +++
components/audio/src/software_encoder.cpp | 2 +-
components/streams/src/receiver.cpp | 26 ++++++++++++++---------
components/structures/src/new_frame.cpp | 3 +++
5 files changed, 25 insertions(+), 12 deletions(-)
diff --git a/applications/tools/simple_viewer/main.cpp b/applications/tools/simple_viewer/main.cpp
index d9d79dd13..80a591b3c 100644
--- a/applications/tools/simple_viewer/main.cpp
+++ b/applications/tools/simple_viewer/main.cpp
@@ -154,7 +154,8 @@ static void run(ftl::Configurable *root) {
}
if (f.has(Channel::AudioStereo)) {
- LOG(INFO) << "Got stereo";
+ const auto &audio = f.get<std::list<ftl::audio::Audio>>(Channel::AudioStereo).front();
+ LOG(INFO) << "Got stereo: " << audio.data().size();
}
}
diff --git a/components/audio/src/software_decoder.cpp b/components/audio/src/software_decoder.cpp
index c5516fa26..8dd11bda1 100644
--- a/components/audio/src/software_decoder.cpp
+++ b/components/audio/src/software_decoder.cpp
@@ -78,6 +78,9 @@ bool SoftwareDecoder::_decodeOpus(const ftl::codecs::Packet &pkt, std::vector<sh
for (size_t i=0; i<pkt.data.size(); ) {
const short *len = (const short*)inptr;
+ LOG(INFO) << "LENGTH = " << *len;
+ if (*len == 0) break;
+
inptr += 2;
i += (*len)+2;
int samples = opus_multistream_decode(opus_decoder_, inptr, *len, outptr, FRAME_SIZE, 0);
diff --git a/components/audio/src/software_encoder.cpp b/components/audio/src/software_encoder.cpp
index 054bb886f..6eeda3dbb 100644
--- a/components/audio/src/software_encoder.cpp
+++ b/components/audio/src/software_encoder.cpp
@@ -105,7 +105,7 @@ bool SoftwareEncoder::_encodeOpus(const std::vector<short> &in, ftl::codecs::Pac
short *len = (short*)outptr;
outptr += 2;
int nbBytes = opus_multistream_encode(opus_encoder_, &in.data()[i], FRAME_SIZE, outptr, MAX_PACKET_SIZE);
- //LOG(INFO) << "Opus encode: " << nbBytes << ", " << (in.size()-i);
+ LOG(INFO) << "Opus encode: " << nbBytes << ", " << (in.size()-i);
if (nbBytes <= 0) return false;
//if (nbBytes > 32000) LOG(WARNING) << "Packet exceeds size limit";
diff --git a/components/streams/src/receiver.cpp b/components/streams/src/receiver.cpp
index c3c59f06a..24c2a54f2 100644
--- a/components/streams/src/receiver.cpp
+++ b/components/streams/src/receiver.cpp
@@ -168,17 +168,23 @@ ftl::audio::Decoder *Receiver::_createAudioDecoder(InternalAudioStates &frame, c
void Receiver::_processAudio(const StreamPacket &spkt, const Packet &pkt) {
// Audio Data
- /*InternalAudioStates &frame = _getAudioFrame(spkt);
+ InternalAudioStates &state = _getAudioFrame(spkt);
+
+ //frame.frame.reset();
+ state.timestamp = spkt.timestamp;
+
+ auto fs = builder(spkt.streamID).get(spkt.timestamp, spkt.frame_number+pkt.frame_count-1);
+ auto &frame = fs->frames[0];
+
+ auto &audiolist = frame.createChange<std::list<ftl::audio::Audio>>(spkt.channel, ftl::data::ChangeType::FOREIGN, pkt);
+ auto &audio = audiolist.emplace_back();
- frame.frame.reset();
- frame.timestamp = spkt.timestamp;
- auto &audio = frame.frame.create<ftl::audio::Audio>(spkt.channel);
//size_t size = pkt.data.size()/sizeof(short);
//audio.data().resize(size);
//auto *ptr = (short*)pkt.data.data();
//for (size_t i=0; i<size; i++) audio.data()[i] = ptr[i];
- ftl::audio::Decoder *dec = _createAudioDecoder(frame, pkt);
+ ftl::audio::Decoder *dec = _createAudioDecoder(state, pkt);
if (!dec) {
LOG(ERROR) << "Could get an audio decoder";
return;
@@ -189,7 +195,7 @@ void Receiver::_processAudio(const StreamPacket &spkt, const Packet &pkt) {
}
// Generate settings from packet data
- ftl::audio::AudioSettings settings;
+ /*ftl::audio::AudioSettings settings;
settings.channels = (spkt.channel == Channel::AudioStereo) ? 2 : 1;
settings.frame_size = 960;
@@ -197,12 +203,12 @@ void Receiver::_processAudio(const StreamPacket &spkt, const Packet &pkt) {
case definition_t::hz48000 : settings.sample_rate = 48000; break;
case definition_t::hz44100 : settings.sample_rate = 44100; break;
default: settings.sample_rate = 48000; break;
- }
+ }*/
- frame.state.setLeft(settings);
- frame.frame.setOrigin(&frame.state);
+ //frame.state.setLeft(settings);
+ //frame.frame.setOrigin(&frame.state);
- if (audio_cb_) {
+ /*if (audio_cb_) {
// Create an audio frameset wrapper.
ftl::audio::FrameSet fs;
fs.id = 0;
diff --git a/components/structures/src/new_frame.cpp b/components/structures/src/new_frame.cpp
index c5a146ff4..3ce0238b1 100644
--- a/components/structures/src/new_frame.cpp
+++ b/components/structures/src/new_frame.cpp
@@ -328,6 +328,9 @@ void Frame::reset() {
for (auto &d : data_) {
d.second.status = ChannelStatus::INVALID;
d.second.encoded.clear();
+
+ // Note: Aggregate channels cannot memory pool and must be cleared.
+ if (ftl::data::isAggregate(d.first)) d.second.data.reset();
}
changed_.clear();
status_ = FrameStatus::CREATED;
--
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