diff options
| -rw-r--r-- | README.md | 10 | ||||
| -rw-r--r-- | src/audio/jitterbuffer.hpp | 82 | ||||
| -rw-r--r-- | src/audio/manager.cpp | 22 | ||||
| -rw-r--r-- | src/audio/manager.hpp | 3 | ||||
| -rw-r--r-- | src/settings.cpp | 2 | ||||
| -rw-r--r-- | src/settings.hpp | 2 |
6 files changed, 11 insertions, 110 deletions
@@ -330,12 +330,10 @@ For example, memory_db would be set by adding `memory_db = true` under the line #### voice -| Setting | Type | Default | Description | -|--------------------------|--------|------------------------------------|----------------------------------------------------------------------------------------------------------------------------| -| `vad` | string | rnnoise if enabled, gate otherwise | Method used for voice activity detection. Changeable in UI | -| `backends` | string | empty | Change backend priority when initializing miniaudio: `wasapi;dsound;winmm;coreaudio;sndio;audio4;oss;pulseaudio;alsa;jack` | -| `jitter_latency_desired` | int | 50 | Desired/Minimum latency for jitter buffer (in milliseconds) | -| `jitter_latency_maximum` | int | 200 | Maximum latency for jitter buffer before frames are discarded (in milliseconds) | +| Setting | Type | Default | Description | +|------------|--------|------------------------------------|----------------------------------------------------------------------------------------------------------------------------| +| `vad` | string | rnnoise if enabled, gate otherwise | Method used for voice activity detection. Changeable in UI | +| `backends` | string | empty | Change backend priority when initializing miniaudio: `wasapi;dsound;winmm;coreaudio;sndio;audio4;oss;pulseaudio;alsa;jack` | #### windows diff --git a/src/audio/jitterbuffer.hpp b/src/audio/jitterbuffer.hpp deleted file mode 100644 index 3da3594..0000000 --- a/src/audio/jitterbuffer.hpp +++ /dev/null @@ -1,82 +0,0 @@ -#pragma once -#include <chrono> -#include <cstdint> -#include <deque> - -// very simple non-RTP-based jitter buffer. does not handle out-of-order -template<typename SampleFormat> -class JitterBuffer { -public: - /* - * desired_latency: how many milliseconds before audio can be drawn from buffer - * maximum_latency: how many milliseconds before old audio starts to be discarded - */ - JitterBuffer(int desired_latency, int maximum_latency, int channels, int sample_rate) - : m_desired_latency(desired_latency) - , m_maximum_latency(maximum_latency) - , m_channels(channels) - , m_sample_rate(sample_rate) - , m_last_push(std::chrono::steady_clock::now()) { - } - - [[nodiscard]] size_t Available() const noexcept { - return m_samples.size(); - } - - bool PopSamples(SampleFormat *ptr, size_t amount) { - CheckBuffering(); - if (m_buffering || Available() < amount) return false; - std::copy(m_samples.begin(), m_samples.begin() + amount, ptr); - m_samples.erase(m_samples.begin(), m_samples.begin() + amount); - return true; - } - - void PushSamples(SampleFormat *ptr, size_t amount) { - m_samples.insert(m_samples.end(), ptr, ptr + amount); - m_last_push = std::chrono::steady_clock::now(); - const auto buffered = MillisBuffered(); - if (buffered > m_maximum_latency) { - const auto overflow_ms = MillisBuffered() - m_maximum_latency; - const auto overflow_samples = overflow_ms * m_channels * m_sample_rate / 1000; - m_samples.erase(m_samples.begin(), m_samples.begin() + overflow_samples); - } - } - -private: - [[nodiscard]] size_t MillisBuffered() const { - return m_samples.size() * 1000 / m_channels / m_sample_rate; - } - - void CheckBuffering() { - // if we arent buffering but the buffer is empty then we should be - if (m_samples.empty()) { - if (!m_buffering) { - m_buffering = true; - } - return; - } - - if (!m_buffering) return; - - // if we reached desired latency, we are sufficiently buffered - const auto millis_buffered = MillisBuffered(); - if (millis_buffered >= m_desired_latency) { - m_buffering = false; - } - // if we havent buffered to desired latency but max latency has elapsed, exit buffering so it doesnt get stuck - const auto now = std::chrono::steady_clock::now(); - const auto millis = std::chrono::duration_cast<std::chrono::milliseconds>(now - m_last_push).count(); - if (millis >= m_maximum_latency) { - m_buffering = false; - } - } - - int m_desired_latency; - int m_maximum_latency; - int m_channels; - int m_sample_rate; - bool m_buffering = true; - std::chrono::time_point<std::chrono::steady_clock> m_last_push; - - std::deque<SampleFormat> m_samples; -}; diff --git a/src/audio/manager.cpp b/src/audio/manager.cpp index f9ffdd5..33730d9 100644 --- a/src/audio/manager.cpp +++ b/src/audio/manager.cpp @@ -26,7 +26,6 @@ const uint8_t *StripRTPExtensionHeader(const uint8_t *buf, int num_bytes, size_t return buf; } -// frameCount is configured to be 480 samples per channel void data_callback(ma_device *pDevice, void *pOutput, const void *pInput, ma_uint32 frameCount) { AudioManager *mgr = reinterpret_cast<AudioManager *>(pDevice->pUserData); if (mgr == nullptr) return; @@ -38,14 +37,12 @@ void data_callback(ma_device *pDevice, void *pOutput, const void *pInput, ma_uin if (const auto vol_it = mgr->m_volume_ssrc.find(ssrc); vol_it != mgr->m_volume_ssrc.end()) { volume = vol_it->second; } - - static std::array<int16_t, 480 * 2> buf; - - if (!pair.first.PopSamples(buf.data(), 480 * 2)) continue; - - for (size_t i = 0; i < 480 * 2; i++) { + auto &buf = pair.first; + const size_t n = std::min(static_cast<size_t>(buf.size()), static_cast<size_t>(frameCount * 2ULL)); + for (size_t i = 0; i < n; i++) { pOutputF32[i] += volume * buf[i] / 32768.F; } + buf.erase(buf.begin(), buf.begin() + n); } } @@ -205,14 +202,7 @@ void AudioManager::AddSSRC(uint32_t ssrc) { int error; if (m_sources.find(ssrc) == m_sources.end()) { auto *decoder = opus_decoder_create(48000, 2, &error); - auto &s = Abaddon::Get().GetSettings(); - m_sources.insert(std::make_pair(ssrc, std::make_pair( - JitterBuffer<int16_t>( - s.JitterDesiredLatency, - s.JitterMaximumLatency, - 2, - 48000), - decoder))); + m_sources.insert(std::make_pair(ssrc, std::make_pair(std::deque<int16_t> {}, decoder))); } } @@ -252,7 +242,7 @@ void AudioManager::FeedMeOpus(uint32_t ssrc, const std::vector<uint8_t> &data) { } else { UpdateReceiveVolume(ssrc, pcm.data(), decoded); auto &buf = it->second.first; - buf.PushSamples(pcm.data(), decoded * 2); + buf.insert(buf.end(), pcm.begin(), pcm.begin() + decoded * 2); } } } diff --git a/src/audio/manager.hpp b/src/audio/manager.hpp index 56882fd..5716fc5 100644 --- a/src/audio/manager.hpp +++ b/src/audio/manager.hpp @@ -21,7 +21,6 @@ #endif #include "devices.hpp" -#include "jitterbuffer.hpp" // clang-format on class AudioManager { @@ -137,7 +136,7 @@ private: mutable std::mutex m_rnn_mutex; #endif - std::unordered_map<uint32_t, std::pair<JitterBuffer<int16_t>, OpusDecoder *>> m_sources; + std::unordered_map<uint32_t, std::pair<std::deque<int16_t>, OpusDecoder *>> m_sources; OpusEncoder *m_encoder; diff --git a/src/settings.cpp b/src/settings.cpp index 6dab229..fc76ddb 100644 --- a/src/settings.cpp +++ b/src/settings.cpp @@ -130,8 +130,6 @@ void SettingsManager::DefineSettings() { AddSetting("voice", "vad", "gate"s, &Settings::VAD); #endif AddSetting("voice", "backends", ""s, &Settings::Backends); - AddSetting("voice", "jitter_latency_desired", 50, &Settings::JitterDesiredLatency); - AddSetting("voice", "jitter_latency_maximum", 200, &Settings::JitterMaximumLatency); } void SettingsManager::ReadSettings() { diff --git a/src/settings.hpp b/src/settings.hpp index 0b0f6e2..5805452 100644 --- a/src/settings.hpp +++ b/src/settings.hpp @@ -52,8 +52,6 @@ public: // [voice] std::string VAD; std::string Backends; - int JitterDesiredLatency; - int JitterMaximumLatency; // [windows] bool HideConsole; |