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audio: let the AJM Opus decoder write part of a packet (#1914)
ajm: buffer and resample Opus output
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Devran Cosmo Uenal authored and 
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#pragma once
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#include "libs/ajm/decoder.h"
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#include <algorithm>
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#include <cmath>
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#include <cstring>
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#include <vector>
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namespace Libs::Audio::Ajm {
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constexpr uint64_t AJM_INSTANCE_FLAG_GAPLESS_LOOP = 1ull << 10u;
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constexpr uint64_t AJM_INSTANCE_FLAG_RESAMPLE = 1ull << 12u;
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constexpr uint64_t AJM_INSTANCE_FLAG_SEAMLESS_LOOP = 1ull << 13u;
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constexpr uint32_t AJM_RESAMPLE_FLAG_RESET = 1u;
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constexpr uint32_t AJM_RESAMPLE_FLAG_END = 2u;
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constexpr uint32_t AJM_RESAMPLE_FLAG_IGNORE_RATIO_PARAMETER = 8u;
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class AjmResampler {
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public:
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int32_t SetParameters(float ratio, float change, uint32_t flags) {
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if ((flags & AJM_RESAMPLE_FLAG_IGNORE_RATIO_PARAMETER) == 0 &&
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(!std::isfinite(ratio) || ratio < 0 || ratio > 4 || !std::isfinite(change))) {
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return AJM_RESULT_INVALID_PARAMETER;
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}
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if ((flags & AJM_RESAMPLE_FLAG_RESET) != 0) {
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m_pcm.clear();
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m_position = 0;
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m_end = false;
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}
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if ((flags & AJM_RESAMPLE_FLAG_IGNORE_RATIO_PARAMETER) == 0) {
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m_ratio = ratio;
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m_change = std::abs(change) < 0.000001f ? 0 : change;
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}
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m_end |= (flags & AJM_RESAMPLE_FLAG_END) != 0;
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return OK;
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}
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[[nodiscard]] AjmSidebandResampleInfo GetInfo() const {
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AjmSidebandResampleInfo info;
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info.ratio = static_cast<float>(m_ratio);
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info.num_samples = m_end && m_position >= Samples()
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? -1
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: static_cast<int32_t>(std::ceil(Samples() - m_position));
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return info;
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}
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AjmDecodeResult Decode(AjmDecoder& decoder, uint64_t flags, const void* input,
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size_t input_size, void* output, size_t output_size,
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bool multiple_frames, AjmGaplessState* gapless) {
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auto result = decoder.MakeResult();
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if (result.format.channel_num == 0) {
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return decoder.Decode(input, input_size, output, output_size, multiple_frames, gapless);
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}
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bool stop_decoding = false;
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for (;;) {
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const auto format = decoder.GetFormat();
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const auto encoding = static_cast<AjmSampleEncoding>(format.sample_encoding);
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const auto sample_bytes = format.channel_num * AjmBytesPerSample(encoding);
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if (!m_pcm.empty() && sample_bytes != m_sample_bytes) {
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result.result = AJM_RESULT_INVALID_PARAMETER;
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break;
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}
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if (output == nullptr || output_size - result.output_written < sample_bytes) {
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if (m_position < Samples() || result.input_consumed < input_size) {
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result.result |= AJM_RESULT_NOT_ENOUGH_ROOM;
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}
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break;
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}
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const auto index = static_cast<size_t>(m_position);
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const auto fraction = m_position - static_cast<double>(index);
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const bool need_next = fraction != 0 && index + 1 >= Samples();
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if (index >= Samples() || need_next) {
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Compact();
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if (!stop_decoding && input != nullptr && result.input_consumed < input_size &&
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(gapless == nullptr || !gapless->IsEnd())) {
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// A decode job can produce up to six 20 ms Opus frames in one packet.
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const auto channels = std::max<uint32_t>(format.channel_num, flags & 0x7fu);
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m_frame.resize(5760 * channels * AjmBytesPerSample(encoding));
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auto decoded =
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decoder.Decode(static_cast<const uint8_t*>(input) + result.input_consumed,
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input_size - result.input_consumed, m_frame.data(),
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m_frame.size(), false, gapless);
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result.input_consumed += decoded.input_consumed;
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result.frames += decoded.frames;
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result.frames_per_packet = decoded.frames_per_packet;
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result.total_decoded_samples = decoded.total_decoded_samples;
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result.format = decoded.format;
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result.result |= decoded.result;
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result.internal_result = decoded.internal_result;
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m_sample_bytes = decoded.format.channel_num * AjmBytesPerSample(encoding);
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m_pcm.insert(m_pcm.end(), m_frame.begin(),
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m_frame.begin() + decoded.output_written);
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stop_decoding =
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!multiple_frames || decoded.result != OK || decoded.input_consumed == 0;
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if (gapless != nullptr && gapless->IsEnd() &&
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(flags & AJM_INSTANCE_FLAG_GAPLESS_LOOP) != 0) {
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gapless->Reset();
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stop_decoding |= (flags & AJM_INSTANCE_FLAG_SEAMLESS_LOOP) == 0;
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}
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continue;
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}
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if (!m_end || m_position >= Samples()) {
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if (!m_end && !stop_decoding && (gapless == nullptr || !gapless->IsEnd())) {
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result.result |= AJM_RESULT_PARTIAL_INPUT;
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}
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break;
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}
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}
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const auto current = static_cast<size_t>(m_position);
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const auto next = std::min(current + 1, Samples() - 1);
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auto* dst = static_cast<uint8_t*>(output) + result.output_written;
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const auto* a = m_pcm.data() + current * sample_bytes;
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const auto* b = m_pcm.data() + next * sample_bytes;
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const auto weight = m_position - static_cast<double>(current);
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if (weight == 0) {
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std::memcpy(dst, a, sample_bytes);
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} else {
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switch (encoding) {
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case AjmSampleEncoding::S16:
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Interpolate<int16_t>(dst, a, b, format.channel_num, weight);
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break;
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case AjmSampleEncoding::S32:
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Interpolate<int32_t>(dst, a, b, format.channel_num, weight);
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break;
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case AjmSampleEncoding::Float:
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Interpolate<float>(dst, a, b, format.channel_num, weight);
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break;
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}
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}
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result.output_written += sample_bytes;
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m_position += m_ratio;
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m_ratio = std::clamp(m_ratio + m_change, 0.0, 4.0);
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if (m_ratio == 0 || m_ratio == 4) {
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m_change = 0;
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}
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}
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Compact();
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result.resample_info = GetInfo();
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return result;
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}
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private:
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template <typename T>
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static void Interpolate(uint8_t* dst, const uint8_t* a, const uint8_t* b, uint32_t channels,
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double weight) {
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for (uint32_t channel = 0; channel < channels; ++channel) {
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T first, second;
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std::memcpy(&first, a + channel * sizeof(T), sizeof(T));
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std::memcpy(&second, b + channel * sizeof(T), sizeof(T));
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const auto value =
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static_cast<T>(first + (static_cast<double>(second) - first) * weight);
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std::memcpy(dst + channel * sizeof(T), &value, sizeof(T));
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}
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}
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[[nodiscard]] size_t Samples() const {
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return m_sample_bytes == 0 ? 0 : m_pcm.size() / m_sample_bytes;
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}
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void Compact() {
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const auto consumed = std::min(static_cast<size_t>(m_position), Samples());
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const auto bytes = consumed * m_sample_bytes;
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if (bytes != 0) {
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std::memmove(m_pcm.data(), m_pcm.data() + bytes, m_pcm.size() - bytes);
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m_pcm.resize(m_pcm.size() - bytes);
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m_position -= static_cast<double>(consumed);
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}
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}
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std::vector<uint8_t> m_pcm;
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std::vector<uint8_t> m_frame;
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size_t m_sample_bytes = 0;
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double m_position = 0;
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double m_ratio = 1;
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double m_change = 0;
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bool m_end = false;
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};
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} // namespace Libs::Audio::Ajm
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