diff options
| -rw-r--r-- | Libraries/LibAudio/ABuffer.h | 47 | ||||
| -rw-r--r-- | Libraries/LibAudio/AClientConnection.cpp | 4 | ||||
| -rw-r--r-- | Libraries/LibAudio/AWavLoader.cpp | 83 | ||||
| -rw-r--r-- | Libraries/LibAudio/AWavLoader.h | 2 | ||||
| -rw-r--r-- | Servers/AudioServer/ASClientConnection.cpp | 2 | ||||
| -rw-r--r-- | Servers/AudioServer/AudioServer.ipc | 2 |
6 files changed, 78 insertions, 62 deletions
diff --git a/Libraries/LibAudio/ABuffer.h b/Libraries/LibAudio/ABuffer.h index 287c0a738f..237ec2b209 100644 --- a/Libraries/LibAudio/ABuffer.h +++ b/Libraries/LibAudio/ABuffer.h @@ -1,6 +1,5 @@ #pragma once -#include <AK/RefCounted.h> #include <AK/ByteBuffer.h> #include <AK/Types.h> #include <AK/Vector.h> @@ -12,19 +11,22 @@ struct ASample { ASample() : left(0) , right(0) - {} + { + } // For mono ASample(float left) : left(left) , right(left) - {} + { + } // For stereo ASample(float left, float right) : left(left) , right(right) - {} + { + } void clip() { @@ -57,38 +59,57 @@ struct ASample { float right; }; +// Small helper to resample from one playback rate to another +// This isn't really "smart", in that we just insert (or drop) samples. +// Should do better... +class AResampleHelper { +public: + AResampleHelper(float source, float target); + + void process_sample(float sample_l, float sample_r); + bool read_sample(float& next_l, float& next_r); + +private: + const float m_ratio; + float m_current_ratio { 0 }; + float m_last_sample_l { 0 }; + float m_last_sample_r { 0 }; +}; + // A buffer of audio samples, normalized to 44100hz. class ABuffer : public RefCounted<ABuffer> { public: - static RefPtr<ABuffer> from_pcm_data(ByteBuffer& data, int num_channels, int bits_per_sample, int source_rate); + static RefPtr<ABuffer> from_pcm_data(ByteBuffer& data, AResampleHelper& resampler, int num_channels, int bits_per_sample); static NonnullRefPtr<ABuffer> create_with_samples(Vector<ASample>&& samples) { return adopt(*new ABuffer(move(samples))); } - static NonnullRefPtr<ABuffer> create_with_shared_buffer(NonnullRefPtr<SharedBuffer>&& buffer) + static NonnullRefPtr<ABuffer> create_with_shared_buffer(NonnullRefPtr<SharedBuffer>&& buffer, int sample_count) { - return adopt(*new ABuffer(move(buffer))); + return adopt(*new ABuffer(move(buffer), sample_count)); } const ASample* samples() const { return (const ASample*)data(); } - int sample_count() const { return m_buffer->size() / (int)sizeof(ASample); } + int sample_count() const { return m_sample_count; } const void* data() const { return m_buffer->data(); } - int size_in_bytes() const { return m_buffer->size(); } + int size_in_bytes() const { return m_sample_count * (int)sizeof(ASample); } int shared_buffer_id() const { return m_buffer->shared_buffer_id(); } SharedBuffer& shared_buffer() { return *m_buffer; } private: explicit ABuffer(Vector<ASample>&& samples) - : m_buffer(*SharedBuffer::create_with_size(samples.size() * sizeof(ASample))) + : m_buffer(*SharedBuffer::create_with_size(samples.size() * sizeof(ASample))), + m_sample_count(samples.size()) { memcpy(m_buffer->data(), samples.data(), samples.size() * sizeof(ASample)); } - explicit ABuffer(NonnullRefPtr<SharedBuffer>&& buffer) - : m_buffer(move(buffer)) + explicit ABuffer(NonnullRefPtr<SharedBuffer>&& buffer, int sample_count) + : m_buffer(move(buffer)), + m_sample_count(sample_count) { } NonnullRefPtr<SharedBuffer> m_buffer; - int m_sample_count { 0 }; + const int m_sample_count; }; diff --git a/Libraries/LibAudio/AClientConnection.cpp b/Libraries/LibAudio/AClientConnection.cpp index 3d9c46405d..0ff43ec198 100644 --- a/Libraries/LibAudio/AClientConnection.cpp +++ b/Libraries/LibAudio/AClientConnection.cpp @@ -18,7 +18,7 @@ void AClientConnection::enqueue(const ABuffer& buffer) { for (;;) { const_cast<ABuffer&>(buffer).shared_buffer().share_with(server_pid()); - auto response = send_sync<AudioServer::EnqueueBuffer>(buffer.shared_buffer_id()); + auto response = send_sync<AudioServer::EnqueueBuffer>(buffer.shared_buffer_id(), buffer.sample_count()); if (response->success()) break; sleep(1); @@ -28,7 +28,7 @@ void AClientConnection::enqueue(const ABuffer& buffer) bool AClientConnection::try_enqueue(const ABuffer& buffer) { const_cast<ABuffer&>(buffer).shared_buffer().share_with(server_pid()); - auto response = send_sync<AudioServer::EnqueueBuffer>(buffer.shared_buffer_id()); + auto response = send_sync<AudioServer::EnqueueBuffer>(buffer.shared_buffer_id(), buffer.sample_count()); return response->success(); } diff --git a/Libraries/LibAudio/AWavLoader.cpp b/Libraries/LibAudio/AWavLoader.cpp index 6315a8c3c2..d64a65ddc6 100644 --- a/Libraries/LibAudio/AWavLoader.cpp +++ b/Libraries/LibAudio/AWavLoader.cpp @@ -1,5 +1,4 @@ #include <AK/BufferStream.h> -#include <LibAudio/ABuffer.h> #include <LibAudio/AWavLoader.h> #include <LibCore/CFile.h> #include <LibCore/CIODeviceStreamReader.h> @@ -14,6 +13,7 @@ AWavLoader::AWavLoader(const StringView& path) } parse_header(); + m_resampler = make<AResampleHelper>(m_sample_rate, 44100); } RefPtr<ABuffer> AWavLoader::get_more_samples(size_t max_bytes_to_read_from_input) @@ -26,7 +26,7 @@ RefPtr<ABuffer> AWavLoader::get_more_samples(size_t max_bytes_to_read_from_input if (raw_samples.is_empty()) return nullptr; - auto buffer = ABuffer::from_pcm_data(raw_samples, m_num_channels, m_bits_per_sample, m_sample_rate); + auto buffer = ABuffer::from_pcm_data(raw_samples, *m_resampler, m_num_channels, m_bits_per_sample); m_loaded_samples += buffer->sample_count(); return buffer; } @@ -80,7 +80,7 @@ bool AWavLoader::parse_header() u16 audio_format; stream >> audio_format; - CHECK_OK("Audio format"); // incomplete read check + CHECK_OK("Audio format"); // incomplete read check ok = ok && audio_format == 1; // WAVE_FORMAT_PCM ASSERT(audio_format == 1); CHECK_OK("Audio format"); // value check @@ -137,64 +137,62 @@ bool AWavLoader::parse_header() return true; } -// Small helper to resample from one playback rate to another -// This isn't really "smart", in that we just insert (or drop) samples. -// Should do better... -class AResampleHelper { -public: - AResampleHelper(float source, float target); - bool read_sample(); - void prepare(); - -private: - const float m_ratio; - float m_current_ratio { 0 }; -}; - AResampleHelper::AResampleHelper(float source, float target) : m_ratio(source / target) { } -void AResampleHelper::prepare() +void AResampleHelper::process_sample(float sample_l, float sample_r) { - m_current_ratio += m_ratio; + m_last_sample_l = sample_l; + m_last_sample_r = sample_r; + m_current_ratio += 1; } -bool AResampleHelper::read_sample() +bool AResampleHelper::read_sample(float& next_l, float& next_r) { - if (m_current_ratio > 1) { - m_current_ratio--; + if (m_current_ratio > 0) { + m_current_ratio -= m_ratio; + next_l = m_last_sample_l; + next_r = m_last_sample_r; return true; } return false; } -template<typename SampleReader> -static void read_samples_from_stream(BufferStream& stream, SampleReader read_sample, Vector<ASample>& samples, int num_channels, int source_rate) +template <typename SampleReader> +static void read_samples_from_stream(BufferStream& stream, SampleReader read_sample, Vector<ASample>& samples, AResampleHelper& resampler, int num_channels) { - AResampleHelper resampler(source_rate, 44100); float norm_l = 0; float norm_r = 0; + switch (num_channels) { case 1: - while (!stream.handle_read_failure()) { - resampler.prepare(); - while (resampler.read_sample()) { - norm_l = read_sample(stream); + for(;;) { + while (resampler.read_sample(norm_l, norm_r)) { + samples.append(ASample(norm_l)); + } + norm_l = read_sample(stream); + + if (stream.handle_read_failure()) { + break; } - samples.append(ASample(norm_l)); + resampler.process_sample(norm_l, norm_r); } break; case 2: - while (!stream.handle_read_failure()) { - resampler.prepare(); - while (resampler.read_sample()) { - norm_l = read_sample(stream); - norm_r = read_sample(stream); + for(;;) { + while (resampler.read_sample(norm_l, norm_r)) { + samples.append(ASample(norm_l, norm_r)); + } + norm_l = read_sample(stream); + norm_r = read_sample(stream); + + if (stream.handle_read_failure()) { + break; } - samples.append(ASample(norm_l, norm_r)); + resampler.process_sample(norm_l, norm_r); } break; default: @@ -236,7 +234,7 @@ static float read_norm_sample_8(BufferStream& stream) // ### can't const this because BufferStream is non-const // perhaps we need a reading class separate from the writing one, that can be // entirely consted. -RefPtr<ABuffer> ABuffer::from_pcm_data(ByteBuffer& data, int num_channels, int bits_per_sample, int source_rate) +RefPtr<ABuffer> ABuffer::from_pcm_data(ByteBuffer& data, AResampleHelper& resampler, int num_channels, int bits_per_sample) { BufferStream stream(data); Vector<ASample> fdata; @@ -248,13 +246,13 @@ RefPtr<ABuffer> ABuffer::from_pcm_data(ByteBuffer& data, int num_channels, int b switch (bits_per_sample) { case 8: - read_samples_from_stream(stream, read_norm_sample_8, fdata, num_channels, source_rate); + read_samples_from_stream(stream, read_norm_sample_8, fdata, resampler, num_channels); break; case 16: - read_samples_from_stream(stream, read_norm_sample_16, fdata, num_channels, source_rate); + read_samples_from_stream(stream, read_norm_sample_16, fdata, resampler, num_channels); break; case 24: - read_samples_from_stream(stream, read_norm_sample_24, fdata, num_channels, source_rate); + read_samples_from_stream(stream, read_norm_sample_24, fdata, resampler, num_channels); break; default: ASSERT_NOT_REACHED(); @@ -265,10 +263,5 @@ RefPtr<ABuffer> ABuffer::from_pcm_data(ByteBuffer& data, int num_channels, int b // don't belong. ASSERT(!stream.handle_read_failure()); - // HACK: This is a total hack to remove an unnecessary sample at the end of the buffer. - // FIXME: Don't generate the extra sample... :^) - for (int i = 0; i < 1; ++i) - fdata.take_last(); - return ABuffer::create_with_samples(move(fdata)); } diff --git a/Libraries/LibAudio/AWavLoader.h b/Libraries/LibAudio/AWavLoader.h index 494649b7a6..34fb0a0305 100644 --- a/Libraries/LibAudio/AWavLoader.h +++ b/Libraries/LibAudio/AWavLoader.h @@ -4,6 +4,7 @@ #include <AK/RefPtr.h> #include <AK/StringView.h> #include <LibCore/CFile.h> +#include <LibAudio/ABuffer.h> class ABuffer; @@ -31,6 +32,7 @@ private: bool parse_header(); RefPtr<CFile> m_file; String m_error_string; + OwnPtr<AResampleHelper> m_resampler; u32 m_sample_rate { 0 }; u16 m_num_channels { 0 }; diff --git a/Servers/AudioServer/ASClientConnection.cpp b/Servers/AudioServer/ASClientConnection.cpp index 227de826ef..0ec2b0d13c 100644 --- a/Servers/AudioServer/ASClientConnection.cpp +++ b/Servers/AudioServer/ASClientConnection.cpp @@ -68,6 +68,6 @@ OwnPtr<AudioServer::EnqueueBufferResponse> ASClientConnection::handle(const Audi if (m_queue->is_full()) return make<AudioServer::EnqueueBufferResponse>(false); - m_queue->enqueue(ABuffer::create_with_shared_buffer(*shared_buffer)); + m_queue->enqueue(ABuffer::create_with_shared_buffer(*shared_buffer, message.sample_count())); return make<AudioServer::EnqueueBufferResponse>(true); } diff --git a/Servers/AudioServer/AudioServer.ipc b/Servers/AudioServer/AudioServer.ipc index c2078afeb9..dfad1dd7fa 100644 --- a/Servers/AudioServer/AudioServer.ipc +++ b/Servers/AudioServer/AudioServer.ipc @@ -8,5 +8,5 @@ endpoint AudioServer SetMainMixVolume(i32 volume) => () // Buffer playback - EnqueueBuffer(i32 buffer_id) => (bool success) + EnqueueBuffer(i32 buffer_id, int sample_count) => (bool success) } |