roc_audio: adjust resampler coef

src/lib/roc/config.h

@@ -105,6 +105,9 @@ typedef struct roc_receiver_config {
 
     //! A bitmask of ROC_FLAG_* constants.
     unsigned int flags;
+
+    //! Number of samples per second per channel.
+    unsigned int sample_rate;
 } roc_receiver_config;
 
 #ifdef __cplusplus

src/lib/roc/receiver.cpp

@@ -28,6 +28,10 @@ bool make_receiver_config(pipeline::ReceiverConfig& out, const roc_receiver_conf
     out.default_session.timeout = in->timeout;
     out.default_session.samples_per_packet = in->samples_per_packet;
 
+    if (in->sample_rate) {
+        out.sample_rate = in->sample_rate;
+    }
+
     switch ((unsigned)in->fec_scheme) {
     case ROC_FEC_RS8M:
         out.default_session.fec.codec = fec::ReedSolomon8m;

src/modules/roc_audio/resampler_updater.cpp

@@ -22,7 +22,8 @@ const core::nanoseconds_t LogRate = 5000000000;
 } // namespace
 
 ResamplerUpdater::ResamplerUpdater(packet::timestamp_t update_interval,
-                                   packet::timestamp_t aim_queue_size)
+                                   packet::timestamp_t aim_queue_size,
+                                   const float sample_rate_coef)
     : writer_(NULL)
     , reader_(NULL)
     , resampler_(NULL)
@@ -34,7 +35,8 @@ ResamplerUpdater::ResamplerUpdater(packet::timestamp_t update_interval,
     , head_(0)
     , has_tail_(false)
     , tail_(0)
-    , started_(false) {
+    , started_(false)
+    , sample_rate_coef_(sample_rate_coef) {
 }
 
 void ResamplerUpdater::set_writer(packet::IWriter& writer) {
@@ -96,13 +98,16 @@ bool ResamplerUpdater::update(packet::timestamp_t time) {
         update_time_ += update_interval_;
     }
 
+    const float adjusted_coef = sample_rate_coef_ * fe_.freq_coeff();
+
     if (rate_limiter_.allow()) {
-        roc_log(LogDebug, "resampler updater: queue_size=%lu fe=%.5f",
-                (unsigned long)queue_size, (double)fe_.freq_coeff());
+        roc_log(LogDebug, "resampler updater: queue_size=%lu fe=%.5f, adjust_fe=%.5f",
+                (unsigned long)queue_size, (double)fe_.freq_coeff(),
+                (double)adjusted_coef);
     }
 
     roc_panic_if(!resampler_);
-    return resampler_->set_scaling(fe_.freq_coeff());
+    return resampler_->set_scaling(adjusted_coef);
 }
 
 } // namespace audio

src/modules/roc_audio/resampler_updater.h

@@ -34,8 +34,11 @@ class ResamplerUpdater : public packet::IWriter,
     //! @b Parameters
     //!  - @p update_interval defines how often to call FreqEstimator, in samples
     //!  - @p aim_queue_size defines FreqEstimator target queue size, in samples
+    //!  - @p sample_rate_coef represents a factor for converting session sample rate to
+    //!       output sample rate of a receiver.
     ResamplerUpdater(packet::timestamp_t update_interval,
-                     packet::timestamp_t aim_queue_size);
+                     packet::timestamp_t aim_queue_size,
+                     const float sample_rate_coef);
 
     //! Set output writer.
     void set_writer(packet::IWriter&);
@@ -76,6 +79,8 @@ class ResamplerUpdater : public packet::IWriter,
     packet::timestamp_t tail_;
 
     bool started_;
+
+    const float sample_rate_coef_;
 };
 
 } // namespace audio

src/modules/roc_pipeline/config.h

@@ -125,7 +125,7 @@ struct ReceiverConfig {
     //! Default parameters for session.
     SessionConfig default_session;
 
-    //! Sample rate, number of samples for all channels per second.
+    //! Number of samples per second per channel.
     size_t sample_rate;
 
     //! Channel mask.
@@ -149,7 +149,7 @@ struct SenderConfig {
     //! Parameters for the port from which repair packets are sent.
     PortConfig repair_port;
 
-    //! Sample rate, number of samples for all channels per second.
+    //! Number of samples per second per channel.
     size_t sample_rate;
 
     //! Channel mask.

src/modules/roc_pipeline/receiver.cpp

@@ -148,9 +148,9 @@ bool Receiver::create_session_(const packet::PacketPtr& packet) {
     }
     const packet::Address src_address = packet->udp()->src_addr;
 
-    core::SharedPtr<ReceiverSession> sess = new (allocator_)
-        ReceiverSession(config_.default_session, src_address, format_map_, packet_pool_,
-                        byte_buffer_pool_, sample_buffer_pool_, allocator_);
+    core::SharedPtr<ReceiverSession> sess = new (allocator_) ReceiverSession(
+        config_.default_session, config_.sample_rate, src_address, format_map_,
+        packet_pool_, byte_buffer_pool_, sample_buffer_pool_, allocator_);
 
     if (!sess || !sess->valid()) {
         roc_log(LogError, "receiver: can't create session, initialization failed");

src/modules/roc_pipeline/receiver_session.cpp

@@ -19,6 +19,7 @@ namespace roc {
 namespace pipeline {
 
 ReceiverSession::ReceiverSession(const SessionConfig& config,
+                                 const size_t out_sample_rate,
                                  const packet::Address& src_address,
                                  const rtp::FormatMap& format_map,
                                  packet::PacketPool& packet_pool,
@@ -28,14 +29,17 @@ ReceiverSession::ReceiverSession(const SessionConfig& config,
     : src_address_(src_address)
     , allocator_(allocator)
     , audio_reader_(NULL) {
+    roc_panic_if(out_sample_rate == 0);
+
     const rtp::Format* format = format_map.format(config.payload_type);
     if (!format) {
         return;
     }
 
     if (config.resampling) {
         resampler_updater_.reset(new (allocator_) audio::ResamplerUpdater(
-                                     config.fe_update_interval, config.latency),
+                                     config.fe_update_interval, config.latency,
+                                     (float)format->sample_rate / out_sample_rate),
                                  allocator_);
         if (!resampler_updater_) {
             return;

src/modules/roc_pipeline/receiver_session.h

@@ -49,6 +49,7 @@ class ReceiverSession : public core::RefCnt<ReceiverSession>, public core::ListN
 public:
     //! Initialize.
     ReceiverSession(const SessionConfig& config,
+                    const size_t out_sample_rate,
                     const packet::Address& src_address,
                     const rtp::FormatMap& format_map,
                     packet::PacketPool& packet_pool,

src/modules/roc_sndio/target_sox/roc_sndio/player.cpp

@@ -16,14 +16,12 @@
 namespace roc {
 namespace sndio {
 
-Player::Player(pipeline::IReceiver& input,
-               core::BufferPool<audio::sample_t>& buffer_pool,
+Player::Player(core::BufferPool<audio::sample_t>& buffer_pool,
                core::IAllocator& allocator,
                bool oneshot,
                packet::channel_mask_t channels,
                size_t sample_rate)
     : output_(NULL)
-    , input_(input)
     , buffer_pool_(buffer_pool)
     , allocator_(allocator)
     , clips_(0)
@@ -34,10 +32,6 @@ Player::Player(pipeline::IReceiver& input,
         roc_panic("player: # of channels is zero");
     }
 
-    if (sample_rate == 0) {
-        roc_panic("player: sample rate is zero");
-    }
-
     memset(&out_signal_, 0, sizeof(out_signal_));
     out_signal_.rate = sample_rate;
     out_signal_.channels = (unsigned)n_channels;
@@ -71,18 +65,51 @@ bool Player::open(const char* name, const char* type) {
         return false;
     }
 
+    unsigned long in_rate = (unsigned long)out_signal_.rate;
+    unsigned long out_rate = (unsigned long)output_->signal.rate;
+
+    if (in_rate != 0 && in_rate != out_rate) {
+        roc_log(LogError,
+                "can't open output file or device with the required sample rate: "
+                "required=%lu suggested=%lu",
+                out_rate, in_rate);
+        return false;
+    }
+
+    roc_log(LogInfo,
+            "player:"
+            " bits=%lu out_rate=%lu in_rate=%lu ch=%lu",
+            (unsigned long)output_->encoding.bits_per_sample, out_rate, in_rate,
+            (unsigned long)output_->signal.channels);
+
     return true;
 }
 
 void Player::stop() {
     stop_ = 1;
 }
 
+void Player::start(pipeline::IReceiver& input) {
+    input_ = &input;
+    core::Thread::start();
+}
+
+size_t Player::get_sample_rate() const {
+    if (!output_) {
+        roc_panic("player: can't get sample rate for non-open output file or device");
+    }
+    return size_t(output_->signal.rate);
+}
+
 void Player::run() {
     roc_log(LogDebug, "player: starting thread");
 
+    if (!input_) {
+        roc_panic("player: thread is started not from the start() call");
+    }
+
     if (!output_) {
-        roc_panic("player: thread is started before open() returnes success");
+        roc_panic("player: thread is started before open() returned success");
     }
 
     loop_();
@@ -115,7 +142,7 @@ void Player::loop_() {
     SOX_SAMPLE_LOCALS;
 
     while (!stop_) {
-        pipeline::IReceiver::Status status = input_.read(frame);
+        pipeline::IReceiver::Status status = input_->read(frame);
 
         if (status == pipeline::IReceiver::Inactive) {
             if (oneshot_ && n_bufs_ != 0) {

src/modules/roc_sndio/target_sox/roc_sndio/player.h

@@ -35,11 +35,9 @@ class Player : public core::Thread {
     //! Initialize.
     //!
     //! @b Parameters
-    //!  - @p input is used to read samples
     //!  - @p channels defines bitmask of enabled channels in input buffers
     //!  - @p sample_rate defines sample rate of input buffers
-    Player(pipeline::IReceiver& input,
-           core::BufferPool<audio::sample_t>& buffer_pool,
+    Player(core::BufferPool<audio::sample_t>& buffer_pool,
            core::IAllocator& allocator,
            bool oneshot,
            packet::channel_mask_t channels,
@@ -60,11 +58,23 @@ class Player : public core::Thread {
     //!  Should be called once before calling start().
     bool open(const char* name, const char* type = NULL);
 
+    //! Start reading samples in a separate thread.
+    //!
+    //! @b Parameters
+    //!  - @p input is used to read samples.
+    void start(pipeline::IReceiver& input);
+
     //! Stop thread.
     //! @remarks
     //!  Can be called from any thread.
     void stop();
 
+    //! Get sample rate of an output file or a device.
+    //!
+    //! @pre
+    //!  Output file or device should be opened.
+    size_t get_sample_rate() const;
+
 private:
     virtual void run();
 
@@ -76,7 +86,7 @@ class Player : public core::Thread {
     sox_format_t* output_;
     sox_signalinfo_t out_signal_;
 
-    pipeline::IReceiver& input_;
+    pipeline::IReceiver* input_;
 
     core::BufferPool<audio::sample_t>& buffer_pool_;
     core::IAllocator& allocator_;