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authorMatthias P. Braendli <matthias.braendli@mpb.li>2017-10-07 11:47:05 +0200
committerMatthias P. Braendli <matthias.braendli@mpb.li>2017-10-07 11:51:46 +0200
commit84febca8b268129cdd79ff0d1c4f8eeed092c5fb (patch)
tree3e75880241e0fcd2951c04aad77d7077b4ec130c
parent9c2615425bb4f35a417eb04b1ceebfc77d8e2c8b (diff)
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Use queue for all inputs and unify interface
This also changes the --fifo-silence option. Instead of inserting silence separately, it uses the drift compensation to do that.
-rw-r--r--src/AlsaInput.cpp21
-rw-r--r--src/AlsaInput.h24
-rw-r--r--src/FileInput.cpp49
-rw-r--r--src/FileInput.h18
-rw-r--r--src/InputInterface.h16
-rw-r--r--src/JackInput.cpp6
-rw-r--r--src/JackInput.h3
-rw-r--r--src/VLCInput.cpp6
-rw-r--r--src/VLCInput.h2
-rw-r--r--src/odr-audioenc.cpp128
10 files changed, 150 insertions, 123 deletions
diff --git a/src/AlsaInput.cpp b/src/AlsaInput.cpp
index af3c284..747814f 100644
--- a/src/AlsaInput.cpp
+++ b/src/AlsaInput.cpp
@@ -137,6 +137,12 @@ void AlsaInputThreaded::prepare()
}
}
+bool AlsaInputThreaded::read_source(size_t num_bytes)
+{
+ // Reading done in separate thread, no normal termination condition possible
+ return true;
+}
+
void AlsaInputThreaded::process()
{
uint8_t samplebuf[NUM_SAMPLES_PER_CALL * BYTES_PER_SAMPLE * m_channels];
@@ -158,14 +164,19 @@ void AlsaInputDirect::prepare()
m_init_alsa();
}
-ssize_t AlsaInputDirect::read(uint8_t* buf, size_t length)
+bool AlsaInputDirect::read_source(size_t num_bytes)
{
- int bytes_per_frame = m_channels * BYTES_PER_SAMPLE;
- assert(length % bytes_per_frame == 0);
+ const int bytes_per_frame = m_channels * BYTES_PER_SAMPLE;
+ assert(num_bytes % bytes_per_frame == 0);
- ssize_t read = m_read(buf, length / bytes_per_frame);
+ const size_t num_frames = num_bytes / bytes_per_frame;
+ vector<uint8_t> buf(num_bytes);
+ ssize_t ret = m_read(buf.data(), num_frames);
- return (read > 0) ? read * bytes_per_frame : read;
+ if (ret > 0) {
+ m_queue.push(buf.data(), ret * bytes_per_frame);
+ }
+ return ret == num_frames;
}
#endif // HAVE_ALSA
diff --git a/src/AlsaInput.h b/src/AlsaInput.h
index 08ccbf6..e90ed36 100644
--- a/src/AlsaInput.h
+++ b/src/AlsaInput.h
@@ -47,10 +47,12 @@ class AlsaInput : public InputInterface
public:
AlsaInput(const std::string& alsa_dev,
unsigned int channels,
- unsigned int rate) :
+ unsigned int rate,
+ SampleQueue<uint8_t>& queue) :
m_alsa_dev(alsa_dev),
m_channels(channels),
- m_rate(rate) { }
+ m_rate(rate),
+ m_queue(queue) { }
AlsaInput(const AlsaInput& other) = delete;
AlsaInput& operator=(const AlsaInput& other) = delete;
@@ -70,6 +72,8 @@ class AlsaInput : public InputInterface
unsigned int m_channels;
unsigned int m_rate;
+ SampleQueue<uint8_t>& m_queue;
+
snd_pcm_t *m_alsa_handle = nullptr;
};
@@ -78,8 +82,9 @@ class AlsaInputDirect : public AlsaInput
public:
AlsaInputDirect(const std::string& alsa_dev,
unsigned int channels,
- unsigned int rate) :
- AlsaInput(alsa_dev, channels, rate) { }
+ unsigned int rate,
+ SampleQueue<uint8_t>& queue) :
+ AlsaInput(alsa_dev, channels, rate, queue) { }
#if 0
AlsaInputDirect(AlsaInputDirect&& other) :
@@ -102,6 +107,8 @@ class AlsaInputDirect : public AlsaInput
virtual bool fault_detected(void) const override { return false; };
+ virtual bool read_source(size_t num_bytes) override;
+
/*! Read length Bytes from from the alsa device.
* length must be a multiple of channels * bytes_per_sample.
*
@@ -117,10 +124,9 @@ class AlsaInputThreaded : public AlsaInput
unsigned int channels,
unsigned int rate,
SampleQueue<uint8_t>& queue) :
- AlsaInput(alsa_dev, channels, rate),
+ AlsaInput(alsa_dev, channels, rate, queue),
m_fault(false),
- m_running(false),
- m_queue(queue) { }
+ m_running(false) { }
virtual ~AlsaInputThreaded()
{
@@ -135,6 +141,8 @@ class AlsaInputThreaded : public AlsaInput
virtual bool fault_detected(void) const override { return m_fault; };
+ virtual bool read_source(size_t num_bytes) override;
+
private:
void process();
@@ -142,8 +150,6 @@ class AlsaInputThreaded : public AlsaInput
std::atomic<bool> m_running;
std::thread m_thread;
- SampleQueue<uint8_t>& m_queue;
-
};
#endif // HAVE_ALSA
diff --git a/src/FileInput.cpp b/src/FileInput.cpp
index 89e1dab..5eb39ee 100644
--- a/src/FileInput.cpp
+++ b/src/FileInput.cpp
@@ -84,31 +84,44 @@ void FileInput::prepare(void)
}
}
-ssize_t FileInput::read(uint8_t* buf, size_t length)
+bool FileInput::read_source(size_t num_bytes)
{
- ssize_t pcmread;
+ vector<uint8_t> samplebuf(num_bytes);
+
+ ssize_t ret = 0;
if (m_raw_input) {
- if (fread(buf, length, 1, m_in_fh) == 1) {
- pcmread = length;
- }
- else {
- //fprintf(stderr, "Unable to read from input!\n");
- return 0;
- }
+ ret = fread(samplebuf.data(), 1, num_bytes, m_in_fh);
}
else {
- pcmread = wav_read_data(m_wav, buf, length);
+ ret = wav_read_data(m_wav, samplebuf.data(), num_bytes);
}
- return pcmread;
-}
+ if (ret > 0) {
+ m_queue.push(samplebuf.data(), ret);
+ }
-int FileInput::eof()
-{
- int eof = feof(m_in_fh);
- clearerr(m_in_fh);
- return eof;
-}
+ if (ret < num_bytes) {
+ if (m_raw_input) {
+ if (ferror(m_in_fh)) {
+ return false;
+ }
+
+ if (feof(m_in_fh)) {
+ if (m_continue_after_eof) {
+ clearerr(m_in_fh);
+ }
+ else {
+ return false;
+ }
+ }
+ }
+ else {
+ // the wavfile input doesn't support the continuation after EOF
+ return false;
+ }
+ }
+ return true;
+}
diff --git a/src/FileInput.h b/src/FileInput.h
index 59e0f0b..c839b42 100644
--- a/src/FileInput.h
+++ b/src/FileInput.h
@@ -31,6 +31,7 @@
#include <stdint.h>
#include <cstdio>
#include <string>
+#include "SampleQueue.h"
#include "InputInterface.h"
class FileInput : public InputInterface
@@ -38,10 +39,14 @@ class FileInput : public InputInterface
public:
FileInput(const std::string& filename,
bool raw_input,
- int sample_rate) :
+ int sample_rate,
+ bool continue_after_eof,
+ SampleQueue<uint8_t>& queue) :
m_filename(filename),
m_raw_input(raw_input),
- m_sample_rate(sample_rate) {}
+ m_sample_rate(sample_rate),
+ m_continue_after_eof(continue_after_eof),
+ m_queue(queue) {}
~FileInput();
FileInput(const FileInput& other) = delete;
@@ -52,17 +57,14 @@ class FileInput : public InputInterface
virtual bool fault_detected(void) const override { return false; };
- /*! Read length bytes into buf.
- *
- * \return the number of bytes read.
- */
- ssize_t read(uint8_t* buf, size_t length);
- int eof();
+ virtual bool read_source(size_t num_bytes) override;
protected:
std::string m_filename;
bool m_raw_input;
int m_sample_rate;
+ bool m_continue_after_eof;
+ SampleQueue<uint8_t>& m_queue;
/* handle to the wav reader */
void *m_wav = nullptr;
diff --git a/src/InputInterface.h b/src/InputInterface.h
index b582c33..d2fcf97 100644
--- a/src/InputInterface.h
+++ b/src/InputInterface.h
@@ -32,6 +32,20 @@ class InputInterface {
*/
virtual void prepare(void) = 0;
- /*! Return true if the input detected some sort of fault */
+ /*! Return true if the input detected some sort of fault or
+ * abnormal termination
+ */
virtual bool fault_detected(void) const = 0;
+
+ /*! Tell the input that it shall read from source and fill the queue.
+ * The num_samples argument is an indication on how many bytes
+ * the encoder needs.
+ * Some inputs fill the queue from another thread, in which case
+ * this function might only serve as indication that data gets
+ * consumed.
+ *
+ * A return value of true means data was read, a return value of
+ * false means a normal termination of the input (e.g. end of file)
+ */
+ virtual bool read_source(size_t num_bytes) = 0;
};
diff --git a/src/JackInput.cpp b/src/JackInput.cpp
index 70c354f..958685d 100644
--- a/src/JackInput.cpp
+++ b/src/JackInput.cpp
@@ -104,6 +104,12 @@ void JackInput::prepare()
}
}
+bool JackInput::read_source(size_t num_bytes)
+{
+ // Reading done in separate thread, no normal termination condition possible
+ return true;
+}
+
void JackInput::jack_process(jack_nframes_t nframes)
{
/*! JACK works with float samples, we need to convert
diff --git a/src/JackInput.h b/src/JackInput.h
index aa33a03..d5e2bcf 100644
--- a/src/JackInput.h
+++ b/src/JackInput.h
@@ -59,8 +59,11 @@ class JackInput : public InputInterface
virtual ~JackInput();
virtual void prepare() override;
+
virtual bool fault_detected(void) const override { return m_fault; };
+ virtual bool read_source(size_t num_bytes) override;
+
private:
jack_client_t *m_client;
diff --git a/src/VLCInput.cpp b/src/VLCInput.cpp
index 9c424bb..494d620 100644
--- a/src/VLCInput.cpp
+++ b/src/VLCInput.cpp
@@ -252,6 +252,12 @@ void VLCInput::prepare()
m_thread = std::thread(&VLCInput::process, this);
}
+bool VLCInput::read_source(size_t num_bytes)
+{
+ // Reading done in separate thread, no normal termination condition possible
+ return true;
+}
+
void VLCInput::preRender_cb(uint8_t** pp_pcm_buffer, size_t size)
{
const size_t max_length = 20 * size;
diff --git a/src/VLCInput.h b/src/VLCInput.h
index 703641d..076e961 100644
--- a/src/VLCInput.h
+++ b/src/VLCInput.h
@@ -117,6 +117,8 @@ class VLCInput : public InputInterface
* the libVLC thread that fills m_samplequeue */
virtual void prepare() override;
+ virtual bool read_source(size_t num_bytes) override;
+
/*! Write the last received ICY-Text to the
* file.
*/
diff --git a/src/odr-audioenc.cpp b/src/odr-audioenc.cpp
index 11978a9..098d700 100644
--- a/src/odr-audioenc.cpp
+++ b/src/odr-audioenc.cpp
@@ -299,12 +299,12 @@ static int prepare_aac_encoder(
return 1;
}
}
- if (aacEncEncode(*encoder, NULL, NULL, NULL, NULL) != AACENC_OK) {
+ if (aacEncEncode(*encoder, nullptr, nullptr, nullptr, nullptr) != AACENC_OK) {
fprintf(stderr, "Unable to initialize the encoder\n");
return 1;
}
- uint32_t bw = aacEncoder_GetParam(*encoder, AACENC_BANDWIDTH);
+ const uint32_t bw = aacEncoder_GetParam(*encoder, AACENC_BANDWIDTH);
fprintf(stderr, "Bandwidth is %d\n", bw);
return 0;
@@ -395,6 +395,7 @@ int main(int argc, char *argv[])
// For the file input
string infile;
+ bool continue_after_eof = false;
int raw_input = 0;
// For the VLC input
@@ -407,17 +408,16 @@ int main(int argc, char *argv[])
unsigned verbosity = 0;
// For the file output
- FILE *out_fh = NULL;
+ FILE *out_fh = nullptr;
string jack_name;
vector<string> output_uris;
int sample_rate=48000, channels=2;
- void *rs_handler = NULL;
+ void *rs_handler = nullptr;
bool afterburner = true;
uint32_t bandwidth = 0;
- bool inFifoSilence = false;
bool drift_compensation = false;
AACENC_InfoStruct info = { 0 };
int aot = AOT_NONE;
@@ -449,7 +449,7 @@ int main(int argc, char *argv[])
int show_level = 0;
/* Data for ZMQ CURVE authentication */
- char* keyfile = NULL;
+ char* keyfile = nullptr;
char secretkey[CURVE_KEYLEN+1];
const struct option longopts[] = {
@@ -519,7 +519,11 @@ int main(int argc, char *argv[])
case 2: // PS
aot = AOT_DABPLUS_PS;
break;
- case 3: // FIFO SILENCE
+ case 3: // FIFO Silence
+ continue_after_eof = true;
+ // Enable drift compensation, otherwise we would block instead of inserting silence.
+ drift_compensation = true;
+ break;
case 4: // DAB channel mode
dab_channel_mode = optarg;
if (not( dab_channel_mode == "s" or
@@ -694,7 +698,7 @@ int main(int argc, char *argv[])
if (not output_uris.empty()) {
for (auto uri : output_uris) {
if (uri == "-") {
- if (out_fh != NULL) {
+ if (out_fh != nullptr) {
fprintf(stderr, "You can't write to more than one file!\n");
return 1;
}
@@ -722,7 +726,7 @@ int main(int argc, char *argv[])
zmq_sock.connect(uri.c_str());
}
else { // We assume it's a file name
- if (out_fh != NULL) {
+ if (out_fh != nullptr) {
fprintf(stderr, "You can't write to more than one file!\n");
return 1;
}
@@ -860,7 +864,7 @@ int main(int argc, char *argv[])
/* symsize=8, gfpoly=0x11d, fcr=0, prim=1, nroots=10, pad=135 */
rs_handler = init_rs_char(8, 0x11d, 0, 1, 10, 135);
- if (rs_handler == NULL) {
+ if (rs_handler == nullptr) {
perror("init_rs_char failed");
return 1;
}
@@ -868,9 +872,9 @@ int main(int argc, char *argv[])
// We'll use one of the tree possible inputs
#if HAVE_ALSA
AlsaInputThreaded alsa_in_threaded(alsa_device, channels, sample_rate, queue);
- AlsaInputDirect alsa_in_direct(alsa_device, channels, sample_rate);
+ AlsaInputDirect alsa_in_direct(alsa_device, channels, sample_rate, queue);
#endif
- FileInput file_in(infile, raw_input, sample_rate);
+ FileInput file_in(infile, raw_input, sample_rate, continue_after_eof, queue);
#if HAVE_JACK
JackInput jack_in(jack_name, channels, sample_rate, queue);
#endif
@@ -992,8 +996,8 @@ int main(int argc, char *argv[])
* without drift compensation) or in a non-blocking way (VLC or ALSA
* with drift compensation, JACK).
*
- * The file input doesn't need the queue at all. But the other inputs
- * do, and either use \c pop() or \c pop_wait() depending on if it's blocking or not
+ * All inputs write samples into the queue, and either use \c pop() or
+ * \c pop_wait() depending on if it's blocking or not
*
* In non-blocking, the \c queue makes the data available without delay, and the
* \c drift_compensation_delay() function handles rate throttling.
@@ -1005,68 +1009,14 @@ int main(int argc, char *argv[])
break;
}
- if (not infile.empty()) {
- read_bytes = file_in.read(&input_buf[0], input_buf.size());
- if (read_bytes < 0) {
- break;
- }
- else if (read_bytes != input_buf.size()) {
- if (inFifoSilence && file_in.eof()) {
- memset(&input_buf[0], 0, input_buf.size());
- read_bytes = input_buf.size();
- usleep((long)input_buf.size() * 1000000 /
- (BYTES_PER_SAMPLE * channels * sample_rate));
- }
- else {
- fprintf(stderr, "Short file read !\n");
- read_bytes = 0;
- }
- }
+ if (not input->read_source(input_buf.size())) {
+ fprintf(stderr, "End of input reached\n");
+ retval = 0;
+ break;
}
-#if HAVE_VLC
- else if (not vlc_uri.empty()) {
-
- if (drift_compensation) {
- size_t overruns;
- size_t bytes_from_queue = queue.pop(&input_buf[0], input_buf.size(), &overruns); // returns bytes
- if (bytes_from_queue != input_buf.size()) {
- expand_missing_samples(input_buf, channels, bytes_from_queue);
- }
- read_bytes = input_buf.size();
- drift_compensation_delay(sample_rate, channels, read_bytes);
-
- if (bytes_from_queue != input_buf.size()) {
- status |= STATUS_UNDERRUN;
- }
-
- if (overruns) {
- status |= STATUS_OVERRUN;
- }
- }
- else {
- const int timeout_ms = 10000;
- read_bytes = input_buf.size();
-
- /*! pop_wait() must return after a timeout, otherwise the silence detector cannot do
- * its job.
- */
- size_t bytes_from_queue = queue.pop_wait(&input_buf[0], read_bytes, timeout_ms); // returns bytes
- if (bytes_from_queue < read_bytes) {
- // queue timeout occurred
- fprintf(stderr, "Detected fault in VLC input! No data in time.\n");
- retval = 5;
- break;
- }
- }
-
- if (not vlc_icytext_file.empty()) {
- vlc_input.write_icy_text(vlc_icytext_file, vlc_icytext_dlplus);
- }
- }
-#endif
- else if (drift_compensation or not jack_name.empty()) {
- size_t overruns;
+ if (drift_compensation) {
+ size_t overruns = 0;
size_t bytes_from_queue = queue.pop(&input_buf[0], input_buf.size(), &overruns); // returns bytes
if (bytes_from_queue != input_buf.size()) {
expand_missing_samples(input_buf, channels, bytes_from_queue);
@@ -1083,17 +1033,31 @@ int main(int argc, char *argv[])
}
}
else {
-#if HAVE_ALSA
- read_bytes = alsa_in_direct.read(&input_buf[0], input_buf.size());
- if (read_bytes < 0) {
+ const int timeout_ms = 10000;
+ read_bytes = input_buf.size();
+
+ /*! pop_wait() must return after a timeout, otherwise the silence detector cannot do
+ * its job. */
+ size_t bytes_from_queue = queue.pop_wait(&input_buf[0], read_bytes, timeout_ms); // returns bytes
+
+ if (bytes_from_queue < read_bytes) {
+ // queue timeout occurred
+ fprintf(stderr, "Detected fault in VLC input! No data in time.\n");
+ retval = 5;
break;
}
- else if (read_bytes != input_buf.size()) {
- fprintf(stderr, "Short alsa read !\n");
- }
-#endif
}
+ /*! \section MetadataFromSource
+ * The VLC input is the only input that can also give us metadata, which
+ * we can hand over to ODR-PadEnc.
+ */
+#if HAVE_VLC
+ if (not vlc_uri.empty() and not vlc_icytext_file.empty()) {
+ vlc_input.write_icy_text(vlc_icytext_file, vlc_icytext_dlplus);
+ }
+#endif
+
/*! \section AudioLevel
* Audio level measurement is always done assuming we have two
* channels, and is formally wrong in mono, but still gives
@@ -1113,7 +1077,7 @@ int main(int argc, char *argv[])
* only useful if the connection dropped, or if no data is available. It is not
* useful if the source is nearly silent (some noise present), because the
* threshold is 0, and not configurable. The rationale is that we want to
- * guard against connection issues, not source level issues
+ * guard against connection issues, not source level issues.
*/
if (die_on_silence && MAX(peak_left, peak_right) == 0) {
const unsigned int frame_time_msec = 1000ul *