#include #include <alsa/asoundlib.h>
#define BUFFER_SIZE 4096
int main() { int err; char *buffer; snd_pcm_t *capture_handle; snd_pcm_hw_params_t *hw_params;
// Open the capture device (microphone)
err = snd_pcm_open(&capture_handle, "default", SND_PCM_STREAM_CAPTURE, 0);
if (err < 0) {
std::cout << "Cannot open capture device: " << snd_strerror(err) << std::endl;
return 1;
}
// Set hardware parameters for capturing
snd_pcm_hw_params_malloc(&hw_params);
snd_pcm_hw_params_any(capture_handle, hw_params);
snd_pcm_hw_params_set_access(capture_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
snd_pcm_hw_params_set_format(capture_handle, hw_params, SND_PCM_FORMAT_S16_LE);
snd_pcm_hw_params_set_channels(capture_handle, hw_params, 2); // Stereo audio
unsigned int sample_rate = 44100; // Sample rate (adjust as needed)
snd_pcm_hw_params_set_rate_near(capture_handle, hw_params, &sample_rate, 0);
snd_pcm_hw_params(capture_handle, hw_params);
snd_pcm_hw_params_free(hw_params);
// Allocate buffer to hold captured audio data
buffer = (char *)malloc(BUFFER_SIZE);
// Open the remote playback device (speaker)
snd_pcm_t *playback_handle;
err = snd_pcm_open(&playback_handle, "default", SND_PCM_STREAM_PLAYBACK, 0);
if (err < 0) {
std::cout << "Cannot open playback device: " << snd_strerror(err) << std::endl;
return 1;
}
// Set hardware parameters for playback
snd_pcm_hw_params_malloc(&hw_params);
snd_pcm_hw_params_any(playback_handle, hw_params);
snd_pcm_hw_params_set_access(playback_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
snd_pcm_hw_params_set_format(playback_handle, hw_params, SND_PCM_FORMAT_S16_LE);
snd_pcm_hw_params_set_channels(playback_handle, hw_params, 2); // Stereo audio
snd_pcm_hw_params_set_rate_near(playback_handle, hw_params, &sample_rate, 0);
snd_pcm_hw_params(playback_handle, hw_params);
snd_pcm_hw_params_free(hw_params);
// Start capturing and playback loop
while (true) {
// Capture audio from the microphone
err = snd_pcm_readi(capture_handle, buffer, BUFFER_SIZE / 4);
if (err != BUFFER_SIZE / 4) {
std::cout << "Capture error: " << snd_strerror(err) << std::endl;
}
// Play captured audio to the speaker
err = snd_pcm_writei(playback_handle, buffer, BUFFER_SIZE / 4);
if (err != BUFFER_SIZE / 4) {
std::cout << "Playback error: " << snd_strerror(err) << std::endl;
}
}
// Clean up and close the audio devices
free(buffer);
snd_pcm_close(capture_handle);
snd_pcm_close(playback_handle);
return 0;
}
Write C++ code for below scenario in linux: Client side holds microphone in Linux, server hold the remote camera and speaker. Client receive video from the remote camera and display in client. client sends audio to remote speaker.
#include #include <gst/gst.h> #include <gst/app/gstappsink.h> #include <alsa/asoundlib.h>
#define BUFFER_SIZE 4096
// Structure to hold the GStreamer pipeline elements and data struct PipelineData { GstElement* pipeline; GstElement* videoSink; GstElement* audioSink; };
// GStreamer bus callback function to handle messages static gboolean busCallback(GstBus* bus, GstMessage* message, gpointer data) { switch (GST_MESSAGE_TYPE(message)) { case GST_MESSAGE_EOS: g_print("End of stream\n"); break; case GST_MESSAGE_ERROR: { gchar* debug; GError* error; gst_message_parse_error(message, &error, &debug); g_printerr("Error: %s\n", error->message); g_error_free(error); g_free(debug); break; } default: break; }
return TRUE;
}
// GStreamer app sink callback function to process video samples static GstFlowReturn videoSampleCallback(GstAppSink* sink, gpointer data) { GstSample* sample = gst_app_sink_pull_sample(sink); if (sample != NULL) { GstBuffer* buffer = gst_sample_get_buffer(sample); GstMapInfo map; gst_buffer_map(buffer, &map, GST_MAP_READ);
// Process video frame data in 'map.data' with 'map.size'
gst_buffer_unmap(buffer, &map);
gst_sample_unref(sample);
}
return GST_FLOW_OK;
}
int main() { // Initialize GStreamer gst_init(NULL, NULL);
// Create the pipeline
PipelineData pipelineData;
pipelineData.pipeline = gst_pipeline_new("client-pipeline");
// Create the video source element
GstElement* videoSource = gst_element_factory_make("tcpclientsrc", "video-source");
g_object_set(videoSource, "host", "SERVER_IP_ADDRESS", NULL); // Replace with the actual server IP address
g_object_set(videoSource, "port", 5000, NULL); // Replace with the actual server port
gst_bin_add(GST_BIN(pipelineData.pipeline), videoSource);
// Create the video decoder element
GstElement* videoDecoder = gst_element_factory_make("decodebin", "video-decoder");
gst_bin_add(GST_BIN(pipelineData.pipeline), videoDecoder);
gst_element_link(videoSource, videoDecoder);
// Create the video sink element (app sink)
pipelineData.videoSink = gst_element_factory_make("appsink", "video-sink");
GstAppSink* videoAppSink = GST_APP_SINK(pipelineData.videoSink);
gst_bin_add(GST_BIN(pipelineData.pipeline), pipelineData.videoSink);
gst_element_link(videoDecoder, pipelineData.videoSink);
gst_app_sink_set_emit_signals(videoAppSink, TRUE);
gst_app_sink_set_drop(videoAppSink, TRUE);
gst_app_sink_set_max_buffers(videoAppSink, 1);
GstAppSinkCallbacks videoCallbacks;
videoCallbacks.eos = NULL;
videoCallbacks.new_preroll = NULL;
videoCallbacks.new_sample = videoSampleCallback;
gst_app_sink_set_callbacks(videoAppSink, &videoCallbacks, NULL, NULL);
// Create the audio source element (ALSA microphone)
GstElement* audioSource = gst_element_factory_make("alsasrc", "audio-source");
gst_bin_add(GST_BIN(pipelineData.pipeline), audioSource);
// Create the audio encoder element
GstElement* audioEncoder = gst_element_factory_make("audioconvert", "audio-encoder");
gst_bin_add(GST_BIN(pipelineData.pipeline), audioEncoder);
gst_element_link(audioSource, audioEncoder);
// Create the audio sink element (ALSA speaker)
pipelineData.audioSink = gst_element_factory_make("alsasink", "audio-sink");
gst_bin_add(GST_BIN(pipelineData.pipeline), pipelineData.audioSink);
gst_element_link(audioEncoder, pipelineData.audioSink);
// Set up the GStreamer bus to handle messages
GstBus* bus = gst_element_get_bus(pipelineData.pipeline);
gst_bus_add_watch(bus, busCallback, NULL);
gst_object_unref(bus);
// Start the pipeline
gst_element_set_state(pipelineData.pipeline, GST_STATE_PLAYING);
// Open the ALSA microphone device
snd_pcm_t* captureHandle;
int err = snd_pcm_open(&captureHandle, "default", SND_PCM_STREAM_CAPTURE, 0);
if (err < 0) {
std::cout << "Cannot open capture device: " << snd_strerror(err) << std::endl;
return 1;
}
// Set hardware parameters for capturing
snd_pcm_hw_params_t* hw_params;
snd_pcm_hw_params_malloc(&hw_params);
snd_pcm_hw_params_any(captureHandle, hw_params);
snd_pcm_hw_params_set_access(captureHandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
snd_pcm_hw_params_set_format(captureHandle, hw_params, SND_PCM_FORMAT_S16_LE);
snd_pcm_hw_params_set_channels(captureHandle, hw_params, 2); // Stereo audio
unsigned int sampleRate = 44100; // Sample rate (adjust as needed)
snd_pcm_hw_params_set_rate_near(captureHandle, hw_params, &sampleRate, 0);
snd_pcm_hw_params(captureHandle, hw_params);
snd_pcm_hw_params_free(hw_params);
// Allocate buffer to hold captured audio data
char* buffer = (char*)malloc(BUFFER_SIZE);
// Open the ALSA speaker device
snd_pcm_t* playbackHandle;
err = snd_pcm_open(&playbackHandle, "default", SND_PCM_STREAM_PLAYBACK, 0);
if (err < 0) {
std::cout << "Cannot open playback device: " << snd_strerror(err) << std::endl;
return 1;
}
// Set hardware parameters for playback
snd_pcm_hw_params_malloc(&hw_params);
snd_pcm_hw_params_any(playbackHandle, hw_params);
snd_pcm_hw_params_set_access(playbackHandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
snd_pcm_hw_params_set_format(playbackHandle, hw_params, SND_PCM_FORMAT_S16_LE);
snd_pcm_hw_params_set_channels(playbackHandle, hw_params, 2); // Stereo audio
snd_pcm_hw_params_set_rate_near(playbackHandle, hw_params, &sampleRate, 0);
snd_pcm_hw_params(playbackHandle, hw_params);
snd_pcm_hw_params_free(hw_params);
// Audio streaming loop
while (true) {
// Capture audio from the microphone
err = snd_pcm_readi(captureHandle, buffer, BUFFER_SIZE / 4);
if (err != BUFFER_SIZE / 4) {
std::cout << "Capture error: " << snd_strerror(err) << std::endl;
}
// Play captured audio to the speaker
err = snd_pcm_writei(playbackHandle, buffer, BUFFER_SIZE / 4);
if (err != BUFFER_SIZE / 4) {
std::cout << "Playback error: " << snd_strerror(err) << std::endl;
}
}
// Clean up and close the audio devices
free(buffer);
snd_pcm_close(captureHandle);
snd_pcm_close(playbackHandle);
// Stop the pipeline and release resources
gst_element_set_state(pipelineData.pipeline, GST_STATE_NULL);
gst_object_unref(pipelineData.pipeline);
return 0;
}
Here's an example of C++ code that demonstrates transmitting audio from a Linux system to a remote speaker over the network using the G.726 algorithm.
#include #include #include #include #include <netinet/in.h> #include <sys/socket.h> #include <arpa/inet.h>
#define MAX_BUFFER_SIZE 1024
// Function to read audio data from a file bool readAudioData(const char* filename, char* buffer, int bufferSize) { std::ifstream audioFile(filename, std::ios::binary); if (!audioFile) { std::cout << "Failed to open audio file." << std::endl; return false; }
audioFile.read(buffer, bufferSize);
if (!audioFile) {
std::cout << "Failed to read audio data from file." << std::endl;
return false;
}
return true;
}
int main() { const char* serverIP = "192.168.0.100"; // Replace with the IP address of the remote speaker const int serverPort = 5000; // Replace with the port number on which the remote speaker is listening const char* audioFile = "audio.wav"; // Replace with the path to your audio file
// Read audio data from file
char audioBuffer[MAX_BUFFER_SIZE];
if (!readAudioData(audioFile, audioBuffer, MAX_BUFFER_SIZE)) {
std::cout << "Failed to read audio data." << std::endl;
return 1;
}
// Create a UDP socket
int socketFd = socket(AF_INET, SOCK_DGRAM, 0);
if (socketFd < 0) {
std::cout << "Failed to create socket." << std::endl;
return 1;
}
// Set up server address
struct sockaddr_in serverAddress{};
memset(&serverAddress, 0, sizeof(serverAddress));
serverAddress.sin_family = AF_INET;
serverAddress.sin_port = htons(serverPort);
if (inet_pton(AF_INET, serverIP, &(serverAddress.sin_addr)) <= 0) {
std::cout << "Invalid server IP address." << std::endl;
return 1;
}
// Transmit audio data
ssize_t numBytesSent = sendto(socketFd, audioBuffer, strlen(audioBuffer), 0,
(struct sockaddr*) &serverAddress, sizeof(serverAddress));
if (numBytesSent < 0) {
std::cout << "Failed to send audio data." << std::endl;
return 1;
}
// Close the socket
close(socketFd);
std::cout << "Audio transmission completed." << std::endl;
return 0;
}