r_submix: Add some tests
Add simple tests to verify remote submix behavior regarding
blocking writes. Currently one of the tests fails (does not finish),
this needs to be fixed.
Also fixed some minor issues in remote submix module code.
Bug: 73175392
Test: r_submix_tests
Change-Id: Ic88d0c385c0102e35b4f751f9c5cd8a6488949c8
diff --git a/modules/audio_remote_submix/audio_hw.cpp b/modules/audio_remote_submix/audio_hw.cpp
index eb6ae92..8c0c097 100644
--- a/modules/audio_remote_submix/audio_hw.cpp
+++ b/modules/audio_remote_submix/audio_hw.cpp
@@ -52,9 +52,9 @@
namespace android {
-// Set to 1 to enable extremely verbose logging in this module.
-#define SUBMIX_VERBOSE_LOGGING 0
-#if SUBMIX_VERBOSE_LOGGING
+// Uncomment to enable extremely verbose logging in this module.
+// #define SUBMIX_VERBOSE_LOGGING
+#if defined(SUBMIX_VERBOSE_LOGGING)
#define SUBMIX_ALOGV(...) ALOGV(__VA_ARGS__)
#define SUBMIX_ALOGE(...) ALOGE(__VA_ARGS__)
#else
@@ -205,7 +205,7 @@
int log_fd;
#endif // LOG_STREAMS_TO_FILES
- volatile int16_t read_error_count;
+ volatile uint16_t read_error_count;
};
// Determine whether the specified sample rate is supported by the submix module.
@@ -467,11 +467,9 @@
rsxadev->routes[route_idx].address);
if (rsxadev->routes[route_idx].rsxSink != 0) {
rsxadev->routes[route_idx].rsxSink.clear();
- rsxadev->routes[route_idx].rsxSink = 0;
}
if (rsxadev->routes[route_idx].rsxSource != 0) {
rsxadev->routes[route_idx].rsxSource.clear();
- rsxadev->routes[route_idx].rsxSource = 0;
}
memset(rsxadev->routes[route_idx].address, 0, AUDIO_DEVICE_MAX_ADDRESS_LEN);
#ifdef ENABLE_RESAMPLING
@@ -816,8 +814,8 @@
static uint8_t flush_buffer[64];
const size_t flushBufferSizeFrames = sizeof(flush_buffer) / frame_size;
size_t frames_to_flush_from_source = frames - availableToWrite;
- SUBMIX_ALOGV("out_write(): flushing %d frames from the pipe to avoid blocking",
- frames_to_flush_from_source);
+ SUBMIX_ALOGV("out_write(): flushing %llu frames from the pipe to avoid blocking",
+ (unsigned long long)frames_to_flush_from_source);
while (frames_to_flush_from_source) {
const size_t flush_size = min(frames_to_flush_from_source, flushBufferSizeFrames);
frames_to_flush_from_source -= flush_size;
@@ -898,7 +896,8 @@
}
SUBMIX_ALOGV("out_get_presentation_position() got frames=%llu timestamp sec=%llu",
- frames ? *frames : -1, timestamp ? timestamp->tv_sec : -1);
+ frames ? (unsigned long long)*frames : -1ULL,
+ timestamp ? (unsigned long long)timestamp->tv_sec : -1ULL);
return ret;
}
@@ -1541,7 +1540,7 @@
audio_bytes_per_sample(config->format);
const size_t buffer_size = max_buffer_period_size_frames * frame_size_in_bytes;
SUBMIX_ALOGV("adev_get_input_buffer_size() returns %zu bytes, %zu frames",
- buffer_size, buffer_period_size_frames);
+ buffer_size, max_buffer_period_size_frames);
return buffer_size;
}
return 0;
@@ -1692,10 +1691,10 @@
reinterpret_cast<const uint8_t *>(device) -
offsetof(struct submix_audio_device, device));
char msg[100];
- int n = sprintf(msg, "\nReroute submix audio module:\n");
+ int n = snprintf(msg, sizeof(msg), "\nReroute submix audio module:\n");
write(fd, &msg, n);
for (int i=0 ; i < MAX_ROUTES ; i++) {
- n = sprintf(msg, " route[%d] rate in=%d out=%d, addr=[%s]\n", i,
+ n = snprintf(msg, sizeof(msg), " route[%d] rate in=%d out=%d, addr=[%s]\n", i,
rsxadev->routes[i].config.input_sample_rate,
rsxadev->routes[i].config.output_sample_rate,
rsxadev->routes[i].address);